Friedom-Tech Posted December 13, 2010 Report Share Posted December 13, 2010 Hi, i am having a very interesting issue but this is an urgent one. The client is running Windows System Version 4.2.0.3961 (Win32) and using all Polycom Phones. The clients receives a call and can talk regularly with the the caller however if they attempt to transfer the caller hears hold music and ringback until the second party picks up the phone but once the second user picks up the phone there is dead air on both ends! please see extract of the logfile below. SIP/2.0 200 OK Via: SIP/2.0/TCP 10.0.3.16:5060;branch=z9hG4bK-df21e7b7aba9f122f5119b08b53f9e9d;rport From: "Susan Currie" <sip:325@localhost>;tag=16259 To: "Doreen Agliata" <sip:319@localhost>;tag=7C41E843-809506C6 CSeq: 28452 INVITE Call-ID: c52aa411@pbx Contact: <sip:319@10.0.3.51;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Supported: 100rel,replaces User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.2.1.0054 Accept-Language: en-us,en;q=0.9 Content-Type: application/sdp Content-Length: 193 v=0 o=- 1292261604 1292261604 IN IP4 10.0.3.51 s=Polycom IP Phone c=IN IP4 10.0.3.51 t=0 0 m=audio 2226 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 [7] 2010/12/13 12:40:13: Call c52aa411@pbx: Clear last INVITE [7] 2010/12/13 12:40:13: SIP Tx tcp:10.0.3.51:51811: ACK sip:319@10.0.3.51;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.0.3.16:5060;branch=z9hG4bK-d174e72aa06f2b9b1b1f8aa27b7cb9d2;rport From: <sip:7328008008@localhost:5060;transport=udp;user=phone>;tag=16259 To: "Doreen Agliata" <sip:319@localhost>;tag=7C41E843-809506C6 Call-ID: c52aa411@pbx CSeq: 28452 ACK Max-Forwards: 70 Contact: <sip:319@10.0.3.16:5060;transport=tcp> Content-Length: 0 [8] 2010/12/13 12:40:14: DNS: Add NAPTR callcentric.com (ttl=60) [8] 2010/12/13 12:40:14: DNS: Add SRV _sips._tcp.callcentric.com (ttl=60) [8] 2010/12/13 12:40:14: DNS: Add SRV _sip._tcp.callcentric.com (ttl=60) [7] 2010/12/13 12:40:19: SIP Rx udp:10.5.4.50:5060: OPTIONS sip:metaswitch@10.128.1.202:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.5.4.50:5060;rport;branch=z9hG4bK-d6deca0e45ed19f515a24c1897b171d7-meta2-o.pins.net-1 Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event Max-Forwards: 70 Call-ID: BB5EF132@meta2-o.pins.net From: <sip:metaswitch@10.5.4.50:5060;transport=udp>;tag=meta2-o.pins.net+1+0+9a600b28 CSeq: 575142591 OPTIONS Organization: Supported: 100rel, resource-priority Content-Length: 0 Contact: <sip:metaswitch@10.5.4.50:5060;transport=udp> To: <sip:metaswitch@10.128.1.202> [7] 2010/12/13 12:40:19: SIP Tx udp:10.5.4.50:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.5.4.50:5060;rport=5060;branch=z9hG4bK-d6deca0e45ed19f515a24c1897b171d7-meta2-o.pins.net-1 From: <sip:metaswitch@10.5.4.50:5060;transport=udp>;tag=meta2-o.pins.net+1+0+9a600b28 To: <sip:metaswitch@10.128.1.202>;tag=14cc23a9cc Call-ID: BB5EF132@meta2-o.pins.net CSeq: 575142591 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [5] 2010/12/13 12:40:27: Did not receive ACK, disconnecting call 46506489-6cfdc1d4-e75f677b@10.0.3.45 [8] 2010/12/13 12:40:27: Hangup: Call 224 not found [7] 2010/12/13 12:40:35: SIP Rx tcp:10.0.3.34:57885: Quote Link to comment Share on other sites More sharing options...
mattlandis Posted December 13, 2010 Report Share Posted December 13, 2010 i'm answering quickly but I'm presuming: your using snom ONE. snom ONE does not facilitate sip REFER for non-snom UA's. pbxnsip does. Matt Quote Link to comment Share on other sites More sharing options...
