-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: 11 March 2015 17:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID Names
Are the
From a softphone (x-lite) the caller id information comes through as
anonymous@anonymous.invalid
These are also valid calls - If I disable outbound CLID on my mobile and call
in -
this happens. However it works fine on calls where I send caller id
information.
Okay, just figured this
Hi,
In my dialplan I have the following line.
same = n,Set(CALLERID(name)=Support)
I am expecting this to always set the caller id name to 'Support' - however,
we are getting calls come in as Anonymous with the number as something like
unknown@unknown
We're using Cisco 7945 phones - I
Hello,
I am playing around with events in asterisk via asterisk manager - i've noticed
it doesnt seem to be emitting events to my connected client. Is there something
that I need to do to receive events?
Also output from 'manager show events'
voip*CLI manager show events
Events:
We run a variety of 5000, 6000, and 7000 series Soundstations running
Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these
registration issues.
Would you be willing to send the configuration from asterisk for this?
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Hello,
I'm having a problem with a few Polycom SoundStation 6000s. Everything works
fine, but they drop registration to asterisk after about maybe 30 minutes - the
phone does not re-try to register and if you try to dial out on the phone it
says URI Dialing is Disabled
Has anyone else had
Apparently this is a known problem past v8 firmware:
http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-
version-9/
I've done some more playing about and what I've noticed is that even when using
TCP SIP on the 8.x Firmware conferencing doesn’t work - making it use UDP
Next step is packet capture to see if there is a clue as to the cause of the
failure in the SIP signalling.
Right, I see the following when running SIP Debug. Looks to me like the phones
are expecting the server to do the conference mixing, which I guess it would do
in CallManager?
--- SIP
Possibly slightly off topic, has anyone ever had Cisco 79xx Series phones come
up with cannot complete conference errors when trying to conference two calls
together?
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We were using G722 - I thought similarly and tried a call with alaw. Same
problem occurred, any other ideas?
I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can
only do a single G729 channel, and if you require G729 for the second leg of a
conference, it will fail.
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