Andrea Deltacom Posted December 16, 2010 Report Posted December 16, 2010 Hi, i'm using Snom ONE Free with TWT Voip Trunks. It was going good, but from yesterday I can't no more dial. After 5 seconds it goes in timeout! I don't know what happened. It's making me crazy I didn't make any changes to my configuration This is a part of Sip LOG [7] 2010/12/16 10:20:05: SIP Rx udp:82.113.194.190:5060: SIP/2.0 183 Call progress v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171 Record-Route: <sip:5698784-192.168.0.133.dialog.cgatepro;lr> Record-Route: <sip:192.168.0.133:5060;lr> Record-Route: <sip:82.113.194.190:5060;lr> f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275 t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1 i: a2d6a018@pbx CSeq: 19295 INVITE m: <sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190> User-Agent: CommuniGatePro-callLeg/5.3.7 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,REFER c: application/sdp l: 267 v=0 o=CGPLeg387906 2482349901 1241174951 IN IP4 82.113.194.196 s=SIP Call c=IN IP4 82.113.194.196 t=0 0 m=audio 20980 RTP/AVP 8 101 c=IN IP4 82.113.194.196 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcpping:F:381081:38108178 [6] 2010/12/16 10:20:05: Codec pcma/8000 is chosen for call id a2d6a018@pbx [6] 2010/12/16 10:20:05: Sending RTP for a2d6a018@pbx to 82.113.194.196:20980, codec pcma/8000 [7] 2010/12/16 10:20:05: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.: RTP pass-through mode [7] 2010/12/16 10:20:05: a2d6a018@pbx: RTP pass-through mode [6] 2010/12/16 10:20:05: Different Codecs (local pcmu/8000, remote pcma/8000), callid MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY., falling back to transcoding [7] 2010/12/16 10:20:09: SIP Tx udp:192.168.2.22:23646: SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 192.168.2.22:23646;branch=z9hG4bK-d8754z-1006586e4773aa76-1---d8754z-;rport=23646 From: "999" <sip:999@192.168.2.254>;tag=3e27cd2c To: "NUMTEL" <sip:NUMTEL@192.168.2.254>;tag=d8791cb712 Call-ID: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY. CSeq: 2 INVITE Contact: <sip:999@192.168.2.254:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/4.2.0.3958 Content-Length: 0 [7] 2010/12/16 10:20:09: SIP Tx udp:82.113.194.190:5060: CANCEL sip:NUMTEL@sip.twt.it;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport From: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275 To: <sip:3928435208@sip.twt.it;user=phone> Call-ID: a2d6a018@pbx CSeq: 19295 CANCEL Max-Forwards: 70 P-Asserted-Identity: "Consorzio Servizi Auto" <sip:NUMTEL@sip.twt.it> Authorization: Digest realm="twtmail.twt.it",nonce="2EF7B69D7C18A493D722",response="1cc255a13c4901611ed21d8306a100b9",username="NUMTEL",uri="sip:NUMTEL@sip.twt.it;user=phone",qop="auth",nc=00000002,cnonce="cbd770c4",opaque="opaqueData",algorithm=MD5 Content-Length: 0 [7] 2010/12/16 10:20:09: SIP Rx udp:82.113.194.190:5060: SIP/2.0 200 OK v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171 f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275 t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=74D52A3E i: a2d6a018@pbx CSeq: 19295 CANCEL Server: CommuniGatePro/5.3.7 l: 0 [7] 2010/12/16 10:20:09: Call a2d6a018@pbx: Clear last request [7] 2010/12/16 10:20:09: SIP Rx udp:82.113.194.190:5060: SIP/2.0 487 Request cancelled v: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport=54084;received=78.7.88.171 f: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275 t: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1 i: a2d6a018@pbx CSeq: 19295 INVITE Server: CommuniGatePro/5.3.7 l: 0 [7] 2010/12/16 10:20:09: Call a2d6a018@pbx: Clear last INVITE [7] 2010/12/16 10:20:09: SIP Tx