similar to: how does GOTO_ON_BLINDXFR work?

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 as if the user had not > dialled any digits. > > >

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:

On 12/20/2014 03:22 PM, sean darcy 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 >>> --

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

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

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", "1") in new stack > -- Executing [266 at internal:3]

Using Digium fax I've tried a simple dialplan: '8447' => 1. Answer() [pbx_config] 2. Set(CALLERID(num)=xxxyyyzzzz) [pbx_config] 3. Dial(DAHDI/g0/1bbbcccdddd,,G(send)) [pbx_config] [send] 4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config] 5. HangUp() But I doesn't work. It executes

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBridge:81 ;;exten => 82,hint,MeetMe:82 exten => 82,hint,ConfBridge:82 ;;exten

On Tue, Apr 26, 2016 at 11:07 AM, Administrator TOOTAI <admin at tootai.net> wrote: > Le 26/04/2016 17:23, Mamadou NGOM a ?crit : > >> Hello, >> >> >> Having installed DAHDI to be able to use the meetme() application , when >> I start the dahdi service it generates me the following error: >> >> -bash: /etc/init.d/dahdi: No such file or

I'm running asterisk 16 on Fedora 31. If I start asterisk as user asterisk, all goes well. But if I start asterisk from systemd: asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 sorcery_config_internal_load: Unable to load config file 'pjsip.conf' Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: config_options.c:710

We just finished an upgrade of our Asterisk system to an HA environment on a pair of servers using Linux-HA. As part of the upgrade, we also moved to Asterisk version 1.8.4.3 Most things are working quite nicely on the new system. However, I?m having trouble getting a paging feature to work. In Asterisk 1.4, we simply used the Page() application like this:

I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? sean

Hello! I need to use n-way call as it described here: http://habrahabr.ru/sandbox/52259/ It is in russian, but dial plan is quite clear. It works in asterisk 11: -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer) priority 1 -- Executing [0 at fromtransfer:1] NoOp("OOH323/7272-6385", "") in new stack -- Executing [0 at fromtransfer:1]

Receiving a fax with 1.6.1: == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [fax at incoming-pstn-line:1] NoOp("DAHDI/4-1", "Fax Detected") in new stack -- Executing [fax at incoming-pstn-line:2] Goto("DAHDI/4-1", "incoming-fax,s,1") in new stack -- Goto (incoming-fax,s,1) --

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

On 1.6.1-beta4: Trying to receive faxes over a pstn line. extensions.conf: [incoming-pstn-line] exten => fax,1,NoOp(Fax Detected) exten => fax,2,GoTo(incoming-fax,s,1) exten => fax,n,Hangup() ............ [incoming-fax] exten => s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM}) exten => s,2,ReceiveFAX(${FAXFILE}.tif) exten =>

Hi All, Running asterisk 11.25.3, the /proc/cpuinfo says Intel(R) Xeon(R) CPU X5675 @ 3.07GHz lsmod | grep dahdi gives dahdi_transcode 16384 1 wctc4xxp dahdi_voicebus 61440 2 wctdm24xxp,wcte12xp dahdi 225280 11 wctdm24xxp,wcfxo,wctdm,dahdi_transcode,oct612x dahdi_voicebus,wcb4xxp,wct1xxp,wct4xxp,wcte11xp,wcte12xp crc_ccitt 16384 2