[asterisk-users] rtptimeout on Asterisk 1.4.x
Hi Folks, Since I upgraded my asterisk box from 1.2.x to 1.4.x (1.4.10.1 now) I noticed some dead calls apparently running for more than 8 hours. I'm using rtptimeout=60 and rtpholdtimeout=120 and found some log messages like this: chan_sip.c: 'SIP/XXX-085a9308' will not be disconnected in 61 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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:
exten =
177,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 177,2,Monitor(wav,${CALLFILENAME},bm)
exten = 177,3,Dial(SIP/17,30,tT)
exten = 177,4,Hangup
exten =
178,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 178,2,Monitor(wav,${CALLFILENAME},bm)
exten = 178,3,Dial(SIP/17,30,tT)
exten = 178,4,Hangup
exten =
179,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 179,2,Monitor(wav,${CALLFILENAME},bm)
exten = 179,3,Dial(SIP/17,30,tT)
exten = 179,4,Hangup
Ex:
A = 177
B = 178
C = 179
A calls to B (Monitor starts recording conversation between A and B) and then
B press flash
and calls C (Monitor starts recording conversation between B and C and A stay
on moh) and then
B hangup the phone bridging A with C.
The first (A to B) and the second (B to C) recording ends when B hangup the
phone so I'd like
to have recorded the conversation between A and C, is that possible?
Thanks for any help!
Telles
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[asterisk-users] Recording/Monitor after xfer
Hi,
I'd like to know if some one knows how to make Asterisk record a call after
xfer (not bxfer).
I tried some ways but it doesn't work at all.
extensions.conf example:
exten =
177,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 177,2,Monitor(wav,${CALLFILENAME},bm)
exten = 177,3,Dial(SIP/17,30,tT)
exten = 177,4,Hangup
exten =
178,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 178,2,Monitor(wav,${CALLFILENAME},bm)
exten = 178,3,Dial(SIP/17,30,tT)
exten = 178,4,Hangup
exten =
179,1,Set(CALLFILENAME=/var/spool/asterisk/monitor/${EXTEN}/${TIMESTAMP})
exten = 179,2,Monitor(wav,${CALLFILENAME},bm)
exten = 179,3,Dial(SIP/17,30,tT)
exten = 179,4,Hangup
Ex:
A = 177
B = 178
C = 179
A calls to B (Monitor starts recording conversation between A and B) and then B
press flash
and calls C (Monitor starts recording conversation between B and C and A stay
on moh) and then
B hangup the phone bridging A with C.
The first (A to B) and the second (B to C) recording ends when B hangup the
phone so I'd like
to have recorded the conversation between A and C, is that possible?
Thanks for any help!
Telles
___
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[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 to debug SIP transactions in a friendly way. http://www.devel-it.org/index.php?modulo=projetoslang=en_US We hope you enjoy our work. Telles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 o sistema operacional SUSE com kernel 2.6.13-15-smp sem atualizações. Eu sou muito grato ao Frederic Jean, Pierre Freire, Caio, Josué Conti, Alexandre Fenzke, Gabriel Sartor e todos que comentaram na lista ou me ajudaram diretamente. Perdoem-me se não falei de alguém, é que são muitos. Eu já havia testado anteriormente com o mesmo sistema operacional, mesmo kernel e as mesmas bibliotecas que menciono a seguir, e não havia funcionado. A diferença foi que dessa vez, selecionei somente alguns pacotes. Antes, pelo fato de eu não ser especialista em Linux, eu estava instalando no gerenciador de pacotes tudo o que era referente a C e C++ e tb um monte de pacotes que são relacionados a usuários avançados, especialistas, não lembro direito, mas se procurarem no gerenciador de pacotes, irão encontrar. Os pacotes que selecionei foram: glibc, glibc-devel, gcc, gcc c++, ncurses, ncurses-devel, cvs, bison, termcap, openssl, openssl-devel, zlib, zlib-devel, libjpeg, libjpeg-devel, libtiff, libtiff-devel, libxml2, libxml2-devel, patch, readline, readline-devel, spandsp, kernel-source, wget Uso o Asterisk 1.2.4, Zaptel 1.2.4, libdtmfr2-0.0.3, libfx-0.0.3, libpri-1.2.2, spandsp-0.0.3, libmfcr2-0.0.3, libsupertone-0.0.2, libunicall-0.0.3. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End Message--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 speak with A again I tried with Asterisk-1.2.2 + bristuff-0.3.0-PRE-1i to use Steal() and StealChan() but that apps hangs up A when Steal() B. Perhaps there is another way to do it by manager interface but I don't know how. Does some one knows how to it using Asterisk? Thanks for any help! Telles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for Call Center (missing reference)
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/045882.html To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk for Call Center
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. To be able to listen (zapbarge, zapscan or chanspy) an specific channel and can talk to one side (the operator). That feature is very usefull in call centers in Brazil so if you want to use Asterisk as a Call Center PBX you have to support it. John Todd post it in May 2004 so perhaps now (Jan 2006) it's possible or there is another app (commercial?) that can support it. John: have you 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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Poor volume on SPA-2100 due to asterisk?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael, I don't know SPA 2100 but I have 2 SPA 2000 and changed the option FXS Port Input Gain: and FXS Port Output Gain: to 5 and the problem was solved. regards. Michael Stahl escreveu: | I just bough a Sipura SPA-2100 to use with Asterisk. When I use the | analog handset plugged into the SPA-2100, the person on the other end | can hardly hear me. | | I check the SPA-2100 setup and their is no mic/spk gain control. Is | this a problem with the SPA-2100 or with Asterisk? Any way for asterisk | to compensate for the poor audio level (if the problem is the SPA-2100)? | | Thanks, | Mike | | | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | 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] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFChQGYiLK8unYgEMQRApQ8AJ9u5odWEVE+QCZNukSE/w+qyhS8owCfRX89 KQFeclRvgVtHHbIyek+6vSQ= =SfGa -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Collect calls
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 James, jltaylor escreveu: | Since you are referring to R2 signaling, it works like this: I'm referring to ISDN PRI channels not R2. | | The E1 R2 Call Blocking feature provides two ways to block incoming collect | calls-category-based and double answer. With category-based call blocking, | collect calls will be blocked based on a specific category. For example, in | Brazil, collect calls arrive with a category II-8, for which the gateway | should send B-7 as a response instead of an answer signal. This approach is | only applicable when switches in the central office support category-based | blocking. | | For legacy switches that do not support category-based blocking, the double | answer method is implemented to support the collect-call blocking. For an | incoming collect call, the gateway will answer the call with a clearback | after one second and re-answer the call after two seconds, causing the | collect call to be dropped and normal calls to stay connected. Can you give me an example using this method with Asterisk? | | This is what the referenced patches are attempting to do. Referenced Patches? What do you mean? Does someone is working with patches to implement this feature in Asterisk? | | This does not work in the U.S. or if you have SS7, you don't need it. Thanks for your answer. | | James | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] Behalf Of Michael D | Schelin | Sent: Tuesday, May 03, 2005 6:06 PM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: Re: [Asterisk-Users] Collect calls | | | You Bring up a great point. I understand these codes and my system | brings them in via ss7 but as youself I don't know how to protect my | network from 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/007500.html | I don't know what does it mean!!! | Can someone 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 [EMAIL PROTECTED] | IVOZ # 1009 | TI Manager | Devel-IT - http://www.devel.it | Bestcom Group | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | 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] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCeOf0iLK8unYgEMQRAnFJAJoDdR07uKNGOyIjtV1lgnrCoS+7xACfTRc/ aaw9DBci1lZfamMxO4PQJdA= =Y/Qc -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Collect calls
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 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/007500.html I don't know what does it mean!!! Can someone 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 [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCd+zUiLK8unYgEMQRAkChAJ4xDYOvl8yZY+Uqn6v5VFZ4tMzicQCfT8+T 5foewh0m/o3ABMqcNHhtQs4= =rsu2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. Telles [EMAIL PROTECTED] IVOZ # 1009 Diretor de Tecnologia Devel-IT - http://www.devel.it Grupo Bestcom -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCciwBiLK8unYgEMQRAqFLAJ449p8tLjyglG+Mt40wUllfDBTyQQCeIAlM 8Q+wdor3HoczTGFxG7Fzdi4= =UKW2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 failure? I don't see anything. The next stable release of | asterisk will be 1.2 and it will have RealTime. The current CVS-HEAD (aka | 1.1) has it now and it is very stable. I don't have any failure I just want to know if the next release will be 1.0.8 or 1.2. | | The eta on 1.2 is unknown. You can help 1.2 along by downloading it and | running it to help fix bugs. Great, I'll do that! Thanks for your answer Matthew. regards. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCcnEaiLK8unYgEMQRAtZzAJ44ELXg1lpQfBh45Aj9gDbj/MqO7wCfTYi0 sqrNa7Lz30lpWcxxP9ciN1E= =3UYk -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-lite for linux
Hi Edgar, Don't use XLite under wine, use native code for Linux: http://sipthat.com/archives/000187.html Cheers Edgar de Leon escreveu: Hello im triying to config xlite on wine for linux, but got problems with the mic test, can anybody tell me how to get the mic config to work with wine or x-lite? TIA Edgar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IVOZ # 1009 Project Manager Devel-IT - http://www.devel.it TDKOM Group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DSLink modem freeze
Hi Eric, Thanks every body that answered about this problem. About change de default SIP port (5060), I tried it at first and the UAC could authenticate but when I made a call and another side pick the phone up DSLink 200E freeze again. ie. there wasn't any port 5060 on transactions. I will have this DSL modem on my LAB asap and I will give feedback to the list. Thanks Eric Wieling aka ManxPower escreveu: On Cisco routers you can do something like no nat sip fixup 5060 and that will disable only the special SIP related nat features, but leave in all of the other NAT features. If a vendor does not include a similar ability in their SIP aware router they should be shot. --Eric C F wrote: I have this problem with Best Data DSL Modems, If I disable NAT (on the router, not in SIP) it works fine. You might be able to do the same just disable NAT and it will work, if you disable NAT 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 Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E with the same results. Turn off SIP support and let the generic NAT deal with it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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://www.devel.it TDKOM Group ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DSLink modem freeze
Hi Folks, I've been having troubles with a DSL router (DSLink 200E) and SIP phones. When I put any SIP phone (software or hardware) to work behind that DSL router, it completely freeze. I ready tech specs of that DSL router and it says that SIP protocol is supported. ie. I tested two DSLink 200E 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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, | | Brancaleoni Matteo wrote: | | Hi, | | | | | |I'm using Asterisk-1.0.2 and voicemail linked with MySQL-4.1.7 library. | |I realised that asterisk is loosing connection with MySQL server and | |inform that user doesn't exist. | |Does anyone is using Asterisk voicemail linked with MySQL 4.1.x library? | | | | | | sure. never lost a connection. | | Using * + mysql 4.1 since when mysql 4.1 was in beta. | | | | So, I'm using MySQL server 4.1.7 for another applications too and I don't | have troubles with them. | I realised that: | When the Asterisk starts, it make a connection with MySQL server. I saw | that | the connection with MySQL server is ESTABLISHED (netstat). | When someone try to access his mailbox, asterisk just send the query | trough the | connection without problems. | In a few minutes, the state of the connection become in CLOSE_WAIT and | in that | moment I tried to access the voicemail and failed, but the second time | it works. | It seems that I've been having problems with connections. | | I don't know exactly what's happening. | I'll try Asterisk-1.0.1 with MySQL 4.0.20 (I never had problem with | them) and then | with Asterisk-1.0.2 with MySQL 4.0.20 and 4.1.7 just 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://www.devel-it.com.br | TDKOM Group | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ~ http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail and MySQL 4.1.x
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Folks, I'm using Asterisk-1.0.2 and voicemail linked with MySQL-4.1.7 library. I realised that asterisk is loosing connection with MySQL server and inform that user doesn't exist. Does anyone is using Asterisk voicemail linked with 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) iD4DBQFBknoZiLK8unYgEMQRAjNOAJd3ppds/2c7wJpxhUSEGSiIj6MaAJ4wt94s bHXcLVHMcGlr+6uNi1j1ZQ== =mk0S -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and MySQL 4.1.x
