# PLEASE HELP!- Asterisk hungs up after 7 seconds **URL:** \ **Category:** Asterisk Support\ **Created:** [December 6, 2011, 3:58pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582 "2011-12-06T15:58:53Z")\ **Posts on this page:** 5\ **Page:** 1
**Author:** ![BorjaGVO](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/borjagvo/32/391_2.png) [@BorjaGVO](https://community.asterisk.org/u/BorjaGVO)\ **Post date:** [December 6, 2011, 3:58pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582/1 "2011-12-06T15:58:53Z")
Hi, first of all I would like to say that I just started with Asterisk a month ago. I’m a completely newbie. I’ve seen lot of people having this same issue. However, none of their solutions worked out for me. Yesterday I decided to reinstall Asterisk thinking that problem would go away (once, I had it working in some scenarios…). Anyway, here is what I’m doing: I have a NATed Asterisk and I’m doing tests everywhere. However, right now I’m testing it from home doing some calls from a SP (softphone-X-lite) inside LAN to another SP inside this same LAN (remember that Asterisk is in another LAN). Whenever a make a call it rings without a problem but there will be 2 problems: - The caller is not heared from the other side (while he does hear the other side) - The call lasts just a few seconds (7-10 sec.) I went out over the sip trace setting debug on and I see two thing that make me suspect that something isn’t going well: -At the beginning: \<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as04a28f41 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="501b3342" Content-Length: 0 - At the end: \<-------------\> — (9 headers 0 lines) — Retransmitting #2 (NAT) to 192.168.1.1:47388: BYE sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:actualize.es”, nonce="", response="a9457c2d90bef8cf33e8333f970336d8" X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 My sip.conf (changed some options but the result is the same): [b][general] context=unauthenticated allowguest=no srvlookup=yes udpbindaddr=0.0.0.0 tcpenable=no localnet=192.168.1.0/255.255.255.0 externaddr=212.0.118.86 ;directmedia=no ofi type=friend context=LocalSets nat=yes reinvite=yes qualify=yes host=dynamic ;directmedia=nonat secret=sip4ctu4l dtmfmode=auto disallow=all allow=alaw allow=ulaw 101 102 103 [/b] Here it is the whole trace (101 calling 103) (Please feel free of asking me ANY question regarding my configuration): [b] \<— SIP read from UDP:192.168.1.1:47388 —\> INVITE [sip:103@actualize.es](mailto:sip:103@actualize.es) SIP/2.0 Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;rport Max-Forwards: 70 Contact: sip:101@192.168.1.1:47388 To: sip:103@actualize.es From: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63215 Content-Length: 378 v=0 o=- 1323185834370550 1 IN IP4 192.168.1.38 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.38 t=0 0 a=ice-ufrag:603986 a=ice-pwd:e9ef4090258b1b29a1aaf2d9e40fa869 m=audio 62146 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.38 62146 typ host a=candidate:1 2 UDP 659134 192.168.1.38 62147 typ host \<-------------\> — (13 headers 13 lines) — Sending to 192.168.1.1:47388 (no NAT) Using INVITE request as basis request - NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. Found peer ‘101’ for ‘101’ from 192.168.1.1:47388 \<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —\> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as04a28f41 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="501b3342" Content-Length: 0 \<------------\> Scheduling destruction of SIP dialog ‘NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.’ in 6400 ms (Method: INVITE) \<— SIP read from UDP:192.168.1.1:47388 —\> ACK [sip:103@actualize.es](mailto:sip:103@actualize.es) SIP/2.0 Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-53192655047c7370-1—d8754z-;rport Max-Forwards: 70 To: sip:103@actualize.es;tag=as04a28f41 From: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 1 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — \<— SIP read from UDP:192.168.1.1:47388 —\> INVITE [sip:103@actualize.es](mailto:sip:103@actualize.es) SIP/2.0 Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;rport Max-Forwards: 70 Contact: sip:101@192.168.1.1:47388 To: sip:103@actualize.es From: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63215 Authorization: Digest [username=“101”,realm=“asterisk”,nonce=“501b3342”,uri="sip:103@actualize.es](mailto:username=%22101%22,realm=%22asterisk%22,nonce=%22501b3342%22,uri=%22sip:103@actualize.es)",response=“46208eb31f4b352ac1a16fd7c70880a0”,algorithm=MD5 Content-Length: 378 v=0 o=- 1323185834370550 1 IN IP4 192.168.1.38 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.38 t=0 0 a=ice-ufrag:603986 a=ice-pwd:e9ef4090258b1b29a1aaf2d9e40fa869 m=audio 62146 RTP/AVP 0 8 3 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.38 62146 typ host a=candidate:1 2 UDP 659134 192.168.1.38 62147 typ host \<-------------\> — (14 