[asterisk-users] enable eyeBeam to accept only one call

call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao Pereira -- __ João Gomes Pereira FCCN Av. do Brasil, nº 101 1700-066 Lisboa tel: +351 218 440 100 - fax: +351 218 472 167 email|SIP: [EMAIL PROTECTED] http

[asterisk-users] multiple PBXs in one box

. Each PBX will have its extensions and outbound/inbound routes... but everything in only one Asterisk. Is this possible? How can I implement it? Creating different contexts? Should I use a special software together with Asterisk? Thanks Regards Joao Pereira

[asterisk-users] dial-out call queue

Is it possible to implement a dial-out call queue in Asterisk? My idea is to give Asterisk a list of numbers, and then he makes the calls and delivers the calls to a call queue. Then, the agents will answer the calls. Is this possible? Thanks Regards Joao pereira

[asterisk-users] Asterisk crash and debug

Regards Joao Pereira ___ 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

Re: [asterisk-users] Call Center SoftPhone with Auto Answer

on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11

Re: [asterisk-users] Call Center SoftPhone with Auto Answer

But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Regards Joao Pereira Thiago Maluf wrote: Ola Joao, tem um modo do Asterisk fazer isso sim. Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações. Abs! Hi List

[asterisk-users] Call Center SoftPhone with Auto Answer

Hello I need a Softphone with auto answer where users can't turn it off. Does someone knows a softphone where users can't turn the auto answer off? Or is there any way Asterisk could force the clients to answer the phone? Thanks Regards Joao Pereira

[asterisk-users] asterisk SIP domain (in LAN or DMZ)?

local network and put a SIP Proxy (like Openser) in the DMZ to implement the SIP domain? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] compile problem with wavelenght

] asterisk-1.2.10]# Whats happening? I already tried with 3 different versions downloaded from asterisk.org site. Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] compile problem with wavelenght

problems in the future :P Thanks regards Joao Pereira Tzafrir Cohen wrote: On Thu, Apr 12, 2007 at 10:25:37AM +0100, Joao Pereira wrote: Hello Im trying to install an old version of Asterisk. But it isnt working: when I run make install: gcc -pipe -Wall -Wstrict-prototypes

Re: [asterisk-users] Maximum retries exceeded on transmission

Hello Thanks a lot for your reply. Im now using asterisk-1.2.10 and the problem disappeared. Thanks regards Joao Pereira Edoardo Serra wrote: Same to me !! Calls from OpenSER to Asterisk It happens only with Asterisk versions = 1.2.14 I'm going to capture some traffic Tnx for help Regards

[asterisk-users] Maximum retries exceeded on transmission

process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] no reply to our critical packet

process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: xxx.xxx.xxx.xxx Thanks for the help Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

Re: [Asterisk-Users] Cell phone dock/switch as Asterisk FXO source

by the local user to PSTN line so, if it cant terminate VoIP calls into the PSTN, it cant forward VoIP calls to the Dock and Talk. Joao Dovid B wrote: There has been talk about it before and I think people have done it. Paging Sam Tam - Original Message - From: Joao Pereira

[asterisk-users] SNOM loses server registration

or 320 ? Thanks Joao Pereira ___ --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] Cell phone dock/switch as Asterisk FXO source

Do you know If its possible to do the same with Dock and Talk and an ATA GrandStream HandyTone 386? Thanks Joao Pereira Jonathan Attwood wrote: I use a Dock-n-Talk in conjuction with a Sipura SPA3000 Asterisk. Because I'm using Asterisk, I cannot use voice dialling, however inbound

Re: [Asterisk-Users] Siemens Gigaset SL75

Do you know if it has 802.1x authentication as it is defined in EDUroam ( http://www.eduroam.org/ ) ? I never found a WiFi phone working with 802.1x I tested ZyXel Prestige 2000 but the sound was bad and it doesnt support 802.1x :( Thanks Joao Pereira [EMAIL PROTECTED] wrote

[asterisk-users] matching the beginning of an EXTEN

Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX ? I want match the first 9 digits does Asterisk has any function for this? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided

Re: [asterisk-users] PRI to SIP

stability 3. You can also use a dedicated router (ex: Cisco) to do that.Its more expensive, but more reliable. Regards Joao Pereira Patrick Fortin wrote: Hi Can someone recommend a PRI to SIP Box that work well with asterisk We are presently testing with a Patton Smartnode 2400 but we

Re: [asterisk-users] matching the beginning of an EXTEN

perfect!!! its now working this way: exten = _.,4,GotoIf($[ ${EXTEN:0:9} = 351217588] ? 20:10) Thanks a lot Joao Pereira Ove Aursand wrote: Use ${EXTEN:0:9} Regards, Ove Joao Pereira wrote: Hello how can I distinguish all the calls that arrive to my Asterisk starting with: 351217588XXX

Re: [asterisk-users] how to define a secure trunk

Can I do the encrypted trunk in SIP? Does Asterisk supports it? Thanks Joao Pereira Pavel Jezek wrote: http://www.voip-info.org/wiki/view/IAX+encryption Joao Pereira wrote: Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its

[asterisk-users] how to define a secure trunk

the trunk in SIP, IAX or something else? Thanks Joao Pereira ___ --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] Snom or Cisco Phones?

). Are there any features that Snom has, that Cisco doesnt? And are these features important? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] defining trunks in sip.conf

1000 and to register in domain.pt I already saw the manuals but the trunks arent still working :( Can someone help me? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] SIP client with video???

The problems with X-Lite 3 are: - just accepts one SIP registration - doesnt send video to other X-Lite or eyeBeam versions - sometimes loses the SIP informations when you reboot the PC . Regards Joao Pereira Blake Krone wrote: What's wrong with X-Lite 3.0? I haven't had any issues

[asterisk-users] Asterisk video support

Hello to all I used SER for SIP calls with video, but now Im trying the same in Asterisk and It doesnt work. I m using X-Lite 3.0 (the same that worked with SER). Do Asterisk needs any special configuration to allow SIP calls with video between its clients? Regards Joao Pereira Asterisk's

[asterisk-users] SIP client with video???

Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___ --Bandwidth and Colocation provided

[Asterisk-Users] planet VIP 152 T

Hello to all Im testing a Planet 152 T phone and Im having some problems. Can someone tell me if this phone does URI dialing? And does it work behind NAT (does it need any special configuration on the SIP server)? Thanks Joao Pereira ___ --Bandwidth

[Asterisk-Users] regexp issue

...) exten = _.,7,Macro(uridial,[EMAIL PROTECTED]) exten = _.,8,HangUp() exten = _.,10,Goto(custom-noturi,${EXTEN},1) exten = h,1,HangUp() How can I say that this code is just for calls to foreign domains? Something like:if (SIPDOMAIN != fccn.pt) Regards Joao Pereira

Re: [Asterisk-Users] using a billing system

php-odbc-4.3.9-3 php-pgsql-4.3.9-3 php-4.3.9-3 php-pear-4.3.9-3 Thanks Joao Pereira Vahan Yerkanian wrote: exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI, a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I

Re: [Asterisk-Users] using a billing system

-numbers and prepaid-invalid-digits are already in the /var/lib/asterisk/mohmp3/acc_* dirs Can you give me a help to understand whats the problem? Thanks Joao Pereira Vahan Yerkanian wrote: Greetings, pcntl is a required module for a2billing. It is vital for ensuring the call

Re: [Asterisk-Users] using a billing system

I think Asterisk2Billing is trying to play some audio file to make the callers put a PIN number. But can I use it without the PIN, and configure Asterisk2billing to check the database to see if the user exists? Thanks Joao Pereira Vahan Yerkanian wrote: Greetings, pcntl is a required

Re: [Asterisk-Users] How to strip a digit

you need to put :1 next to ${EXTEN} something like: exten = _91NXXNXX,1,AGI(call_log.agi,${EXTEN:1}) exten = _91NXXNXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o) exten = _91NXXNXX,3,Hangup Joao Pereira Erick Perez wrote: I have the following extension to dial outside via SIP it's

Re: [Asterisk-Users] using a billing system

= _,5,Hangup but when I place the call, he fails to authenticate with my-telco :( How can I use the registration information that is in sip.conf and continue to use Asterisk2Billing ? Thanks Joao Pereira William Piper wrote: You need to specify which context to use

