what does sip show peers say?
On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons mat...@jekabsons.lv wrote:
Thats my issue, i hope someone could suggest something:
Phone A - Phone B
== Using SIP RTP CoS mark 5
-- Executing [01@default:1] Dial(SIP/00-0076, SIP/01)
in new stack
== Using SIP RTP CoS mark 5
-- Called 01
-- SIP/01-0077 is ringing
-- SIP/01-0077 answered SIP/00-0076
-- Locally bridging SIP/00-0076 and SIP/01-0077
== Spawn extension (default, 01, 1) exited non-zero on
'SIP/00-0076'
Phone B - phone A
== Using SIP RTP CoS mark 5
-- Executing [00@default:1] Dial(SIP/01-0078, SIP/00)
in new stack
[Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [00@default:2] Hangup(SIP/01-0078, ) in new
stack
== Spawn extension (default, 00, 2) exited non-zero on
'SIP/01-0078'
--
--
Best regards
Matiss Jekabsons
Procerto Ltd.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
