Hi, I have tried with a snom 300 and and all incoming call made from Bria or Jitsi are working so it's looks to be related to Avaya.
So that's mean that Avaya phone 4620 with latest firmware doesn't support direct RTP communication, how can I prove it ? Best Regards. 2012/4/30 Cyril Constantin <[email protected]> > Hi, > > Are you talking about the incorrect header checksum shown in the ACK ? > (see in attachment) because I have this for Jitsi and Bria. > > Best Regards > > 2012/4/27 Tony Graziano <[email protected]> > >> This may be a silly question, but what happens if you use a different >> hardphone other than the Avaya? >> >> The quick glances I gave at the pcap files said to me it was going to FS >> and it showed an invalid header from the UA. >> >> >> >> On Fri, Apr 27, 2012 at 11:34 AM, Cyril Constantin < >> [email protected]> wrote: >> >>> I have tried with 3CX phone and it's the same issue. >>> >>> >>> >>> The call is made from my softphones on my computer (3CX, Bria, Jitsi) >>> and always arrive on Avaya phone and ring it but below it's whats happens: >>> >>> -- When it doesn't work: >>> >>> - I answer it from Avaya Phone and in my headset connected to my >>> softphone I can hear the voicemail from sipxecs and RTP stream coming >>> from >>> Avaya mic in the mean time but no sound on Avaya speaker >>> >>> --When it works: >>> >>> - I answer it from Avaya Phone then I have both way audio. >>> >>> >>> Best Regards >>> >>> 2012/4/27 Cyril Constantin <[email protected]> >>> >>>> I have already only G711 Alaw / Ulaw but with video. >>>> >>>> I have deactivated the video and G711 ulaw, to keep only G711 alaw but >>>> it still the same. >>>> >>>> I'm facing the same issue with Bria latest release. >>>> >>>> Best Regards >>>> >>>> >>>> 2012/4/27 Tony Graziano <[email protected]> >>>> >>>>> turn off all other codecs except g711 in jitsi and see if it is any >>>>> different... >>>>> >>>>> >>>>> On Fri, Apr 27, 2012 at 9:16 AM, Cyril Constantin < >>>>> [email protected]> wrote: >>>>> >>>>>> HI Tony, >>>>>> >>>>>> There is no ICE or STUN activated, there is also no port range >>>>>> configuration on Jitsi, I'm in internal network. >>>>>> >>>>>> Regards >>>>>> >>>>>> 2012/4/27 Tony Graziano <[email protected]> >>>>>> >>>>>>> Have you enetered anything in jitsi to enable stun or ice? do you >>>>>>> have anything that defines rtp port range in jitsi? >>>>>>> >>>>>>> On Fri, Apr 27, 2012 at 6:04 AM, Cyril Constantin < >>>>>>> [email protected]> wrote: >>>>>>> >>>>>>>> Hi All, >>>>>>>> >>>>>>>> I have converted my Avaya 4621 phone form H.323 to SIP with latest >>>>>>>> firmware in order to migrate all my Avaya users to Sipxecs :) but I'm >>>>>>>> facing a strange issue that I couldn't understand, I'm able to >>>>>>>> register the >>>>>>>> 4621 phone to sipxecs without issue, when I'm making a call from Avaya >>>>>>>> to >>>>>>>> any other phone registered to sipxecs I have both way audio, but when I >>>>>>>> call 4621 phone from Bria or Jitsi or whatever I have both way audio >>>>>>>> once >>>>>>>> on three or two calls, the rest of the time it's one way audio, I can >>>>>>>> hear >>>>>>>> sound coming from Avaya on my softphone but there is no sound coming >>>>>>>> into >>>>>>>> Avaya phone from my softphone >>>>>>>> >>>>>>>> All my phone are directly connected to Sipxecs and they are on the >>>>>>>> same switch. >>>>>>>> >>>>>>>> I have tried to register my softphone and Avaya phone to Asterisk >>>>>>>> so see if the issue was related to Avaya phone but it works like a >>>>>>>> charm >>>>>>>> (maybe due to the fact that Asterisk do B2BUA) >>>>>>>> >>>>>>>> I couldn't explain why it doesn't work all the time but I suspect >>>>>>>> the fact that I had redundant server that I deleted some weeks ago >>>>>>>> maybe >>>>>>>> there is some