# MusicOnHold not working **URL:** \ **Category:** Asterisk Support\ **Created:** [August 3, 2012, 11:50pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046 "2012-08-03T23:50:36Z")\ **Posts on this page:** 17\ **Page:** 1
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 3, 2012, 11:50pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/1 "2012-08-03T23:50:36Z")
Hello, This is my first post in this forum -yeah- I have a problem with MusicOnHold and my sip account ([sipgate.de](http://sipgate.de)). If I have an external call from my sip account and hold it the caller hears a few of seconds the MOH but then is silence. If I have an internal call it works fine. Curios is that when the extensions.ael is like this: `context voip { MyVoipNumber => { Answer(); MusicOnHold(); } } ` then it works fine too from the sip account. Only when I hold a call this problem is there. Anyone a solution? ---
**Author:** ![navaismo](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/navaismo/32/2481_2.png) [@navaismo](https://community.asterisk.org/u/navaismo)\ **Post date:** [August 4, 2012, 2:37am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/2 "2012-08-04T02:37:05Z")
Its DAHDI installed? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 4, 2012, 8:54am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/3 "2012-08-04T08:54:30Z")
What version are you using?? As already hinted, older versions require dahdi to provide timing for audio generators when there is no inbound RTP traffic. The very latest versions can be configured to use alternative timing sources. These alternatives were present, but broken, in some intermediate versions. You also need to enable internal timing. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 4, 2012, 1:11pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/4 "2012-08-04T13:11:05Z")
DAHDI ist installed: `||/ Name Version Description +++-==============-==============-============================================ ii dahdi 1:2.2.1.1-1 utilities for using the DAHDI kernel modules` I use Asterisk 1.8.15.0. I’ve activated internal\_timing in the asterisk.conf and reload it but nothing changes. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 4, 2012, 1:49pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/5 "2012-08-04T13:49:39Z")
That package looks like dahdi-tools, not dahdi. Are you sure that your Asterisk package was built with dahdi support. The best way of verifying the proper set of dahdi it to try using the Asterisk CLI dahdi commands. 1.8.5 probably has alternative timing sources; which source is selected? ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 4, 2012, 8:46pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/6 "2012-08-04T20:46:50Z")
You’re right. Its dahdi-tools, not dahdi. I can’t install dahdi because it’s on a vServer and I can’t modify the kernel. I don’t know which timing source is selected. Where can I check this? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 4, 2012, 11:09pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/7 "2012-08-04T23:09:34Z")
We use an older version, which doesn’t have the option. As 1.8.15’s asterisk.conf doesn’t seem to have the option, I suspect you will either have to change it in make menuconfig - which probably means that you have to change from a package install to a source code install, or it the option isn’t available for that version. By the way, if you do not have enough control of your virtual machine environment to be able to load kernel modules on the guest, you are unlikely to have enough control of it to run real time applications, like VoIP on it. I would expect you to get stuttering and pauses in MoH and voice announcements, even with a suitable timing source. I can load dahdi drivers on our VMWare system, even though that is too overloaded to support VoIP in a production environment. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 5, 2012, 10:29pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/8 "2012-08-05T22:29:30Z")
I know that a virtual machine isn’t the best way but I would like to test asterisk. I’ve changed the verbose and debug today. This is the output: ` -- SIP/2000-00000001 is ringing -- SIP/2000-00000001 answered SIP/ext-sip-account-00000000 -- Stopped music on hold on SIP/ext-sip-account-00000000 -- Remotely bridging SIP/ext-sip-account-00000000 and SIP/2000-00000001 -- Started music on hold, class 'default', on SIP/ext-sip-account-00000000 [2012-08-06 00:12:45] WARNING[5673]: chan_sip.c:20526 handle_response_invite: just did sched_add waitid(28) for sip_reinvite_retry for dialog 1a810c700967c8770a1695304bb08047@MyIP:5060 in handle_response_invite -- Stopped music on hold on SIP/ext-sip-account-00000000 ` This confirms your guess? For the install I use the source code. Do you know where the option in the menuselect is? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 6, 2012, 6:33am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/9 "2012-08-06T06:33:02Z")
That suggests that MoH was explicitly stopped, which is not consistent with a timing source issue, although you do need a timing source for reliable MoH. You need to enable SIP debugging to try and explain those messages. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 6, 2012, 8:17am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/10 "2012-08-06T08:17:31Z")