headers 13 lines) — Sending to 192.168.1.1:47388 (NAT) Using INVITE request as basis request - NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. Found peer ‘101’ for ‘101’ from 192.168.1.1:47388 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/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 192.168.1.38:62146 Looking for 103 in LocalSets (domain [actualize.es](http://actualize.es)) list\_route: hop: sip:101@192.168.1.1:47388 \<— Transmitting (NAT) to 192.168.1.1:47388 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Length: 0 \<------------\> Audio is at 29466 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.1:36842: INVITE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Date: Tue, 06 Dec 2011 15:37:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 276 v=0 o=root 224739662 224739662 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 29466 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=sendrecv * * * \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 102 INVITE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — list\_route: hop: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c \<— Transmitting (NAT) to 192.168.1.1:47388 —\> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Length: 0 \<------------\> \<— SIP read from UDP:192.168.1.1:47388 —\> \<-------------\> \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK0912e4b4;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 377 v=0 o=- 12967659441998509 1 IN IP4 192.168.1.34 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.34 t=0 0 a=ice-ufrag:4ba8a2 a=ice-pwd:729e3ff03866436c713d52d18176b88d m=audio 60798 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.34 60798 typ host a=candidate:1 2 UDP 659134 192.168.1.34 60799 typ host \<-------------\> — (12 headers 13 lines) — Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 192.168.1.34:60798 set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Transmitting (NAT) to 192.168.1.1:36842: ACK sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK1724c7ae;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Content-Length: 0 * * * Audio is at 19120 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (NAT) to 192.168.1.1:47388 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv \<------------\> set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Audio is at 29466 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.1:36842: INVITE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK60d8facd;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M 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: 274 v=0 o=root 224739662 224739663 IN IP4 192.168.1.38 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.38 t=0 0 m=audio 62146 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=sendrecv * * * \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK60d8facd;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 377 v=0 o=- 12967659441998509 2 IN IP4 192.168.1.34 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.34 t=0 0 a=ice-ufrag:4ba8a2 a=ice-pwd:729e3ff03866436c713d52d18176b88d m=audio 60798 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.34 60798 typ host a=candidate:1 2 UDP 659134 192.168.1.34 60799 typ host \<-------------\> — (12 headers 13 lines) — Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 192.168.1.34:60798 set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Transmitting (NAT) to 192.168.1.1:36842: ACK sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK09e3c707;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Content-Length: 0 * * * Retransmitting #1 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * Retransmitting #2 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * Retransmitting #3 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * Retransmitting #4 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * Retransmitting #5 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * Retransmitting #6 (NAT) to 192.168.1.1:47388: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.38:33176;branch=z9hG4bK-d8754z-6545cc5ef76dd324-1—d8754z-;received=192.168.1.1;rport=47388 From: sip:101@actualize.es;tag=92ad2c71 To: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r344965M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:103@192.168.1.199:5060 Content-Type: application/sdp Content-Length: 278 v=0 o=root 1482553346 1482553346 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * set\_destination: Parsing sip:101@192.168.1.1:47388 for address/port to send to set\_destination: set destination to 192.168.1.1:47388 Audio is at 19120 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.1:47388: INVITE sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK1e888176;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Contact: sip:103@192.168.1.199:5060 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M 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: 278 v=0 