andrewgroup Posted June 22, 2009 Report Share Posted June 22, 2009 We are seeing these from a recently installed CS410 The call between sip:123-456-0000@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5931 packets have been received/sent We are seeing these on all three analog ports... My guess is the call is ended, but the PBX is not detected the hangup and timing out... Investigating this, the first attempt to reach the WEB interface failed, and in a moment or two we recieved the "System start" email from the PBX we just tried to reach. It's as if the HTTP stream forced the reset and this isn't the first time these two events seemed related.. Running 3.4.0.3194 (Linux) MSP 828_v2_03_01 Release 2.4.2 The FXS interface is a Arris Touchstone TM504G/NA cable DOCSIS 4 port FXS media Gateway And the installation tech had no details on CPC timing,,, This device is commonly used by Comcast and other Cable providers when selling dial-tone.. Does anyone have the details on the CPC settings for this device? Cheers, Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted June 22, 2009 Author Report Share Posted June 22, 2009 The FXS interface is a Arris Touchstone TM504G/NA cable DOCSIS 4 port FXS media Gateway The Cable Carrier in question has agreed to get me the answers to our questions regarding how the default settings of the FXS media gateway is provisioned from the switch. What are the specific questions to ask, so that we can program the CS410 for best performance? IE Disconnect Wink Timing? We'll Post what we learn for all to know.. Cheers Quote Link to comment Share on other sites More sharing options...
pbx support Posted June 23, 2009 Report Share Posted June 23, 2009 The call between sip:123-456-0000@localhost;user=phone and sip:anonymous@localhost;user=phone has been disconnected because of media timeout (120 seconds), 0/5931 packets have been received/sent The default CPC value is 400 ms. This works with most of the cases. Also, you can change value for "Detect polarity change" and try the problem goes away. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted June 23, 2009 Author Report Share Posted June 23, 2009 The above is in place.. we are awaiting detail from the cable company on all issues for the settings in the ARRIS 504G media gateway. Hopefully we can delay until we get this information... How do we get to the advanced PSTN interface settings? Quote Link to comment Share on other sites More sharing options...
pbx support Posted June 24, 2009 Report Share Posted June 24, 2009 The above is in place.. we are awaiting detail from the cable company on all issues for the settings in the ARRIS 504G media gateway. Hopefully we can delay until we get this information... How do we get to the advanced PSTN interface settings? The PSTN settings are in /etc/sipfxo.conf file. There aren't too many settings though. Quote Link to comment Share on other sites More sharing options...
andrewgroup Posted June 24, 2009 Author Report Share Posted June 24, 2009 The PSTN settings are in /etc/sipfxo.conf file. There aren't too many settings though. In the WIKI / Support site on the CS410 appliance a screen shows the frequency / cadences, is that screen still accessible? Quote Link to comment Share on other sites More sharing options...
pbx support Posted June 24, 2009 Report Share Posted June 24, 2009 In the WIKI / Support site on the CS410 appliance a screen shows the frequency / cadences, is that screen still accessible? I believe, those settings(and the screen shots) were part of 2.x and may be early 3.0.x version. We did not update he wiki pages. Quote Link to comment Share on other sites More sharing options...
sudo Posted November 21, 2012 Report Share Posted November 21, 2012 Im getting a similar report: The call between sip:5555551234@cust_domain:5060;user=phone and sip:900@cust_domain has been disconnected because of media timeout (3600 seconds), 1036/2178 packets have been received/sent I get a few of these everyday. "900@cust_domain" is the Auto Attendant. I should mention that this customer is using Polycoms (IP331's mostly and PnP supported with a pbxnsip license). Any suggestions on getting these alerts cleaned up? Is this a setting on the Auto Attendant? Maybe its not acknowledging the BYE signal? Is this a compatibility issues with the polycoms? (I have had RTP timeouts on all snom device domains as well, though not nearly as frequent). Should I lower the media timeout to something smaller, like 60 seconds? Thanks in advance. Sudo Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 21, 2012 Report Share Posted November 21, 2012 What version are you on? I remember there was an issue when the Polycom muted the call. Then the regular RTP traffic is suspended and the PBX receives only SID packets. I believe that was fixed years ago, thus my question what version you are running. Quote Link to comment Share on other sites More sharing options...
sudo Posted November 21, 2012 Report Share Posted November 21, 2012 What version are you on? I remember there was an issue when the Polycom muted the call. Then the regular RTP traffic is suspended and the PBX receives only SID packets. I believe that was fixed years ago, thus my question what version you are running. Version: 2011-4.5.0.1050 Coma Berenicids (CentOS64) Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 22, 2012 Report Share Posted November 22, 2012 In that version the mute problem is definitively fixed. Maybe you can pay attention how many seconds it takes before the call gets disconnected. Also, you can set the flag in the admin part of the PBX to send out an email with the SIP trace in the attachment, so that you get a better idea what is going on without the need to go to PCAP. Quote Link to comment Share on other sites More sharing options...