udp:82.113.194.190:5060: ACK sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK-8065394e3b6d25dec7891a3c3b8933f7;rport Route: <sip:82.113.194.190:5060;lr> Route: <sip:192.168.0.133:5060;lr> Route: <sip:5698784-192.168.0.133.dialog.cgatepro;lr> From: "Deltacom Andrea" <sip:NUMTEL@localhost;user=phone>;tag=1302926275 To: <sip:NUMTEL@sip.twt.it;user=phone>;tag=B11E799E-387906-FBB87516_kmbcnrx-23A1 Call-ID: a2d6a018@pbx CSeq: 19295 ACK Max-Forwards: 70 Contact: <sip:NUMTEL@192.168.2.254:5060;transport=udp> P-Asserted-Identity: "Consorzio Servizi Auto" <sip:509184860231@sip.twt.it> Authorization: Digest realm="twtmail.twt.it",nonce="2EF7B69D7C18A493D722",response="b1e68d419c86b8de84ad041c2df03ccb",username="509184860231",uri="sip:signode-387906-FBB87516_kmbcnrx-23A1@82.113.194.190",qop="auth",nc=00000003,cnonce="49364538",opaque="opaqueData",algorithm=MD5 Content-Length: 0 [5] 2010/12/16 10:20:09: INVITE Response 487 Request cancelled: Terminate a2d6a018@pbx [7] 2010/12/16 10:20:09: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY.: Media-aware pass-through mode [7] 2010/12/16 10:20:09: SIP Rx udp:192.168.2.22:23646: ACK sip:NUMTEL@192.168.2.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.22:23646;branch=z9hG4bK-d8754z-1006586e4773aa76-1---d8754z-;rport To: "NUMTEL" <sip:3928435208@192.168.2.254>;tag=d8791cb712 From: "999"<sip:999@192.168.2.254>;tag=3e27cd2c Call-ID: MjQ4MTZiOGEwYmNlZjJiMGMwMjlkYjNiZjhhZDgzYzY. CSeq: 2 ACK Content-Length: 0 Quote
Vodia PBX Posted December 16, 2010 Report Posted December 16, 2010 Check if your trunk has something set in the failover section and turn it off. Maybe it is programmed to failover after 5 seconds, and a couple of days ago that was fine (because the provider was answering quick) and now the response takes a little bit longer. Quote
Andrea Deltacom Posted December 16, 2010 Author Report Posted December 16, 2010 I had put it 10s, but then i tried removing it but nothing changed ... the phone rings but after some second it goes down before the calling reply Quote
Vodia PBX Posted December 16, 2010 Report Posted December 16, 2010 Does it work fine if you take the failover out completely? Quote
Andrea Deltacom Posted December 16, 2010 Author Report Posted December 16, 2010 ah ok i checked again and seems i didn't saved the changes. Without failover works good, I have written "For all codes error" so if a user don't reply in 10 seconds is a error? But why i can make a call against another twt number and it doesn't go in error? Quote
Vodia PBX Posted December 16, 2010 Report Posted December 16, 2010 Well if you dont really need the failover then definitevely turn it off. The definition for a failover timeout is that the PBX received a code >= 180 within the specified time. Then the PBX acts as if it has received a 408 Request Timeout error code. If the trunk sends an error code before that, then it depends on the error code what will happen. The critical case here is 486 Busy, because technically that means that the user is busy (not the network) and failover will not change that. But the way the 486 code is used in the real life is different, that is why the error code is a little bit special in the dropdown list. Quote
Andrea Deltacom Posted December 16, 2010 Author Report Posted December 16, 2010 thanks for your explanations, I used failover in old snom one version 4.0 (before goes free) and doesn't give me this problem. Quote
Vodia PBX Posted December 16, 2010 Report Posted December 16, 2010 Yea, actually we fixed a problem in the failover area and that would explain the changed behavior. But it should be "right" now. Quote
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