When im enable SIP debugging there is the same error. I can’t see other errors, warnings or soemthing else: [code]\<-------------\> — (7 headers 0 lines) — set\_destination: Parsing sip:2000@CallerIP:5060;transport=UDP;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;transport=UDP;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK635d8543;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as466ed077 To: sip:2000@CallerIP;ob;tag=IHCypA.yaHvFhBM368IleTLc1PvxoLRk Contact: sip:anonymous@ServerIP:5060 Call-ID: 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * [2012-08-06 10:03:10] WARNING[7066]: chan\_sip.c:20526 handle\_response\_invite: just did sched\_add waitid(80) for sip\_reinvite\_retry for dialog 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 in handle\_response\_invite \<— SIP read from UDP:CallerIP:5060 —\> ACK sip:anonymous@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPjurmH90KTcVEW4U2Lr-4q1jfvxDtc2DlN Max-Forwards: 70 From: sip:2000@CallerIP;ob;tag=IHCypA.yaHvFhBM368IleTLc1PvxoLRk To: “anonymous” sip:anonymous@ServerIP;tag=as466ed077 Call-ID: 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 CSeq: 10233 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — set\_destination: Parsing sip:2000@CallerIP:5060;transport=UDP;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Audio is at 12214 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;transport=UDP;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK72160e4c;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as466ed077 To: sip:2000@CallerIP;ob;tag=IHCypA.yaHvFhBM368IleTLc1PvxoLRk Contact: sip:anonymous@ServerIP:5060 Call-ID: 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 265 v=0 o=root 1669423909 1669423913 IN IP4 217.116.117.12 s=Asterisk PBX 1.8.15.0 c=IN IP4 217.116.117.12 t=0 0 m=audio 19256 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly * * * \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK72160e4c Call-ID: 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as466ed077 To: sip:2000@CallerIP;ob;tag=IHCypA.yaHvFhBM368IleTLc1PvxoLRk CSeq: 105 INVITE Contact: sip:2000@CallerIP:5060;transport=UDP;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS upported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553228945 3553228949 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 0 101 a=rtcp:4003 IN IP4 192.168.178.23 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4002 set\_destination: Parsing sip:2000@CallerIP:5060;transport=UDP;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;transport=UDP;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK2f431ff9;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as466ed077 To: sip:2000@CallerIP;ob;tag=IHCypA.yaHvFhBM368IleTLc1PvxoLRk Contact: sip:anonymous@ServerIP:5060 Call-ID: 5386adb850e7426975c1170021e0ebdd@ServerIP:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * ``` -- Stopped music on hold on SIP/ext-sip-account-00000002 -- Started music on hold, class 'default', on SIP/ext-sip-account-00000002 ``` set\_destination: Parsing sip:sipgateIP;lr;ftag=as298aa57a for address/port to send to set\_destination: set destination to sipgateIP:5060 Audio is at 12094 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to sipgateIP:5060: INVITE [sip:anonymous@217.116.117.12](mailto:sip:anonymous@217.116.117.12) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK49089775;rport Route: sip:sipgateIP;lr;ftag=as298aa57a,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as298aa57a Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as777b8fca To: “anonymous” sip:anonymous@sipgate.de;tag=as298aa57a Contact: sip:Nr@ServerIP:5060 Call-ID: [71052d21459b63e664439ae34b5a931b@sipgate.de](mailto:71052d21459b63e664439ae34b5a931b@sipgate.de) CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 235 v=0 o=root 399190703 399190707 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 12094 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK49089775;rport=5060 From: sip:00493069205575@sipgate.de;tag=as777b8fca To: “anonymous” sip:anonymous@sipgate.de;tag=as298aa57a Call-ID: [71052d21459b63e664439ae34b5a931b@sipgate.de](mailto:71052d21459b63e664439ae34b5a931b@sipgate.de) CSeq: 105 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK49089775;rport=5060 From: sip:00493069205575@sipgate.de;tag=as777b8fca To: “anonymous” sip:anonymous@sipgate.de;tag=as298aa57a Call-ID: [71052d21459b63e664439ae34b5a931b@sipgate.de](mailto:71052d21459b63e664439ae34b5a931b@sipgate.de) CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:anonymous@217.116.117.12 Content-Type: application/sdp Content-Length: 260 v=0 o=root 616223999 616224003 IN IP4 217.116.117.12 s=sipgate VoIP GW c=IN IP4 217.116.117.12 t=0 0 m=audio 19256 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.12:19256 set\_destination: Parsing sip:sipgateIP;lr;ftag=as298aa57a for address/port to send to set\_destination: set destination to sipgateIP:5060 Transmitting (NAT) to sipgateIP:5060: ACK [sip:anonymous@217.116.117.12](mailto:sip:anonymous@217.116.117.12) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK1daff39b;rport Route: sip:sipgateIP;lr;ftag=as298aa57a,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as298aa57a Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as777b8fca