[asterisk-users] seems like call is picked and returned to me

Hi all I hope that someone of you can solve this. Right now I'm stuck! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182) All I can see in CLI is: == Using SIP RTP CoS mark 5 --

[asterisk-users] Asterisk 1.8, busylevel and CCBS

My question is so complex and I try to explain well. We have a customer that he wants limits incoming calls to his extensions to only one. That's not complicated with GROUPCOUNT, DEVICE_STATE or SIPPEER with curcalls option.But the problem is when you want implement CCBS service. If we have

RE: [Asterisk-Users] Re: Asterisk Redundency

A good solution is use a program that use sipsack for SIP, something like sipsack for IAX and Linux-HA for asterisk. In this way you check if SIP or IAX is OK, and if these technologies are bad, you can kill asterisk and linux-HA will do the rest. In PSTN Field, you can check rxhooksig in struct

RE: [Asterisk-Users] E1/T1 failover hardware

http://www.junghanns.net/en/ISDNguard_produkt.html srsergio -Mensaje original- De: John Daragon [mailto:[EMAIL PROTECTED] Enviado el: jueves, 20 de octubre de 2005 17:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] E1/T1 failover hardware

RE: [Asterisk-Users] TDM400P off-hook detection problem

Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? regards, srsergio -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves,

RE: [Asterisk-Users] TDM400P off-hook detection problem

Discussion Asunto: RE: [Asterisk-Users] TDM400P off-hook detection problem On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote: Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports

RE: [Asterisk-Users] Quad PRI Problems

You can't put four span in timing, because only one must be like nmaster sincronization. If one of your telco provide time for your card. Put second value in all span to 0. regards, srsergio -Mensaje original- De: Ronald Hartmann [mailto:[EMAIL PROTECTED] Enviado el: martes, 04 de

[Asterisk-Users] VideoConference with UMTS

Hi Srs., Do you know if it's possible make a videocall from asterisk to UMTS mobile phone?. Both technologies use H.263 like videocodec. Any idea? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list

RE: [Asterisk-Users] call center software and asterisk

www.inconcertCC.com has a solution based on Asterisk. regards, srsergio -Mensaje original- De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Enviado el: jueves, 29 de septiembre de 2005 17:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion CC: Commercial and

RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)

You must install libncurses5-dev regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: martes, 27 de septiembre de 2005 9:20 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Termcap missing (compile

RE: [Asterisk-Users] IBM x306

Hi all, we have same problem with a x346. Mainly, TE410P shares IRQ with network card and if you change IRQ for this slot, automatically change IRQ in network card. Any idea? srsergio -Mensaje original- De: George Pajari [mailto:[EMAIL PROTECTED] Enviado el: lunes, 26 de

RE: [Asterisk-Users] Queues

show application Queue is your friend. De: Sander [mailto:[EMAIL PROTECTED] Enviado el: viernes, 23 de septiembre de 2005 13:11Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'Asunto: [Asterisk-Users] Queues Hi there i need to know if there is a wayto play a ringing sound

RE: [Asterisk-Users] Asterisk in Spanish

Try in www.asterisk-es.org -Mensaje original- De: Sebastian Milioto [mailto:[EMAIL PROTECTED] Enviado el: lunes, 19 de septiembre de 2005 15:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Asterisk in Spanish Hi all, I've been installing

RE: [Asterisk-Users] E1 configuration problem

Please, send us zaptel.conf and zapata.conf and say us what card you have(TE110P, TE410P...). And what is your country. Regards, srsergio -Mensaje original- De: manish kumar [mailto:[EMAIL PROTECTED] Enviado el: lunes, 19 de septiembre de 2005 6:32 Para:

RE: [Asterisk-Users] Manipulate CALLERIDNUM

Hi, Try SetCIDNum application before VoiceMail application regards, srsergio -Mensaje original- De: Chad Brown [mailto:[EMAIL PROTECTED] Enviado el: miércoles, 31 de agosto de 2005 8:48 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Manipulate

RE: [Asterisk-Users] unresolved symbol when loading ztdummy

This option is under Library routines in your kernel configuration. Regards, srsergio -Mensaje original- De: Christoph Eicke [mailto:[EMAIL PROTECTED] Enviado el: miércoles, 31 de agosto de 2005 10:59 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re:

RE: [Asterisk-Users] queue - ringing members in order

Hi Sr. rrmemory i same like roundrobin, but this policy store which is the next when a call get into your system. For example with next queue: SIP/1 SIP/2 SIP/3 and roundrobin, all calls stars with SIP/1 and with rrmemory first call starts with SIP/1, second call with SIP/2 and so on.

