similar to: pbx.c:1836 ast_pbx_run: Channel 'Zap/1-1' WARNING

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hi I have reason to believe that I have errors in my configuration because when I make a call I can see the H323 call executed ok but not being processed by Zap. I am using R2 signaling ( which I know is incomplete but should I not see it when I debug Zap channel?). I think there is a problem with my Zapata and zaptel configs . I understand that R2 can work with R2 China and R2 Argentina.

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hi, I've got an x100P and I'm able to dial out and make phone calls with it ok but I just want to set it up to answer the phone and be a simple answering machine but it doesn't seem to want to answer the phone. I keep getting this: on the console when the phone rings: -- Starting simple switch on 'Zap/1-1' Nov 28 08:55:09 NOTICE[29298]: chan_zap.c:5458 ss_thread: Got

i am trying to make calls between two workstations using netmeeting and asterisk. i get the popup on both when i call the extensions 665 and 667 but when accept, i get this error *CLI> 0:18.190 H225 Caller:8112978 H225 Received connect PDU. 0:18.288 H245:810b388 H245 Read error: Bad file descriptor 0:18.318 H323 Cleaner H323

I get the following message when I dial 74#. Does anyone have any ideas on what might be going on? If I don't require numbers to be terminated with # everything works as expected, but you have to wait for the digit timeout, of course. MESSEGE: DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got something to jump out with ('#')! -- Invalid extension '#' in

Hi, I noticed that it takes around 5 sec before the phone hang up immediately if SIP response 486 "Busy Here" was received. How to change it so that it will hangup immediately. >From the asterisk CLI, I am seeing ocalhost*CLI> -- Executing Macro("SIP/6200-70bb", "oneline|SIP/6203") in new stack -- Executing

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:AsteriskA@10.0.0.121/100") in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Hi all, I am either going daft or not reading things right. I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I have followed the examples for the conf files to the letter. I can call the pots extensions OK from IAX clients, SIP clients and from the incoming X100P cards. But, if I pick up the handset to make a call all I get is the engaged tone and the following message.

asterisk-users-request@lists.digium.com is believed to have said: > >Hey Dan!! > >Give us a clue as to what hardware/setup & network provider you have there, >and we might be able to help :) > >Paul > Hello Paul, hello everybody! I have, too, an inbound call problem. I am using an ISDN Fritz Card PCI 2.00, together with chan_capi 3.5.x . As I call my number I get

Ok, at the bottom of my h323.conf file on my 1st server I have this: ; --------------------- [test] type=user host=209.237.227.185 context=termination-test incominglimit=10 accountcode=005 ; --------------------- Using an Asterisk at the other IP, I have this: exten => _1NXXNXXXXXX,1,Dial(H323/${EXTEN}@64.135.11.85,,o) This should send a call from the test-server to the IP of the 1st server;

Hi, On Friday we changed our Telco-Provider (from German Telekom to Mnet) and recieved new Numbers. I changed the extensions in extension conf to match the new numbers. But i always get: Jul 18 12:10:39 WARNING[245776]: pbx.c:1780 ast_pbx_run: Channel 'CAPI[contr1/89064934]/0' sent into invalid extension 's' in context 'default', but no invalid handler I only changed the

I have tried to set up app_rxfax to receive faxes over IP. I realise there are mixed stories about how reliable this is at the best of times, but at this point all I'm after is some guidance in interpreting the log below. What does "RTP: Received packet with bad UDP checksum" suggest? Here is the full log: -- Executing SetVar("SIP/0892130888-b27c",

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>