MusicOnHold not working

Hello,

This is my first post in this forum -yeah-

I have a problem with MusicOnHold and my sip account (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?

Its DAHDI installed?

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.

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.

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?

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?

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.

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?

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.

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 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
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
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
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 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
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
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”, 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 is 120 sec (Scheduling reregistration in 105 s)

[/code]

The trace starts too late.

Asterisk initiating with a=recvonly looks new to me.

Hm. And what can I do now?

Where do you see that the trace starts too late?

It starts on the ACK before the first warning message. It needs to start on the first INVITE of the call.

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
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”, 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 is 120 sec (Scheduling reregistration in 105 s)
v2201205126208248CLI> 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
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
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
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
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
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 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
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
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
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 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
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 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
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
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
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 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
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 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
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
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
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 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
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 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
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
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
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 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
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
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
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]

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.

I have found the mistake:

It was because of the Dial command. The parameters “tT” were missing.

It would still be good practice to set directmedia to no, rather than relying on T and t being incompatible with direct media.