similar to: google voice

Some calls I get from google voice, I just send myself an email about the call and want to hangup. But I can't seem to make gv know I've hung up. extensions.conf: same => n,GoToIf($["${CALLERID(num)}"="office"]?email) ................. same => n(email),System(/usr/local/bin/emailme........) same => n,Answer() ; also tried without this same =>

I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean

I've got a confbridge set up which works if dialed locally: -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing

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

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload => res_jabber.so noload => chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had no errors, but making outgoing calls from clients just kept ringing even though the other side

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

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

Hi, I have a piece of code originally written for s-plus - I am trying to run it in R now. The code was obtained from someone who is now not available to give any pointers and I am a beginner in R. Here is where it is getting stuck: > +names(good.motifs[,1]) Error in +names(good.motifs[, 1]) : Invalid argument to unary operator here is now names(good.motifs,1]) looks: >

Hi list, I have a big doubt!, I have some users with ejabberd and am using motif to make some calls to extensions, here works fine, the problem is when I want to send a message to another user on ejabberd and asterisk take this message as part him, like a sip message , the other user does not receive this message xmpp User A xmpp == Chat to == User B xmpp (not receive the message) look cli

I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1<called-num>@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean

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

On Fedora 29, nbdkit-devel at 1.10.4 is too old for any of the tests/Makefile.am actions to run, but this also means we don't create any tls pki/ or keys.psk files that we then want to reuse during qemu testing, as evidenced by 'make check' failures such as: FAIL: interop-qemu-nbd-tls-certs ================================ qemu-nbd: Unable to access credentials

I know you that GV won't respect CALLERID from motif, but is there a way have GV show Unknown on outgoing calls. I don't want to have people think my GV number is really my number. sean

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

Hi, I am somewhat new to R and RSPerl, but I think this particular problem has to do with RSPerl and so I am not sure if this is the right forum to ask for help. Nevertheless I am quite sure that many of you would have used RSPerl with R. My hardware platform is a Sun/Solaris V440 server running Solaris 9 operating system I use the gcc-3.4.0 compiler to compile R without any problems. My

This is slightly redundant with just trying nbd_set_tls(nbd, 2) then checking for failure; however, this function does not set errors and looks more similar to nbd_supports_uri. --- This is borderline enough that I figured I'd post it to check if we want it. generator/generator | 45 ++++++++++++++++++++++++++++++++++++++------- interop/interop.c | 4 ++++ lib/handle.c | 12

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear