[Freeswitch-users] No SIP UPDATE will send

[Alexander Haugg]

Hi,

the problem is the same.

What can I do to send a SIP Update to a call leg. Manual or automatic after a dialplan-, or uuid_bridge command?
Just for info, the first leg is coming from a trunk and the second leg go back to the trunk...

Thanks a lot

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Alexander Haugg
Gesendet: Montag, 20. Juli 2015 09:15
An: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Betreff: [Freeswitch-users] No SIP UPDATE will send

Hi all,
I have the problem, that the freeSwitch don't send a SIP UPDATE to the PBX.

Scenario:
PBX (Siemens H3K) is connected via SIP trunking to the FreeSwitch.
If I have two Legs from the H3K to the Freeswitch I'm using the uuid_bridge command to connect this two legs (works fine), but the call legs will not updated via SIP UPDATE. The PBX have no information to update the displays on the telephones.

The target number (toHeader) was the routing number from the H3K to the freeswitch. Now I need a SIP UPDATE from FreeSwitch to PBX to actualize the toHeader with the new partner number (the partner number is the fromHeader of the other leg).

In the sofia profile I was try it with the parameter <param name="send-display-update" value="true"/> and without this parameter (in the source code the default is "true")

What can I do to force an UPDATE to the call legs?

Version of the FreeSwitch
FreeSWITCH Version 1.5.15b+git~20150121T080154Z~90ab1d16f5~64bit (git 90ab1d1 2015-01-21 08:01:54Z 64bit)

In the INVITE from the PBX the ALLOW header includes the UPDATE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE

Many thanks!

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20151014/fd59597e/attachment-0001.html