To: “anonymous” sip:anonymous@sipgate.de;tag=as298aa57a Contact: sip:Nr@ServerIP:5060 Call-ID: [71052d21459b63e664439ae34b5a931b@sipgate.de](mailto:71052d21459b63e664439ae34b5a931b@sipgate.de) CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * [2012-08-06 10:03:20] NOTICE[7066]: chan\_sip.c:13151 sip\_reregister: – Re-registration for [Nr@sipgate.de](mailto:Nr@sipgate.de) REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to sipgateIP:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK2b1f5cd1;rport Max-Forwards: 70 From: sip:Nr@sipgate.de;tag=as75a1c012 To: sip:Nr@sipgate.de Call-ID: 4e6ac1691f0b75933009df81082d9388@ServerIP CSeq: 106 REGISTER User-Agent: Asterisk PBX 1.8.15.0 Authorization: Digest username=“Nr”, realm=“[sipgate.de](http://sipgate.de)”, algorithm=MD5, uri=“sip:sipgate.de”, nonce=“501f7b65c1e4917d307c761d66af85c98ceff195”, response="2f2842e0cfcc8df774f30de57e355913" Expires: 120 Contact: sip:Nr@ServerIP:5060 Content-Length: 0 * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK2b1f5cd1;rport=5060 From: sip:Nr@sipgate.de;tag=as75a1c012 To: sip:Nr@sipgate.de;tag=fbf1d80521ea9f98078b6998e7669f9b.3b4f Call-ID: 4e6ac1691f0b75933009df81082d9388@ServerIP CSeq: 106 REGISTER Contact: sip:Nr@ServerIP:5060;expires=120 Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Scheduling destruction of SIP dialog ‘4e6ac1691f0b75933009df81082d9388@ServerIP’ in 32000 ms (Method: REGISTER) [2012-08-06 10:03:20] NOTICE[7066]: chan\_sip.c:20905 handle\_response\_register: Outbound Registration: Expiry for [sipgate.de](http://sipgate.de) is 120 sec (Scheduling reregistration in 105 s) [/code] ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 6, 2012, 9:22am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/11 "2012-08-06T09:22:04Z")
The trace starts too late. Asterisk initiating with a=recvonly looks new to me. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 6, 2012, 9:30am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/12 "2012-08-06T09:30:42Z")
Hm. And what can I do now? Where do you see that the trace starts too late? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 6, 2012, 9:32am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/13 "2012-08-06T09:32:51Z")
It starts on the ACK before the first warning message. It needs to start on the first INVITE of the call. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 6, 2012, 9:57am UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/14 "2012-08-06T09:57:41Z")
Ok. Sorry but this was an excerpt of the sip debug…I think my Putty had wrong setting and it delete the old lines. The full sip debug: I call the Number, answer the call, set the call on hold, get the call back and hang up: [code]— (7 headers 0 lines) — \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK094c39be Call-ID: 1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as34125ff0 To: sip:2000@CallerIP;ob;tag=eCxgGVvVAU2LgHAjSgMmi.iVRzwPyxI7 CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK094c39be;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as34125ff0 To: sip:2000@CallerIP:5060;ob;tag=eCxgGVvVAU2LgHAjSgMmi.iVRzwPyxI7 Contact: sip:anonymous@ServerIP:5060 Call-ID: 1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * Scheduling destruction of SIP dialog ‘1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060’ in 32000 ms (Method: INVITE) Really destroying SIP dialog ‘3ef8b2763aaf72746610a54554629d50@sipgate.de’ Method: BYE [2012-08-06 11:42:57] NOTICE[27943]: chan\_sip.c:13151 sip\_reregister: – Re-registration for [Nr@sipgate.de](mailto:Nr@sipgate.de) REGISTER 11 headers, 0 lines Reliably Transmitting (NAT) to sipgateIP:5060: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK792dd0d5;rport Max-Forwards: 70 From: sip:Nr@sipgate.de;tag=as6f1be703 To: sip:Nr@sipgate.de Call-ID: 5b77d58a4585f32c52dd1d00657b433c@ServerIP CSeq: 104 REGISTER User-Agent: Asterisk PBX 1.8.15.0 Authorization: Digest username=“Nr”, realm=“[sipgate.de](http://sipgate.de)”, algorithm=MD5, uri=“sip:sipgate.de”, nonce=“501f9390e91a7b775e34a39bdf76a205534dad00”, response="a03eedafba2f5e98ca579aa73bd1934e" Expires: 120 Contact: sip:Nr@ServerIP:5060 Content-Length: 0 * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK792dd0d5;rport=5060 From: sip:Nr@sipgate.de;tag=as6f1be703 To: sip:Nr@sipgate.de;tag=fbf1d80521ea9f98078b6998e7669f9b.46e9 Call-ID: 5b77d58a4585f32c52dd1d00657b433c@ServerIP CSeq: 104 REGISTER Contact: sip:Nr@ServerIP:5060;expires=120 Content-Length: 0 \<-------------\> — (8 headers 0 lines) — Scheduling destruction of SIP dialog ‘5b77d58a4585f32c52dd1d00657b433c@ServerIP’ in 32000 ms (Method: REGISTER) [2012-08-06 11:42:57] NOTICE[27943]: chan\_sip.c:20905 handle\_response\_register: Outbound Registration: Expiry for [sipgate.de](http://sipgate.de) is 120 sec (Scheduling reregistration in 105 s) v2201205126208248_CLI\> clear No such command ‘clear’ (type ‘core show help clear’ for other possible commands) Really destroying SIP dialog ‘1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060’ Method: INVITE v2201205126208248_CLI\> Reliably Transmitting (NAT) to sipgateIP:5060: OPTIONS sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK34ea94ab;rport Max-Forwards: 70 From: “asterisk” sip:Nr@ServerIP;tag=as206eec2c To: sip:sipgate.de Contact: sip:Nr@ServerIP:5060 