RE: [Asterisk-Users] looking for failover ideas

How do you do monitoritng? How Server B knows that Servar A is down? I just do a rsync and MySQL Replication, but I try to do a C program that monitor Server. If you know how can I do this monitoring I will be pleasant with you. regards, srsergio -Mensaje original- De: Senad J

RE: [Asterisk-Users] looking for failover ideas

If I use hearbeat I need a failover system for ISDN Lines, not? I waould like that if Server A crashes, Server B Control SIP Registration and ISDN Lines. Do you know about this? regards, srsergio -Mensaje original- De: Senad J [mailto:[EMAIL PROTECTED] Enviado el: martes, 23 de agosto

[Asterisk-Users] Please, excuse me

Title: Mensaje I'm sorry for my holidays message, but I think it's too hard span me from list, don't you think? Could admin return to list, please? Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: asunto_mensaje_entrante

Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: asunto_mensaje_entrante

Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: asunto_mensaje_entrante

Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Sorry

Title: Mensaje I'm sorry for the several messages with holidays message. Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options

RE: [Asterisk-Users] outgoing call routing

Please, send us zapata.conf. It's possible that you don't have well configure zapata.conf, because in your trace you try to dial through g0 group and your Zap/4(I understand is your Zap connected to PSTN) must be into the 0 group. Regards, srsergio -Mensaje original- De:

RE: [Asterisk-Users] outgoing call routing

: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from-pstn channel = 4 At 10:10 AM 6/20/2005, Sergio Serrano wrote: Please, send us zapata.conf. It's possible that you don't have well configure zapata.conf, because in your trace you try to dial through g0 group and your Zap/4(I

RE: [Asterisk-Users] ztcfg server crash

Before change OS try to do next steps: first, stop asterisk. Second, you must do ztcfg -s to shutdown span. Unload modules, load modules if you need and do ztcfg -vv again. Start asterisk Regards Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En

[Asterisk-Users] TE410P and Siemens HIPATH 3750

Title: Mensaje Hi all, I have to interconnect Asterisk with a Siemens HIPATH 3750. In siemens we can configure ECMA-QSIG Master, ISO-QSIG Master,Point to Point link withCRC4 and Point to Point link withouthCRC4): Siemens has BNC connector. I use a balun with BNC and RH45 connectro. I

RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk

Fantastic!! Thanks to your works regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Areski Enviado el: miércoles, 26 de enero de 2005 18:05 Para: Asterisk-Users Mailing-list Asunto: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard

RE: [Asterisk-Users] Different EXT lines for different users?

You can try to set one context for each extension. regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Alen Salamun Enviado el: martes, 25 de enero de 2005 16:04 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto:

RE:[Asterisk-Users] Problems with loading TE110 module

Title: Mensaje Have you solve your Problem?, I have same problem after with recompile kernel. Regards, srsergio Monday, December 20, 2004, 12:44:36 PM, Matt wrote:MR Have you tried doing a modprobe -r first?Before reboot I did rmmod wcte11xp. If you mean that.now modprobe -r wcte11xp

[Asterisk-Users] TE110P doesn't appear in /proc/zaptel

Title: Mensaje Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. My zaptel.conf is the next: span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=esdefaultzone=es and cat /proc/pci throguh

RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel

] Enviado el: miércoles, 29 de diciembre de 2004 18:45 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 11:59:58AM +0100, Sergio Serrano wrote: Hi all, I have installed

RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel

: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 09:17:45PM +0100, Sergio Serrano wrote: Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules

RE: [Asterisk-Users] Link an Asterisk Box with a PBX (E1 connection)

Try exten= _X.,1, Dial(Zap/g2) If your 2nd TE110 has defined a group. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jeremy SALMON Enviado el: miércoles, 22 de diciembre de 2004 15:01 Para: asterisk-users@lists.digium.com

[Asterisk-Users] Asterisk and HylaFax

Title: Mensaje Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Any idea woluld be great idea? regards, srsergio --

RE: [Asterisk-Users] Error when install E100P

Please, could you send us cat /proc/pci?. Could you compile libpri? Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ning Zhou Enviado el: martes, 23 de noviembre de 2004 16:10 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Error when

RE: [Asterisk-Users] hangup()???

