RE: [Asterisk-Users] hierarchical VoIP system
You should take a look to ENUM protocol: http://www.voip-info.org/wiki/view/ENUM. It could provide a decentralized and simple solution for your requirements. Regards -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Joao Pereira Enviado el: miércoles, 30 de noviembre de 2005 18:45 Para: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Asunto: [Asterisk-Users] hierarchical VoIP system Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want 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 Pereira www.fccn.pt ___ --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] threewaycalling
Hello, It's possible to provide threewaycalling service in asterisk (nor in terminals) for SIP users? I would like to be able to join to calls in a threewaycall sending some dtfm. Thank you in advance for the information G.
RE: [Asterisk-Users] Using Ser and Asterisk together
Hi,
It's possible, we're using a configuration like that.
1. Configure diferent sip listening ports for SER (/etc/ser/ser.cfg) and
Asterisk (/etc/asterisk/sip.conf).
2. Configure SER (/etc/ser/ser.cfg) for forwarding calls based in
destination. For example adding:
if (uri=~^sip:[EMAIL PROTECTED]) {
forward( 10.10.10.10, 5070 ); //Where local asterisk is
listening
break;
}
(Documentation in SER admin guide
http://www.iptel.org/ser/doc/seruser/seruser.html)
3. Configure Asterisk as PSTN gateway. I haven't experience in this point.
Good luck.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Aiden Chew
Enviado el: martes, 25 de mayo de 2004 9:21
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Using Ser and Asterisk together
Hi all,
I would like to know if it is possible to use asterisk and ser together in a
single computer system using ser as a sip proxy and forwarding any voice
call request to asterisk for calling into the pstn gateway. (or any other
alternative that is possible is also welcomed for suggestions). If it is
possible can someone kindly show me the necessary configuration files or
refer me to any page that can show me how to do it ? Thanks a lot in
advance.
Kevin
__
Do You Yahoo!?
Log on to Messenger with your mobile phone!
http://sg.messenger.yahoo.com
___
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] max asterisk load
Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G ___ 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] max asterisk load
We would like to use asterisk as a SIP phone PBX with voicemail support only. The hardware could be a typical modern PC. Thank you very much. .G -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de mattf Enviado el: martes, 17 de febrero de 2004 16:48 Para: '[EMAIL PROTECTED]' Asunto: RE: [Asterisk-Users] max asterisk load That is extremely depandant upon what you want to do. First we have to know what the job of the asterisk server is, will it be an inbound ACD, a SIP phone PBX, a T1-only IVR, a conference call system, a voicemail system, and office phone system with H323 phones, etc.. Also, we need to know what hardware you were planning on using. The load is extremely depandant upon what you are doing with it. For example, a simple IVR/Zap-T1-channels-only system can handle 10 times the number of consecutive calls of a SIPZap conference call system(at least in my experience). Let us know what you plan on doing with it and we'll give you our best guesses as to the capacity. MATT--- -Original Message- From: Gustavo García Bernardo [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 17, 2004 9:54 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] max asterisk load Hi, We're evaluating asterisk, somebody has measured maximum asterisk load (simultaneus calls, calls per seconds...)? Are there any stimation? Thx. Best regards. .G ___ 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 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] LDAP authentication
Hi, I'm using SIP channel, and i would like to authenticate users with a LDAP server. Is this feature implemented in Asterik? I have read some posts about it, but i don't know if it's currently available. Thank you very much. .G ___ 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] SIP Messages (SIMPLE)
Hi How can i configure Asterisk for proxing SIP/SIMPLE Messages when the target is registered? How can the user retrieve the waiting-messages? Thank you very much. .G ___ 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