[Asterisk-Users] using a billing system

Regards Joao Pereira ___ --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: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

they want us to buy. In Portugal I already did 3G VoIP calls from TMN and Vodafone. I would really like to try this phone :) Regards Joao Pereira Steve Kennedy wrote: On Tue, May 23, 2006 at 02:50:33AM +0800, Sam Tam wrote: Well it is incorrect to say that. In places like USA or London

Re: [Asterisk-Users] Best VoIP provider for Asterisk

Hi I dont know if it's the best, but for Portugal and to place calls throwout Europe, www.startel.pt has a good service. Regards Joao Kerry Garrison wrote: Depends on your location and your requirements. A generic post like this generally turns into a flame war. Please be MUCH more specific.

[Asterisk-Users] [EMAIL PROTECTED] doing SIP URI calls

:5060: SIP/2.0 404 Not Found If I have _. in [from-internal-custom] why do the call fails? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] LDAPget

Hello to all Im using [EMAIL PROTECTED] 2.7 and I would like to do LDAP querys. Can I simply use LDAPget or do I need to install Asterisk::LDAP from Alkaloid Networks? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] do extensions must be numbers in [EMAIL PROTECTED]

Hello to all In Asterisk, SIP clients can be registered with numbers (2001, 2002, ...) or with names (manuel, maria,...). But [EMAIL PROTECTED] only allows SIP registers to be done with numbers... Is there any way of register SIP users with names and then give them a numeric alias? Thanks

[Asterisk-Users] SIP domain in Asterisk

(and the help of SRV records) -the possibility of dialing [EMAIL PROTECTED] and route the calls through the Internet Can this be done? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] CDRs and billing

in the CDR table? Or they connect directly to Asterisk? Or is Asterisk that has, before the Dial command, to put the information on the Asterisk2Billing tables? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

Re: [Asterisk-Users] CDRs and billing

Ok, no problem, Ill do it with the AGI. Do I need to re-compile asterisk to support the AGI writing? or it goes by default? Thank you Joao Pereira Chris Mason (Lists) wrote: Joao Pereira wrote: Hello I configured Asterisk to put CDRs in the database like it was explained in: www.voip

[Asterisk-Users] billing with PostgreSQL

Hello to all Im looking for a billing tool for Asterisk, that works with PostgreSQL. All the tools I found in www.asteriskbilling.com just work with MySQL :( Do you know a nice billing tool for Asterisk with PostgreSQL? Thanks Joao Pereira

[Asterisk-Users] the best billing tool for Asterisk

Hello to all I would like to know some opinions of people that are using billing tools for Asterisk. Can you please advise me in wich billing tool to I use? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] Routing SIP calls via URI

But is there a way of doing this without a prefix? because people should dial without prefixes: [EMAIL PROTECTED] , not like: [EMAIL PROTECTED] How can we make this without a prefix? something like: if( !uri=~@mydomain.pt ){ forward the all to the Internet } :) Thanks Joao Pereira Shad

[Asterisk-Users] Asterisk behind NAT

to the Internet? And he can still receive calls from the Internet? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [Asterisk-Users] Asterisk behind NAT

successfully a SIP domain in Asterisk behind NAT? Thanks Joao Pereira Kerry Garrison wrote: Yes. In Sip.conf you need the following lines: externip=xxx.xxx.xxx.xxx ; put public ip address here localnet=192.168.10.0/255.255.255.0 ; edit as appropriate In your firewall, add the following mappings

Re: [Asterisk-Users] OT - force Cisco phones to reboot

I dont have this cisco-check-cfg exten command in my asterisk... Did you installed some extra module or channel? Thanks Joao Pereira Aaron Daniel wrote: It really depends on the number of phones you're wanting to reboot. Whenever we do a reconfiguration of our phones, I have a script

[Asterisk-Users] OT - force Cisco phones to reboot

Hello to all Does someone knows how to force the Cisco IP phones (7940 and 7960) to reboot weekly or monthly? I think this would be useful because sometimes we change the configuration settings in the TFTP, but the phone just check the TFTP when he restarts... Thanks João

Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)

And about the 802.1x ? The phones can work as passthrough and force the PC to use 802.1x ? What configuration do we put in the switches? Do we put the switch as access (with 802.1x) or trunk (without 802.1x) ? Thanks Joao Pereira Greg Oliver wrote: It actually depends on the switch model

Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs(with 802.1x)

Ok, but the PC has an 802.1x client that configures the VLAN when he authenticates. Is this going to pass through the phone? And will the switch accept it? Thanks Joao Pereira Wojciech Tryc wrote: Your pc has to able to support tagged vlans. The switch on the phone will pass through both

[Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)

? Thanks Joao Pereira ___ --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] Deploying VoIP on a WAN

://www.juniper.net/solutions/literature/solutionbriefs/351085.pdf) Can anyone drop me some lines about this? I urgently need some feedback on this. Thanks! Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] adress book

? Thanks Joao Pereira ___ --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] PBX making ENUM lookups

) Thanks Joao Pereira ___ --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] ENUM trees

just exist one ENUM root? Thanks Joao Pereira ___ --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] hierarchical VoIP system

SIP redirects as the central server then can handle a lot of calls as its only doing the routing decisions. Best regards jan --On Wednesday, November 30, 2005 05:45:21 PM + Joao Pereira [EMAIL PROTECTED] wrote: Hello Im managing a WAN with a lot of Universities. Some of them already

[Asterisk-Users] hierarchical VoIP system

that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao

[Asterisk-Users] voicemail clients

Hello to all I have clients registered with names (joao, manuel, etc...) and clients registered with numbers (123, 120,...). To make the number clients receive voicemail, I have this: exten = _X,1,Answer exten = _X,2,Wait(1) exten = _X,3,VoiceMail(u${EXTEN}) exten =

[Asterisk-Users] Voicemail configuration

) exten = pereira,5,Hangup But how do I force this rule to be applied to all calls? instead of writing these 5 lines for each of my clients ? If I used numbers, I could do _ ... but how do I write the rule for client names? Thanks Joao Pereira

Re: [Asterisk-Users] OT: SIP firmware image for Cisco 7940 or 7960

You can download a new SIP firmware and force the Cisco IP phone to use it. Some interesting links about it: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Joao

[Asterisk-Users] Cisco phones port range

, and assumes the defaults (16384-32766). Even when I put these ports directly in the phone configuration, he doesnt accept them. How can I change the RTP ports in the Cisco IP phone? ( Like in Xlite we do: System Settings- Network - Listen RTP port ) Thanks Joao Pereira

Re: [Asterisk-Users] Eicon Diva Server query

These cards are very good, the only problem is the price... I bought one Diva Server 4BRI for 1300 Euros... its a lot... The configuration of the board is a bit hard but check this link for help: http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI Joao Armin Schindler wrote:

[Asterisk-Users] Cisco IP phone NAT config

Hello, I have SER in bridging mode with two IPs (private and public). To dial the world, my Cisco IP phones must contact the SER private IP, and the call is then proxyed by SER. All other SIP clients can do it, but the Cisco phones dont What should I put in the configuration file? For now I

[Asterisk-Users] Illegal redirection

Asterisk, and it says Illegal redirection 10.0.0.135-10.0.11.240. How can the firewall know that the INVITE was going to be redirected by Asterisk to PhoneB(10.0.11.240) Joao Pereira ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk

Re: [Asterisk-Users] dialplan defenition (closer)

the 74 part is being eaten somewere. Joao Pereira Armin Schindler wrote: On Wed, 10 Aug 2005, Joao Pereira wrote: Ok, I m getting to the point, This route: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Isn't working because the dialed number isnt maching _74XXX I putted Asterisk in capi

Re: [Asterisk-Users] dialplan defenition (goooooooal)

. Cheers Joao Pereira Joao Pereira wrote: The IP - pbx extension calls are already workin fine. Now Im just configuring the pbx extension - IP calls this way: [pbx extensions] --- [SIEMENS PBX] [ASTERISK] --- [SER] --- [sip clients] Thats why the Dial is for SIP only. Now Im going to try

Re: [Asterisk-Users] dialplan defenition

this is realy close: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) because it seems that is everything right... but It always answer: pbx.c:1877 ast_pbx_run: Channel 'CAPI[contr1/118]/1' sent into invalid extension 's' in context 'default', but no invalid handler Joao Pereira Moises Silva