Hi, this call is from? Zap channel, Capi channel or other channel? It is possible that you don't detect well hangup from incoming channel. Regards. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Altus Snyman Enviado el: lunes, 22 de noviembre

[Asterisk-Users] CAPI 0x3301 Problem

Hi all, I have a PBX working for a year with an Eicon Diva Server 4BRI. One day it was a storm and nothing occurs, but after a a few days I can't send and receive any calls. I have connected TEIs to Asterisk and other PBX and when I try to dial, I hear correct tone two times, but then line

[Asterisk-Users] app_icd compile problem

Hi all, I try to compile app_icd to test it but I can't compile it. I have installed asterisk 1.0.2 and I download ICD and put files into /usr/src/asterisk/apps/icd directory. I think that make.conf in icd directory is ok but when I try to compile icd I obtain next error: === Compile:

RE: [Asterisk-Users] Software SIP Phones

Hi, Voicemoil capabilities are in Asterisk. You can use Asterisk voicemail from any SIP Software. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ashling O'Driscoll Enviado el: miércoles, 17 de noviembre de 2004 18:28 Para:

RE: [Asterisk-Users] Chan zap not loaded(ast_pickup_call)

Hi all, I'm sorry, but I'm stupid because I haven't load res_parking.so. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: viernes, 10 de septiembre de 2004 9:35 Para: 'Asterisk Users Mailing List - Non

[Asterisk-Users] SIP Swissvoice de-register

Hi all, I'm trying to configure a swissvoice IP10S but after a minutes this phones appears like UKNOWN in sip show peers and it is unaccesible. This phone can make call but it can't receive calls. Any idea? Regards, srsergio ___ Asterisk-Users

RE: [Asterisk-Users] SIP Swissvoice de-register

SIP version IP10 SP v0.0.1 (Build 5) Regards, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Florian Overkamp Enviado el: lunes, 06 de septiembre de 2004 13:42 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users]

RE: [Asterisk-Users] Asterisk SIP between two networks

had same problem? Could anyone help me with this problem? Best regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: miércoles, 01 de septiembre de 2004 0:46 Para: 'Asterisk Users Mailing List - Non-Commercial

RE: [Asterisk-Users] Asterisk SIP between two networks

, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: miércoles, 01 de septiembre de 2004 12:51 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks

RE: [Asterisk-Users] Asterisk SIP between two networks

19:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks Sergio Serrano [EMAIL PROTECTED] wrote: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first

RE: [Asterisk-Users] Asterisk SIP between two networks

Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 It is a bug? Why if I put bindaddr=0.0.0.0 packet received by asterisk is broken? Could anyone help me? Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano

[Asterisk-Users] Asterisk SIP between two networks

Hi all, I have next configuration: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have

RE: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone

Title: Mensaje Push send after you number, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de James DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer Problems with Grandstream

RE: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn

Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication. Regards,

RE: [Asterisk-Users] debian install zaptel

Title: Mensaje It's more easy download tarball and compile it. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de YanEnviado el: jueves, 22 de julio de 2004 13:31Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] debian install zaptel Hi:

RE: [Asterisk-Users] Problems with festival

Title: Mensaje I have the same problem.I'm usinr asterisk-1.0-RC1. Anyone could help us? regards, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Dan FernandezEnviado el: viernes, 16 de julio de 2004 20:42Para: [EMAIL PROTECTED]Asunto:

RE: [Asterisk-Users] Chan_Capi 0.3.4a error

Try to compile with lastest CVS srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Martin List-Petersen Enviado el: jueves, 15 de julio de 2004 1:12 Para: [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Chan_Capi 0.3.4a error

RE: [Asterisk-Users] Help with chan_capi

, S.L. Sergio Serrano RevueltoRD Manager Avda. Juan López de Peñalver 17Edificio

[Asterisk-Users] Transfer with Budgetone

Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the

RE: [Asterisk-Users] Re: Transfer with Budgetone

: Transfer with Budgetone Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person

RE: [Asterisk-Users] Re: Transfer with Budgetone

I have just to talk with Grandstream and they say to me that they ar working in 3-way conferencing for BT-100 series. I hope they have FW soon. One question more? How can I do parking call with Budgetone. Before # works fine, but Now it doesn't work. -Mensaje original- De: [EMAIL

[Asterisk-Users] Codec G729 uninstall

Hi all, Are there any way to clean codec_g729b license ffrom Asterisk. I would like to clean a license to install other more big, but when I do ../codec_g729b/Registration --XX I obtain a segmentation fault. Any idea? srsergio

[Asterisk-Users] Chan_capi and modem-fax

Hi all, I have just put a message from a few days with a problem with CAPI hangup. I have noticed that line with 97% of hangs, is a line connected with a ATA286 with a modem-fax. Could it be the problem? Regards, srsergio ___ Asterisk-Users

RE: [Asterisk-Users] multiplle isdn card

First thing you must is read next url http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO and if you hav done this, please attach your capi.conf. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de massimo Enviado el: martes, 04 de mayo de 2004

