similar to: Description for each sound files

Hello all, I am running Asterisk-1.4.5 on my Debian GNU/Linux Etch here and I want to send the voicemails as attachment to e-mails and delete the voicemails from my PBX once it has been sent. But, I don't have a running MTA here even on the PBX itself. I just want to send the e-mails to my GMail account from my PBX. Can I just use the mail or mailx command to send the e-mail and attach the

Hello all, I am planning to setup a MeetMe conference functionality on Asterisk-1.4.13without having a Zaptel card. All users will be calling through SIP only. AFAIK, the said application needs a timer which makes use of the ztdummy module. I have basically two (2) problems I am encountering here that [1] I can't load the ztdummy.ko module and [2] Asterisk don't run when running it

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Hello, I am trying to Bridge 2 E1 interfaces over a long distance link exactly the same way Redfone does. How can asterisk be configured to do that? Best regards Arinze Izukanne ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html -------------- next

Hello Since TDM cards are known for being particular when it comes to motherboards (PCI 2.2, etc.), I was wondering if there is a utility that can check that the Zaptel driver works OK and can tell if the TDM card is compatible? That way, if an FXO module is not reporting an incoming call, we'd know it's because of the Zaptel driver, and not something elsewhere. Are "dmesg",

Hi List; What the following mean: CONSOLE=Phone/phone0 CONSOLE=Console/dsp TRUNK=Zap/g2 I know SIP/John and Zap/1 but I do not know above (I do not know also the difference between Zap/2 and Zap/g2)? Any advise? Regards Bilal ____________________________________________________________________________________ Got a little couch potato? Check out fun summer activities for kids.

I understood that a timing device (ztdummy if no zaptel hardware is present) was not necessary anymore with linux kernel 2.6. When I enable iax2 trunking I get this warning chan_iax2.c:8908 build_user: Unable to support trunking on user 'xxxxxx' without zaptel timing The linux kernel is 2.6.22-14-386 Can I ignore this message, and is trunking working despite this warning? The ztdummy

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

i am have x100P clone, and install asterisk 1.4 and out call normaly and hangup in xlite to zap but call to asterisk for zap channel nop pass to xlite and the caller hangup the asterisk not detect. what is the problem ??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070720/39d4c1e8/attachment.htm

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't

And if that ain't confusing I don't know what would be. I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago and ended up never using it. Passed it along to a friend who is having some problems with it. (He isn't on this list.) We've both tried searches using Google but haven't been able to find anything that helps. So this is more a question of

Hi all, I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Also using something like this is not working as it still displays unknown number. I want set the callerid on the 1.call which is made. exten

I need help on my zaptel.conf and Zapata.conf for a TE207P I'd like Span 1 to receive a PRI from the phone company(US PRI). I'd like Span 2 to interface with a Nortel Phone system as a PRI(acting as the phone company) Essentially my asterisk box is a man in the middle intercepting calls from the PRI passing certain DID to the Nortel, also intercepting calls from the Nortel passing them

Hello, I am missing any description of zaptel/DAHDI alarms. The TE200 series user manual contains only a description of LEDs states. These alarms states are visible in zttool/dahditool or in astersick CLI (zap show status) and I wonder what is the real meaning of these alarms for E1 channel. Possible alarm states (based on zaptel.h 1.2): 1. No alarms 2. Recovering from alarm 3. In loopback

Hi everybody, I need to know about sound quailty issues from those who have experience with Tormenta2 PRI Interface. Also how to make it work with new versions of Asterisk and Zaptel. And also suggestion if it is a good idea to switch to some newer card from Sangoma or Digium, or Tormenta should work fine. I have sound breaking many times a day over the trunks. Server is AMD Athlon 2.4 GHz with

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Hi! I have Asterisk 1.6.1 installed on OpenSuSE 11.0 running with choppy sound. One specialist on the forums asked me if I have DAHDI configured, he assumed that this could be cause of choppy sound problem. > dahdi_test Unable to open dahdi interface: No such file or directory Do I need to configure DAHDI even if I do not have any Zaptel devices? Is there any guide for configuring

Dear all, Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, the memory in use starts off at 50% and continues to climb until it hits 99%. When memory usage ratio become 50% or more, the quality of calls become extremely noisy. The call quality goes back to being perfect once I reboot the machine, but I was to try and avoid having to reboot the machine every week. the following

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in

Any idea what causes the warning "Unable to open pseudo channel for timing... Sound may be choppy."? Any ideas what I need to resolve this? I do have the zaptel module installed but don't have a zaptel card. I'm guessing this has to do with ztdummy? I'm running Debian and installed asterisk, zaptel, and zaptel-source from the backports. Any information appreciated!