nathans Posted February 11, 2012 Report Share Posted February 11, 2012 Hi, I'm having a tough time getting a Vitelity trunk to work. It used to work but something happen and it no longer receives calls or makes them. Vitelity has checked it thousands of time but no luck. Tried both Sip registration and getaway. Now i'm in the point where calls out ring the phone but with only 1 way audio (snom to remote phone) Incoming calls do nothing. This is the log of an incoming one: 5] 2012/02/11 01:05:28: Identify trunk (IP address/port and domain match) 6 [6] 2012/02/11 01:05:28: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 INVITE Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Rx udp:64.2.142.15:5060: CANCEL sip:36144423xx@192.168.88.127:5060 SIP/2.0 Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060> Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Content-Length: 0 [5] 2012/02/11 01:05:36: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Contact: <sip:nsandler@75.149.181.125:5060;transport=udp> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [3] 2012/02/11 01:05:36: Via and source address are empty for SIP/2.0, cannot send reply The call should route to extension 70 (as it is correctly trying to) but nothing happens. Any ideas what "Via and source address are empty for SIP/2.0, cannot send reply" means? Thanks Nathan Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted February 13, 2012 Report Share Posted February 13, 2012 Hi, I'm having a tough time getting a Vitelity trunk to work. It used to work but something happen and it no longer receives calls or makes them. Vitelity has checked it thousands of time but no luck. Tried both Sip registration and getaway. Now i'm in the point where calls out ring the phone but with only 1 way audio (snom to remote phone) Incoming calls do nothing. This is the log of an incoming one: 5] 2012/02/11 01:05:28: Identify trunk (IP address/port and domain match) 6 [6] 2012/02/11 01:05:28: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 INVITE Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Rx udp:64.2.142.15:5060: CANCEL sip:36144423xx@192.168.88.127:5060 SIP/2.0 Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060> Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Content-Length: 0 [5] 2012/02/11 01:05:36: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [6] 2012/02/11 01:05:36: SIP Tx udp:64.2.142.15:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK6821a68f;rport=5060 From: "+305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as6f146510 To: <sip:36144423xx@192.168.88.127:5060>;tag=919722800a Call-ID: 5bf4290630ae7cd1583fc2da7809b23f@64.2.142.15 CSeq: 102 CANCEL Contact: <sip:nsandler@75.149.181.125:5060;transport=udp> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [3] 2012/02/11 01:05:36: Via and source address are empty for SIP/2.0, cannot send reply The call should route to extension 70 (as it is correctly trying to) but nothing happens. Any ideas what "Via and source address are empty for SIP/2.0, cannot send reply" means? Thanks Nathan We have many vitelity trunks without a problem. Sounds to me like a NAT/FIREWALL issue. make sure you have all ports forwarded and if you are using nat, you may need to adjust the routting in admin/ports. Also make sure you don't have a sip alg in the way. BTW what firewall/router are you using? Quote Link to comment Share on other sites More sharing options...
nathans Posted February 13, 2012 Author Report Share Posted February 13, 2012 Thanks Moishe for the answer. The snomone is indeed behind a firewall but it has all of the ports forwarded to it (kind of DMZ) We are using a Peplink router. For the NAT issue, we have the local/public IP in the port part of the configuration. Playing around with some settings in the trunk I managed to make outgoing calls. The incoming ones are still failing. Form the log it seems like the snomone is getting and recognizing the incoming call, even sending it to the right AA account but nothing happens: [5] 2012/02/12 22:39:19: Domain trunk vitelity@sxxxx.beta-brain.com sends call to 70 in domain snom.beta-brain.com [6] 2012/02/12 22:39:19: SIP Tx udp:64.2.142.xxx:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 64.2.142.15:5060;branch=z9hG4bK3f21e420;rport=5060 From: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.xx>;tag=as44e70cd8 To: <sip:361xxxxxxx@75.149.181.xxx:5060>;tag=288c11fec4 Call-ID: 50499c6c0b573e8b62cdae0201ab3d25@64.2.142.15 CSeq: 102 CANCEL Contact: <sip:nsandler@75.149.181.xxx:5060;transport=udp> User-Agent: snom-PBX/2011-4.3.0.5021 Content-Length: 0 [3] 2012/02/12 22:39:19: Via and source address are empty for SIP/2.0, cannot send reply [1] 2012/02/12 22:39:21: Timeout: Call 44 not found Any chance you can share with me your trunk definitions for any a Vitelity line? Also the LAG your are referring is in the router or in the snomone? Thanks a lot!! Nathan Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 13, 2012 Report Share Posted February 13, 2012 [3] 2012/02/12 22:39:19: Via and source address are empty for SIP/2.0, cannot send reply That would also concern me. Maybe there is a UDP fragmentation problem? If you have Wireshark on the PBX host, it would be easy to see if that is the case. Quote Link to comment Share on other sites More sharing options...
shopcomputer Posted February 13, 2012 Report Share Posted February 13, 2012 That would also concern me. Maybe there is a UDP fragmentation problem? If you have Wireshark on the PBX host, it would be easy to see if that is the case. See here http://www.peplink.com/index.php?view=faq&id=124&path=16 how to turn off sip alg in your router. With most routers I find a need to use the IP routing list in admin, settings, ports, sip. We try to avoid NAT whenever possible. also port forward 5060/5061 tcp and udp, your SnomOne's rtp port range udp, and vitelity's rtp port range udp. I beleive their default is 10000-20000. Quote Link to comment Share on other sites More sharing options...
