I need to be a little quicker on my replies :-) FS tells me "FreeSWITCH Version 1.0.trunk (16619M)". I installed it this past Saturday from SVN trunk as I recall. I can supply anything you need, but I am pretty new to FS and definately wet behind the ears. Thanks- Anthony Minessale wrote:
Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled.
On Feb 17, 2010 1:51 AM, "Doc" <ledoktre@meanie.us <mailto:ledoktre@meanie.us>> wrote:
Greetings one more time,
I just remembered. I also ran into one other issue in my testing.
When I route the incoming skype call to an extension (Sipura SPA 1001), it plays hold music on the callers side (good....), and when I pick up the extension, calling party drops, internal phone is left with a fast busy signal :
2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel [sofia/internal/sip:1001@10.24.72.12:5060 <http://sip:1001@10.24.72.12:5060>] has been answered 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use ptime 3 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 sofia/internal/sip:1001@10.24.72.12:5060 <http://sip:1001@10.24.72.12:5060> has no read codec.
First thing I notice is this unusual error about the ptime? and the one that hangs me up I think is the "sofia... has no read codec". I have been poking around my configuration - any suggestions?
Thanks,
Doc
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