similar to: SPA3102 interdigit timers bug?

Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only "333" What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha

Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP->PSTN

Hi. Does anyone knows how to trigger a phone call from a bash command? Thx! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha Contra La Pobreza - PADEP Av. S?nchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email:

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hi, I just want to deny FOP originated calls in TRIXBOX. All remaining operations (hanging up, transferring, etc) should go. Where is that option in TRIXBOX (already googled, nothing, checked config files but cant find that option)? Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica

client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo < rodolfo.alcazar at padep.org.bo> wrote: > Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. > Could be what you want. > > Greets! > > Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: > > TianLun, > > > > I

Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/a278cbbd/attachment.htm

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

Ok, I downgraded spa3102 to 3.3.6. Now when I make a call from pstn and call is established asterisk seems to drop the call. However I still hearing ringback on pstn side, call is established again, and asterisk drops the call again, like a loop. -- Executing [preat_admin at nodo:1] Playback("SIP/PSTN-08214948", "horario-atencion/our-business-hours-are") in new stack

Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear R list, I've read the function writing sections on both "An introduction to R" and "R language Definition" manuals but still don't understand why the following gives an error message: fun <- function(x, ...) x + y fun(1, y=2) I get: Error in fun(1, y = 2) : object "y" not found I'd appreciate any help in understanding this. R version 2.4.1

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->