This is a problem that is completely stumping me, and my understanding of
Asterisk dialplans tells me this should never be a problem. Moreover, this
scenario works on Asterisk 1.4 but not 1.6.
We have a customer with several Aastra 6731 phones. They want incoming
calls from the PSTN to work and they also want to be able to call each
other "internally" on a special non-DID number (like extensions 311, 312,
313, etc).
In the dialplan, both the extensions for their DID and their internal
extensions use the same Dial() command. The only difference that I can see
is that we make changes to the CallerID Name field and do a little dance
with SIPAddHeader() to make the Aastra phones ring differently. This
doesn't appear to have any effect on Asterisk, but when the call is made,
the phone responds back with "SIP response 400 "Bad Request"".
Here's the two dialplans (private details redacted):
Internal calls:
exten => _312,1,Set(CALLERID(name)="Internal call")
exten => _312,n,SIPAddHeader(Alert-Info: info=<Bellcore-dr2>)
exten => _312,n,Dial(SIP/username2,20)
exten => _312,n,Voicemail(312,u)
exten => _312,n,Macro(handle-hangup)
Calls from the PSTN:
[Somecompany-IVR-day]
exten => s,1,Dial(SIP/username1&SIP/username2&SIP/username3,20)
exten => s,n,Goto(Somecompany-IVR-night,s,1)
The errors from Asterisk when internal calls are made:
-- Executing [311@somecompany:1] Set("SIP/username3-000001b0",
"CALLERID(name)="Internal call"") in new stack
-- Executing [311@somecompany2] SIPAddHeader("SIP/username3-000001b0",
"Alert-Info: info=<Bellcore-dr2>") in new stack
-- Executing [311@somecompany3] Dial("SIP/username3-000001b0",
"SIP/username1,20") in new stack
== Using SIP RTP CoS mark 5
-- Called username1
-- Got SIP response 400 "Bad Request" back from XX.XXX.XXX.X
-- SIP/username1-000001b1 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [311@somecompany4] VoiceMail("SIP/username3-000001b0",
"311,u") in new stack
--
_____________________________________________________________________
-- 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
