similar to: ALL incoming Zap channel calls are getting picked up as FAX calls!

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

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

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

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 :-).

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

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

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 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'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

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

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

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

Hi all, I'm using asterisk CVS-08/14/03-22 on a box with a digium T1 connected to a channel bank and a digium E1 connected to the PSTN. I get occasional warnings from asterisk: WARNING[37909]: File chan_zap.c, Line 3197 (zt_read): zt_rec: Unknown error 500 This happens mosttimes in a loop like this: [netland_helpdesk] exten =>

Hello, let's assume we have: [$someshare] path = /some/path create mask = 0644 directory mask = 0755 Within /some/path there's directory named dir1 for example (/some/path/dir1). Is it possible to define alternative masks for that particular directory without presenting it as a different share. My second question - is it possible to define write list for /some/path/dir1 only. --

ehlo list, I''m willing to shape ppp users e.g. each pppX interface to get XXX kbits I classify traffic going to ppp+ interfaces like this: iptables -t mangle -A POSTROUTING -o ppp+ -j CLASSIFY --set-class 0002:0020 then i have a file which is executed when a ppp inteface is up and first argument parsed to that file ($1) is the ppp number (e.g. 35 for ppp35). it generally looks like

Hi, I have successfully compiled spandsp 0.0.2pre10 with * 1.05 which can accept inbound fax calls. However, all fax received are incomplete (the first 10% of an A4 page is fine, the remaining is either missing or garbled). I suspect this is due to 'training error' (see below) which, according to Steve Underwood's postings, cannot be resolved further. I wonder if it would help to

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Le 26 juil. 07 à 13:55, Georgi Alexandrov a écrit : > Vincent Dautremont wrote: > >> Hi, >> here I have another newcomer question :-) >> in the section 9.6.1 of this how too >> http://tldp.org/HOWTO/Adv-Routing-HOWTO/lartc.qdisc.filters.html >> we can read commands about filters : >> ------------------------------------------ >> >> # tc