On Monday 19 Jan 2015, ricky gutierrez wrote:
Hi list, I write on the list looking for help, buy a openvox gw gsm
for four channels and I'm a little disappointed with the support
openvox, for some reason , The call doesn´t get trough
support tells me it was my asterisk server, but does not
I have a situation that I need help with. I have 2 phone systems, 1 Asterisk
and 1 Avaya. All voicemail is kept on the Avaya system. Whenever a call comes
into an extension that the Asterisk server owns, I re-direct it to a different
number that is owned by the Avaya System. If that Avaya
On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog sgriepent...@digium.com
wrote:
I would recommend capturing traffic outside your Asterisk server with
Wireshark, then running the Telephony/Rtp/Analysize Streams option to
determine if you have packet loss at that point in the network.
On
Next step is packet capture to see if there is a clue as to the cause of
the failure in the SIP signalling.
On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
We were using G722 - I thought similarly and tried a call with alaw. Same
problem occurred,
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
Apparently this is a known problem past v8 firmware:
http://kaa.kiev.ua/blog/asterisk-and-cisco-7945g-after-sip-firmware-update-version-9/
On Tue, Jan 20, 2015 at 11:16 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
Next step is packet capture to see if there is a clue as to
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.
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.
On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks
jordan.c...@gyron.net wrote:
Possibly slightly
Hello,
I am struggling with what seems a common unresolved problem, changing the
password from voicemailman when using a realtime engine (adaptive_odbc in
my case, connected to mysql).
I have seen messages dating back to 2007 with this problem and the last one
was bug 5168, reported as closed,
I'm using chan_motif with Asterisk 11. It still works. I actually
received an email from google yesterday that there had been no traffic on
my number lately so the number would be reclaimed. I had switched my
outgoing away from GV several months ago when they were supposed to
discontinue the
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