If I call a number from the phone of my wife, I get this warning:
[Jun 15 20:50:18] WARNING: translate.c:206 framein: no samples for gsmtolin
(more time per seconds). I didn't found any help in Google with this message... Someone wrote about "turning off silence suppression", that it's already turned off...
I tried to change the settings for the users, allowing just ulaw and alaw, but it's the same...
Can someone say me what does this message mean and how can I suppress it?
Thanks Luca Bertoncello (firstname.lastname@example.org)
We are seeing the message " PRI Error on span 1: Received MDL/TEI management message, but configured for mode other than PTMP" on one of our Asterisk boxes on a PRI. A Google search turns up a number of hits for this error, but they are all for BRI not PRI.
I'm reasonably sure there are no recent updates or config changes.
Running Asterisk 1.4.37. We can update to 1.4.44 if it will help. At this time we cannot upgrade to anything later.
Does anyone know how I might troubleshoot this?
I've try to search Google about this without any chance. I want to know if it's possible to use a mobile phone application for redirect automatically incoming calls of a GSM phone connected to Wifi network to a Sip phone. I've try to use different mobile phones SIP clients without any success. No one of them can redirect calls automatically. I've got Android and BlackBerry phones. Thanks. Sil
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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.
Thank you, Vladimir
Given the recent announcement about Google slimming their support for public interconnection with XMPP, can anybody comment on where this leaves the XMPP support in Asterisk?
In particular, I notice many of the references to XMPP on the wiki link to https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
which seems to suggest that XMPP support and Google Talk support are one and the same.
Is the XMPP support only tuned for Google variation of XMPP/ICE/TURN, or is it supported for all open Jabber servers? I currently run 1.8 (before chan_motif) against ejabberd
Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender like Google), and am looking for suggestions based on experience with different calendar applications/servers available for Linux that you have integrated with Asterisk. If you can give a quick, simple list of what worked and what didn't I would be very grateful. Thank You, John
I just wanted to send out some information that will hopefully help others. I don't know, maybe I'm the only one that's been having problems with this. I've been pulling my hair out for a while wondering why Google would not send my incoming calls to my Asterisk box. The calls would just roll to voice mail and no packets ever reached Asterisk. This has happened on two separate Asterisk boxes and three different GoogleVoice numbers. I've been all through the GV web page settings but nothing I did changed anything. I figured it had to be something simple, and…