similar to: Busy problem

Hello, I have a problem with Asterisk, on 10% of incoming calls the IP Phone ring but I don't hear the caller and the caller doesn't hear me (all IP Phones have the same problem). This problem appear also if the call is directly send to the second E1 of the digium card who is connected to an IVR. It does not depand on the charge of the server (I have the problem with only one call).

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

hello list i have installed 2 diguim card in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

getting some errors on my test system. this is 1.4 (Asterisk SVN-branch-1.4-r214194) with a 4 port T412p card. Three of the ports are connected: Span 1 to the PSTN on a 10 channel bearer line, ports 2 and 3 are cross-overed (!) to each other. Port 4 is not plugged in. This has been working fine for several months. I updated a few days ago to the latest 1.4 branch. However, now I cannot dial into

Ok Guys, I've done a tonne of hunting around on this problem, but can't find much help. I'm running: asterisk 1.4.19.1 libpri 1.4.3 and zaptel 1.4.9.2 which I believe has been modified by RedFone to add the ztd-ethmf module. My interface is a RedFone foneBridge2 4 Span; and I'm connecting to a BT E1 PRI / ISDN30e with 15 lines on span 1, and a legacy Panasonic PBX on span 4. Upon

hello list, i have installed 2 diguim cards in my server using asterisk 1.4 (i use the old version with zapata.conf and zaptel.conf) i want to use the span 1 for group 1 and span 2-span 6 for the group 2 (i want to active the round-robin for span 2 and 6) in order to activate the WIMAX and FH please see the configuration below and tell me if there is anything wrong question 2: what is

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to

Hi below are my configs: pstn(e1)--->asterisk (span1)----->legacy pbx(connected via span2)-----> legacy pbx analog extensions. my dial plan is like callers dial into asterisk(span1) , hear an IVR option and they are connected to the agents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk

Hi! I have the chance to play with a couple of E400P cards, each installed in a IBM e330 XSeries servers (2 x 1GHz P-III CPU 2 Gb RAM, 36Gb SCSI HDD with RH8.0 2.4.18-smp kernel), and I'm trying to test/benchmark this e330/E400P combo generating calls thru /var/spool/asterisk/outgoing One e400P if doing the carrier work making calls and the other just receives the calls: Server#1

I have a two port TE205P Digium card. I have set everything up to create a native zap bridge between the two spans. Everything works perfectly except one thing. Our telco has a "password" that has to be entered as soon as a long distance call is made. So if I dial a long distance call from my meridian system, asterisk bridges the call between two channels, my telco picks up and gives

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Hi, How would one activate/deactivate hardware echo cancellation on the TE406 card? Can it be done per channel? I'm going to run TE406 in the following scenario: ISDN -> TE406 -> PABX I understand from Steve Underwood's site that echo cancellation is not good for faxes (and they do that themselves). So what I want to do and bypass echo cancellation for selected extensions before

Hello Team I have installed the new DL580 and used the new TE420B to add capacity on our ivr. Before I put new E1?s I decided to first move the old e1 from the old system to this new one but it has errors which not only affect the audio quality, but also cause the asterisk to refuse any call after sometime even though the channels seems up and active {seems d-channel fails}.. When processing

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78

Hi all, I still haven't seen my first post (cry for help) so I'm not sure if these are going through or not ... I've played with the box all day today ... I got it to quit seg faulting finally by playing with the config files. Currently I have 4 spans configured for a Digium TE420B (With Octastic DSP) Here is my zaptel.conf: (I know the device connected to spans 1-3 are providing

Hi, I'm having problems with a TE420P card, in which I cannot make calls using spans 2 through 4. After a couple of days of working correctly, spans 2, 3 and 4 start failing (can not make calls). The system is configured to work with SS7. After the ACM message goes out, immediately a REL message is returned. I searched for error messages in /var/log/messages and could not find a clue

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

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)

-----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