similar to: Help needed - Zap Transfer Failing...

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hello, Hopefully I'm posting to the correct list, but if not, please shun me ;). I'm running Asterisk 1.4, with Festival 1.4.1. I've got a test extension setup, Festival configured and for some reason, when I dial that extension I get this: [Jan 10 17:16:05] WARNING[9082]: app_festival.c:511 festival_exec: Festival returned ER See the full debug below: [Jan 10 17:16:05]

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

All, I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all* of my incoming calls are coming up as FAXes. I had to disable my fax extension because every call to my POTS line was getting redirected to my FAX machine. After removing the FAX extension, if I call my POTS line from my cell phone, I get the following: *CLI> -- Starting simple switch on 'Zap/1-1'

I've got a new setup (different building) where Asterisk is sitting between the PBX and phone company on a E&M T1 line. Mitel PBX <-> Asterisk <-> Phone company Inbound works. Asterisk gets the in-band digits from the phone company and hands the call off to the Mitel just fine. Outbound is weird. Asterisk seems to expect that the mitel will send routing information

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

After sometime I got my E1 PRI configured correctly in /etc/zaptel.conf and I now don't see any alarms on the E1, but I can't still dialout correctly, I enable every debug that I could though of and this is what I see: At the end the D-channel is down and I cannot even try to connect another call because it tell me that the zap channel is unavailable ,the other this that I notice is that I

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hi, I'm starting testing 1.6.2 beta. CentOs 5.2 I found my first crash, first I have [Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql: Attempted to update column 'useragent' in table 'sip', but column does not exist! [Mar 20 20:30:41] ERROR[11201]: res_config_mysql.c:581 update_mysql: MySQL RealTime: Updating on column 'lastms', but

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Hi, Here is my first step (call it a proof of concept) in using the "hint" priority with dynamic values. Background - this works exten => 12345,hint,SIP/12345-1 To make this a little dynamic, I used a web page to return to me the value of the sip registration. In other words, http://www.somepage.com/test.html returns the following (without quotes): "SIP/12345-1" I

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

We're still having problems with DTMF detection on our X100P cards. Incoming callers that hold down the "1" button for too long are being connected to extension 11. One would think fat fingers were uncommon, but it happens to alot of people. I suspected this was related to our having to increase the txgain, but I tried turning it down with no effect. I also tried disabling the

Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug

I've been using spandsp and rxfax to receive faxes for a while now and over the past few days I've looked into the other side of things - txfax. I can't seem to get it working properly. I've included debug logs below of both the tx and rx side of things. I've tried three different servers, asterisk v1.0.6, 1.0.9 and CVS, FC2 and FC4, libtiff 3.5.7 and 3.7.1, TE405P and

I have found that fax detection is returning an error saying that no fax extension is present when I have defined one. The console returns this error: Aug 26 10:58:41 NOTICE[1112745536]: chan_zap.c:3989 zt_read: Fax detected, but no fax extension extensions.conf has: [default] exten => fax,1,Hangup exten => fax,2,Congestion exten => fax,102,Congestion exten => f,1,Hangup exten =>

I have a number of Zap/ extensions defined in a queue with ringall strategy. When this queue is called sometimes Asterisk seems to think that one of these channels is busy, while it is NOT. The following is shown on the console: --Called 44 -- Called 36 -- Called 41 -- Called 35 -- Called 38 -- Zap/44-1 is ringing -- Zap/36-1 is ringing -- Zap/41-1 is ringing

Hi, I compiled and am running the latest CVS but strange things are now happening.. it looks like asterisk is randomly declaring my calls to be fax calls, complaining and then sending the calls into a black hole... if I hangup the calls below (soft hangup) asterisk locks up and I have to kill the process. NOTICE[21526]: File chan_zap.c, Line 3520 (zt_read): Fax detected, but no fax

A caller to me was this afternoon detected as a fax machine: Jan 15 15:31:17 NOTICE[41997]: File chan_zap.c, Line 3564 (zt_read): Fax detected, but no fax extension ... and then redirected to voicemail. An extract from extensions.conf is attached below. Is there any way to stop * even considering an incoming call on a line as a fax call? Iain bell] include => mailboxes include