[asterisk-users] rtptimeout on Asterisk 1.4.x

seconds because it is directly bridged to another RTP stream I can kill that calls using 'soft hangup channel' but I'd like to know if its a new BUG introduced in 1.4.x releases and if possible, how to fix this? Thanks in advance. Rodrigo P. Telles

[asterisk-users] Re: Recording/Monitor after xfer

Hi, Does any one knows some thing about this issue? I'll appreciate any comments! Telles Rodrigo P. Telles wrote: Hi, I'd like to know if some one knows how to make Asterisk record a call after xfer (not bxfer). I tried some ways but it doesn't work at all. extensions.conf example

[asterisk-users] Recording/Monitor after xfer

Hi, I'd like to know if some one knows how to make Asterisk record a call after xfer (not bxfer). I tried some ways but it doesn't work at all. extensions.conf example: exten = 177,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP}) exten =

[Asterisk-Users] ANNOUNCE: WIST - Web Interface for SIP Trace

Hi Folks, I'm glad to announce WIST for SIP debug/trace dialogs. This software born as a prof concept of the idea to capture SIP traffic from a remote host (SIP Proxy, Gateway, etc) and show up alive SIP messages about an specific dialog (filtered by From SIP user) to help our tech support team

Re: [Asterisk-Users] MFCR2

Darlon, Looks like you did a mistake! Try to send your e-mail to [EMAIL PROTECTED] Telles Darlon wrote: Oi pessoal. É com muita alegria que venho escrever pra vcs hj. Após uma longa jornada de testes, mais de um mês quebrando a cabeça tentando fazer o R2 funcionar, enfim consegui utilizando

[Fwd: Re: [Asterisk-Users] Asterisk for Call Center (missing reference)]

John Todd, Can you please answer that question or just give me your feedback about it? I'll be very thankfull to hear something from you! regards, Telles ---BeginMessage--- Hi, Does any body knows some thing about it? Thanks in advance. Telles Rodrigo P. Telles wrote: Hi Folks, I've

[Asterisk-Users] Steal with MusicOnHold

Hi, I have the following situation on my Asterisk PBX: 1) A (caller) is talking to B (called) 2) C (supervisor) want to Steal/Pickup/Speak with side B without hanging up A (possible put A on MusicOnHold) 3) C or B hangup the phone and then B start to ring, when B pickup the phone starts to

Re: [Asterisk-Users] Asterisk for Call Center (missing reference)

Hi, Does any body knows some thing about it? Thanks in advance. Telles Rodrigo P. Telles wrote: Hi Folks, I've been searching for an specific feature on asterisk and I found an e-mail from John Todd asking for the same thing. http://lists.digium.com/pipermail/asterisk-users/2004-May

[Asterisk-Users] Asterisk for Call Center (missing reference)

found a solution for your question? if so, please let me know! Thanks in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group

[Asterisk-Users] Asterisk for Call Center

in advance, -- Rodrigo P. Telles [EMAIL PROTECTED] IT Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Poor volume on SPA-2100 due to asterisk?

| To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users - --- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group

Re: [Asterisk-Users] Collect calls

these charges. I will follow this post to see if anybody | has a fix. | | | Rodrigo P. Telles wrote: | | | Hi Folks, | | Does someone knows how to identify and block collect calls on Asterisk | using PRI | channels? | I googled it and found this: | http://lists.digium.com/pipermail/asterisk-dev/2004-November

[Asterisk-Users] Collect calls

help me to understand this? I tried to apply that way too, using Flash() but Flash() complains and looks like just work with FXO channels: http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html Thanks in advance. - -- Rodrigo P. Telles

[Asterisk-Users] Realtime feature

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, Does someone knows if the next release of Asterisk (1.0.8?) will have Realtime support and when we will have the next Asterisk release with Realtime features? Thanks in advance. - -- Rodrigo P

Re: [Asterisk-Users] Realtime feature

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Matthew, Matthew Boehm escreveu: | Rodrigo P. Telles wrote: | | |Does someone knows if the next release of Asterisk (1.0.8?) will have |Realtime support and when we will have the next Asterisk release |with Realtime features? | | | Where is your

Re: [Asterisk-Users] X-lite for linux

-- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 Project Manager Devel-IT - http://www.devel.it TDKOM Group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

Re: [Asterisk-Users] DSLink modem freeze

then you will have to get a different router to be able to share the same IP, and if you use PPPoE you might not be able to do it, in which case you will have to get a different DSL modem. On Wed, 29 Dec 2004 20:00:28 -0600, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote: Rodrigo P. Telles wrote: Hi

[Asterisk-Users] DSLink modem freeze

with the same results. Does anyone has any idea? Thanks in advance. -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel.it TDKOM Group

Re: [Asterisk-Users] Voicemail and MySQL 4.1.x

Hi Folks, I think that I found the reason for that problem. MySQL server was starting with options: -O wait_timeout=100 -O interactive_timeout=200 I restarted MySQL server without these options, and apparently the problem was solved. Thanks for the answers. Rodrigo P. Telles wrote: | Hi Matteo

[Asterisk-Users] Voicemail and MySQL 4.1.x

library? Best Regards - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux

Re: [Asterisk-Users] Voicemail and MySQL 4.1.x

to check. Thanks for your answer. Best regards. | also used asterisk + mysql cluster for a while, only | in the lab, but never lost connection also | in that case. | | Matteo. - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http

Re: [Asterisk-Users] Astricon Brazil. Why not ?!

PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http

Re: [Asterisk-Users] ZTdummy

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Mark, You need to uncomment this part: MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \ ztdynamic ztd-eth wct1xxp wct4xxp # ztdummy to look like this: MODULES=zaptel tor2 torisa wcusb wcfxo wcfxs \ ztdynamic ztd-eth

Re: [Asterisk-Users] Zap ANSWER the Call

was answered or not? Someone have any idea to solve that? Is that problem is specific for TDM400P? Thanks for your answer. | Lyle | | - Original Message - | From: Rodrigo P. Telles [EMAIL PROTECTED] | To: Asterisk Users Mailing List - Non-Commercial Discussion | [EMAIL PROTECTED] | Sent: Tuesday

Re: [Asterisk-Users] Zap ANSWER the Call

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Nobody knows about that strange behaviour of Zap channels or at least if is that right? Thanks in advance. Rodrigo P. Telles wrote: | Hi, | | I'm using a TDM400 with one FXS and one FXO module (developer kit) and | I've been testing termination

[Asterisk-Users] Zap ANSWER the Call

. - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBL8ZdiLK8unYgEMQRAuR/AKCDfvkFnJ2/EFSZLD9dwDeXXtuDEwCfVrfn Fi

Re: [Asterisk-Users] Grandstream 1.0.5.30 available

-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Gerente de Projetos Devel-IT - http://www.devel-it.com.br Grupo TDKOM

Re: [Asterisk-Users] SIP Softphone

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arve, I've been using kphone (http://www.wirlab.net/kphone/) with success. It's simple and works fine :-) []'s Arve Rasmussen wrote: | Hi, | | What is the best SIP softphone to use with Asterisk? | | I have a hard time finding OpenSource SIP soft