Comments on: How to set up a SIP trunk in the Asterisk PBX http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx thoughts, more or less unorganized Fri, 06 Jan 2012 03:38:36 +0000 hourly 1 http://wordpress.org/?v=3.3.1 By: Jungli http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-19754 Jungli Fri, 06 Jan 2012 03:30:58 +0000 http://www.beardy.se/?p=451#comment-19754 Its really helpful for the beginners like me i love read yor articles bro. :) Its really helpful for the beginners like me i love read yor articles bro. :)

]]> By: tamaso http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-19106 tamaso Tue, 20 Dec 2011 03:49:47 +0000 http://www.beardy.se/?p=451#comment-19106 Hello, I have been having some serious problem trying to get my asterisk system to register with my sip provider. Could you please help me figure out why I am not able to connect to my sip provider? [general] register => username:password@sip.fooprovider.com [flowroute] ;keep this lowercase, do not change format type=friend secret=passworkd username=username host=sip.fooprovider.com dtmfmode=rfc2833 context=inbound ;change to 'ext-did' or 'from-trunk' for asterisk@home canreinvite=no allow=ulaw allow=g729 insecure=port,invite fromdomain=sip.fooprovider.com Your input on this will be much appreciated. Thanks Tamaso Hello,

I have been having some serious problem trying to get my asterisk system to register with my sip provider. Could you please help me figure out why I am not able to connect to my sip provider?

[general]
register => username:password@sip.fooprovider.com

[flowroute] ;keep this lowercase, do not change format
type=friend
secret=passworkd
username=username
host=sip.fooprovider.com
dtmfmode=rfc2833
context=inbound ;change to ‘ext-did’ or ‘from-trunk’ for asterisk@home
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
fromdomain=sip.fooprovider.com

Your input on this will be much appreciated.

Thanks
Tamaso

]]> By: beardy http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-13069 beardy Tue, 28 Jun 2011 21:24:00 +0000 http://www.beardy.se/?p=451#comment-13069 Magne, It sounds like you put your "register =>" first in the file? You can't do that, it has to be within the "[general]" section, within a context preferebly, like in my example above. Leon, The configuration for outgoing calls is the section under the "; Register and get calls from Foo Provider" comment in the sip.conf example above. The definition of "fooprovider" (an example provider) is directly underneath, in the "[fooprovider]" section. These are then used in extensions.conf to make outgoing calls. In my example above, any number dialed starting with a "9" and having more than 5 digits will be routed out to the provider (see the bottom of the example extensions.conf above). Magne,
It sounds like you put your “register =>” first in the file? You can’t do that, it has to be within the “[general]” section, within a context preferebly, like in my example above.

Leon,
The configuration for outgoing calls is the section under the “; Register and get calls from Foo Provider” comment in the sip.conf example above. The definition of “fooprovider” (an example provider) is directly underneath, in the “[fooprovider]” section.
These are then used in extensions.conf to make outgoing calls. In my example above, any number dialed starting with a “9″ and having more than 5 digits will be routed out to the provider (see the bottom of the example extensions.conf above).

]]> By: Leon http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-8586 Leon Tue, 05 Apr 2011 11:31:06 +0000 http://www.beardy.se/?p=451#comment-8586 Hi, I followed the instructions above. I can only received incoming call. What is the configuration for outgoing calls? Hi,

I followed the instructions above. I can only received incoming call.
What is the configuration for outgoing calls?

]]> By: Magne http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-7041 Magne Tue, 08 Feb 2011 02:51:15 +0000 http://www.beardy.se/?p=451#comment-7041 Hi, Nice article! But i got a problem. When i put register => blablabla at the top of sip.conf. I get a error message that the sip.conf is invalid. If i remove the line, it works. I know iam writing my line correct. What to do? Hi,

Nice article!

But i got a problem. When i put register => blablabla at the top of sip.conf. I get a error message that the sip.conf is invalid. If i remove the line, it works. I know iam writing my line correct. What to do?

