similar to: facing problems with TDM400P

This has got to be a stupid error I'm making... I have been experimenting with different hardware and software configurations before I decide what to use as a production platform. Up until just recently things were going well. But now it appears I'm unable to get access to my TDM400p from Asterisk. I know the TDM card works fine, used it in another machine where it performed flawlessly.

Hi there, I have a TDM400P (1 x fxs, 1 x fxo) which I'm attempting to run on linux 2.6 (gentoo), without much success at the moment. I have previously had it working on a 2.4 installation, but when I moved to a new box and installed a 2.6-based system, it failed to work. In both cases I'm using whatever (libpri, zaptel, asterisk) is checked out by default (I assume that means HEAD)

Have an intersting issue whem migrating a site from Zap on 1.3 to DAHDI on 1.4.. Nothing special about the hardware - older TDM400 card, 2 red modules fitted... Both channels work fine under 1.2/Zaptel. With 1.4/DAHDI both channels still work OK, but only for one line - the 2nd line causes it to refuse to dial-out no matter which port it's plugged into. The Lines are bog-standard BT

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

what is the most stable linux that we can build business on it, i mean the best linux a linux without problems . __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi, We've got a dedicated T1 with two trunks running into our ILECs selective router for 911. Split out of the T1 into two MF CAMA trunks on ILEC side. I'm trying to use asterisks e911 signaling, but I'm having trouble with the dial command. (== Everyone is busy/congested at this time (1:1/0/0)) I'm missing something and I'm thinking it has to do with the hookstate

Hi few weeks ago I read about redundancy (HA) of asterisk boxes using DNS, DUNDi, so I decided to give it a try. OS FreeBSD 6.1-RELEASE, asterisk 1.2.7.1 on one peer I get: lk110*CLI> dundi show peers EID Host Model AvgTime Status 00:11:43:3d:69:e6 195.28.109.37 (S) Symmetric Unavail OK (1 ms) 1 dundi peers [1 online, 0 offline, 0

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi all, my local channel seems to be not working properly. im doing this: exten=> s,1,Dial(Local/123@users,,Tt) some times it rings the phone at extension 123, and sometimes it doesn`t. When it doesnt, it actually displays a msg that it could not find that extension. [May 1 16:54:02] NOTICE[4658]: chan_local.c:563 local_alloc: No such extension/context 12129339038@users creating local

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796 at internal:1] Set("SIP/1001-0000000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796 at internal:2] Dial("SIP/1001-0000000b", "SIP/00919885497796 at sonetel")

Am getting netsock error like this when using IAX2, Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid = 4270) == Using SIP RTP CoS mark 5 -- Executing [2001 at Test:1] Dial("SIP/4090-00000005", "SIP/2001 at IAX2/IND-MAN,30") in new stack [Jun 23 06:31:36] NOTICE[4383][C-00000005]: chan_sip.c:29491 sip_request_call: Conflicting extension values