Re: [Asterisk-Users] dialplan defenition

explicitly write Dial(SIP/[EMAIL PROTECTED],30,r) ?? João Matt Riddell wrote: Joao Pereira wrote: Hello list, Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line

Re: [Asterisk-Users] dialplan defenition

yes, I know, in my extensions.conf is writen correctly. Thanks Joao Bryce Chidester wrote: On Wed, 2005-08-10 at 15:51 +0100, Joao Pereira wrote: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Just an observation that you have an invalid address there; you have 1193 instead

Re: [Asterisk-Users] dialplan defenition (closer)

Ok, I m getting to the point, This route: exten = _74XXX,1,Dial(SIP/[EMAIL PROTECTED],30,r) Isn't working because the dialed number isnt maching _74XXX I putted Asterisk in capi debug mode and when I dial 74118 he says: gnugk*CLI capi debug CAPI Debugging Enabled -- CONNECT_IND ID=001

[Asterisk-Users] dialplan defenition

couldnt find it until now... Thanks Joao Pereira ___ 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

Re: [Asterisk-Users] dialplan defenition

and route the call to [EMAIL PROTECTED] Thanks Joao Christian Victor wrote: Joao Pereira schrieb: Im writing my dial plan, in witch every SIP phone begins with 74 and has more 3 numbers (like 74XXX). So, I want to route all 74XXX calls to my sip channel. For this I wrote this line: exten

[Asterisk-Users] Asterisk as Gateway

${EXTEN}|90 Does someone have an ideia of what is missing? The Siemens PBX should forward the call to its 116 extension... but there's no way I can debug it... Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Diva Server 4BRI + Asterisk ------(QSIG)------ PBX

. not in CAPI. Does someone have a solution for this? Are any of my assumptions wrong? Did someone ever putted a Diva Server with Asterisk and QSIG? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

[Asterisk-Users] ETSI or QSIG

Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? I had my Asterisk connecting to a Siemens PBX with ETSI and it was working fine, but peolpe said to me that QSIG could implement more features and

Re: FW: [Asterisk-Users] ETSI or QSIG

] [mailto:[EMAIL PROTECTED] Behalf Of Joao Pereira Sent: Wednesday, July 06, 2005 7:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ETSI or QSIG Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have

Re: [Asterisk-Users] ETSI or QSIG

-07-06 at 15:05 +0100, Joao Pereira wrote: Hello to all, Does Asterisk support QSIG? and the configuration is in capi.conf? and if it supports it, do you have samples of the configuration? QSIG is not an option in capi.conf. It is an option in the configuration of my Eicon Diva Server BRI

Re: [Asterisk-Users] ETSI or QSIG

you re using an Eicon Diva Server BRI, what are you using to connect? ETSI, QSIG or someting else? I use my Eicon Diva Server card to hook up my ISDN/BRI line through ETSI and capi.conf to asterisk. I had that configuration too, but isnt QSIG better? because QSIG can send the

Re: [Asterisk-Users] routing in extensions.conf

,DoStandardThings Of course, this is only the minimum, there are much more possibilities (especially if you want to do more than one thing in an extension). Bye Stefan sth==Originalnachricht== sthVon: Joao Pereira [EMAIL PROTECTED] sthDatum: 2005-04-22 18:25:17 sthAn: Asterisk Users Mailing List - Non

[Asterisk-Users] routing in extensions.conf

Hello all, Im using chan_capi to connect from a Siemens High Path to a Aterisk, when I call from the Asterisk clients to the Siemens PBX, it works, when I call from a Siemens client to a SIP(Asterisk) client, it doesnt work and says this: == Starting CAPI[contr1/930]/1 at default,930,1 failed

[Asterisk-Users] Diva Server configuration

Hello Can someone tell me how do I configure a Eicon Diva Server BRI with Asterisk? Should I use CAPI? And how do I tell Asterisk to use QSIG? Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com

Re: [Asterisk-Users] wiki down?

yeah. and it would me cool to come up more up to date. Joao Steve Totaro wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:

[Asterisk-Users] EICON DIVA prices

Hi to all my local reseller gave me this price for the Eicon DIVA server boards... Diva Server BRI-2M 749 Euros Diva Server 4BRI-8M ..1927 Euros Diva Server PRI E1/T1 3796 Euros I think that they are expensive. Is this the normal price? I just hope that Asterisk and my

Re: : [Asterisk-Users] QSIG, Asterisk and Eicon DIVA

Are you using a Diva SERVER board or just a Diva PCI? I also whant to connect Asterisk with a Siemens HH3000, but I whant to know if it can be done with an Eicon PCI or with a Digium board, because the Eicon DIVA Server 4BRI is very expensive. Joao Pereira - Original Message - From

Re: [Asterisk-Users] sip wifi phone?

