hosted Posted November 8, 2007 Report Share Posted November 8, 2007 I have a CS410 that after they transfer they transfer to another extension midstream conversation is just drops them to VM. any ideas? i just enabled the logging. Quote Link to comment Share on other sites More sharing options...
Kristan Posted November 9, 2007 Report Share Posted November 9, 2007 We've had this happen a couple of times on full 2.1 systems (only 2.1 thou.) Sequence goes; transfer call, talk to person, then person gets sent to VM and call to phone is killed. Is there some kind of timeout going on? We've never managed to catch it in the logs. Quote Link to comment Share on other sites More sharing options...
joeh Posted November 9, 2007 Report Share Posted November 9, 2007 We've had this happen a couple of times on full 2.1 systems (only 2.1 thou.) Sequence goes; transfer call, talk to person, then person gets sent to VM and call to phone is killed. Is there some kind of timeout going on? We've never managed to catch it in the logs. Just happened again today. Verbose logging was off though. Call came in, answered on a Polycom, transferred to me (Polycom) - I answered, talking away for 5 minutes, call cut off and the caller was transferred into my voicemail.. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 9, 2007 Report Share Posted November 9, 2007 Are you mixing UDP with TCP? Could be we found something there... Quote Link to comment Share on other sites More sharing options...
joeh Posted November 9, 2007 Report Share Posted November 9, 2007 Are you mixing UDP with TCP? Could be we found something there... All UDP as far as I'm aware. It also seems to happen in some cases when we enable call divert using the * codes. We dial the star-code, then after performing the divert we are notified of a message. When we log into voicemail, the message left is the system saying "you have enabled call divert to..." Which is odd. Quote Link to comment Share on other sites More sharing options...
hosted Posted November 9, 2007 Author Report Share Posted November 9, 2007 This is my scenerio exactly. Polycoms doing transfers. mine are all UDP. I turned logging on but it hasnt happened today. happened 3 times yesterday. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 9, 2007 Report Share Posted November 9, 2007 IF someone could catch the SIP traffic it would be extremly useful... Quote Link to comment Share on other sites More sharing options...
joeh Posted November 9, 2007 Report Share Posted November 9, 2007 IF someone could catch the SIP traffic it would be extremly useful... I'll see if I can get WireShark running for a couple of days to reproduce it. Quote Link to comment Share on other sites More sharing options...
hosted Posted November 14, 2007 Author Report Share Posted November 14, 2007 did you ever get a capture? Quote Link to comment Share on other sites More sharing options...
hosted Posted November 15, 2007 Author Report Share Posted November 15, 2007 I dont know if this has anything to do with it.. BUT. on this box i did 'df' and it said: Filesystem 1K-blocks Used Available Use% Mounted on /dev/mtdblock2 253952 237148 16804 94% / **BUT it was 100% full when i logged in I deleted a couple files** rebooted now its: Filesystem 1K-blocks Used Available Use% Mounted on /dev/mtdblock2 253952 56404 197548 23% / fsck didnt work for me and i did a du -a and it didnt show anything abnormal. So I assume the reboot fixed the file system errors. Dont know if it was just this box that im having an issue with or not. others look just fine. upgrading to 2.1.1 now. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted November 15, 2007 Report Share Posted November 15, 2007 **BUT it was 100% full when i logged in I deleted a couple files** Did you leave the logging on high level on? It is easy to forget to turn logging off after fixing a problem - then a time bomb is ticking. Quote Link to comment Share on other sites More sharing options...
hosted Posted November 22, 2007 Author Report Share Posted November 22, 2007 yea fixed. the new 2.1.2 is still having the same issue. Quote Link to comment Share on other sites More sharing options...
Kristan Posted November 29, 2007 Report Share Posted November 29, 2007 yea fixed. the new 2.1.2 is still having the same issue. Really? We're running on 2.1.2.2292 and haven't had this at all. And we definitely would have heard about it. Quote Link to comment Share on other sites More sharing options...
hosted Posted November 29, 2007 Author Report Share Posted November 29, 2007 OK I just installed a 14 day trial... Polycoms and what do you know! scenerio incoming call to a ring group. receptionist answers xfers to a Polycom phone. Polycom talks for 3 minutes and the PBX send back a BYE command. and the PBX voicemail for that user immediatly picks up. this is a bad bug. luck for me i reproducted it and got a wireshark capture (its a big log) http://www.nexsip.com/files/nexsip_wire.zip the BYE command is at location# 47975 Beware! Quote Link to comment Share on other sites More sharing options...
Kristan Posted December 3, 2007 Report Share Posted December 3, 2007 That's really strange - we've got a 50 user polycom system and not had a transfer to vm since running 2292 (fingers crossed this seems to be running smoothly). We also run it here on our office one and we've not seen it happen since upgrading to 2292. Are you using the out of the box provisioning for the polycoms and the 2.2 sip application on them? I found this didn't work too well and I needed to mangle the config files from the 2.2 firmware and the pbxnsip ones together. Drop me a PM if you want a copy of my configs. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 3, 2007 Report Share Posted December 3, 2007 Well, the transfer happens when the other call is not connected yet. We have a patch for that, but it is not part of a released version yet. In Wireshark, enter as filter: sip.Call-ID == "998aeed0-22a00782-71608e9b@192.168.31.111" to see the unconnected call, and sip.Call-ID == "9d5c5862@pbx" to see the call that sends the REFER. Quote Link to comment Share on other sites More sharing options...
hosted Posted December 3, 2007 Author Report Share Posted December 3, 2007 Kristian: I am using 1.6.4 version of firmware. Interesting that 2.2 doesnt have this issue. pbxnsip: The call actually does connect and you are talking for several minutes when it happens. sounds like i should upgrade to 2.2. Just have to disable the buddie watch. (its annoying to burn a key on a 3 button phone) Quote Link to comment Share on other sites More sharing options...
hosted Posted December 8, 2007 Author Report Share Posted December 8, 2007 UPDATE: so far no issues with Polycom firmware 2.2 Quote Link to comment Share on other sites More sharing options...
hosted Posted December 12, 2007 Author Report Share Posted December 12, 2007 Well one of my sites with Polycom 2.2 called and said it happened to them today. is there a beta i can run? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted December 13, 2007 Report Share Posted December 13, 2007 Is there a beta i can run? Well, there is a new release 2.1.3 image that should avoid the problem. Quote Link to comment Share on other sites More sharing options...
hosted Posted December 13, 2007 Author Report Share Posted December 13, 2007 whats the url for windows and cs410 image? Quote Link to comment Share on other sites More sharing options...
cfcs Posted December 13, 2007 Report Share Posted December 13, 2007 Well, there is a new release 2.1.3 image that should avoid the problem. I'm having a similar problem only with Polycom phones: My VM has been recording conversations between the reception and when an SCC client calls. It appears that at times my VM does not pick up and the caller’s call rings back to the front desk. It is at this time that the entire conversation is recorded on my VM. Quote Link to comment Share on other sites More sharing options...
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