similar to: help with DAHDI hangup on calling out.

I was running : asterisk 1.2.24 zaptel 1.2.21 libpri 1.2.6 I remove zaptel and compiled asterisk 1.4.22 libpri 1.4.7 dahdi 2.1.0 dahdi_cfg -vvv DAHDI Tools Version - 2.1.0 DAHDI Version: 2.1.0 Echo Canceller(s): Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel

Hi Listers, I'm running 1.4.25-rc1 on opensuse 11.0 with dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and snapdsp.0.0.2. Incoming calls work fine. Outgoing calls made directly (exten => s,1,Dial(DAHDI/G1) then number work fine. The problem I have is trying to let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r). If I use "m" (moh) the

[Dec 23 17:58:49] ERROR[3091]: chan_dahdi.c:8413 dahdi_pri_error: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX I am seeing the above error on DAHDI 2.1.0, asterisk 1.4.22 and libpri 1.4.7 I am using a TE120P card. I am also getting this VERY frequently: -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'DAHDI/0-1' Versus a normal hangup:

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

I am current running on a production system zaptel 1.4.12.1 libxpri 1.4.1 asterisk 1.4.25 The above configuration works. I tried to update to dahdi 2.2.0, libpri 1.4.7 and asterisk 1.4.25 This did not work. calls came in but not out. I dropped back to the initial configuration. Today, I dried only updating libpri on the original configuration. I removed /usr/lib/libpri* installed libpri 1.4.7,

Anyone using 5ess with success? I have a customer that wants to use it. Seems to WORK for a short time then there are errors on the D-channel and their side (nortel switch) shuts down the channel. They were saying something about a slip error. I dont know this term. I am using dahdi 2.1.0.4 and asterisk 1.4.22 and libpri 1.4.7 Any ideas? jerry

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes

Hi all, I am having trouble setting the signalling method for the B410P using DAHDI. Asterisk complains that it has never heard of 'bri_cpe' or 'bri_net' - but it doesn't mind having 'pri_cpe' etc. ERROR[4294]: chan_dahdi.c:11327 process_dahdi: Unknown signalling method 'bri_net' Dahdi - dahdi-linux-complete-2.1.0.4+2.1.0.2 Asterisk - 1.4.23.1 Libpri - 1.4.9

I am connecting 1.4.22 and dahdi 2.1.0.3+2.1.0.2 to a siemens hipath 300 and siemens hipath 4000. (2 channels to each switch) with a TE210p card setup as T1 with em_w. When the call is initiated to either switch the phone rings, when its answered then nothing... I hear no audio etc... After the timeout period the call is hung up. The phone switch 300 needs the T1 reset as the channel is not

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>

Hi, I've read that the new dahdi channel supporting B410P doesn't support PtmP when in NT mode. Does it support PtmP when in TE mode ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081209/f422b5d0/attachment.htm

Hey folks, I have a Digium TE205P working as a man in the middle: PRI line -------- Asterisk/TE205P -------- PBX The PBX is a Panasonic KX - TVP 100. Everything is working great except for one little issue. Asterisk isn't hanging up the PRI B channel when the PBX channel is hung up. I don't want to overload you with information but please ask if more is needed. I suspect I'm

-- Attempting call on DAHDI/1wwwwww for smvoice_callprogress at smvoice-dialout:1 (Retry 1) [Oct 16 14:36:42] WARNING[16408]: chan_dahdi.c:8132 dahdi_request: Unknown option 'w' in '1wwwwww' [Oct 16 14:36:43] WARNING[16408]: chan_dahdi.c:1481 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device) Does DAHDI not know about the W ??? I think zaptel used

I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I "rm /usr/lib/libpri*" then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do

I am running 1.4.22. I am doing a simple call into the dialplan and am getting a strange error: [Nov 3 08:32:27] NOTICE[8022]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user "404" <sip:404 at 192.168.1.8>;tag=547521CB-DB0D6130 This is the only line that prints on the console... Typically I get a few lines like: -- Executing [33 at smvoice-sip:1]

I googled about a bit without success, so... Is there a version matrix available? Something that would say: for kernel version w, you can run up to version x of Asterisk, DAHDI version y, and libpri version z? For example, I have a bunch of remote hosts running kernel 2.6.26, Asterisk 11.6.0, and DAHDI 2.7.0.1. We experience occasional Asterisk crashes, so I'd like to get as up to date

Hi All; I have an E1 card with two ports for ISDN PRI. Do I need to install DAHDI in addition to LIBPRI? For placing outside calls (outgoing) via the PRI, then in the extension.exe file, I will use the Dial function? But how can I determine that I need to use the PRI channels and not the analoge channels? Last point: how can I know that asterisk is containing libpri? In other words, how can I

Hi All, I must be doing something really stupid as I can't get DAHDi working in Asterisk. It is loaded and working in Linux fine. *CLI> module load chan_dahdi Unable to load module chan_dahdi Command 'module load chan_dahdi' failed. [2010-03-08 14:05:34 CST] WARNING[26926]: loader.c:393 load_dynamic_module: Error loading module 'chan_dahdi': libpri.so.1.4: cannot

Hi i am upgrading an asterisk server from 1.2.4 to 1.4. i've installed libpri 1.4 i've installed zaptel 1.4 I've installed the new version of misdn with the script of beronet. i use this configure script: ./configure --with-postgres=/usr/local/pgsql then: make menuselect [*] 1. cdr_csv [*] 2. cdr_custom [*] 3. cdr_manager XXX 4. cdr_odbc [*] 5. cdr_pgsql XXX 6. cdr_radius

Hy guys i am having so much hard time to setup asterisk on a virtual machine that i got , i just want to know if i really need to use Dahdi and libpri on a complete Digital PBX i just gonna use sip and iax. I will never use any kind of analog line on this machine. Wait for a feed back. Daniel Abreu.