o=root 1482553346 1482553347 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 19120 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=sendrecv * * * set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Audio is at 29466 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.1.1:36842: INVITE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK437eb9b2;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M 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: 276 v=0 o=root 224739662 224739664 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 29466 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=sendrecv * * * Scheduling destruction of SIP dialog ‘3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060’ in 6400 ms (Method: INVITE) Scheduling destruction of SIP dialog ‘NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.’ in 6400 ms (Method: INVITE) \<— SIP read from UDP:192.168.1.1:47388 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK1e888176;rport=5060;received=212.0.118.86 Contact: sip:101@192.168.1.1:47388 To: sip:101@actualize.es;tag=92ad2c71 From: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63215 Content-Length: 376 v=0 o=- 1323185834370550 2 IN IP4 192.168.1.38 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.38 t=0 0 a=ice-ufrag:603986 a=ice-pwd:e9ef4090258b1b29a1aaf2d9e40fa869 m=audio 62146 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.38 62146 typ host a=candidate:1 2 UDP 659134 192.168.1.38 62147 typ host \<-------------\> — (12 headers 13 lines) — Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 192.168.1.38:62146 set\_destination: Parsing sip:101@192.168.1.1:47388 for address/port to send to set\_destination: set destination to 192.168.1.1:47388 Transmitting (NAT) to 192.168.1.1:47388: ACK sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK01829e93;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Contact: sip:103@192.168.1.199:5060 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Content-Length: 0 * * * set\_destination: Parsing sip:101@192.168.1.1:47388 for address/port to send to set\_destination: set destination to 192.168.1.1:47388 Reliably Transmitting (NAT) to 192.168.1.1:47388: BYE sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:actualize.es”, nonce="", response="a9457c2d90bef8cf33e8333f970336d8" X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 * * * Scheduling destruction of SIP dialog ‘NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.’ in 6400 ms (Method: INVITE) Retransmitting #1 (NAT) to 192.168.1.1:36842: INVITE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK437eb9b2;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M 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: 276 v=0 o=root 224739662 224739664 IN IP4 192.168.1.199 s=Asterisk PBX SVN-branch-1.8-r344965M c=IN IP4 192.168.1.199 t=0 0 m=audio 29466 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=sendrecv * * * \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK437eb9b2;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 104 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 377 v=0 o=- 12967659441998509 3 IN IP4 192.168.1.34 s=CounterPath X-Lite 4.1 c=IN IP4 192.168.1.34 t=0 0 a=ice-ufrag:4ba8a2 a=ice-pwd:729e3ff03866436c713d52d18176b88d m=audio 60798 RTP/AVP 8 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:1 1 UDP 659136 192.168.1.34 60798 typ host a=candidate:1 2 UDP 659134 192.168.1.34 60799 typ host \<-------------\> — (12 headers 13 lines) — Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/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 192.168.1.34:60798 set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Transmitting (NAT) to 192.168.1.1:36842: ACK sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK66a4db0f;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Contact: sip:101@192.168.1.199:5060 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 104 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Content-Length: 0 * * * set\_destination: Parsing sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c for address/port to send to set\_destination: set destination to 192.168.1.1:36842 Reliably Transmitting (NAT) to 192.168.1.1:36842: BYE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53225668;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 105 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 * * * Scheduling destruction of SIP dialog ‘3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060’ in 6400 ms (Method: INVITE) Retransmitting #1 (NAT) to 192.168.1.1:47388: BYE sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:actualize.es”, nonce="", response="a9457c2d90bef8cf33e8333f970336d8" X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 * * * Retransmitting #1 (NAT) to 192.168.1.1:36842: BYE sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53225668;rport Max-Forwards: 70 From: “101” sip:101@192.168.1.199;tag=as65397079 