similar to: How to change Zap channel negotiation/exclusive etc..?

Hi, I have a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>

Hi I have this escenario: |SIP or H323 phone|---->|Cisco2600|----E1-pri---->|Asterisk|------>IVR, A2Billing, etc... The problem is that I can not hear any audio when call from 'sip or H323 phone' and configure something like: exten => _01XXXXXXX,1,Playback(thank-you-for-calling|noanswer) ... It works if I remove the 'noanswer' parameter but in this case it connects

Hi Srs. we have to integrate a Nortel MATRA M6501-L with Asterisk with a TE410P. All call from outside get into asterisk and asterisk send to Nortel in a correct way. My problem is when a call is made from Nortel to Asterisk. If we digit a national Number in Spain([98]ZXXXXXXX or 6XXXXXXXX) all work find. But if we digit an international number call doesn't progress. I Have seen in

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

Hello, I have a busy problem with Asterisk when I try to transfer a call from PRI directly to IVR. This problem appear sometime after 2 hours or 2 minutes. The log file contain : Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) When this problem appear I must restart Asterisk to solve it. Another thing, I don't know why the alarm is set to NOP on SPAN

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

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)

Hello! I am trying to install a TN405P on a P4-3GHz-HT machine running Debian Sarge with kernel 2.4.27. When I start Asterisk in -vvvvc mode it always shows == D-Channel on span 1 up == Restart on requested on entire span 1 == D-Channel on span 3 up == D-Channel on span 2 up == Restart on requested on entire span 3 == Restart on requested on entire span 2 == D-Channel on span 4 up == Restart

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

Regarding this message, I've actually been told one caller who has consistently had this problem was using Vonage, but calling from his Verizon line, it worked. This skewed my survey. Therefore I do believe it's the same callers having the issue, and in which case, I think Vonage is to blame. I found this thread:

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 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).

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route

Hi, I am trying to validate a setup with a sangoma A116 card (16PRI in one card). I currently have two machines set up, each with a sangoma A116 card. Those are interconnected with crossed PRI cables. One of them is in NT mode, the other in TE. The ports are configured in E1 mode, asterisk is all set up and I can make calls between them, filling up all 480 (16 * 30) channels. Now, I learned

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

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

Hi, I've connected Asterisk with 4 PRI to a Siemens HiPath 4000. For CALLERID(name) feature I wanna use Q.SIG as switchtype. Cause Siemens PBX orders Channels logical I need the parameter qsigchannelmapping=logical. Here is my chan_dahdi.conf trunkgroups] [channels] language=de context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

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

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