]]> By: Gilles http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-6135 Gilles Wed, 15 Dec 2010 14:30:40 +0000 http://www.beardy.se/?p=451#comment-6135 Great article! I did have a problem getting it to work with my VOSP and Asterisk 1.4 (Asterisk and SIP clients behind a NAT router), though: In sip.conf, I had to have two sections (Outgoing and Incoming), and the Outgoing section had to be located before Incoming or I would get a BUSY signal when calling the VOSP number from a cellphone: <code> ;========== sip.conf ... register => myaccount:mypasswd@myvosp.com [vosp_outgoing] type=peer host=myvosp.com username=myaccount secret=mypasswd fromuser=myaccount fromdomain=myvosp.com nat=yes canreinvite=no [vosp_incoming] type=peer host=myvosp.com context=from_vosp nat=yes canreinvite=no [1234] type=friend context=my-phones secret=5678 host=dynamic qualify=yes nat=no ;========== extensions.conf [from_vosp] exten => s,1,Dial(SIP/1234) exten => s,n,Hangup() [my-phones] exten => 1234,1,Dial(SIP/1234) exten => 1234,n,Hangup() exten => _0.,1,Dial(SIP/vosp_outgoing/${EXTEN}) exten => _0.,n,Hangup() </code> I don't know the cause of this, but it took me a couple of hours to finally figure out why I could call out but not call in. Great article!

I did have a problem getting it to work with my VOSP and Asterisk 1.4 (Asterisk and SIP clients behind a NAT router), though: In sip.conf, I had to have two sections (Outgoing and Incoming), and the Outgoing section had to be located before Incoming or I would get a BUSY signal when calling the VOSP number from a cellphone:


;========== sip.conf
...
register => myaccount:mypasswd@myvosp.com

[vosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypasswd
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no

[vosp_incoming]
type=peer
host=myvosp.com
context=from_vosp
nat=yes
canreinvite=no

[1234]
type=friend
context=my-phones
secret=5678
host=dynamic
qualify=yes
nat=no

;========== extensions.conf
[from_vosp]
exten => s,1,Dial(SIP/1234)
exten => s,n,Hangup()

[my-phones]
exten => 1234,1,Dial(SIP/1234)
exten => 1234,n,Hangup()

exten => _0.,1,Dial(SIP/vosp_outgoing/${EXTEN})
exten => _0.,n,Hangup()

I don’t know the cause of this, but it took me a couple of hours to finally figure out why I could call out but not call in.

]]> By: MW http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-5735 MW Mon, 29 Nov 2010 18:57:13 +0000 http://www.beardy.se/?p=451#comment-5735 Great article! I really hope you'll write some more! Easy to understand and follow. I'd really like to see an article like this about all the config files and options. Bookmarked immediately! It's hard to find information that's to the point like this for us who don't have much experience. Thank You! Great article! I really hope you’ll write some more! Easy to understand and follow. I’d really like to see an article like this about all the config files and options. Bookmarked immediately! It’s hard to find information that’s to the point like this for us who don’t have much experience.

Thank You!

]]> By: Asterisk Server with SIP Account Routing to Cell Phones http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-5577 Asterisk Server with SIP Account Routing to Cell Phones Wed, 24 Nov 2010 10:27:26 +0000 http://www.beardy.se/?p=451#comment-5577 [...] about how to do this at the voip-info.org site: sip.conf and extension.conf). This other link seems to be a good example of what you [...] [...] about how to do this at the voip-info.org site: sip.conf and extension.conf). This other link seems to be a good example of what you [...]

]]> By: beardy http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-5046 beardy Thu, 09 Sep 2010 15:58:28 +0000 http://www.beardy.se/?p=451#comment-5046 Thank you for your comment Jp. You are absolutely right. I must have made a copypaste error or just not checking properly when assembling the full config at the end. I have corrected it now, the full config now corresponds to the rest. Thanks for observing and pointing it out, I appreciate it. I must have been tired when finishing the article. Apologies to anyone following it not getting correct information. I hope you return and see the correction. Thank you for your comment Jp.
You are absolutely right. I must have made a copypaste error or just not checking properly when assembling the full config at the end. I have corrected it now, the full config now corresponds to the rest.
Thanks for observing and pointing it out, I appreciate it. I must have been tired when finishing the article.

Apologies to anyone following it not getting correct information. I hope you return and see the correction.

]]> By: Jp http://www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx/comment-page-1#comment-4980 Jp Mon, 23 Aug 2010 10:57:29 +0000 http://www.beardy.se/?p=451#comment-4980 Is it just me or is there a discrepancy between the article and the full config files at the bottom? They don't seem to match up and the full config doesn't seem to have any rules for incoming? -jp Is it just me or is there a discrepancy between the article and the full config files at the bottom? They don’t seem to match up and the full config doesn’t seem to have any rules for incoming?

-jp

]]>