To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 105 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 * * * \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53225668;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 105 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Really destroying SIP dialog ‘3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060’ Method: INVITE \<— SIP read from UDP:192.168.1.1:36842 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK53225668;rport=5060;received=212.0.118.86 Contact: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c To: sip:103@192.168.1.1:36842;rinstance=ffff719e5d52082c;tag=8cf2350a From: "101"sip:101@192.168.1.199;tag=as65397079 Call-ID: 3a1286613c6f547d67c0a89401966ab3@192.168.1.199:5060 CSeq: 105 BYE User-Agent: X-Lite 4 release 4.1 stamp 63214 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Retransmitting #2 (NAT) to 192.168.1.1:47388: BYE sip:101@192.168.1.1:47388 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport Max-Forwards: 70 From: sip:103@actualize.es;tag=as09cec867 To: sip:101@actualize.es;tag=92ad2c71 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r344965M Proxy-Authorization: Digest username=“101”, realm=“asterisk”, algorithm=MD5, uri=“sip:actualize.es”, nonce="", response="a9457c2d90bef8cf33e8333f970336d8" X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 * * * \<— SIP read from UDP:192.168.1.1:47388 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport=5060;received=212.0.118.86 Contact: sip:101@192.168.1.1:47388 To: sip:101@actualize.es;tag=92ad2c71 From: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: X-Lite 4 release 4.1 stamp 63215 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — Really destroying SIP dialog ‘NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU.’ Method: INVITE \<— SIP read from UDP:192.168.1.1:47388 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport=5060;received=212.0.118.86 Contact: sip:101@192.168.1.1:47388 To: sip:101@actualize.es;tag=92ad2c71 From: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: X-Lite 4 release 4.1 stamp 63215 Content-Length: 0 \<-------------\> — (9 headers 0 lines) — \<— SIP read from UDP:192.168.1.1:47388 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.199:5060;branch=z9hG4bK55cea076;rport=5060;received=212.0.118.86 Contact: sip:101@192.168.1.1:47388 To: sip:101@actualize.es;tag=92ad2c71 From: sip:103@actualize.es;tag=as09cec867 Call-ID: NTQ4Zjk0YzRiNWNjYzYwYWU2ZDg0MmVmNWRiODZmMmU. CSeq: 103 BYE User-Agent: X-Lite 4 release 4.1 stamp 63215 Content-Length: 0 [/b] ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [December 6, 2011, 4:17pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582/2 "2011-12-06T16:17:10Z")
Unauthorised is normal. X-Lite version 3 (build 56125) has completely broken directmedia support (it doesn’t even reject it properly). I suspect, although it is giving the correct initial response, that version 4 still has flakey directmedia support. ---
**Author:** ![BorjaGVO](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/borjagvo/32/391_2.png) [@BorjaGVO](https://community.asterisk.org/u/BorjaGVO)\ **Post date:** [December 6, 2011, 5:23pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582/3 "2011-12-06T17:23:27Z")
[quote=“david55”]Unauthorised is normal. X-Lite version 3 (build 56125) has completely broken directmedia support (it doesn’t even reject it properly). I suspect, although it is giving the correct initial response, that version 4 still has flakey directmedia support.[/quote] IThanks for the reply! I see…well, I used Linphone but I don’t like it too much. Do you recommend any particular client? ---
**Author:** ![lorenzo\_s](https://avatars.discourse-cdn.com/v4/letter/l/eb8c5e/32.png) [@lorenzo\_s](https://community.asterisk.org/u/lorenzo_s)\ **Post date:** [December 12, 2011, 3:47pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582/4 "2011-12-12T15:47:59Z")
[quote]I see…well, I used Linphone but I don’t like it too much. Do you recommend any particular client? [/quote] hi! you can try “zoiper” or there is a new that im evaluating: (opensource and seems still under some good development) named “jitsi”. you can google the name for have some more info and download 8) 8) 8) 8) 8) ---
**Author:** ![BorjaGVO](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/borjagvo/32/391_2.png) [@BorjaGVO](https://community.asterisk.org/u/BorjaGVO)\ **Post date:** [December 12, 2011, 4:39pm UTC](https://community.asterisk.org/t/please-help-asterisk-hungs-up-after-7-seconds/37582/5 "2011-12-12T16:39:10Z")
[quote=“lorenzo\_s”][quote]I see…well, I used Linphone but I don’t like it too much. Do you recommend any particular client? [/quote] hi! you can try “zoiper” or there is a new that im evaluating: (opensource and seems still under some good development) named “jitsi”. you can google the name for have some more info and download 8) 8) 8) 8) 8)[/quote] I will give it a try! Thanks.