One of their guys said by email they tested on Git within the last few weeks and found calls were hanging up after 2m40s due to something about how we handle timers... I'm going to test that next. Were there any versions in Git recently that might have had problems?
There does appear to be an issue here. Calls of 4mins duration with no problems with enable-timer=false With enable-timer=true the call times out. The reason is they send an INVITE and we reply 400 Bad Session Description. http://pastebin.freeswitch.org/14957 shows those two packets. -Steve On 8 January 2011 15:56, Steven Ayre <steveayre@gmail.com> wrote:
<param name="extension" value="auto_to_user"/> was what did the trick though. Thanks! :)
There's a minor issue from it where the trunk shows unregistered on their website (they expect the Contact header to be username@ip") but incoming calls are still sent through fine so it doesn't actually matter at all. They can still see FS registered on their switch.
I've added the configuration to the SIP Provider Examples section of the Wiki.
One of their guys said by email they tested on Git within the last few weeks and found calls were hanging up after 2m40s due to something about how we handle timers... I'm going to test that next. Were there any versions in Git recently that might have had problems?
-Steve
On 8 January 2011 02:50, Brian West <brian@freeswitch.org> wrote:
Also if you set the extension to auto_to_user it'll do it also.
/b
On Jan 7, 2011, at 5:24 PM, Steven Ayre wrote:
I'm currently handling it with:
<param name="context" value="numbergroup"/>
and:
<context name="numbergroup"> <extension name="numbergroup"> <condition field="destination_number" expression="^USERNAME$"> <action application="transfer" data="${sip_to_user} XML default"/> </condition> </extension> </context>
Is there a better way of handling it? Such as a param that tells Sofia to use the To not the URI to populate destination_number?
-Steve
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