similar to: channel offset between Asterisk and PBX

Hi all, most of the time deleted messages aren't moved into trash folder but go into nirwana. Sometimes it works correctly. But a restart does not guarantee this function. This effect exists with different imap clients. Can anybody help please. Rainer Sigl -- ----------------- Rainer Sigl, | email: sigl at mpe.mpg.de MPI f. extraterrestrische Physik, | phone: +49

Hello, We have been trying to add asterisk to our Ascom Ascotel 2050 PBX. We have a AVM Fritz!PCI Card connected to an S0 bus extension from the PBX. The fritz card is configured to use chan_capi, and we can make calls SIP->SIP SIP->PBX extension PBX extension->SIP all successfully, we have assigned more than one PBX extension number to the S0 port in the Ascom PBX (it has 8 positions)

Hello all, I have some questions about the libpri and zap lib. I've sent mail to Mark for help. Also I wonder whether anyone else would be interested and helpful in them. I'm going to migrate a program to Wildcard E400P card. Our original program is based on C API similar to the functions provided in libpri.h and zap.h, so I want to use the E400P card in the following way: 1. install and

Hello ! I 've some questions how bridging of ISDN calls is done. Assume an asterisk system with a TE405 card equipped. (PRI1 - PRI4) An incoming ISDN call on PRI1 is transfered back to PRI3. Unless there is DTMF detection or other things involved, the bridging is done without Asterisk. Does this card have a some sort of cross connection ? Does the PCM leave the card ? Or is there some DMA

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

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, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

Hello! I have a special problem with a zaptel PRI card (E100P) with Asterisk STABLE CVS (zaptel and libpri also from stable branch). I use iaxComm softphone for outgoing calls (ulaw currently). Calls are going out through PRI. After about 5-6 calls, when the 7th call arrives, the voice volume on the called party's side is decreased, so they can't hear much (the caller's volume kept

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

I've been having trouble matching variable extensions on a zap channel (an E1 line). Doing it the extensions.conf way: [pri1] ; Match 8078078- calls include => m807nat include => m807mob include => m807oth [m807nat] exten => _80780782XXXXXXXXX,1,StripMSD(7) exten => _2XXXXXXXXX,1,SetVar,clidest=${EXTEN} exten => _2XXXXXXXXX,2,Goto(cli,s,1) [m807mob] exten =>

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Hello, I am looking to buy a large number of TE410P cards for a big international project. I have studied the documentation and zaptel.conf and as result I have the following questions: 1. How do you switch the card from ISDN PRI TE to NT? This means from being configured as User Equipment (the PABX) to be Network Equipment (the Exchange). 2. How do you configure the card for E1 or T1? 3. In

Hi, I am trying to use Asterisk cmd ZapRAS (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS), I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I try to use it, I obtain the following error: Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid = 7242) Verbosity was 0 and is now 44 -- Accepting call from '123456789' to

First of all sorry for the little offtopic post :-) For one of our customers i need to make a vpn between the Netherlands and Hungary. Over this vpn two * machines are gonna talk IAX and employees in Hungary are gonna use the Exchange server located in the Netherlands. So far no problem... The problem: I'm from in the Netherlands and don't understand the .hu websites :-( First question:

Hi guys, I've been trying to get my ISDN-10 line up for the past few days, but its been going up and down. I am using OpenVox D110P card on asterisk version 1.4.21. It seems to me like a cable problem. I tried using Ethernet straight cable (12, 45, 36, 78) and also a "straight" cable where the twisted pairs are on 12, 34, 56 and 78. The problem remains the same.