[Asterisk-Users] CAPI Eicon Diva Server 4BRI

Hi all, I have a PC working with a DIVA Eicon Server 4BRI during a lot of time. Now I can't make call but I can receive calls. I load diva with command: divactrl load -c 1 -f ETSI -u -t 0 Country: Spain Isdnmode: point to point My capi.conf is the next: [global] mode=immediate

RE: [Asterisk-Users] CAPI Eicon Diva Server 4BRI

? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Revuelto Enviado el: miércoles, 19 de mayo de 2004 12:00 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] CAPI Eicon Diva Server 4BRI Hi all, I have a PC working

[Asterisk-Users] chan_capi Fax

Hi all, I would like to know if chan_capi is prepared to receive faxes. I have a eicon deiva server 4bri with chan_capi and Grandstream HandyTone connected to a Fax, but this fax can't receive faxes. Any idea? Thanks, srsergio ___ Asterisk-Users

RE: [Asterisk-Users] Asterisk + GrandStream SIP phones

Try to add a qualify= to sip.conf, and try to exec a sip show peers. In spite of phones appears like register, if you use NAT, your firewall can cut communication. Try the next: Just after phone register call to it, and then wait for a minutes and try to call again. Could you call first time

[Asterisk-Users] G.729 and SCSI

Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED]

RE: [Asterisk-Users] G.729 and SCSI

] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki

[Asterisk-Users] CAPI

Hi, I have a problem with a Eicon Diva Server 4 BRI. I have 4 BRI ISDN and 11 number for these 4 ISDN. At first I have connected one of these 4 ISDN. When I try to call I receive the next trace: -- Executing ChanIsAvail(SIP/716-b0cd,

RE: [Asterisk-Users] PCphoneline FXO to FXS box??

We are going to do this test next week. I will say the result Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jim Rosenberg Enviado el: domingo, 29 de febrero de 2004 1:15 Para: Asterisk Asunto: [Asterisk-Users] PCphoneline FXO to FXS

RE: [Asterisk-Users] GS Budgetone 101 canot receive calls

If your BG 101 is in intranet, try to adjust your qualify parameter to 60. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew B Marlowe Enviado el: viernes, 27 de febrero de 2004 2:08 Para: [EMAIL PROTECTED] Asunto: RE:

[Asterisk-Users] SIP Extrange Problem

Title: Mensaje Hi all, For a few days we have a veryextrange problem. We have an intranet with Budgetone and others SIP Phones. In the extranet We HaveBudgetone Phones. The whole system was working well between the extranet and the intranet until a few days ago. When we try to speak

[Asterisk-Users] RE: [Asterisk-Users] Spanish indications configurationº

www.avanzada7.com Sergio Serrano RevueltoRD Manager Avanzada 7

RE: [Asterisk-Users] Eicon Diva Server

Hi all, I will ptobe your answers tomorrow. I'll say the results. Thanks for all. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sascha Knific Enviado el: martes, 10 de febrero de 2004 22:08 Para: [EMAIL PROTECTED] Asunto: AW: [Asterisk-Users]

[Asterisk-Users] Eicon Diva Server

Hi all, anyone could help me with capi.conf?. I have installed an Eicon Diva Server 4BRI. I have 2 EuroISDN BRI lines, First line number: 951014943 Second line number: 951014944 I try to do 4 calls but, I can't do more than two call. My capi.conf is the next:

RE: [Asterisk-Users] Call recording

You must use Monitor Application Happy New Year, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Edoardo Borghesi [fabbricadigitale] Enviado el: viernes, 02 de enero de 2004 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Call recording

RE: [Asterisk-Users] RxFAX application

PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Masakazu Nakano Enviado el: domingo, 21 de diciembre de 2003 5:37 Para: [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] RxFAX application Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 Sergio Serrano Revuelto [EMAIL PROTECTED

[Asterisk-Users] RxFAX application

Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetMusicOnHold(Zap/1-1, random) in new stack -- Executing

[Asterisk-Users] Asterisk as SIP Server

Hey Srs. I have a little problem with the next scenario: Internal Phone(801)--Asterisk(public IP) --INTERNET--ADSL Router--Budgetone(716) |--ADSL Router--Budgetone(717) My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr =

RE: [Asterisk-Users] Still TDM400P problem

Next configuration must work: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de JanM Enviado el: jueves, 20 de

RE: [Asterisk-Users] Open Source Linux PBX!