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Matteo, Brancaleoni Matteo wrote: | Hi, | | |I'm using Asterisk-1.0.2 and voicemail linked with MySQL-4.1.7 library. |I realised that asterisk is loosing connection with MySQL server and |inform that user doesn't exist. |Does anyone is using Asterisk voicemail linked with MySQL 4.1.x library? | | | sure. never lost a connection. | Using * + mysql 4.1 since when mysql 4.1 was in beta. | So, I'm using MySQL server 4.1.7 for another applications too and I don't have troubles with them. I realised that: When the Asterisk starts, it make a connection with MySQL server. I saw that the connection with MySQL server is ESTABLISHED (netstat). When someone try to access his mailbox, asterisk just send the query trough the connection without problems. In a few minutes, the state of the connection become in CLOSE_WAIT and in that moment I tried to access the voicemail and failed, but the second time it works. It seems that I've been having problems with connections. I don't know exactly what's happening. I'll try Asterisk-1.0.1 with MySQL 4.0.20 (I never had problem with them) and then with Asterisk-1.0.2 with MySQL 4.0.20 and 4.1.7 just 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://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBkqGGiLK8unYgEMQRAvryAJ9GgC95nBin1wqmMrFEhEKbsIjABQCeJdZL MhyQ+lG52tfhTDd9pFuiLpY= =+poC -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Brazil. Why not ?!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jefferson, Jefferson Carvalho wrote: | Hello list , | | I'm looking for partners in Brazil to discuss a possible way | to have in Brazil an Official Conference regards Asterisk. Yes, it will be wonderfull ! | It'll includes a hardware/workshop and tech-seminars. | Would be nice if we could include in this conference , Anatel's | presence and a seminar about the lawful aspects of VoIP in Brazil. | I'm 100% sure that in Brazil , we have enough resources to | become a large and active Asterisk community. :) I'm glad to say that you are right and more, there are a lot of people in Brazil working with Asterisk. Contact me in private if you want. Best regards. | | Best Regards, | | -Jefferson Carvalho | Jeff Networks Consulting Ltda. | Teresina-PI-Brazil | ___ | Asterisk-Users mailing list | [EMAIL 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://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBj19BiLK8unYgEMQRAqG4AJ0azCrspMj2Ca0m/bc6FERBf2lP6QCfYKEZ oguuMN/B5xP8WofrQppKI6Y= =9dRq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 wct1xxp wct4xxp ztdummy Exactly in the line 52 for zaptel version 1.0.0. If you have doubts, use vi +52 Makefile Best regards Telles Mark Halverson wrote: | Okay - I will be honest here, I have no idea what I am doing.. | | Ever watch the CDW commercials the IT guy's worse nightmare That's Me. | | A friend told me to look at the makefile for in the zaptel directory and | uncomment the line for ztdummy so that it would compile. (we need to make a | conference room on a server, no access to put a x100p card in) | | Well I opened the makefile and I see the following line: | | ztdummy.o:ztdummy.c ztdummy.h | $(HOSTCC) $(KFLAGS) -c ztdummy.c | | So is it already uncommented and complied by default? | | I just use CVS this morning to get the zaptel, libpri and asterisk so I | assume that I am compiling 1.0 | | Thanks in advance, | | Mark Halverson | 707-735-1038 | | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBhqD/iLK8unYgEMQRAq4mAJ9ZBsjezYUmENKO4ilJC/dyE9zr7QCfQGCJ O8RjT+vySSPpCHIEjxEDipg= =XcFC -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zap ANSWER the Call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Lyle,
Lyle Giese wrote:
| The standard for loop start does not send answer supervision, so * and all
| other telcom devices that do CDR records have to 'assume' that the call was
| answered.
So, it means that is impossible to say if 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, August 31, 2004 5:31 PM
| Subject: Re: [Asterisk-Users] Zap ANSWER the Call
|
|
|
| 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 from SIP phones to PSTN and it works fine,
| | but
| | asterisk accounting is doing something strange (for me).
| | Scenario:
| | 1 - extension 1009 (SIP phone - BT101)
| | 2 - Zap/4-1 (TDM400 FXO module)
| |
| | extensions.conf:
| | [dialout]
| | exten = _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| | exten = _9.,2,Congestion
| |
| | [sip]
| | include = dialout
| | exten = 1009,1,Dial(SIP/1009,20,rt)
| |
| | So, when I dial 9something from 1009, something rings and then I
| | hangup the phone.
| | I realised that asterisk thought that something ANSWERED the call:
| |
| |
|
| ,1009,9something,sip,Tests,SIP/1009-42fb,Zap/4-1,Dial,
|
| | Zap/4/something|25,2004-08-27 20:15:34,200
| | 4-08-27 20:15:36,2004-08-27 20:15:43,9,7,ANSWERED,BILLING
| |
| | Is that right?
| |
| | Version: Asterisk 0.9.0
| |
| | Thanks in advance.