residue of the previous configuration, there is currently >>>>>>>> only one primary server configured without HA. >>>>>>>> >>>>>>>> IP address of the computer where softphone are installed: >>>>>>>> 10.147.116.102 >>>>>>>> IP Address of the Avaya phone: 10.147.116.202 >>>>>>>> IP Address of Sipxecs: 10.147.113.221 >>>>>>>> >>>>>>>> I have attached traces from a call from softphone to 4621 working >>>>>>>> and not working, there is tcpdump from my computer, from sipxecs, >>>>>>>> merged.xml, sipXproxy.log in debug. >>>>>>>> >>>>>>>> There is also a screenshot from Wireshark where I have opened the >>>>>>>> both capture made from my computer, we can see the difference between >>>>>>>> both >>>>>>>> but I couldn't explain why. >>>>>>>> >>>>>>>> I don't think that it comes from my DNS like it's managed by >>>>>>>> Sipxecs automatically, I have attached the zone as well. >>>>>>>> >>>>>>>> Any help will be greatly appreciated. >>>>>>>> >>>>>>>> Best Regards. >>>>>>>> >>>>>>>> Cyril CONSTANTIN >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> sipx-users mailing list >>>>>>>> [email protected] >>>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>> Tony Graziano, Manager >>>>>>> Telephone: 434.984.8430 >>>>>>> sip: [email protected] >>>>>>> Fax: 434.465.6833 >>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>> Linked-In Profile: >>>>>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>>>>> Ask about our Internet Fax services! >>>>>>> ~~~~~~~~~~~~~~~~~~ >>>>>>> >>>>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>>>> Telephone: 434.984.8426 >>>>>>> sip: >>>>>>> [email protected].**net<[email protected]> >>>>>>> >>>>>>> Helpdesk Customers: >>>>>>> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> >>>>>>> Blog: http://blog.myitdepartment.net >>>>>>> >>>>>>> _______________________________________________ >>>>>>> sipx-users mailing list >>>>>>> [email protected] >>>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> sipx-users mailing list >>>>>> [email protected] >>>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> ~~~~~~~~~~~~~~~~~~ >>>>> Tony Graziano, Manager >>>>> Telephone: 434.984.8430 >>>>> sip: [email protected] >>>>> Fax: 434.465.6833 >>>>> ~~~~~~~~~~~~~~~~~~ >>>>> Linked-In Profile: >>>>> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >>>>> Ask about our Internet Fax services! >>>>> ~~~~~~~~~~~~~~~~~~ >>>>> >>>>> LAN/Telephony/Security and Control Systems Helpdesk: >>>>> Telephone: 434.984.8426 >>>>> sip: >>>>> [email protected].**net<[email protected]> >>>>> >>>>> Helpdesk Customers: >>>>> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> >>>>> Blog: http://blog.myitdepartment.net >>>>> >>>>> _______________________________________________ >>>>> sipx-users mailing list >>>>> [email protected] >>>>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>>>> >>>> >>>> >>> >>> _______________________________________________ >>> sipx-users mailing list >>> [email protected] >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >>> >> >> >> >> -- >> ~~~~~~~~~~~~~~~~~~ >> Tony Graziano, Manager >> Telephone: 434.984.8430 >> sip: [email protected] >> Fax: 434.465.6833 >> ~~~~~~~~~~~~~~~~~~ >> Linked-In Profile: >> http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 >> Ask about our Internet Fax services! >> ~~~~~~~~~~~~~~~~~~ >> >> LAN/Telephony/Security and Control Systems Helpdesk: >> Telephone: 434.984.8426 >> sip: [email protected].**net<[email protected]> >> >> Helpdesk Customers: >> http://myhelp.myitdepartment.**net<http://myhelp.myitdepartment.net> >> Blog: http://blog.myitdepartment.net >> >> _______________________________________________ >> sipx-users mailing list >> [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> > >
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