Re: [asterisk-users] sip and extensions
Hello, If you would like to make out bound call (from Asterisk to SIP provider), it is fine. But if you want have inbound call (from SIP provider to Asterisk). I think you are supposed to have something like this sip.conf register = 5552530146:your_password@sip3.voipvoip.com/5552530146 [5552530146] ... context=incoming extensions.conf [incoming] ;first creating extensions for your local users exten = 5552530146,1,Goto(5552530146_incomming,s,1) [5552530146_incomming] ;more logic wish it would help. On Jul 5, 2012, at 11:44 PM, Thomas Perron wrote: I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip and extensions
I am new. Here is the code that I am playing with on CentOS 6.x When I dial the number that corresponds w/ my SIP account I get a recording: reached a non-working number I built Asterisk a few times last year and am now back working on a similar project. In my view, there is something wrong in sip.conf I don't remember using a file that long to get a basic call set up. The format was provided to me by voipvoip.com (the SIP provider). Does anyone have any comments please? I just want a very simple config to get my machine to recognize a call to the SIP provider. Here is results of sip show registry: Hostdnsmgr Username Refresh StateReg.Time sip3.voipvoip.com:5060 N 5552530146 285 Registered Thu, 05 Jul 2012 21:39:56 1 SIP registrations. Here is sip and extensions.conf sip.conf [general] register = 5552530146:funnytiger...@sip3.voipvoip.com ; [sip3.voipvoip.com] [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no extensions.conf [general] ; ; [incoming] ;first creating extensions for your local users exten= s,1,Dial(SIP/1703717) exten= s,2,Hangup() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip and extensions
- Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip and extensions
Hi, I changed these codes to not coincide with actual account info. Thanks On Thu, Jul 5, 2012 at 5:48 PM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip phone extensions at a remote site
If you're using SIP I think what you want is canreinvite=yes which means the two remote user clients can talk directly to each other. Asterisk disappears from the loop which means no accounting. I think NAT causes problems in this scenario also. More details on the wiki Regards Cameron - Original Message - From: cmould [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, April 09, 2005 7:48 PM Subject: [Asterisk-Users] sip phone extensions at a remote site I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. ___ 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] sip phone extensions at a remote site
I am in the proscess of integrating a clients remote and head office phone systems. Currenty each office has their own PBX and trunk lines. I am recommending that they put in an Asterisk server at the Head office with a WAN link to the remote office and switch to IP phones. Trunk lines at the remote site would be returned to the TELCO. External calls over the PSTN from the remote office would be routed over the WAN to the head office and through Asterisk to the PSTN trunk lines. All phones would then become extensions (both remote and head office locations). I want Person A in the remote office to dial an extension number and get Person B in the head office. What I am unsure about is if person A and Person B are both at the remote site and Asterisk PBX is at the head office, can A and B talk directly to each pther without traversing the WAN link? Has anyone done this before? What is the quality of the call if they have? Any information is useful. begin:vcard fn:Carey Mould n:Mould;Carey org:E2 Systems Limited adr:237 Old hope Road;;Suite 11 12, technology Innovation Centre;Kingston;;Kgn 6;Jamaica email;internet:[EMAIL PROTECTED] title:CEO/Consultant tel;work:(876) 512-2680 x-mozilla-html:FALSE url:http://www.e2team.com version:2.1 end:vcard ___ 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