sudo Posted November 26, 2012 Report Share Posted November 26, 2012 Here is the sip trace: 2012/11/26 12:00:36 Tx: tcp:192.168.1.1:13886 (1017 bytes) INVITE sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com> Call-ID: 3236d169@pbx CSeq: 24571 INVITE Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Type: application/sdp Content-Length: 382 v=0 o=- 325608696 325608696 IN IP4 cust_ip_add s=- c=IN IP4 cust_ip_add t=0 0 m=audio 56626 RTP/AVP 0 8 9 18 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/11/26 12:00:36 Rx: tcp:192.168.1.1:13886 (446 bytes) SIP/2.0 100 Trying Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (512 bytes) SIP/2.0 180 Ringing Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Allow-Events: talk,hold,conference Accept-Language: en Require: 100rel RSeq: 8193 Content-Length: 0 2012/11/26 12:00:37 Tx: tcp:192.168.1.1:13886 (446 bytes) PRACK sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 Call-ID: 3236d169@pbx CSeq: 24572 PRACK Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> RAck: 8193 24571 INVITE Content-Length: 0 2012/11/26 12:00:37 Rx: tcp:192.168.1.1:13886 (441 bytes) SIP/2.0 200 OK Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-0bb1d0d9c3f074fe2fbe2e5ccd5a1fbd;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24572 PRACK Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Length: 0 2012/11/26 12:00:43 Rx: tcp:192.168.1.1:13886 (766 bytes) SIP/2.0 200 OK Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-2b12b9dccafb724fe8e2c614ebfa276a;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 CSeq: 24571 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Content-Type: application/sdp Content-Length: 193 v=0 o=- 1353881399 1353881399 IN IP4 10.1.1.1 s=Polycom IP Phone c=IN IP4 10.1.1.1 t=0 0 m=audio 2236 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 2012/11/26 12:00:43 Tx: tcp:192.168.1.1:13886 (417 bytes) ACK sip:200@10.1.1.1;transport=tcp SIP/2.0 Via: SIP/2.0/TCP cust_ip_add:5060;branch=z9hG4bK-a35f384c9690c01159e767ce6f4dcee1;rport From: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 To: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 Call-ID: 3236d169@pbx CSeq: 24571 ACK Max-Forwards: 70 Contact: <sip:200@cust_ip_add:5060;transport=tcp> Content-Length: 0 2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (864 bytes) INVITE sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 1 INVITE Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 205 v=0 o=- 1353881399 1353881400 IN IP4 10.1.1.1 s=Polycom IP Phone c=IN IP4 10.1.1.1 t=0 0 a=sendonly m=audio 2236 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 2012/11/26 12:01:35 Tx: tcp:192.168.1.1:13886 (843 bytes) SIP/2.0 200 Ok Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK58b41ef7F5AAB32;rport=13886;received=192.168.1.1 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 Call-ID: 3236d169@pbx CSeq: 1 INVITE Contact: <sip:200@cust_ip_add:5060;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Type: application/sdp Content-Length: 237 v=0 o=- 325608696 325608696 IN IP4 cust_ip_add s=- c=IN IP4 cust_ip_add t=0 0 m=audio 56626 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=recvonly 2012/11/26 12:01:35 Rx: tcp:192.168.1.1:13886 (557 bytes) ACK sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bKd2d05b4862B5D81B From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 1 ACK Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Max-Forwards: 70 Content-Length: 0 2012/11/26 12:01:38 Rx: tcp:192.168.1.1:13886 (568 bytes) REFER sip:200@cust_ip_add:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 CSeq: 2 REFER Call-ID: 3236d169@pbx Contact: <sip:200@10.1.1.1;transport=tcp> User-Agent: PolycomSoundPointIP-SPIP_601-UA/3.1.3.0439 Accept-Language: en Refer-To: sip:210@cust_domain.com:5060;user=phone Referred-By: <sip:200@cust_domain.com> Max-Forwards: 70 Content-Length: 0 2012/11/26 12:01:38 Tx: tcp:192.168.1.1:13886 (431 bytes) SIP/2.0 202 Accepted Via: SIP/2.0/TCP 10.1.1.1;branch=z9hG4bK7bb88c1e4E0E8A51;rport=13886;received=192.168.1.1 From: <sip:900@cust_domain.com>;tag=C901125A-23F0AE97 To: "John Smith" <sip:5554441234@cust_domain.com:5060;user=phone>;tag=1784396399 Call-ID: 3236d169@pbx CSeq: 2 REFER Contact: <sip:200@cust_ip_add:5060;transport=tcp> User-Agent: snomONE/2011-4.5.0.1050 Coma Berenicids Content-Length: 0 Any Thoughts? I think I remember reading somewhere that Polycoms have issues with the REFER method. Dont know if its the same for PRACK? Thanks again - Sudo Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 27, 2012 Report Share Posted November 27, 2012 Yes there were changes in the transfer area. The big question is who disconnects the call after a blind transfer. I guess we need to re-visit the topic with the Polycoms. Quote Link to comment Share on other sites More sharing options...
sudo Posted November 28, 2012 Report Share Posted November 28, 2012 So whats the fix here? Is there anyway to lower the media timeout? At least the call wont be hung up for 3600 seconds. Quote Link to comment Share on other sites More sharing options...
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