Call-ID: 6645e4490327183f56037244615d04d1@ServerIP:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.15.0 Date: Mon, 06 Aug 2012 09:43:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK34ea94ab;rport=5060 From: “asterisk” sip:Nr@ServerIP;tag=as206eec2c To: sip:sipgate.de;tag=64c295986a77a1f756ad49f3e6513d0d.5ae7 Call-ID: 6645e4490327183f56037244615d04d1@ServerIP:5060 CSeq: 102 OPTIONS Accept: _/_ Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 \<-------------\> — (11 headers 0 lines) — Really destroying SIP dialog ‘6645e4490327183f56037244615d04d1@ServerIP:5060’ Method: OPTIONS v2201205126208248\*CLI\> Really destroying SIP dialog ‘5b77d58a4585f32c52dd1d00657b433c@ServerIP’ Method: REGISTER \<— SIP read from UDP:sipgateIP:5060 —\> INVITE sip:Nr@ServerIP:5060 SIP/2.0 Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5 Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060 Max-Forwards: 67 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de Contact: sip:anonymous@217.116.117.69 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 387 v=0 o=root 1143069018 1143069018 IN IP4 217.116.117.69 s=sipgate VoIP GW c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 8 0 3 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (18 headers 17 lines) — Sending to sipgateIP:5060 (NAT) Using INVITE request as basis request - [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) Found peer ‘ext-sip-account’ for ‘anonymous’ from sipgateIP:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 112 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format G729 for ID 18 Found audio description format G726-32 for ID 112 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.69:18490 Looking for Nr in von-voip-provider (domain ServerIP) list\_route: hop: sip:sipgateIP;lr;ftag=as0c7de2d8 list\_route: hop: sip:172.20.40.3;lr=on list\_route: hop: sip:sipgateIP;lr;ftag=as0c7de2d8 \<— Transmitting (NAT) to sipgateIP:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0;received=sipgateIP;rport=5060 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5 Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060 Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:Nr@ServerIP:5060 Content-Length: 0 \<------------\> – Executing [Nr@von-voip-provider:1] Answer(“SIP/ext-sip-account-00000002”, “”) in new stack Audio is at 11222 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (NAT) to sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0;received=sipgateIP;rport=5060 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5 Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060 Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 Record-Route: sip:172.20.40.3;lr=on Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de;tag=as160de318 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:Nr@ServerIP:5060 Content-Type: application/sdp Content-Length: 284 v=0 o=root 1630035855 1630035855 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 11222 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv \<------------\> \<— SIP read from UDP:sipgateIP:5060 —\> ACK sip:Nr@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.2 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.2 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK781b2e7f Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK781b2e7f;rport=5060 Max-Forwards: 67 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de;tag=as160de318 Contact: sip:anonymous@217.116.117.69 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 ACK Content-Length: 0 X-hint: rr-enforced \<-------------\> — (13 headers 0 lines) — – Executing [Nr@von-voip-provider:2] Dial(“SIP/ext-sip-account-00000002”, “SIP/2000&SIP/2001,m”) in new stack == Using SIP RTP CoS mark 5 Audio is at 19112 Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6e0a50d4;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP:5060;ob Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15.0 Date: Mon, 06 Aug 2012 09:43:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 282 v=0 o=root 270956748 270956748 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 19112 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * ``` -- Called SIP/2000 ``` Really destroying SIP dialog ‘6a1b82dd713d3cf9179df0fd5c7a7474@ServerIP:5060’ Method: INVITE [2012-08-06 11:43:31] WARNING[518]: app\_dial.c:2341 dial\_exec\_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown) – Started music on hold, class ‘default’, on SIP/ext-sip-account-00000002 [2012-08-06 11:43:31] NOTICE[518]: channel.c:4176 \_\_ast\_read: Dropping incompatible voice frame on SIP/ext-sip-account-00000002 of format ulaw since our native format has changed to 0x8 (alaw) \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob CSeq: 102 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 102 INVITE Contact: sip:2000@CallerIP:5060;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 \<-------------\> — (9 headers 0 lines) — list\_route: hop: sip:2000@CallerIP:5060;ob – SIP/2000-00000003 is ringing \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 102 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: sip:2000@CallerIP:5060;ob Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553235010 3553235011 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 0 