similar to: Debugging DTMF Detection

Hi guys, I've been experiencing a very strange issue with my Digium Card TDM400 as of this week. It has two FXS and two FXO. The FXO modules (both of them) never goes on-hook after hanging up in Asterisk. It had worked perfectly well for over four years. I put an ammeter in series with the line and the card, and immediately after plugging the connector to the card, I got 26mA in the circuit

Hi, we upgraded an Asterisk 1.4 with mISDN to 1.6 with chan_dahdi. Due to problems with iax channel posted earlier, we wanted to switch back to 1.4 version. Server has 2 HBA cards, everything is running fine with 1.6, bri_cpe is recognized and the 7 euroISDN channels are running well, ingoing and outgoing. Now we installed 1.4.38 version and no more ISDN. In logs we found this: [2010-12-24

Hi Bruno, actually vicidial is working on top of asterisk, vicidial doesn't know what asterisk using in layer 2. SS7, ISDN stack, Unicall/mfcr2 is working with asterisk. vicidial uses asterisk application to deliver call center functionalities. Regards, Vidura. ================ Dear Bruno, I had the experience of using the Vcidial with the boards of Digivoice. It worked very well!

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

The Asterisk Development Team has announced the release of libpri 1.4.11.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.4 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are some of the issues resolved in this release:

The Asterisk Development Team has announced the release of libpri 1.4.11.4. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.4 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are some of the issues resolved in this release:

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

Hi, Recently, I made issues after upgrading libpri from 1.4.11.2 to 1.4.11.5. After coming back to 1.4.11.2, everything ran successfully. Unfortunately, I didn't have much time to spend to investigate but I hope I'll soon get an opportunity to try 1.4.11.5 again on another telco line. Basically, the symptoms were that pri show spans kept showing lines to be either down or in alarm. My

Hello, I have already installed and active the icecast2 server. But how to use libcurl with icecast. Already downloaded to the windows version and tried everything, but could not. Please help me. Thanks in advance for your time. Flavio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Thomas, We want to be seen on the "http://dir.xiph.org/" portal, and after some research we noticed that we would have to install libcurl. We are already broadcasting, but we are not included in the portal. Our link: http://170.254.33.29:8080/stream. Taking your attention, we want to know how we can have historical statistical data, with IP / local data, date, time and connection

Hi All Does anyone know about any tool that does to Asterisk what mod_jk does for JBoss/Tomcat: a load-balance/failover server that is constantly connected to Asterisk backend servers and is capable of identify loaded or down servers? Regards Ant?nio The?philo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Hi All, I have just latest branch of asterisk 1.8 and i didn't found dahdi command in CLI everything seem fine. am i missing something ? campbx2*CLI> dahdi <tab tab> No such command 'dahdi' (type 'core show help dahdi' for other possible commands) campbx2*CLI> root at campbx1:/etc/wanpipe# wanrouter hwprobe ------------------------------- | Wanpipe Hardware

I am changing a system from zap to DAHDI. I removed everything zap. when doing the command: sh -x /etc/init.d/dahdi start, I see initlog -q -c 'modprobe wct4xxp' sh: /sbin/ztcfg: No such file or directory FATAL: Error running install command for wct4xxp doing updatedb then, locate zap returns me /var/lib/asterisk/sounds/spy-zap.gsm /usr/share/groff/1.18.1.1/font/devps/zapfdr.pfa

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers

Dear list members, Even though I found extremely reasonable not sending any audio when a PROGRESS message is received on a PRI channel (isn't it an early-media session or one-way audio session?), nevertheless some Italian IVRs expect the user to select the proper option by sending DTMF. Now my asterisk box understands correctly the DTMFs on the caller SIP channel, but it doesn't forward

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

HI, I'm using asterisk-1.4.24, dahdi-linux-complete-2.4.0+2.4.0 and libpri-1.4.11.4 When dial, when 492131 answer, in console appear some error messages -- AGI Script Executing Application: (DIAL) Options: (DAHDI/g1/492131|60) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/492131 [Dec 4 11:15:59] WARNING[7669]: chan_dahdi.c:1776 dahdi_enable_ec: Unable to enable

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were