Re: [Asterisk-Users] Asterisk as an internal pbs for a samall company
On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk as an internal pbs for a samall company
Well, U right, many missing informations. The case is quite simple(I guess), we have dids, and each call to these dids has to be routed to the right handset thru Asterisk, no Ivr at this time, at least an answering machine in case of busy or not available users. For the rest, we need to be able to have external calls to pstn, or even to other sip phones form other providers. Is that enough? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de trixter aka Bret McDanel Envoyé : mercredi 2 novembre 2005 13:48 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk as an internal pbs for a samall company On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let
me know).
You will want voicemail for each user. This is configured in
voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of
them as well.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in
from the DID it will then dial the appropriate user and if busy/no
answer goto voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail
extension defined in voicemail.conf and 18005551212 is the extension
that a given did goes to (ie last part of the register line).
Hope this helps
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Re: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company
Thankyou, this was a great primer for me also.
Chris
trixter aka Bret McDanel wrote:
On Wed, 2005-11-02 at 13:58 +0100, Olivier Taylor wrote:
Well,
U right, many missing informations.
The case is quite simple(I guess), we have dids, and each call to these dids
has to be routed to the right handset thru Asterisk, no Ivr at this time, at
least an answering machine in case of busy or not available users.
For the rest, we need to be able to have external calls to pstn, or even to
other sip phones form other providers.
Is that enough?
Not for 100% setup, but enoughto at least get you started. From what I
understand this is what it appears you want (I may be wrong, if I am let
me know).
You will want voicemail for each user. This is configured in
voicemail.conf
http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf
You will need to edit sip.conf for the voip provider (register and
context) and if the office workers use sip to asterisk one for each of
them as well.
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
Lastly you will want to create a dialplan so that when a call comes in
from the DID it will then dial the appropriate user and if busy/no
answer goto voicemail. This is done from extensions.conf.
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf
You may want a macro like:
[macro-dialvmb]
exten = s,1,Dial(${ARG1},20,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup
Then for each inbound DID something like:
exten = 18005551212,1,Macro(dialvmb,SIP/user1,1234)
where user1 is the user defined in sip.conf, 1234 is the voicemail
extension defined in voicemail.conf and 18005551212 is the extension
that a given did goes to (ie last part of the register line).
Hope this helps
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RE: [Asterisk-Users] Asterisk as an internal pbs for a samall company
Why do you want to use a SIP provider instead of a PSTN connection? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Taylor Sent: Wednesday, November 02, 2005 4:59 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE : [Asterisk-Users] Asterisk as an internal pbs for a samall company Well, U right, many missing informations. The case is quite simple(I guess), we have dids, and each call to these dids has to be routed to the right handset thru Asterisk, no Ivr at this time, at least an answering machine in case of busy or not available users. For the rest, we need to be able to have external calls to pstn, or even to other sip phones form other providers. Is that enough? -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de trixter aka Bret McDanel Envoyé : mercredi 2 novembre 2005 13:48 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Asterisk as an internal pbs for a samall company On Wed, 2005-11-02 at 13:36 +0100, Olivier Taylor wrote: Hello all, We'd like to use asteriek as an internal pbx connected to an external sip provider to make outbound/inbound calls to pstn. We have the provider and have installed an asterisk at the office. Does anyone have a sample config? We need 25 telephone numbers(dids), to be registerd to the provider and be able to ceceive calls. Any advice is welcome. Sorry for the noob question, Olivier What you want to do depends largely on what you want to do. While that seems like a cylic statement I will try to explain. You have said that you want to route calls between your asterisk box and the PSTN via a VoIP provider that you have. So far that seems simple, but how are those calls going to go bewteen the office workers and asterisk? You will need configurations for that. How are the inbound calls going to be routed? Via an IVR? Well you will have to configure that. There is a lot of information that is missing from this setup. www.voip-info.org has a lot of asterisk examples including configuration files. You may find something there that does what you want. I cant easily help you solve this problem (and suspect that no one else can either) until you provide more information on exactly what you want. If you wish to discuss this offl ist feel free to email me directly. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.12.7/155 - Release Date: 11/1/2005 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
