Hi Gents, I have just upgraded to the latest trunk from 1.0.4Pre7 and found the Contact header in the 200Ok messages has changed from sip:mod_sofia@123.123.123.123:5060 to dialed_number@sip.evolutiontel.net. Have looked at Jira and in old emails from the list and could not find anything, was there a variable that needs to added to sofia config to make sure the Contact header has an IP address instead of a URI? Thanks, Jim
This behavior only takes place on <param name="manage-shared- appearance" value="true"/>, Which Is required in some cases for things like call pickup and barge when using TLS/TCP. Can you tell me why the contact with the hostname is giving you a problem? /b On May 28, 2009, at 11:40 PM, Jim Burke wrote:
Have looked at Jira and in old emails from the list and could not find anything, was there a variable that needs to added to sofia config to make sure the Contact header has an IP address instead of a URI?
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
Thanks Brian, will check it out. I am using FS as Voicemail behind Opensips. As we have 2 Opensips servers if FS responds with a Contact header with a URI value we cannot route the call back to the correct FS server and the call is eventually dropped. For some reason this occurs even though we have record_route set in Opensips. Regards, On Fri, May 29, 2009 at 3:25 PM, Brian West <brian@freeswitch.org> wrote:
This behavior only takes place on <param name="manage-shared-appearance" value="true"/>, Which Is required in some cases for things like call pickup and barge when using TLS/TCP. Can you tell me why the contact with the hostname is giving you a problem? /b On May 28, 2009, at 11:40 PM, Jim Burke wrote:
Have looked at Jira and in old emails from the list and could not find anything, was there a variable that needs to added to sofia config to make sure the Contact header has an IP address instead of a URI?
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
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try enabling the Path header too we fully support that On Fri, May 29, 2009 at 1:37 AM, Jim Burke <jim@evolutiontel.net> wrote:
Thanks Brian, will check it out.
I am using FS as Voicemail behind Opensips. As we have 2 Opensips servers if FS responds with a Contact header with a URI value we cannot route the call back to the correct FS server and the call is eventually dropped. For some reason this occurs even though we have record_route set in Opensips.
Regards,
On Fri, May 29, 2009 at 3:25 PM, Brian West <brian@freeswitch.org> wrote:
This behavior only takes place on <param name="manage-shared-appearance" value="true"/>, Which Is required in some cases for things like call pickup and barge when using TLS/TCP. Can you tell me why the contact with the hostname is giving you a problem? /b On May 28, 2009, at 11:40 PM, Jim Burke wrote:
Have looked at Jira and in old emails from the list and could not find anything, was there a variable that needs to added to sofia config to make sure the Contact header has an IP address instead of a URI?
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I would recommend not turning on the SLA option then... I had to add that in because when using TLS the phone would try to call the IP which would fail because the SSL cert wouldn't match and the poor phone would kill over, lock up and reboot sometimes :P GO POLYCOM! With that option not set on the profile the default/old behavior should return. /b On May 29, 2009, at 1:37 AM, Jim Burke wrote:
Thanks Brian, will check it out.
I am using FS as Voicemail behind Opensips. As we have 2 Opensips servers if FS responds with a Contact header with a URI value we cannot route the call back to the correct FS server and the call is eventually dropped. For some reason this occurs even though we have record_route set in Opensips.
Regards,
Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com
participants (3)
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Anthony Minessale
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Brian West
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Jim Burke