I have 2 Zyxel Prestige and I m happy with them. In the beginning Its not very easy to use, but when you get used to It, Its nice and easy. The batery lasts long. He isnt so good behind NATs. Joao - Original Message - From: Kurt Fankhauser [EMAIL PROTECTED] To: 'Asterisk Users Mailing

[Asterisk-Users] softphone that registers in 2 or more SERs

Hi all Do someone know about a softphone that can register in 2 or more SIP servers? It would be useful for me to have a softphone registered in my company´s SER and in my nacional SIP server. I think X-lite can't do it. Thanks Joao ___ Asterisk-Users

[Asterisk-Users] RDIS board for gatewaying

Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a RDIS entry in the Siemens HiPath, but the PC with Asterisk doesnt have any RDIS board, can someone tell me about good and cheap PCI RDIS boards that supports QSIG? The Eicon boards

[Asterisk-Users] ISDN board for gatewaying

Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk doesnt have any ISDN board, can someone tell me about good and cheap PCI ISDN boards that supports QSIG? The Eicon

[Asterisk-Users] ISDN board for gatewaying

Hi all I want to connect Asterisk with my Siemens HiPath PBX, to use it as a Gateway to the PSTN. I already have a ISDN entry in the Siemens HiPath, but the PC with Asterisk doesnt have any ISDN board, can someone tell me about good and cheap PCI ISDN boards that supports QSIG? The Eicon

[Asterisk-Users] free pocketPC softphone (toshiba e750)

Hi all I have a pocketPC Toshiba e750 and I want to make SIP calls from it, but I didnt found any free softphones for my Toshiba. X lite's versions for pocketPC isnt free :( Did someone used before a free softphone for pocketPC? witch one? Thanks Joao Pereira www.fccn.pt

Re: [Asterisk-Users] Tie web application to VOIP

- Original Message - From: K J [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 24, 2004 10:06 PM Subject: [Asterisk-Users] Tie web application to VOIP I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users

[Asterisk-Users] B2BUA

Hello to all Im using SER as SIP registrar and Asterisk as GW and billing system but I m not sure if Asterisk can interupt calls when a client is out of credit. Is there any way of doing it or I need to use B2BUA ? Thanks Joao Pereira ___ Asterisk

[Asterisk-Users] Signaling / Streaming

Hi When I forward calls from SER (or GNUGK) to Asterisk, the SER ( or GNUGK) are just used for signaling, but the call streaming passes from the endpoint directly to Asterisk, isnt it? Or does the streming passes from the Endpoint to SER and then to the Asterisk? Thanks Joao Pereira

Re: [Asterisk-Users] Signaling / Streaming

. Regards Lamine - Original Message - From: Joao Pereira [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, January 07, 2005 10:21 AM Subject: [Asterisk-Users] Signaling / Streaming Hi When I forward calls

[Asterisk-Users] softphones

a way to convince peaple in my company to use them. Thanks Joao Pereira ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Re: [Serusers] softphones

- From: Walter Carter [EMAIL PROTECTED] To: 'Joao Pereira' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com; [EMAIL PROTECTED] Sent: Friday, January 07, 2005 3:17 PM Subject: RE: [Serusers] softphones Try Xten: http://www.xten.com/index.php

[Asterisk-Users] chan_cornet

? Thanks Joao Pereira - Original Message - From: Luís Palma [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 04, 2005 10:30 PM Subject: Re: [Asterisk-Users] connect Asterisk with Siemens HiPath HG1500 Hi

Re: [Asterisk-Users] chan_cornet

chan_cornet will change this. Are there any news about this project?Joao Pereira [EMAIL PROTECTED] wrote: Hibut did anyone have ever used a Siemens HiPath PBX with Asterisk?If you made it, please tell me how...I read that chan_cornet does exist...http://lists.digium.com/pipermail

  1   2   >