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Quick and dirty question: for the TDM410P I must use the wctdm24xxp driver? Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

...risk box, app_fax works wonderfully well transmitting at full 9600bps. Just to rule out Asterisk's T.38 handling, I noticed that Fax For Asterisk works perfectly with T.38, also on that same box. I'll be happy to provide you any info that could help solve this issue. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

...in a call (InUse), the call enters the queue and stays there. I would like the call NOT to enter the queue and the ${QUEUESTATUS} variable to be set to something different. Am I missing something or it's just not possible? I'm using SIP phones for the agents and Asterisk 1.4.15. Att Vin?cius Fontes Desenvolvimento Canall Tecnologia em Comunica??es Ltda.

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Hi, I just got a Xorcom Astribank with 8 FXS but it does not work. So I tried resetting and loading the firmware. But loading just times out. /usr/share/zaptel/xpp_fxloader usb --------- FIRMWARE LOADING: (usb) [1 devices] 'xpp_fxloader'[11561]: USB Firmware /usr/share/zaptel/USB_FW.hex into /dev/bus/usb/002/003

...h,n,System(echo "Velocidade de transmissao: ${FAXBITRATE} bps" >> /tmp/${FAXFILE}.txt) exten => h,n,System(echo "Resolucao: ${FAXRESOLUTION}" >> /tmp/${FAXFILE}.txt) exten => h,n,System(mutt -s "Allvo FAX" -a /var/spool/asterisk/fax/${FAXFILE}.pdf vinicius at canall.com.br < /tmp/${FAXFILE}.txt) exten => h,n,System(rm /tmp/${FAXFILE}.txt) I'm using here app_fax that comes with Asterisk, not the res_fax and res_fax_digium that comes with FFA. What happens is sometimes the T.38 negotiation goes well and others it fails completely. That'...

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come over the other lines. -- ?Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director

...ock sample intervals (100.000%) 8192 samples in 8192.000 system clock sample intervals (100.000%) 8192 samples in 8192.000 system clock sample intervals (100.000%)^C --- Results after 21 passes --- Best: 100.000 -- Worst: 100.000 -- Average: 100.000000, Difference: 100.000000 Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a PAP2T I get one way audio, the remote site can hear me but I cannot hear them. If I do an "rtp

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

I'm running dahdi on the host system, and have added the /dev/dahdi/ devices to the guest vserver as recommended in Beave's "Virtual Private Asterisk" whitepaper (http://www.telephreak.org/papers/vpa/). I tried copying libtonezone.so and libtonezone.h to the guest, but I couldn't anything to replace zaptel.h in DAHDI souce (it seems dahdi.h was deprecated?). I need to

...l=yes t38pt_usertpsource=yes [9204] type=friend secret=****** callerid="" <9204> host=dynamic nat=no canreinvite=yes insecure=port,invite qualify=yes context=ddi callgroup=1 pickupgroup=1 call-limit=10 disallow=all allow=g729 allow=alaw t38pt_udptl=yes t38pt_usertpsource=yes Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP

Thank you Vinicius for quick answer, problem is i've my product ISO which i want to use to install in a headless way, while these tutorials are talking about interactive way of installing the basic debian system - which works fine. Headless installation example i'm missing, if you have come across any such ex...

...ething that I'm doing terribly wrong, that would break everything and make the universe collapse into itself when I apply the same principle on production? I'll be happy to provide more details in case there are any doubts. I really appreciate your feedback, no matter what is it. :) Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

...rrupts RES: 16326632 9071480 Rescheduling interrupts CAL: 303 529 function call interrupts TLB: 1521462 4810165 TLB shootdowns TRM: 0 0 Thermal event interrupts SPU: 0 0 Spurious interrupts ERR: 0 MIS: 0 Vin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP

I've spent a week playing with Asterisk 1.6 and I love it. What a brilliant piece of software! Progress and learning have been reasonably good. I have external SIP provider calls coming in and have put together a little call platform and I'm stunned at the flexibility. There is one issue for me. I took me a while to click that ZAPTEL now equals Dahdi, but now I'm there I have an

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <span class="postbody">Greetings,<br> I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a </span>VPMADT032<span