similar to: Background command not interrupted as desired

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hi List! I got this error while upgrading zaptel: make -C firmware hotplug-install DESTDIR= make[1]: Entering directory `/usr/src/zaptel-1.4.7.1/firmware' Attempting to download zaptel-fw-oct6114-064-1.05.01.tar.gz --10:53:09-- http://downloads.digium.com/pub/telephony/firmware/releases/zaptel-fw-oct6114-064-1.05.01.tar.gz Resolving downloads.digium.com... failed: Temporary

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Hi, I have problem using Asterisk.I have isdn-pri and openvox d110p card in my computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all pins to the isdn done by telco workers). I got green led on isdn which is sign that isdn is working and that is connected to openvox, right ? I compiled newest versions of libpri zaptel and asterisk and had no problems during that. When I started

hi! I have a question of ISDN PRI "calling number" setting for outgoing call. I need the receiver to see the CLI that I have set. Eventough I've set the CLI as below, the receiver keep on getting a fixed number( used with the PRI to receive incoming calls). My ISDN PRI E1 provider says he has not done any restricions on the custom CLI side.

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 Folks, I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the CLEC to bring up the PRI and inbound calls are hanging up at his end after a few seconds. I ran PRI debug but it only gives me minimal insight. " Ext: 1 Cause: Unknown (16), class = Normal Event (1)" He ran a trace and the only difference he is seeing is a "ISDN interface explicitly

Hi, I facing a problem with flash hook. When ever I do a flash hook to place an extsing call on hold, the call gets disconnected. The debugs on Asterisk shows that 'on hook event detected' when I press the flash button on the phone. The setup is like this Asterisk box with T1 cards and FXS cards. The T1 card is connected to an IAD and configured for ISDN PRI lines. Analog phones come

I've got some weird behavior on my HFC-s cards. asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel 20040623 When i pick up my ISDN phone on Zap5-1 ("3987") and call the external number "1901" it will do so, connect me and everything is fine. In the second, where it tries to attempt the native bridge, the audio will disappear. Using another card

Don't you just hate it when something was working and when you come to use it in anger it's broken :-( Something in the, fairly, recent series of Asterisk updates has broken DIGITAL call passthrough. I've an ISDN PBX behind my Asterisk Box (PRI ISDN comes into port 1 of a Digium Wildcard and the PBX it connected to port 2 via an ISDN crossover cable). This PBX used to be able to

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

I have a serious problem that repeats very often, after 30 - 50 calls and I can only solve it by stopped and restarting * :-( After a while, * seems to loose track of something. When an ISDN call from PBX needs to go to the Telco, I get 'Ring requested on unconfigured channel 255/255 span 2' It's always channel 255/255 ??, the 'span' number is random Setup: * between Local

Hi everybody, I had an ISDN card with winbond chipset and isdn4linux, but there was a lot of echo when call from SIP to ISDN, so I buy an HFC chipset card (Conceptronic c128i). I downloaded and compiled bristuff-0.2.0-RC5 and everything is going fine. The sound quality is excellent, there is no echo, but i have a strange problem. Sometimes calls made from SIP phones to ISDN just finished

Hi All, I am installing a Zaptel Quad E1 card with ISDN PRI in Brazil and I?m with this problem: Asterisk CLI : Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! No channel map, no channel, and no ds1? W hat am I supposed to identify? Mar 27 01:49:57 WARNING[2612]: chan_zap.c:7143 zt_pri_error: PRI: !! Unable to add IE 'Channel Identification' == Restart

Hi, Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls just rings and rings. We never get the "The number you are trying to reach...". If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get the message on the 1st ring. Here is the PRI debug of such a call that just rings and rings. Any ideas? PRI debug sur CPL: -- Executing

Hi, I'm trying to build a SIP / ISDN BRI gateway. I'm using asterisk and zaptel 1.0.6 with the bristuff patches. I have a Billion HFC card connected to a BRI ISDN line. Unfortunately each time asterisk tries to make a call on that channel, I get NOTICE[6236]: app_dial.c:759 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time When I try

Dear Gurus, i am facing some unknown problem here.. first, let me describe my case. my office is using a nortel merridian option 11c PBX. and it is connected to the provider thru a E1 card, which is working fine with no problem. my plan it to slot in a server with 2 TE11XP card to intercept the outgoing calls. Provider <--te11xp---> asterisk <---te11xp------> nortel merridian

Hey all, I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware, configured with a help from Sangoma Tech Support, running fine. It is connected to a PRI circuit split from Cisco MC 3810, which in turn is connected to a Converged T from CTC Communications. While Asterisk works fine and I can call in/out on my BV account, I am only able to dial in through CTC. I have spent

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends