similar to: E1 not synchronized

Hi All I have problem with LIBMFCR2 for once Exchange I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2. my system: Asterisk CVS 1.1.X LibMFCR2 pre.005 and unicall-pre.005 this's my setting and my setting wanpipe1.conf

Can anybody recommend a good web interface for asterisk that actually works. I am looking for a web interface that can show how many callers are on the phone, should be able to transfer the calls and disconnect. I have tried using the flash operator but has been unsuccessful in making it work. thanks -----Original Message----- From: asterisk-users-admin@lists.digium.com

Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 "T2XXP

Hi I have an asterisk box with digium hardware connected to a Siemens EWSD version 15 using a crossed E1 cable. The asterisk is serving as a h323 media gateway. When i start asterisk, the Siemens gives me this alarm: REPORTES GENERADOS EN LA EWSD AES/V15SBOL/BOLCBK1V51327079/013 05-06-06 11:25:38 7773 3062/03728 HF.ARCHIVE-80040

Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this

Hi everybody, In an Asterisk 1.6.0.5, dahdi-linux-2.1.0.4, dahdi-tools-2.1.0.2, with a Digium B410P and an OpenVox A400E, I can't make "dahdi_scan" to recognize the OpenVox. This card was working correctly but suddenly stopped working and I cannot make it work again. Both ?lspci? and ?dahdi_hardware? detect it but ?dahdi_scan? not and I cannot use it. >lsppci: *0a:00.0

As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too).

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

hi, i'm having problems getting asterisk spliced between an E1 PRI (german Telco Arcor) and an Ericsson Business Phone 250 digital PBX. The Asterisk Server has a TE405P with it's port 1 connected to the E1 PRI provided by our telecommunications provider Arcor and port 2 connected to the E1 PRI of our Ericsson BP250. the setup before: Arcor TelCo PRI(E1)

Hi ! I'm trying to configurate my first asterisk system. My test plant has an E100P Card and a 4 FXS TDM 400P card. I've an E1 configured for 15 channels. Telco says that crc4 is configurated so my zaptel.conf is: loadzone = it defaultzone=it span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 unused=17-31 fxoks=32-35 zttool shows no alarms. I've also tried changing the timing

Hi all, We have been fighting with telco for a entire week. Today they came here with a LITE3000 to analyze what is going on. When I configure zaptel with no external clock, E1 gets aligned/synchronized with bit rate in 2048000 bps, both me and telco. span=4,0,0,ccs,hdb3,crc4 But when I configure span4 to get clock source from telco they become unsynchronized. TElco bit rate stays in

Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks & Regards, Vidura Senadeera, Network Engineer,

Hi everybody, I have a 1.6.0.5 asterisk system and I?m having many problems with my pstn cards. This system has a Digium B410P with 4 BRIs configured, and an OpenVox A400E. The issue is that it?s impossible to make the system recognize the A400E, despite the ?lspci? output is: 0a:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface 11:00.0 ISDN controller: Digium,

Hi all, Our asterisk PBX, randomly restarts all the channels of the E1 connection. It sends this message "There is no D-Channel, using channel 16 anyway".Then the asteisk recive (or it thinks it recives) yellow alarms at all the B-channels, after that it restart all the channels. When restarting the B-channels it cut all the conversations that is handling at that moment. Does anyone

Hello Im from Denmark and i've just got my Digium TE405P. But i have some problems when i connect it to my E1 connection (ISDN30). My telco delivered a alcatel box witch have a G.703 120ohm (DB9 with a serial to rj45) and a 75ohm coax connection. I've tried to connect using the 120ohm with rj45 and a ordinary utp cable. But it dosent seem to work. I've tried several zaptel.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello List, since some days i run into the problem that one span on a TE407P is not comming up correctly. With intense debug on that span i get: < [ 02 01 7f ] < Unnumbered frame: < SAPI: 00 C/R: 1 EA: 0 < TEI: 000 EA: 1 < M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] < 0 bytes of data

hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'.

Hi all, i have configured incoming voip traffic as follows: [voipin] exten => _X.,1,SetCallerID(033283077734) exten => _X.,2,Dial,Zap/g4/${EXTEN} exten => _X.,3,Hangup If the call going out the pri dials with an additional '0' before the dialed number. This is for caller number AND called number. But i can't see anything that says set a '0' more in front of the

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

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