similar to: E1 Card emulator?

Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/b80fa496/attachment.htm

Hello all, is anyone aware of any open source call accounting software for Asterisk? Something that can parse out Asterisk's call detail records and generate on-demand reports? Thanks, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060105/ea63b84e/attachment.htm

I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001

Hello everybody, i'm from Mexico, at the time i?m working on a production server with asterisk 1.2.25 + spandsp-0.0.4 + libmfcr2-0.0.3+libsupertone-0.0.2+libunicall-0.0.3 and zaptel-1.2.22. I installed this version of astunicall that i downloaded from http://www.moythreads.com/astunicall/ Everything works fine, i'm able to make outgoing calls and recive incoming calls with all ANI and

Hi! Im having some troubles trying to configure * as a bridge between a telco and a pbx with R2, the scenario is this: ------------ E1/R2 --------- E1/R2 ------------ | Telco |-------------| * |-------------| PBX | | (Telmex) | --------- | | ------------ ------------ I can receive calls from the telco

Dears, After weeks trying to contact support of my telecom about 'Seize Ack' because that is not returned, was a lock for make calls on my E1s. Now I receive back de Ack and get ready to make calls, but the technical support reports to me that my attempts to call do not send any digits to the oder site (telecom station). 8 seconds after start 'Unicall event Dialing' the line

Dial(UniCall/g1|300|) Where is the number you want to reach? I'd expect to see Dial(Unicall/g1/1234567890|300) To reach number 1234567890 - Mois=E9s Silva On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins <rogerberaldi at gmail.com> wrote: > Dears, > > After weeks trying to contact support of my telecom about 'Seize Ack' > because that is not returned, was a

hello, is R able to suppress tick labels (not tick marks)? i know there is a way around this with axes=F and then draw new axes, but it would be easier to suppress them in the first place. -- ------------------------------------------------ Sebastian Leuzinger Institute of Botany, University of Basel Sch??nbeinstr. 6 CH-4056 Basel ph 0041 (0) 61 2673511 fax 0041 (0) 61 2673504 email

Dear list, I need to setup asterisk to send and receibe fax. I just looking about SpanDSP, Hylafax/Iaxmodem, AsterFax,...etc. The asterisk box has Digium hardware, one TE420B and one TDM2402 (8 FXO ports). I just read the SpanDSP (txfax and rxfax) makes the system more unstable that Hylafax/Iaxmodem. And the Asterfax solution does dislike cause its licensing. The TE420B, is configured in E1

We've just had the problem where our DNS server went down, and * started to act "funny". Is the best solution to install a local DNS server on the * box, and have no other DNS servers ? - this is an internal app, no need for any external DNS resolution at all. Julian. ______________________________________________________________________ This email has been scanned by the

Dear all I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on is asterisk and it is working fine but i got this DTMF error on asterisk CLI what is it ?? -- Zap/36-1 is ringing -- Zap/36-1 answered SIP/5406-9fa59770 -- Channel 0/1, span 2 got hangup request, cause 31 [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to forward voice or

Hi: I want to use asterisk1.2 but I don't know which version of asterisk1.2 and zaptel1.2 is best.Please offer me one version of asterisk and zaptel and libpri.How about asterisk1.2.24 and zaptel1.2.20.1 and libpri1.2.5?And do they work togather well? Best regards. --------------------------------- Pinpoint customers who are looking for what you sell. -------------- next part

Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure

I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey

Hello every boy again I have some problems with modprobe. When I type "modprobe zaphfc", this error happens "FATAL: Module zaphfc not found." And when I tyoe "ztcfg -vv" this error happens: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected Someone can help me??? Thanks to all.

Hello all, as good? I am trying to use the package astunicall-1.2.21.0.1 with a Sangoma A104D card and 04 links E1 mfc/r2 in Brazil. The compilation occurred, normally and links is UP if I place in Loop and I obtain to effect called in Loop, but when I extend for the PSTN, links reports the following messages, without asterisk start calls: Dec 5 22:27:12 WARNING[3868]: chan_unicall.c:627

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a astunicall-1.4 setup with a te110p to a nortel pbx in Mexico. (Hate R2!). This is what I get when trying to call to * box using testcall: ./testcall Chan 31, class 'mfcr2', variant 'mx,20,4', end 0, caller 0, from '' to '' Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 31: Call control(9) MFC/R2

---------- Forwarded message ---------- From: Juan Carlos Huerta <juancarloshuerta at gmail.com> Date: 07-abr-2009 13:41 Subject: Re: [asterisk-r2] MFCR2 With Unicall - can unicall affect zaptel commands? To: asterisk-r2 at lists.digium.com Please wirte to asterisk-users at lists.digium.com to get help about this problem. Juan Carlos ~ Lo que no te mata te fortalece ~ On Tue, Apr 7,

Hi, I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. I tried with and without answeronpolarityswitch=yes but it didn't change anything at all. With callprogress=yes answer get never detected. With