[asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread Administrator TOOTAI

Hi list,

I face the following problem on incoming calls from my provider which 
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are 
not sended to the context set in provider sip.conf definition, but are 
going to the default context setted in [general].


Provider uses few IP's for incoming calls which are not the one used for 
register.



Here are the logs:

[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: --- (15 headers 22 
lines) ---
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Sending to 
85.xx.xx.2:5060 (no NAT)
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: Using INVITE request as 
basis request - 07403bb3412fc5206dec905b4eb26...@85.xx.xx.2
[2012-11-26 14:18:45] VERBOSE[1785] chan_sip.c: No matching peer for 
'0033x' from '85.xx.xx.2:5060'

...
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Peer audio RTP is at 
port 85.x.xx.2:16566
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: Looking for 027xxin 
default-guest (domain 217.yy.yy.yy)
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: list_route: hop: 
sip:0033xx...@85.xx.xx.2
[2012-11-26 14:38:48] VERBOSE[1785] chan_sip.c: RDNIS for this call is 
027xx (reason )



Our asterisk is registered with the provider, registerer IP from the 
provider being 85.xx.xx.3:



Sip.conf

[general]
context=default-guest;where incoming calls ended
...

register = 01234567:mysec...@sip.provider.net/01234567

[01234567]
type=peer
defaultuser=01234567
secret=mysecret
host=sip.provider.net
deny=0.0.0.0/0.0.0.0
permit=85.xx.xx.0/255.255.255.0
directmedia=no
qualify=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw,alaw
context=from-Provider
insecure=port,invite
fromdomain = sip.provider.net
fromuser=01234567
sendrpid = yes
nat=yes

What is wrong?

Thanks for any hint

--
Daniel

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Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread Joshua Colp

Administrator TOOTAI wrote:

Hi list,


Hola,


I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition, but are
going to the default context setted in [general].

Provider uses few IP's for incoming calls which are not the one used for
register.


You will need to create separate SIP peers that match on each IP address 
and direct them accordingly to the correct context. A secondary option 
is to enable anonymous guest support, but I would not recommend that as 
it can pose a security risk.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

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Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread isrlgb
Hi,
If were on this subject I'll throw in my question

Does named acl lists  in asterisk 11 help for this or only for registrations?

Thanks,

-Original Message-
From: Joshua Colp jc...@digium.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Mon, 26 Nov 2012 10:28:05 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] No matching peer for 'callerID' from
'85.xx.xx.2:5060'

Administrator TOOTAI wrote:
 Hi list,

Hola,

 I face the following problem on incoming calls from my provider which
 uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
 not sended to the context set in provider sip.conf definition, but are
 going to the default context setted in [general].

 Provider uses few IP's for incoming calls which are not the one used for
 register.

You will need to create separate SIP peers that match on each IP address 
and direct them accordingly to the correct context. A secondary option 
is to enable anonymous guest support, but I would not recommend that as 
it can pose a security risk.

Cheers,

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://www.asterisk.org/hello

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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread Joshua Colp

isr...@gmail.com wrote:

Hi,
If were on this subject I'll throw in my question

Does named acl lists  in asterisk 11 help for this or only for registrations?


ACLs don't control SIP peer matching, so no.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org

--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread isrlgb
Thought so but hoped other wise

Thanks

--Original Message--
From: Joshua Colp
To: ? ??
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No matching peer for 'callerID' from  
'85.xx.xx.2:5060'
Sent: Nov 26, 2012 4:40 PM

isr...@gmail.com wrote:
 Hi,
 If were on this subject I'll throw in my question

 Does named acl lists  in asterisk 11 help for this or only for registrations?

ACLs don't control SIP peer matching, so no.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com   www.asterisk.org




--
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Re: [asterisk-users] No matching peer for 'callerID' from '85.xx.xx.2:5060'

2012-11-26 Thread Administrator TOOTAI

Le 26/11/2012 15:28, Joshua Colp a écrit :

Administrator TOOTAI wrote:

Hi list,


Hola,


I face the following problem on incoming calls from my provider which
uses Asterisk 1.6.1.25, our asterisk being 1.8.17.0. Incoming calls are
not sended to the context set in provider sip.conf definition, but are
going to the default context setted in [general].

Provider uses few IP's for incoming calls which are not the one used for
register.


You will need to create separate SIP peers that match on each IP 
address and direct them accordingly to the correct context. A 
secondary option is to enable anonymous guest support, but I would not 
recommend that as it can pose a security risk.


Second option was the one I used till you gave me the solution ;-)

Thanks for your support

--
Daniel

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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