similar to: Ringduration problem when calling out via Sip

Hi Folks, I've Asterisk Bristuffed up and running behind an Auerswald Commander Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works marvelleous for outgoing calls (as the parallely installed avm fritzcard with chan_capi does), but when I'm trying to call in, I get a short ring signal and then the connection is terminated. This does not happen with chan_capi and

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

I'm using a Grandstream IP phone to call someone through our asterisk pbx. The PBX is running "Asterisk 1.0.3-BRIstuffed-0.2.0-RC3" and uses 2x ZAP-HFC cards. When I call someone, if the call isn't answered and then I hang up, I get "487" coming up on the grandstream phone. If I pick up the receiver again and then hang up, the PBX starts calling me back and when I

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis <----> ISDN Phones |

Hi all, I have a hangup problem when i get incoming calls on my ISDN interface. I use ISDN network controller [HFC-PCI] and asterisk with florz patch. Logs when the hangup happens follows: May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' not posted May 6 20:52:47 NOTICE[11532] cdr.c: CDR on channel 'Zap/1-1' lacks end May 6 20:52:47 DEBUG[11532] pbx.c: Expression

Hello, I have a system with two cards: a HFC-PCI ISDN and a TDM21B (2 FXO and 1 FXS), running Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l with freePBX beta5 dialplan. I have connected an analog phone to TDM FXS port, but when I pickup the phone to make a call, Asterisk "hangs up" the call. Let me explain: In another system, when I pickup the phone, Asterisk give me tone to dial: >---

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

Hi all, I just posted this question before last week. Meanwhile after upgrading Asterisk 1.0.7-BRIstuffed-0.2.0-RCg to 1.0.8-BRIstuffed-0.2.0-RCh the same problem occurs, but seems to be more seldom. Attached is now the output of "zap show channel" . - I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns OctoBRI running in NT-mode.

This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on

Hi all, I'm running a stable Asterisk on a HP DL380G2 1.4Ghz 0,5GB RAM equipped with 1x TE410P and 2xJunghanns QuadBRI running in NT-mode. Connected to the BRI-Ports are 12 Fax-Modems (Elsa MicroLink ISDN/TL V.34) which are only operating in dial out analog mode to deliver fax messages. After a while of running fine (50-200 dial out connections) on some S0 spans the following message occurs

I think I found another interesting 'feature' with threeway calling. If you hang up while on a 3 way call with both parties on outside lines, Asterisk ends up removing the conference initiator and leaving the outside trunks bridged together. Is this a good idea? This could cause congestion problems on small configurations with limited outgoing lines. Maybe we should add an option to

Hi, I have installed a quadBri card, with Asterisk-1.0.10 and the bristuff-0.2.0-RC8s (* 1.0.10) When calling 0207654321 the following happens: -- Executing Goto("Zap/1-1 ", " salsa-helpdesk-day|s|1 ") in new stack -- Goto (salsa-helpdesk-day,s,1) -- Executing Dial ("Zap/1-1 ", "Zap/g1/0201234567|30 ") in new stack -- Requested transfer capability:

I am working on this app to dial two external numbers. The second is dialed after the first hangs up. I have simplified things down to: exten => 3852,1,Dial(zap/g1/3964,10,g) exten => 3852,2,Wait(2) exten => 3852,3,Dial(zap/g1/7757,10,g) exten => 3852,4,Hangup Here is the debug: -- Accepting call from '0000000000' to '3852' on channel 0/23, span 1 --

Hi All ... I'm running Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v which has a Junghanns QuadBri card in it (lspci reports Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01)) I have a regular KPN ISDN2 Line connected which works fine with the the zapata.conf below. However, I also have a new KPN ISDN Mutli (meervoudig) line (PBAX-Group), which works fine for incoming calls,

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

Hi, (Apologies for readers of Bristuff mailing as I already posted this message to the list) My setup is: SIP hardphone ------------- <LAN> ----------- Asterisk server ---------<ISDN> Asterisk server is : Gentoo enabled with 1.0.8 bristuffed Asterisk equipped with Junghanns Quad BRI with 2 BRI ports connected to ISDN >From log files (with line numbering enabled) I've got :

I am having issues with dialing out. I can dial in to the system just fine. I am hoping some one has some ideas.... Here are my conf's. ZAPTEL.CONF (All is correct for our T1) span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us ZAPATA.CONF [channels] group=1 rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});