[Asterisk-Users] variable limit time on Dial

Hi, Is there a way to set a variable time on the execution of the Dial command in an AGI script? The L option uses fixed times for warnings. regards, -=Raul=- ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] phonejack and linejack in the same system

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am

[Asterisk-Users] fax autoanswer

-o-o-o-o-o-o- Raul Elizondo http://www.acuari.com FWD# 486533 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

RE: [Asterisk-Users] fax autoanswer

Hi, I am a little bit confused about how extensions.conf recognice a fax tone. For testing i got this in my home pbx: [opciones] exten = s,1,wait(1) exten = s,2,Background(new/opciones) exten = 1,1,Goto(explicacion,s,1) exten = 2,1,Goto(extensions,101,1) exten = 3,1,Goto(acuari,s,1) exten =

RE: [Asterisk-Users] fax autoanswer

Goryachev Sent: Monday, September 20, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] fax autoanswer On Tue, 2004-09-21 at 11:52, Raul Elizondo (wizardteam) wrote: Hi, I am a little bit confused about how extensions.conf recognice

[Asterisk-Users] pulsedial on fxo signalling

Hi, On 1.0.9, an old handset with pulse disk was working and now with 1.2.2 is not working. I used pulsedial=yes on the specific channel, but it seems to be only for fxs signalling. How do i enable it on v1.2.2? Regards -=Raul=- ___ --Bandwidth and

[Asterisk-Users] unable to use addpac-ap200 (sip | h323)

Hi, I am testing an AP200 from addpac i m trying to make it register with Asterisk. It manages 3 protocols (sip, h323 and mgcp). If i use sip, i keep getting this messages: May 2 00:15:21 NOTICE[30545]: chan_sip.c:7711 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for

[Asterisk-Users] RSA question

Hi, Acording with http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, both sides should have auth=rsa in their respective section at iax.conf. But i've found that if server includes this option, the client keeps saying No way to send secret to peer. My links with FWD and Iaxtel are working

[Asterisk-Users] trunk timing on 2.6.x

Two asterisk, one in a 2.4.x and another one in a 2.6.x are connection ok using IAX, before i upgraded one of them (the one with 2.6.x) Trunk was working ok. Since i upgraded the one that its now 2.6.x, i m getting this message: chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only

RE: [Asterisk-Users] trunk timing on 2.6.x

Unless something about your setup is causing data corruption of some kind, then I don't think the O/S or kernel has anything to do with this. It appears that one side is using CVS-HEAD, and sending trunktimestamps, and the other side is using stable, which doesn't understand them. So, you need

[Asterisk-Users] problems compiling zaptel on FC5

Hi, I just updated to FC5, and used zaptel-1.2.1 as it was with my last version, it show this error: CC [M] /usr/local/src/zaptel/wcusb.o /usr/local/src/zaptel/wcusb.c:1452: error: unknown field ‘owner’ specified in initializer /usr/local/src/zaptel/wcusb.c:1452: warning: initialization from

[Asterisk-Users] FC4 + ztdummy + timming + trunking

Hi, Using a FC4 system without a digium card to centralize a few other systems with digium cards makes sound to break when zaptel module is loaded. Even if ztdummy is present. If i unload zaptel module, it works fine, but when i try to do a call with trunking, asterisk does not even answer. On

[Asterisk-Users] fc4 + iax + trunking

Hi, I m using asterisk on a system without a digium hardware, and when i try to use trunk=yes for my iax2 links, i just get this message while debugging: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 6ms SCall: 16384 DCall: 16384

[Asterisk-Users] iax to iax link

Hi guys, I got 2 different linux boxex each one with a TDM22B (2 fxs and 2fxo). Bot linuxes are connected to the same dsl company with a dynamic ip and both are doing an ip_gre tunnel VPN. I could work with zapta.conf zapata.conf and extensions.conf locally on each linux. Each linux is

[Asterisk-Users] AGI dtmf problems (with x-lite)

Hi guys, May be this is a subject already disscused some where before, but i cant find a solution. By using x-lite, i can dial in menus, or even in voicemail during the process of any of both. But when i run my own AGI (using tcl), it does not detect DTMF when GET DATA function. I got no more

RE: [Asterisk-Users] AGI dtmf problems (with x-lite) (solved)

Hi Steven, IT is interesting you are even that far along with your AGI application when you haven't even figured out your mail client. Do you mean my email client? or my voicemail? Voicemail is working fine on digim extensions, i even changed the language to spanish (btw, there are lot of

[Asterisk-Users] R2MFC cards

Hi guys, I've been searching for a R2MFC E1 card that works with asterisk. As far as i could find, i got the Dialogic DTI/301SC card. Is there a way to make it work with asterisk or anyone else can recomend me another brand that actually works with asterisk? Regards -=Raul=-

[Asterisk-Users] sip does not bind all addreses

Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and

[Asterisk-Users] sip does not bind all addreses

Hi, My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next time i reload, it opens the specific ip, changing back to 0.0.0.0 and

[Asterisk-Users] IAX to IAX connect question

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it

RE: [Asterisk-Users] IAX to IAX connect question

Hi Benjamin, Thanks for answering, now i got some other questions. Acording to http://www.voip-info.org/wiki-Asterisk+iax+rsa+auth, and my understanding of peer and user, [FWD-service] in the sample you provide me should be type=peer as it will be the master/server, and [FWD-gw] should be

RE: [Asterisk-Users] IAX to IAX connect question

I think i got the solution for what i was planing to set. Here is a ontheway sample (not what i got but its about the same) Office iax.conf --- register = 123456:[EMAIL PROTECTED] jitterbuffer=no tos=lowdelay [iaxfwd] type=user context=fromiaxfwd auth=rsa inkeys=freeworlddialup

RE: [Asterisk-Users] IAX to IAX connect question

and the answer for my own question is: using notransfer=yes on the iax.conf contexts - Raul Elizondo FWD: 486533 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Raul Elizondo (wizardteam) Sent: Thursday, September 16, 2004 5:24 AM