nathans Posted August 8, 2012 Author Report Share Posted August 8, 2012 After a few months, we have yet to make this work. I'm seeing something strange on the logs that maybe a clue to somebody out there. Its seems like the incoming call is detected and recognized and it fails to be tranfer to the extension 70 which is the incoming send all extension. Have no clue why....the calling party juts hears silence for about 10 seconds and then gets a fast busy. Any ideas anyone? Thanks!! [8] 2012/08/08 11:58:02: Trunk: Check if the call to +1361xxxxxx comes from the cell phone +1305xxxxxxx [8] 2012/08/08 11:58:02: To is <sip:+1361xxxxxxx@75.149.181.121:5060;user=phone>, user 0, domain 1 [8] 2012/08/08 11:58:02: Send call to extension ERE returned 70 [5] 2012/08/08 11:58:02: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [5] 2012/08/08 11:58:04: SIP Tx udp:64.2.142.15:5060: INVITE sip:305xxxxxxx2@64.2.142.15 SIP/2.0 Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;rport From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15 CSeq: 30655 INVITE Max-Forwards: 70 Contact: <sip:nsandler@75.149.181.121:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1090 Epsilon Geminids Content-Type: application/sdp Content-Length: 314 v=0 o=- 1472081629 1472081629 IN IP4 75.149.181.121 s=- c=IN IP4 75.149.181.121 t=0 0 m=audio 60702 RTP/AVP 18 0 8 101 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2012/08/08 11:58:04: SIP Rx udp:64.2.142.15:5060: SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 75.149.181.121:5060;branch=z9hG4bK-14b4687fd863a2e50da8ec5032dfbeaa;received=75.149.181.121;rport=5060 From: <sip:361xxxxxxx@75.149.181.121:5060>;tag=15a98ce6fa To: "+1305xxxxxxx" <sip:305xxxxxxx@64.2.142.15>;tag=as1430a5bc Call-ID: 5bcde1726c80fb7f212923382857158a@64.2.142.15 CSeq: 30655 INVITE User-Agent: packetrino Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
pbx support Posted August 9, 2012 Report Share Posted August 9, 2012 [5] 2012/08/08 11:58:02: Domain trunk vitelity@snom.beta-brain.com sends call to 70 in domain snom.beta-brain.com [5] 2012/08/08 11:58:04: SIP Tx udp:64.2.142.15:5060: INVITE sip:305xxxxxxx2@64.2.142.15 SIP/2.0 What is 305xxxxxxx2@64.2.142.15? Is this an internal number or something else? If it is an internal number then either it should ring or answer (if it is some AA/ACD/Hunt etc). The response is sent by "User-Agent: packetrino". Does this ring any bell? Quote Link to comment Share on other sites More sharing options...
nathans Posted August 9, 2012 Author Report Share Posted August 9, 2012 305xxxxxx is the external number calling in to the trunk number (361xxxxxx) No idea what "packetrino" is Quote Link to comment Share on other sites More sharing options...
pbx support Posted August 9, 2012 Report Share Posted August 9, 2012 Ok, not very clear about the call flow here. This log that you attached is the outbound INVITE from PBX. Can you check if some call forwarding (or service flag) is set on the account 70. Maybe the call is being forwarded to somewhere. In any case, please enable complete logging (at level 9 for pretty much everything & SIP call logging, no need of REGISTER, SUB/NOT) and PM the log to me. Quote Link to comment Share on other sites More sharing options...
nathans Posted August 9, 2012 Author Report Share Posted August 9, 2012 Hi, I'm trying to PM but get an error: The member pbx support cannot receive any new messages Quote Link to comment Share on other sites More sharing options...
pbx support Posted August 9, 2012 Report Share Posted August 9, 2012 Please try now. Quote Link to comment Share on other sites More sharing options...
nathans Posted August 10, 2012 Author Report Share Posted August 10, 2012 perfect. Message sent. Thanks! Quote Link to comment Share on other sites More sharing options...
Vodia support Posted August 27, 2012 Report Share Posted August 27, 2012 For future Vitelity trunk customers you will have to add the IP address of sipXX.vitelity.net and outbound.vitelity.net in the "Explicitly list addresses for inbound traffic:" also set Secure: to yes , for inbound calls to work. To find the IP addr of these DNS do a ping on the cmd line. Quote Link to comment Share on other sites More sharing options...
Pablo Posted April 16, 2013 Report Share Posted April 16, 2013 For future Vitelity trunk customers you will have to add the IP address of sipXX.vitelity.net and outbound.vitelity.net in the "Explicitly list addresses for inbound traffic:" also set Secure: to yes , for inbound calls to work. To find the IP addr of these DNS do a ping on the cmd line. Does the "Explicitly list addresses for inbound traffic:" setting accept IP address blocks; b/c, according to Vitelity, Media & SIP signalling for calls may originate from within the following IP ranges: 64.2.142.0/24 66.241.96.0/24 66.241.99.0/24 66.241.111.0/24 Quote Link to comment Share on other sites More sharing options...
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