similar to: Dial out DAHDI Channel?

Does anyone know what this error message means? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) I've upgraded to 1.6.0 with dahdi 2.0. For some reason my outbound dahdi calls are not going through. At some point, it starts to work, but I don't know what the trigger is. Out of the blue, outbound calls start to work. I had been using asterisk-1.6-beta9 with zaptel

Hi everyone, our setup includes a NEC PBX connected to our asterisk via bri lines. The NEC has a doorphone feature, which is just an extension that calls you when someone rings. When connected to this extensions, a 'flash' signalling opens the door. So now, i'd like to trigger this from asterisk, too, but unfortunately wasn't able to do so. Setup: asterisk

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Dear all, i have a problem with a system running - Ubuntu 10.04 ( all updates done ) - ii asterisk 1:1.8.5.0-1digium1~lucid Open Source Private Branch Exchange (PBX) - ii asterisk-dahdi 1:1.8.5.0-1digium1~lucid DAHDI devices support for the Asterisk PBX I also use freepbx 2.9 for the configuration. Hardware is a Dell R410 and a Digium

AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: --

Under 1.4.35 I get this message printed MANY times [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000 (g722)(4096) [May 3 21:41:21] WARNING[21567] chan_sip.c: Asked to transmit frame type 4, while native formats is 0x1000 (g722)(4096) read/write = 0x1000 (g722)(4096)/0x1000

I currently have the following outbound-local config in my setup.... I can call SOME of the numbers (like 337xxxx, and 998xxxx, and 323xxxx).. but when I try to dial say like 601xxxx I get a 404.. any thoughts, I can't see any difference in the config. Also, I seem to be able to dial any number that starts with a 9.. such as 977, 990, 903.. [outbound-local] ;exten =>

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]

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not know about the W ??? I think zaptel used

Dear, I have this scenario: - PBX Asterisk 1.6.2.10 with private IP 192.168.0.10 - Behind a Cisco ASA firewall that connects to Internet - SIP trunk to Net2Phone with these parameters (nat=no): host=200.58.113.60 username=DOLLY secret=123456 port=5060 type=peer dtmfmode=rfc2833 disallow=all allow=alaw&ulaw nat=no canreinvite=no qualify=yes -Softphones Xlite The PBX can't register to

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Hi All, I have searched the various post and not able to find the solution. I am using Asterisk 1.4.21.2 for outgoing calls. Earlier i used ZAP trunk and it works fine. Now i need to move to SIP trunk and configured the same. When i try from softphone i got error as "Call rejected" and in the asterisk i got error as

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try. 2015-04-07 20:33 GMT-03:00 Alex Villac??s Lasso <a_villacis at palosanto.com>: > El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > > I am trying to collect enough information about an

hello list i have an issue related to outbound calls i can contact all the number except on number given by our provider in trunk the issue just when i configure my trunk in our server but when i configure the trunk directly in x-lite i can contact this number without issue below the cli == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0149xxxxxx at

Greetings *`s, I am manually creating call files and dropping them into /var/spool/asterisk/outgoing to be picked up by *. Presently, when I use local/internal parameters using SIP it works..ie I make an internal call from device to device. However, when I try dial an outside number which I have set up in a custom conf file, it bombs out with the following message :

Hi people, I have a problem with some extensions. The calls are ended after 31/35 seconds, also, it depends on the number which I call. This is the log, but I've not been able to find something wrong... Any ideas? [Jun 11 15:50:46] DEBUG[26071] app_macro.c: Executed application: ExecIf [Jun 11 15:50:46] VERBOSE[26071] logger.c: -- Executing [s at macro-dialout-trunk:16]

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