similar to: how to configure E400P card?

asterisk-users Any help will be appreciated. This card did not connect with E1 line how to loop E400P card to test ? now I loop the card. span 1 ---span2 RJ45 pins 1--4 2--5 but show : When calling ,showing error: app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' Asterisk Ready. *CLI> -- Registered SIP '2002' at 192.168.139.59 port 3289 expires 120

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

Hi, I successfully installed 2 avm card in my asterisk box but I'm unable to make call. My capi.conf is: msn=0721111,07211115 incomingmsn=* controller=1,2 softdtmf=1 context=default echocancel=yes callgroup=1 devices=2,2 my capi info : Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. my extensions.conf : exten =>

we got this installation : WinSip(demo version) -> ser(radius accounting) -> asterisk(from sip to h323 channel) -> gsm gateway(with 32 sims in it) we configured winsip to make 28 calls like from 28 different sip accounts, to 28 different cellular phones numbers after the first ten : -- Executing Dial("SIP/5060-081925b0", "OH323/33xxxxxx@gsm.gateway.ip") in

Hi Asterisk-ians! Need all of your help. I am stuck with this issue for last few days. I have one X100P installed in my system. My Asterisk is registered with another Asterisk Server/SIP provider as client and the call is successfully received by my Asterisk server (named as starwars). Now, the extentions.conf file states, the incoming INTO * should go out to fxo as below: exten =>

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

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

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

Hi everybody. We have a problem with the configuration of the card, the cards work and we receive incoming calls but asterisk don't receive dnid. We have 5 servers with 1 E400P with the same problem and the telco told us that we need to configure the card to request it, how we can do this? Can you help me to solve the problem. Best regards, Carlos Fernández Puente carlos.fernandez@alisys.net

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type

I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi, I found E400P quad PRI card quite cheap (749USD): http://www.govarion.com/product_info.php?cPath=1&products_id=2&osCsid=68cdd6e3d08754 in comparison to te410p (approx 1500 USD ) http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TE410P Now newer generation with HW echo canceling emerged (te411p). I'm not sure in what things those two cards

Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi, i'm new to * I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; when i try to call outside i get: -- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024 -- Executing Dial("IAX2[pippo@pippo]/2", "Zap/g1/0123456") in new stack Jul 13 13:42:49

http://lists.digium.com/pipermail/asterisk-users/2004-October/065540.html I saw this post about the M(x) option for the Dial command, but I could not find a reply questions posed here. I am wanting to pass the Zap channel that the original call came from to my macro embedded in the Dial command. I've tried to add arguments to the macro by using the syntax M(x,arg1), and I always get the