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[asterisk-users] SIP domain different than provider's


 
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asterisk at net153.net
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PostPosted: Fri Aug 21, 2015 12:53 am    Post subject: [asterisk-users] SIP domain different than provider's Reply with quote

Hello,

I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains before. My user '140@4354766787.com' is
handled by a 3rd party provider: 'sip.provider.com' and my local domain
on my asterisk box is the hostname 'mypbxdomain.com'.

My normal extension I use for everything is just '111'. I figured the
best way of joining my asterisk box was to just hard code in the
extensions I would need to dial for my remote office work (there are
only a couple of extensions so shouldn't be a big deal).

However I struggled to get authentication working for outgoing calls to
the few new extensions at the remote office through their provider.
Looking at debug logs it was clear that the sip 'To' address was wrong.
It had the provider: "To: <sip:139@sip.provider.com>" instead of the
domain which should look like: "To: <sip:139@4354766787.com>" (right?)

In the end, after hours of googling, reading the docs on sip.conf
several times revealed a little spoke of '!' dialplan option. Simply
changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to
'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.

But this seems really hackish. Is this the right/only way? Or is just
having a provider and mismatched domains not really the norm?

I have an an anonymized log here: http://tinyurl.com/ouy2ajr



Regards,
Sam

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asterisk at net153.net
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PostPosted: Sun Aug 23, 2015 12:53 pm    Post subject: [asterisk-users] SIP domain different than provider's Reply with quote

On 08/21/2015 12:52 AM, Sam wrote:
Quote:
Hello,

I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains before. My user '140@4354766787.com' is
handled by a 3rd party provider: 'sip.provider.com' and my local domain
on my asterisk box is the hostname 'mypbxdomain.com'.

My normal extension I use for everything is just '111'. I figured the
best way of joining my asterisk box was to just hard code in the
extensions I would need to dial for my remote office work (there are
only a couple of extensions so shouldn't be a big deal).

However I struggled to get authentication working for outgoing calls to
the few new extensions at the remote office through their provider.
Looking at debug logs it was clear that the sip 'To' address was wrong.
It had the provider: "To: <sip:139@sip.provider.com>" instead of the
domain which should look like: "To: <sip:139@4354766787.com>" (right?)

In the end, after hours of googling, reading the docs on sip.conf
several times revealed a little spoke of '!' dialplan option. Simply
changing my dialplan from 'Dial(SIP/workphone/${EXTEN})' to
'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.

But this seems really hackish. Is this the right/only way? Or is just
having a provider and mismatched domains not really the norm?

I have an an anonymized log here: http://tinyurl.com/ouy2ajr



Regards,
Sam



So since no one has responded, this either means one of two things:
Either I am an idiot and missed something obvious and therefore no one
wants to deal with me. Or this is indeed not something typical of an
asterisk config.

Can someone at least point me to which? Smile
As I mentioned, everything is working, it just doesn't "feel" right.


Regards,
Sam

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_____________________________________________________________________
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royj at yandex.ru
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PostPosted: Mon Aug 24, 2015 2:58 am    Post subject: [asterisk-users] SIP domain different than provider's Reply with quote

I think you chose the right solution in your case. It might doesn't "feel" right if you corrected source code, I think.

23.08.2015, 20:53, "Sam" <asterisk@net153.net>:
Quote:
On 08/21/2015 12:52 AM, Sam wrote:
Quote:
šHello,

šI have what I would think would be a common situation: I run asterisk at
šhome simply as a land line. I started a new job working remotely and
šthey gave me a SIP account with user name, domain, and proxy. I've never
šhad to deal with sip domains before. My user '140@4354766787.com' is
šhandled by a 3rd party provider: 'sip.provider.com' and my local domain
šon my asterisk box is the hostname 'mypbxdomain.com'.

šMy normal extension I use for everything is just '111'. I figured the
šbest way of joining my asterisk box was to just hard code in the
šextensions I would need to dial for my remote office work (there are
šonly a couple of extensions so shouldn't be a big deal).

šHowever I struggled to get authentication working for outgoing calls to
šthe few new extensions at the remote office through their provider.
šLooking at debug logs it was clear that the sip 'To' address was wrong.
šIt had the provider: "To: <sip:139@sip.provider.com>" instead of the
šdomain which should look like: "To: <sip:139@4354766787.com>" (right?)

šIn the end, after hours of googling, reading the docs on sip.conf
šseveral times revealed a little spoke of '!' dialplan option. Simply
šchanging my dialplan from 'Dial(SIP/workphone/${EXTEN})' to
š'Dial(SIP/workphone/${EXTEN}!${EXTEN}@4354766787.com)' fixed the issue.

šBut this seems really hackish. Is this the right/only way? Or is just
šhaving a provider and mismatched domains not really the norm?

šI have an an anonymized log here: http://tinyurl.com/ouy2ajr

šRegards,
šSam

So since no one has responded, this either means one of two things:
Either I am an idiot and missed something obvious and therefore no one
wants to deal with me. Or this is indeed not something typical of an
asterisk config.

Can someone at least point me to which? Smile
As I mentioned, everything is working, it just doesn't "feel" right.

Regards,
Sam

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
šššššššššššššššhttp://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
šššhttp://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
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