mattlandis Posted December 13, 2010 Report Share Posted December 13, 2010 as a work around: snom ONE built in star code will transfer call i understand. put call on hold and dial the * code for transfer. the whole idea is that non snom devices that are allowed are meant for things like doorphone, cameras, etc. I'm not saying I like this but is how it currently works. If you have a suggestion here: http://snomone.ideascale.com/ Matt Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted December 15, 2010 Report Share Posted December 15, 2010 as a work around: snom ONE built in star code will transfer call i understand. put call on hold and dial the * code for transfer. the whole idea is that non snom devices that are allowed are meant for things like doorphone, cameras, etc. I'm not saying I like this but is how it currently works. If you have a suggestion here: http://snomone.ideascale.com/ Matt Hi I Just tried it on pnxnsip version 4.2.0.3961 (Win32) and i had the same bug on the polycom phone Quote Link to comment Share on other sites More sharing options...
Friedom-Tech Posted December 15, 2010 Author Report Share Posted December 15, 2010 Have not heard back? Quote Link to comment Share on other sites More sharing options...
pbx support Posted December 15, 2010 Report Share Posted December 15, 2010 Have not heard back? We were not able to reproduce it in a slightly later version (.3966). But there seems to be no change in that area. Only difference is that one of the phone used is snom & the other one is Polycom 331 (just had only 1 Poly) Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted December 16, 2010 Report Share Posted December 16, 2010 We were not able to reproduce it in a slightly later version (.3966). But there seems to be no change in that area. Only difference is that one of the phone used is snom & the other one is Polycom 331 (just had only 1 Poly) please post a link for (.3966). for windows 32 and Linux (CentOS 5 32) Thank You Quote Link to comment Share on other sites More sharing options...
pbx support Posted December 16, 2010 Report Share Posted December 16, 2010 please post a link for (.3966). for windows 32 and Linux (CentOS 5 32) Thank You Not sure if you wanted snomone or pbxnsip versions. But here they are anyways - snomone Win32 – http://pbxnsip.com/snomone/beta/win32/pbxctrl-2011-4.2.0.3966.exe CentOS32 - http://pbxnsip.com/snomone/beta/centos32/pbxctrl-centos5-2011-4.2.0.3966 pbxnsip Win32 – http://pbxnsip.com/protect/beta/win32/pbxctrl-4.2.0.3966.exe CentOS32 - http://pbxnsip.com/protect/beta/centos32/pbxctrl-centos5-4.2.0.3966 Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted January 10, 2011 Report Share Posted January 10, 2011 I upgraded to 4.2.0.3966 (Win32) and below in the problems that i have Call Transfer ( Polycom) when trying to transfer a call from ext 2088 to ext 2089 not blind and when ringing i press Transfer again before the other side will receive the call when no audio (this problem is only if you transfer between 2 polycom phones Quote Link to comment Share on other sites More sharing options...
mattlandis Posted January 10, 2011 Report Share Posted January 10, 2011 snom ONE or pbxnsip? snom ONE doesn't gracefully transfer non-snom devices. Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted January 10, 2011 Report Share Posted January 10, 2011 pbxnsip snom ONE or pbxnsip? snom ONE doesn't gracefully transfer non-snom devices. pbxnsip Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 11, 2011 Report Share Posted January 11, 2011 Is this a transfer with early media (the call not connected yet)? This is tricky in SIP. If you can get us a PCAP or a LOG with the SIP messages (attach it, please) we can find out what the problem is. Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted January 11, 2011 Report Share Posted January 11, 2011 Is this a transfer with early media (the call not connected yet)? This is tricky in SIP. If you can get us a PCAP or a LOG with the SIP messages (attach it, please) we can find out what the problem is. I sent you a email with the trace please advise Thank You Quote Link to comment Share on other sites More sharing options...
polycom2080 Posted January 16, 2011 Report Share Posted January 16, 2011 on new version 4.2.0.3973 the transfer will go through, but the sender will still get an busy signal as it wouldn't went through Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 16, 2011 Report Share Posted January 16, 2011 Well, thats a SIP problem. How do you successfully disconnect a call that is not connected yet? We could send 487, right now we are sending 486. Not sure if this would make a big difference. Quote Link to comment Share on other sites More sharing options...
YSJ3010 Posted January 17, 2011 Report Share Posted January 17, 2011 Well, thats a SIP problem. How do you successfully disconnect a call that is not connected yet? We could send 487, right now we are sending 486. Not sure if this would make a big difference. I checked 4.1.0.4026 (Win32) and you are send 486 and i looks like its the best we can do with the polycom for now Quote Link to comment Share on other sites More sharing options...
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