similar to: uninstalling zaptel

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. When a call is incoming, I do a ztmonitor to check the rx and tx values, but nothing appears on screen. Also changed the pci slot where the board is. The

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI

Hi, Is it possible to create a clear zaptel channel which doesn't require to be picked up? The requirement of my client is to open a clear channel to a recorder which starts recording certain message. Currently the channel which is created by zaptel requires the other end to answer the call, and the recorded can't answer, so the channel get hung up after a certain number of rings. Zaptel

Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can anyone help me start to diagnose why a Sangoma A200 wouldn't dial out LD? Local calls are fine, incoming is fine, just no LD. Bell tech has been on site and plugged into lines with his test set and was able to dial LD just fine, so it's not a LEC issue. No errors in asterisk console, using zaptel 1.4.11 and sangoma drivers 3.2.6, asterisk 1.4.18 ________________________________

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Is there a way to ask asterisk from a shell script if its connection (SIP) is valid to another system. Lets say for example to cisco call manager? Thanks, Jerry

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

I suppose this is rather an informative e-mail than a question. However if people had similar experiences or could comment what the differences are in other countries or with business analog lines, it would be interesting. It took me a week until a BT engineer was sent to my home home, since BT tech support was unable to provide information about the problem. Problem: Calling party controls how

Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm

Hello List, our setup: Callcenter IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular providers on the xircom analog port, ~60 agents Debian 5.0.1 (Lenny) Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue segfault fix Zaptel 1.4.11 Debian Package My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a duration of over 4 hours. I am