Yet another question!
On two different trunks, I can route calls to internal points if I use
the trunk auth/username as an alias, eg. T-W02171-001. When I try to
use the destination phone number (have tried both international and UK
versions) eg 03450950110 instead the call fails and I see
Wow. Yeah I
see it sending a load of invites and never coming back. I've tried it a
few more times (my other trunk works once I activate the route to by
thing) and in another trace the ITSP sends back an OPTIONS during the
AUTH handshake. I'll ask them why!--
Ewan McLean
02034684428
ser
Starting at frame 19, see the invite, it goes from the proxy, then to
the ITSP, it asks the be authorized, and acks the auth, then the
invite goes and the itsp never responds. period. The ITSP times out.
On Wed, Sep 28, 2011 at 7:21 PM, Ewan McLean
wrote:
>
> Doh attached.
>
> __
Hmm, too rush.
There's already a JIRA open for that:
http://track.sipfoundry.org/browse/XTRN-1072
-
MM
On Wed, Sep 28, 2011 at 21:02, Melcon Moraes wrote:
> Does anyone got issues with the new 320HD phones, regarding MWI?
>
> It seems that the SIP URI in SUBSCRIBE used by the phone contains t
HiI've picked
up a polycom soundpoint 550 as I was fed up with issues caused by Cisco
7960 (which I've had since I got it due to it's poor SIP
implementation). I'm impressed by the sound quality of the new 550 on
internal calls. However I'm having issues trying to make outbound
external calls
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No, that should be at the beginning of the trace. it is at
the end. It (the invite) is not being sent on port 5
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yeah, but the system answering the call is different than a
phone (UA). Ah... the ITSP is not sending to you on
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Then in this case I would check that to see if that makes a
difference. It should not.
Oh wait... hmmm, these
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The proxy is responding. The invite says "4499". If that is
a valid user alias can you dial it internally?
If
I'm not even sure if this is in the realm of possibility or my end users aren't
telling me the truth, but:
I got someone calling me saying that they received a voicemail today and called
the patient back. The patient was very confused on why she was getting a call
back, as she left the voicema
I have to make a correction. The external person could hear the
internal person but the internal person could not hear the external
person. Not sure it mattered if the internal person was the caller or
the callee.
Anyway, I've switched to a different SIP server from the ITSP and
hopefully th
I've been having lots of one-way audio issues recently and initially, I
saw some evidence from one sip trace that it might be caused by the
ITSP. I saw the following error in the trace:
X-Asterisk-HangupCause: Switching equipment congestion
X-Asterisk-HangupCauseCode: 42
That particular call f
Well, in 4.2.x and earlier wasn't there a problem with multiple
authorization headers? Maybe it was corrected with 4.0.5 and the newer
builds of 4.2.1 but that was relatively recent as compared with the version
of code being questioned here.
Time for an upgrade...
On Tue, Sep 27, 2011 at 10:30 P
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