similar to: 11.4: motif can only handle one channel at a time?

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 =>

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'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

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:

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",

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'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

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

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

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

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > >

To deal with google dropping xmpp for voice, I've gotten a callcentric number. The cc number connects to asterisk, and all works fine. Then I set up the cc number as the gvoice forwarding number. If I'm on the gvoice site, I can make a call and it will ring my cc number and then the outside number. That also works fine. BUT, when an outside call comes into gvoice it forwards the call

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

anybody know the motif driver if the integration with google voice still work or not? What's the best way for the interop with google voice? Thanks. George wu -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141118/d0a6ed6d/attachment.html>

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

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_by=username ; I also tried

I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)}) ; PJSIP_HEADER(read,To) same=>n,.... But when a call comes in to the gv-voice context, "s" doesn't match the extension: res_pjsip_session.c:2991 new_invite: Call from

To connect to google voice with xmpp, I've had to turn on the "less secure apps" switch. > You recently changed your security settings so that your Google Account xxxxxxx at gmail.com is no longer protected by modern security standards. > > Please be aware that it is now easier for an attacker to break into your account. My xmpp.conf : type=client

Hi, I am very new to R and wanted to know if there is a package that, given very long nucleotide sequences, searches and identifies short (7-10nt) motifs.. I would like to look for enrichment of certain motifs in genomic sequences. I tried using MEME (not an R package, I know), but the online version only allows sequences up to MAX 60000 nucleotides, and that's too short for my needs..

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