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4006 list\_route: hop: sip:2000@CallerIP:5060;ob set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK7210ac68;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * ``` -- SIP/2000-00000003 answered SIP/ext-sip-account-00000002 -- Stopped music on hold on SIP/ext-sip-account-00000002 -- Remotely bridging SIP/ext-sip-account-00000002 and SIP/2000-00000003 ``` set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Audio is at 11222 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to sipgateIP:5060: INVITE [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6394f884;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 1630035855 1630035856 IN IP4 192.168.178.23 s=Asterisk PBX 1.8.15.0 c=IN IP4 192.168.178.23 t=0 0 m=audio 4006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Audio is at 19112 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK5bc39143;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 239 v=0 o=root 270956748 270956749 IN IP4 217.116.117.69 s=Asterisk PBX 1.8.15.0 c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK6394f884;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK6394f884;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:anonymous@217.116.117.69 Content-Type: application/sdp Content-Length: 262 v=0 o=root 1143069018 1143069019 IN IP4 217.116.117.69 s=sipgate VoIP GW c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.69:18490 set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Transmitting (NAT) to sipgateIP:5060: ACK [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK70251ad5;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK5bc39143 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 103 INVITE Contact: sip:2000@CallerIP:5060;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553235010 3553235012 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 0 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4006 set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK4ba46e24;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * \<— SIP read from UDP:CallerIP:5060 —\> INVITE sip:anonymous@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo- Max-Forwards: 70 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Contact: sip:2000@CallerIP:5060;ob Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16255 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple r1108 / GT-I9100-15 Content-Type: application/sdp Content-Length: 394 v=0 o=- 3553235010 3553235013 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 9 107 106 105 0 8 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:9 G722/8000 a=rtpmap:107 speex/32000 a=rtpmap:106 speex/16000 a=rtpmap:105 speex/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly \<-------------\> — (15 headers 17 lines) — Sending to CallerIP:5060 (NAT) Found RTP audio format 9 Found RTP audio format 107 Found RTP audio format 106 Found RTP audio format 105 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format G722 for ID 9 Found unknown media description format speex for ID 107 Found audio description format speex for ID 106 Found audio description format speex for ID 105 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x20000120c (ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4006 \<— Transmitting (NAT) to CallerIP:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo-;received=CallerIP;rport=5060 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16255 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:anonymous@ServerIP:5060 Content-Length: 0 \<------------\> Audio is at 19112 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (NAT) to CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo-;received=CallerIP;rport=5060 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16255 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:anonymous@ServerIP:5060 Content-Type: application/sdp Content-Length: 263 v=0 o=root 270956748 270956750 IN IP4 217.116.117.69 s=Asterisk PBX 1.8.15.0 c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly \<------------\> set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Audio is at 11222 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to sipgateIP:5060: INVITE [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK500526ab;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 1630035855 1630035857 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 11222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * ``` -- Started music on hold, class 'default', on SIP/ext-sip-account-00000002 ``` \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK500526ab;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK500526ab;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:anonymous@217.116.117.69 Content-Type: application/sdp Content-Length: 262 v=0 o=root 1143069018 1143069020 IN IP4 217.116.117.69 s=sipgate VoIP GW c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.69:18490 set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Transmitting (NAT) to sipgateIP:5060: ACK [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK57ef9aa3;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Audio is at 11222 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to sipgateIP:5060: INVITE [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK69d13ef5;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 1630035855 1630035858 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 11222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK69d13ef5;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 104 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK69d13ef5;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:anonymous@217.116.117.69 Content-Type: application/sdp Content-Length: 262 v=0 o=root 1143069018 1143069021 IN IP4 217.116.117.69 s=sipgate VoIP GW c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.69:18490 set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Transmitting (NAT) to sipgateIP:5060: ACK [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6a4ca94e;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * \<— SIP read from UDP:CallerIP:5060 —\> ACK sip:anonymous@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPju.FKdbREse-IBM5e98lXY0IRPZb2WfrF Max-Forwards: 70 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16255 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Audio is at 19112 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK351d5160;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 270956748 270956751 IN IP4 217.116.117.69 s=Asterisk PBX 1.8.15.0 c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly * * * \<— SIP read from UDP:CallerIP:5060 —\> INVITE sip:anonymous@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL Max-Forwards: 70 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Contact: sip:2000@CallerIP:5060;ob Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16256 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553235010 3553235014 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 8 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly \<-------------\> — (14 headers 12 lines) — \<— Reliably Transmitting (NAT) to CallerIP:5060 —\> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL;received=CallerIP;rport=5060 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16256 INVITE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 \<------------\> Sending to CallerIP:5060 (NAT) \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 491 Another INVITE transaction in progress Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK351d5160 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 104 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK351d5160;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * [2012-08-06 11:43:37] WARNING[27943]: chan\_sip.c:20526 handle\_response\_invite: just did sched\_add waitid(51) for sip\_reinvite\_retry for dialog 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 in handle\_response\_invite \<— SIP read from UDP:CallerIP:5060 —\> ACK sip:anonymous@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL Max-Forwards: 70 From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 16256 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Audio is at 19112 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK1a367da5;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 270956748 270956752 IN IP4 217.116.117.69 s=Asterisk PBX 1.8.15.0 c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly * * * \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK1a367da5 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 105 INVITE Contact: sip:2000@CallerIP:5060;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553235010 3553235014 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 0 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4006 set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK216abeba;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * ``` -- Stopped music on hold on SIP/ext-sip-account-00000002 -- Started music on hold, class 'default', on SIP/ext-sip-account-00000002 ``` set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Audio is at 11222 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to sipgateIP:5060: INVITE [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK374031f1;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 1630035855 1630035859 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 11222 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv * * * \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK374031f1;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 105 INVITE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — \<— SIP read from UDP:sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK374031f1;rport=5060 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 105 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:anonymous@217.116.117.69 Content-Type: application/sdp Content-Length: 262 v=0 o=root 1143069018 1143069022 IN IP4 217.116.117.69 s=sipgate VoIP GW c=IN IP4 217.116.117.69 t=0 0 m=audio 18490 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 217.116.117.69:18490 set\_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to set\_destination: set destination to sipgateIP:5060 Transmitting (NAT) to sipgateIP:5060: ACK [sip:anonymous@217.116.117.69](mailto:sip:anonymous@217.116.117.69) SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK56ba5450;rport Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8 Max-Forwards: 70 From: sip:00493069205575@sipgate.de;tag=as160de318 