Title: Mensaje try to cvs srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Quan Le TrungEnviado el: jueves, 13 de noviembre de 2003 10:43Para: [EMAIL PROTECTED]CC: [EMAIL PROTECTED]; [EMAIL PROTECTED]Asunto:

RE: [Asterisk-Users] 2 X100Ps give error

Try to load module manually: modprobe wcfxo; ztcfg - srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Cameron Palmer Enviado el: viernes, 31 de octubre de 2003 6:27 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] 2 X100Ps give error I

RE: [Asterisk-Users] 2 X100Ps give error

or address (6) cameron. On Fri, 31 Oct 2003, Sergio Serrano Revuelto wrote: Try to load module manually: modprobe wcfxo; ztcfg - srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Cameron Palmer Enviado el: viernes, 31 de

[Asterisk-Users] Channel Bank with E1

I need connect up to 100 analog phone to a H.323 network through *. I think use TE410P, But I need to know what channel bank is better. I use E1 lines Any idea? Thanks in advance, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de DUSTIN WILDES

RE: [Asterisk-Users] ISDN BRI card

Title: Mensaje AVM Fritz it good for Asterisk. A little difficult to configure but not impossible. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tomica CrnekEnviado el: jueves, 16 de octubre de 2003 14:36Para: [EMAIL PROTECTED]Asunto:

[Asterisk-Users] SIP phone hangs after some hours

Hi, I have a problem with sip.conf. After some hours my sip phone(netergy) hangs. In clonse appears the next logs repeatly: 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.155 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80 From: asterisk sip:[EMAIL

RE: [Asterisk-Users] Using Asterisk in an netted scenario

Title: Mensaje Yes yo can do it. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de e-smithEnviado el: miércoles, 24 de septiembre de 2003 15:02Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Using Asterisk in an netted scenario Hi, Just

RE: [Asterisk-Users] how to dial a h323 destination ?

exten=XXX,1,Dial(h323/3|17|tTm) srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el: martes, 23 de septiembre de 2003 11:07 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] how to dial a h323 destination ? Please, can

RE: [Asterisk-Users] how to dial a h323 destination ?

Could you send me your h323.conf and you gnugk.ini? Sergio Serrano Revuelto Responsable de Consultoría Avanzada 7, S.L. Teléfono / Fax: +34 951 01 49 47 / +34 951 01 09 22 www.avanzada7.com -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger

RE: [Asterisk-Users] how to dial a h323 destination ?

Try to add gwprefix in oh323.conf after your alias. You must know that you can configure * gw in gnugk.ini or in oh323.conf. I recommend you put in your oh323.conf. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el:

RE: [Asterisk-Users] how to dial a h323 destination ?

-Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Sergio Serrano Revuelto Gesendet: Dienstag, 23. September 2003 12:27 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] how to dial a h323 destination ? Try to add gwprefix in oh323.conf after your alias

RE: [Asterisk-Users] ISDN BRI hardware

You can try AVM FRITZ with chan_capi from kapejod. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de YO Internet Information Enviado el: lunes, 22 de septiembre de 2003 0:03 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] ISDN BRI hardware

[Asterisk-Users] SIP Registration NOTIFY EVENT

Hi all, when I try register my netergy SIP Phone with *, I can't do it due to the next message: 1 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f To:

[Asterisk-Users] SIP stage

Title: Mensaje Hi, I would like to configure a stage for SIP phones. This stage would be the next: two netergy SIP phones connected to Asterisk through chan_sip. one X100P or AVM FRITZ to outside lines. I think that sip.conf would be the next: ;; SIP Configuration for

RE: [Asterisk-Users] SIP registration

I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70

RE: [Asterisk-Users] SIP registration

];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0

RE: [Asterisk-Users] SIP registration

];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0

RE: [Asterisk-Users] chan_capi and poor voice quality

HI, I am probing chan_modem_i4l again with AVM FRITZ but I can hear nothing in phone outside of asterisk, I explain Phone 1-- AVM_FRITZ--Asterisk-- Phone 2 From Phone1 to Phone 2 I can hear, but From phone 2 to phone 1 I can't hera nothing. Any idea? srsergio -Mensaje original- De:

[Asterisk-Users] CDR question

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

[Asterisk-Users] Two Question

Hi, I would like to know how do two things. First, it is possible simulate PBX scenary?, I explain. I would like that when an user press 9(outgoing key) asterisk will generate a new dial tone in H.323 EP and then the user could press number for dial. Second, it's possible modify time interdigit.

[Asterisk-Users] Two Question

Hi, I would like to know how do two things. First, it is possible simulate PBX scenary?, I explain. I would like that when an user press 9(outgoing key) asterisk will generate a new dial tone in H.323 EP and then the user could press number for dial. Second, it's possible modify time interdigit.

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