| |
| | --
| |
| | Rodrigo P. Telles [EMAIL PROTECTED]
| | Project Manager
| | Devel-IT - http://www.devel-it.com.br
| | TDKOM Group
| |
| ___
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| ~ http://lists.digium.com/mailman/listinfo/asterisk-users
|
| --
|
| Rodrigo P. Telles [EMAIL PROTECTED]
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
|
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| [EMAIL PROTECTED]
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|http://lists.digium.com/mailman/listinfo/asterisk-users
- --
Rodrigo P. Telles [EMAIL PROTECTED]
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
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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 from SIP phones to PSTN and it works fine,
| but
| asterisk accounting is doing something strange (for me).
| Scenario:
| 1 - extension 1009 (SIP phone - BT101)
| 2 - Zap/4-1 (TDM400 FXO module)
|
| extensions.conf:
| [dialout]
| exten = _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
| exten = _9.,2,Congestion
|
| [sip]
| include = dialout
| exten = 1009,1,Dial(SIP/1009,20,rt)
|
| So, when I dial 9something from 1009, something rings and then I
| hangup the phone.
| I realised that asterisk thought that something ANSWERED the call:
|
| ,1009,9something,sip,Tests,SIP/1009-42fb,Zap/4-1,Dial,
| Zap/4/something|25,2004-08-27 20:15:34,200
| 4-08-27 20:15:36,2004-08-27 20:15:43,9,7,ANSWERED,BILLING
|
| Is that right?
|
| Version: Asterisk 0.9.0
|
| Thanks in advance.
|
| --
|
| Rodrigo P. Telles [EMAIL PROTECTED]
| Project Manager
| Devel-IT - http://www.devel-it.com.br
| TDKOM Group
|
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- --
Rodrigo P. Telles [EMAIL PROTECTED]
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
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[Asterisk-Users] Zap ANSWER the Call
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm using a TDM400 with one FXS and one FXO module (developer kit) and
I've been testing termination from SIP phones to PSTN and it works fine, but
asterisk accounting is doing something strange (for me).
Scenario:
1 - extension 1009 (SIP phone - BT101)
2 - Zap/4-1 (TDM400 FXO module)
extensions.conf:
[dialout]
exten = _9.,1,Dial(Zap/4/${EXTEN:1}|25|r)
exten = _9.,2,Congestion
[sip]
include = dialout
exten = 1009,1,Dial(SIP/1009,20,rt)
So, when I dial 9something from 1009, something rings and then I hangup the phone.
I realised that asterisk thought that something ANSWERED the call:
,1009,9something,sip,Tests,SIP/1009-42fb,Zap/4-1,Dial,
Zap/4/something|25,2004-08-27 20:15:34,200
4-08-27 20:15:36,2004-08-27 20:15:43,9,7,ANSWERED,BILLING
Is that right?
Version: Asterisk 0.9.0
Thanks in advance.
- --
Rodrigo P. Telles [EMAIL PROTECTED]
Project Manager
Devel-IT - http://www.devel-it.com.br
TDKOM Group
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=ukfg
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Re: [Asterisk-Users] Grandstream 1.0.5.30 available
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, You can access web browser only using its LAN network connection and getting (or setting) a IP by builtin DHCP. IMHO its bad. Telles Hekuran Doli wrote: | I just installed it on handytone, and I cant access web based | administration. any idea how to get it back? | | |On Fri, 02 Jul 2004 17:33:30 +1000 |Master Abi [EMAIL PROTECTED] wrote: | | |New firmware version at http://www.hellofone.com/downloads.html. |Might fix the no register issue and others. | | I tried this a few days back, totally hosed my phone, has to back to |Grandstream, don't touch it! | | -Max. | |I just installed it, and it seems to be working fine on my phone. Has |registered, and is working nicely. | |Andrew | | | | | ___ | Asterisk-Users mailing list | [EMAIL 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] Gerente de Projetos Devel-IT - http://www.devel-it.com.br Grupo TDKOM -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA5WzsiLK8unYgEMQRAq82AJ9yPEornjkmYjWg3PGUnf4NehDjoACeOWlB 4JDEOZhvfa3K6y5zCY/YYYA= =v2pp -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 phone. | | Regards | | Arve5 | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA4Iq4iLK8unYgEMQRAmDpAJ49AAIqNUN5t1uhvPL0dwt/bub8PgCeOkZn QhjZWvivXvlwdYCO+mz0tWE= =Kslk -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