To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 Contact: sip:Nr@ServerIP:5060 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * \<— SIP read from UDP:sipgateIP:5060 —\> BYE sip:Nr@ServerIP:5060 SIP/2.0 Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKe479.a7d3cb64.0 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKe479.a7d3cb64.0 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK0cc06cb6 Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK0cc06cb6;rport=5060 Max-Forwards: 67 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de;tag=as160de318 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 BYE Content-Length: 0 X-hint: rr-enforced \<-------------\> — (12 headers 0 lines) — Sending to sipgateIP:5060 (NAT) Scheduling destruction of SIP dialog ‘50b55e844c4028657506bebf100a5a1e@sipgate.de’ in 6400 ms (Method: BYE) \<— Transmitting (NAT) to sipgateIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKe479.a7d3cb64.0;received=sipgateIP;rport=5060 Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKe479.a7d3cb64.0 Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK0cc06cb6 Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK0cc06cb6;rport=5060 From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8 To: sip:00493069205575@sipgate.de;tag=as160de318 Call-ID: [50b55e844c4028657506bebf100a5a1e@sipgate.de](mailto:50b55e844c4028657506bebf100a5a1e@sipgate.de) CSeq: 103 BYE Server: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 \<------------\> – Stopped music on hold on SIP/ext-sip-account-00000002 set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Audio is at 19112 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to CallerIP:5060: INVITE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK5c5f7955;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 106 INVITE User-Agent: Asterisk PBX 1.8.15.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 235 v=0 o=root 270956748 270956753 IN IP4 ServerIP s=Asterisk PBX 1.8.15.0 c=IN IP4 ServerIP t=0 0 m=audio 19112 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly * * * Scheduling destruction of SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ in 32000 ms (Method: ACK) == Spawn extension (von-voip-provider, Nr, 2) exited non-zero on ‘SIP/ext-sip-account-00000002’ \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK5c5f7955 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 106 INVITE Contact: sip:2000@CallerIP:5060;ob Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 257 v=0 o=- 3553235010 3553235015 IN IP4 192.168.178.23 s=pjmedia c=IN IP4 192.168.178.23 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 0 101 a=rtcp:4007 IN IP4 192.168.178.23 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly \<-------------\> — (11 headers 12 lines) — Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.178.23:4006 set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Transmitting (NAT) to CallerIP:5060: ACK sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK060ea50b;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Contact: sip:anonymous@ServerIP:5060 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.15.0 Content-Length: 0 * * * set\_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to set\_destination: set destination to CallerIP:5060 Reliably Transmitting (NAT) to CallerIP:5060: BYE sip:2000@CallerIP:5060;ob SIP/2.0 Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK0bcd7924;rport Max-Forwards: 70 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 CSeq: 107 BYE User-Agent: Asterisk PBX 1.8.15.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 * * * Scheduling destruction of SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ in 32000 ms (Method: ACK) \<— SIP read from UDP:CallerIP:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK0bcd7924 Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75 To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic CSeq: 107 BYE Content-Length: 0 \<-------------\> — (7 headers 0 lines) — Really destroying SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ Method: ACK Really destroying SIP dialog ‘50b55e844c4028657506bebf100a5a1e@sipgate.de’ Method: BYE \<— SIP read from UDP:CallerIP:5060 —\> \<-------------\> [/code] ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 9, 2012, 1:04pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/15 "2012-08-09T13:04:20Z")
You’ve done a direct media re-invite of sipgate to the non-routeable address 192.168.178.23. As I don’t know whether ServerIP is routable, or not, I don’t know if the problem is a misconfiguration of Asterisk, of what is behind ServerIP. ---
**Author:** ![Marcoo](https://avatars.discourse-cdn.com/v4/letter/m/d78d45/32.png) [@Marcoo](https://community.asterisk.org/u/Marcoo)\ **Post date:** [August 9, 2012, 10:19pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/16 "2012-08-09T22:19:34Z")
I have found the mistake: It was because of the Dial command. The parameters “tT” were missing. ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [August 9, 2012, 11:12pm UTC](https://community.asterisk.org/t/musiconhold-not-working/40046/17 "2012-08-09T23:12:35Z")
It would still be good practice to set directmedia to no, rather than relying on T and t being incompatible with direct media.