similar to: Asterisk <--> Cisco 2620

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

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

Hi all I'm trying to configure a TE410P in Europe with three E1s and a T1 channel bank (already bought in the US) to receive and make calls through *. I managed to get the E1s to work, but I'm having trouble with the channel bank (a Rhino). I have tried the bank in spans 1 and 4 changing jumpers in the card (always with no PRI connected, as I have no PRI where I'm testing now,

Hi, I have been trying to get a Wildcard TE122 card running here the last couple of days. libpri and zaptel are all installed and configured to E1 specs. The jumper on the card is on, so configured for E1 (I'm in Norway). When running zttool, I get 'Alarms: RED' on the single card installed. Other information in zttool: Current alarms: Red Alarm Sync source: Internally clocked

Hi All, I'm having problems trying to get a TE120P operational in Canada. I keep getting a congestion error when I try to make a call. I'm not sure if my switching, parity, etc is correct. I'm hoping that someone will be able to verify my config. The Telco is SaskTel, with a 10 channel 50 DDI service. Zap show channels show and ztcfg -vv looks ok and the zttool show

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

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)

I've been running Asterisk for months with no problems. I have grown to the point where I need an aditional TE cards. After many attempts I was able to add the second card without affecting the performance of the first. However, the second card is not working properly. Setup ===== - Running on a Dell Server. - RedHat 8 - Asterisk 1.0 - I have two TE410Ps on that box, card 0 assigned to

Hello list , I?m putting to work a new asterisk box. I?m running * 1.0.3 with (one ) Wildcard TDM ( 2FXS* 2FXO) and (one) E-100P. Both boards are working well. ztcfg don?t show me any error. zttool list both cards as "OK" But when i run asterisk on verbose mode i get those errors : [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

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

Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's lite green. Message type: RELEASE COMPLETE (90) < [08 02 80 ac] < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) < Ext: 1 Cause:

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 all, I am trying to setup an E100P for use on Swisscom E1-PRI here in Switzerland. Swisscom seems to use Siemens hardware. Here are my configs (cvs from a few hours ago) : zaptel.conf loadzone=fr ; tried "de" but got warning at modprobe defaultzone=fr span=1,1,0,cas,hdb3,crc4,yellow ; Seems to be right, as the Siemens SDSL-G703 modem switches its LEDs off ... not sure

Any ideas? 51] logger.c: [chan_zap.so] => (Zapata Telephony) Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing '/etc/asterisk/zapata-auto.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing

Hi guys, I've been installing and configuring a TE110p card. The compile and install went very well. I'm using this on FC4 and I compile with linux26 as well checked I on the udev configs. zttool and ztcfg both indicate that the card is ready. But when I try to "load chan_zap.so" then I get the following Unable to load module chan_zap.so Jan 5 21:43:33 ERROR[6808]:

Hi, I am trying to track down the reason to my problems with sending and reciving fax with my PRI and 2 TDM400P Cards: PSTN <-> PRI (E100P) <-> * <-> TDM400P <-> Fax Machine I have used Zapbarge to listen to the data stream, but I can't say if it really have some time slips - fax kinda noisy in itself. Using the zttool i saw the Sync source for the TDM are

Hi, Using Xorcom's bristuff-0.4.0-RC4-xr7.tar.gz (ie asterisk 1.4.25) with an old Junghanns OctoBRI (ie not the 2.0 version), zttool shows every port in TE mode, though half of them are in NT. Alarm column shows OK for every port, such as : OK octoBRI PCI ISDN Card 1 Span 1 [TE] Lay Globally, this system doesn't work (can't dial out through zaptel) Replacing this board

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

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]

Hola! I'm using asterisk as H.323 -> PRI gateway. First call goes thru ok, second concurrent call fails with: Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to context [ip2pri] -- Executing Dial("H323/ip$192.168.32.25:60271/984", "Zap/1/9541163107100") in new stack Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to