similar to: Asterisk 11.2.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.20.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.20.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 10.12.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.12.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Dear All, I have different Asterisk Servers most of them are version 1.8 - I have recently upgrade to Asterisk version 11 on 2 servers. I have Jabber ( chan_gtalk ) configured on Asterisk 1.8 version and it is working perfect within all 1.8 version servers. I have XMPP ( chan_motif ) configured on Asterisk 11 version and it is working with all 11 versions servers. When I try to call from

Dear All, I have different Asterisk Servers most of them are version 1.8 - I have recently upgrade to Asterisk version 11 on 2 servers. I have Jabber ( chan_gtalk ) configured on 1.8 version and it is working within all 1.8 version servers. I have XMPP ( chan_motif ) configured on 11 version and it is working with all 11 versions servers. When I try to call from version 11 ( usiing xmpp -

So starting up asterisk-11.5.1-2.fc19.x86_64.rpm I get: [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_mgcp.so': /usr/lib64/asterisk/modules/chan_mgcp.so: undefined symbol: ast_pktccops_gate_alloc [Sep 17 21:09:07] WARNING[8606]: loader.c:423 load_dynamic_module: Error loading module 'chan_iax2.so':

Hi all, For one of my inverstigations it looks like i'm back to "square one" I'm trying to accept an incoming xmpp call and forward it conditionally to a sip, isdn, or voicemail. No google is involved as i use a local xmpp server (ejabberd) I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but some suggested me to have a look at asterisk11,so i did... I

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

2015-06-08 13:09 GMT+02:00 Bastien Nocera <hadess at hadess.net>: > On Mon, 2015-06-08 at 10:13 +0200, Arnaud Quette wrote: > > -ATTRS{idVendor}=="06da", ENV{UPOWER_VENDOR}="Liebert" > > > > +ATTRS{idVendor}=="06da", ENV{UPOWER_VENDOR}="Phoenixtec Power Co., Ltd" > > I'm guessing this would require changes in NUT, so I

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

Does the channel chan_motif and res_xmpp still work? I heard that Google had blocked this technology. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150117/f148cad7/attachment.html>

Is it me or Google just blocked Asterisk's chan_motif? I get "violation of terms of service" audio message whenever I send a call. Philip -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130520/7f0148f3/attachment.htm>

I have configured support for ice in sip.conf, and made a connection with motif to jingle, but does not work for me [Jul 15 12:03:32] ERROR[21758]: chan_motif.c:1955 jingle_interpret_ice_udp_transport: Received ICE-UDP transport information on session '8b4hdffbt37vg' but ICE support not available -- Executing [s at xmpp-in:1] NoOp("Motif/allan-ce76", " llamada de

I have a multi-homed machine (quite a few IP addresses on one of the interfaces) For SIP I found that using externaddr in sip.conf would make it much more reliable with ICE and RTP using the correct IP Is there an equivalent setting for XMPP / motif.conf? I saw bindaddr in gtalk.conf but it doesn't appear to be mentioned in the source code for chan_motif

"Jeremiah Gowdy" <jgowdy@home.com> wrote: | I've recently switched my company from a Windows NT fileserver to | FreeBSD/Samba. It's working really great and I absolutely love Samba. The | only reason I'm asking about speed issues and tweaking, is I heard someone | say something about it that implied that there are tricks/tweaks you can do | to increase the speed of

On Mon, 2015-06-08 at 13:21 +0200, Arnaud Quette wrote: > > > 2015-06-08 13:09 GMT+02:00 Bastien Nocera <hadess at hadess.net>: > > On Mon, 2015-06-08 at 10:13 +0200, Arnaud Quette wrote: > > > -ATTRS{idVendor}=="06da", ENV{UPOWER_VENDOR}="Liebert" > > > > > > +ATTRS{idVendor}=="06da",

This question isn't directly Samba related, but Samba has helped me figure out part of the problem. When I CreateFile on an existing file like so: CreateFile(FileName, GENERIC_READ, FILE_SHARE_READ | FILE_SHARE_DELETE, NULL, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, NULL); Samba shows that I have the file opened with this type of lock DENY_NONE 0x20089 RDONLY When the file is locked in

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I