Re: [asterisk-users] Mag Gam

2013-07-21 Thread Mag Gam
http://dks.shrikrishnaent.com/jarhfjf/jkhiymwm.zkcupfeoseu





Mag Gam


7/21/2013 7:26:35 AM

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[asterisk-users] Chris ym

2013-07-21 Thread asterisk_pri_ss7
http://trigonometria.org/mtdn/ghph.ahdffxifcxwtqmbn

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[asterisk-users] Google Voice Calls Fail

2013-07-21 Thread Vladimir Mikhelson
Hi All:

Has anybody tackled the latest Google Voice issue where incoming and
outgoing calls for certain Google Voice accounts fail?

I have filed the bug report with details
https://issues.asterisk.org/jira/browse/ASTERISK-22176

For incoming calls Asterisk does not reply to the initial XML request
coming from Google Voice. Detailed comparison to a successful call
initiation shows the lack of the nick: structure in the failed request.

Outgoing calls connect intermittently, but no sound path gets established.

Any ideas?

Thank you,
Vladimir



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[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

2013-07-21 Thread Zoltán Fekete
Hi!

I have exactly the same problem on asterisk 1.8.22.0  and also on separate
11.2.1 when sending fax to PSTN.
Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone.
SpanDsp also works without any problem on my box.

As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater
was sent as maxBitRate. Without capital M.

Are you closer to the solution?
I have tryed almost anything and I don't understand why sends the
T38MaxBitRate:2400 parameter.

regards,

Blaxy

 On 06/03/2013 05:03 PM, Larry Moore wrote:
  Have you checked the installed version of firmware against the latest
  available from Cisco?
 Oh! I didn't guess to check. The firmware was not fresh, but upgrading
 doesn't help.
  Looking at your SIP information when your ITSP initiated a T.38
  session it did not indicate a maxmimum bitrate, it would appear your
  spa112 attempted to negotiate a connection at 2400bps.
 Whether there is a way to force my provider to indicate maximum bitrate?
  Do you have a sip debug session when you sent a fax from your Asterisk
  box to the PSTN, it would be interesting to see if it sends it as a
  t.38 or reverts to G711 audio.
 I have collect a set of debugs (with fresh SPA112 firmware) and actual
 config files:

 == spa112 — cmd ReceiveFax
 https://gist.github.com/anonymous/5701032

 == cmd SendFax — PSTN
 https://gist.github.com/anonymous/5701150

 == spa112 — PSTN
 https://gist.github.com/anonymous/5701207

 == sip.conf
 https://gist.github.com/anonymous/5701231

 == udptl.conf
 https://gist.github.com/anonymous/5701247


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