Summary:	ASTERISK-12354: No voice joining snom 190 through asterisk to cisco voice gateway occationally.
Reporter:	Kamil Czajko (kactus)	Labels:
Date Opened:	2008-07-10 03:48:42	Date Closed:	2011-06-07 14:08:21
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_sip/CodecHandling
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) gateway_debug_cleaned_final.zip ( 1) iax_cleaned.txt ( 2) pbx_debug_cleaned_final.txt ( 3) sip_broken_-_Cleaned.txt ( 4) sip_working_-_Cleaned.txt
Description:	Hello I have a asterisk 1.4.17 box which I have configured to troubleshoot case 12708 and while it resolves that issue, I have come across a concerning bug that has now occured 3 times in 1.5 days of testing. I have a snom 190 talking to asterisk via sip, talking to a cisco voice gateway via sip. Everything is in realtime talking via odbc and freetds to a mssql db. Calling out works fine most of the time however occationally (3 times) when I call, the call connects, but no voice traverses in either direction. I will attach two sip debugs as I was able to catch it in action, but please note that rtp debug and rtcp debug was not turned on until part way through on the broken one. Things I noticed: when the first "invite sip" occurs on the broken one, it does not pass media  or media atributes. Later on it adds ilbc to the supported codecs and sets mode to 30ms. ****** ADDITIONAL INFORMATION ****** Please note, I am aware that this may have been fixed in later versions, but at the moment I am just trying to find a version that works with all components that we need to replace a geriatric server that if it fell over we would not even be able to get components for any more. We have current issues with both SVN (issue 12508) and with asterisk 1.6 (issue 12708)and new versions of 1.4 (also 12508) If this bug was fixed in a later version of 1.4 please let me know so I can test that extensively. Thank you
Comments:	By: Joshua C. Colp (jcolp) 2008-07-10 18:09:59 I see what is going on here... there is no SDP in the INVITE, which is something we don't currently handle very well. By: Kamil Czajko (kactus) 2008-07-11 00:32:00 Hi File, Is that something the Snom is doing wrong or is that an issue with asterisk? I recall reading something about not sending SDP in the invite and requesting it later on with options was a valid implementation under the RFC but I can't find that link atm. Would removing the incompatible options (in this case ILBC which the snom does not support) from the allow list in the SIP configuration alleviate this or we will have issues occationally regardless of what we do when the phones invite without SDP? If this is just limited to the old snoms it won't be as much of an issue for us, the 190 is just my test phone atm. We primarily use linksys 9xx, with some snom 3xx. We have had too many nat issues with the cisco 7xxx phones to use anywhere other than on local networks. Is this something I should worry about using as a production system? By: Kamil Czajko (kactus) 2008-07-13 20:51:16 Hmm it appears that its not an issue with the snom talking sip to the asterisk box after all. I setup the office pbx (trixbox) to call out via the asterisk 1.4.17 as the default outbound trunk via IAX. We only have a skeleton crew on weekends so thought this would not effect too many people. Cisco 7960/7940 handsets -SIP-> trixbox 2.6 -IAX2-> Asterisk 1.4.17 -SIP-> Cisco call gateway -PRI-> The world. I'm going to upload a debug that shows 4 calls going through. The first works, the next three are to the same number. The call at 10:09 and 10:36 comes through as silence once it stops ringing. The call at just before 10:39 works without issue. We had one call to a different number provide the same behaviour but this was before I turned on sip and iax debugging. Just to clarify something that might throw you in the debug, the outbound number is prefixed with an 8 for all calls. This is including the workingOutboundNumber even though it doesn't show it (as after I had replaced all I realised it might be useful to see when it is working with the pre and post processed number. Let me know how you would like me to proceed or if there is anything else you would like me to test. I've since reverted to our old gateway box so I can do anything to this one. By: Kamil Czajko (kactus) 2008-07-15 06:03:53 Hello I've uploaded pbx and gateway debugs as requested by jsmith in the #asterisk-bugs channel. I had to zip the gateway file as at 1844 KB it was throwing database size errors on upload. The number that goes dead is DeadNum in the configs. pbx debugs are from our office trixbox. Gateway debugs are from the 1.4.17 box running as a gateway to the cisco isdn gateway. This was all done with Verbose at least 4, debug at least 4, sip debug, iax debug and rtp debug. Please let me know what you require further. If I have ommited something and you would like more around debug from around the call, I still have the original log files 714MB and 138MB respectively. I have had to reformat the debugs as the putty loggging seemed to have difficulty as it progressed and started substituting characters. Thank you By: Kamil Czajko (kactus) 2008-07-24 21:51:28 Hello everyone, Just to update you I have been in contact with our ISDN provider to replace the IAD they deliver the circuit over. This week the unit became unresponsive and was unable to process calls at all. It may have been faulty for an extend period of time which could cause intermitent issues. This may be responsible for the issue (obviously not the sip SDP issue, but possibly why it happened over IAX as well.) The IAD will be replaced today and I will swap over the office pbx to talk over this link over the weekend to see if it has resolved the problem. I'll post an update early next week and if it works successfuly for an extended period of time we can close this case. Thank you By: Kamil Czajko (kactus) 2008-07-24 23:29:38 Further update I've just been notified that the ISDN provider needs to ship the unit so it will not be replaced until next week. Thank you By: Kamil Czajko (kactus) 2008-08-06 02:31:07 Hello everyone, Have replaced the hardware and left it running for a while. We do not get the dead calls running IAX to asterisk 1.4.17 sip gateway. Feel free to close this case unless you want to keep it here for the sip sdp invite issue which occured above, though this may have been resolved in later versions. We have worked around that issue by simply forcing the codecs to what the end device supports. We were previously allowing alaw,ulaw,gsm,ilbc and I beleive since the snom 190 does not support ilbc that when it invited without advertising its available codecs this is where the problem stemed from. Removing ILBC from the list has meant that we do not get the dead air on the sip line. By: Jason Parker (jparker) 2008-08-13 15:03:22 Closing per reporter.