[Asterisk-Users] Originate calls but can't receive them on a SIP trunk

[Amaury BOSSE]

Hi list,

I have a problem with a SIP trunk on my * box: I can originate calls but I
can't receive them.

The * box is behind a modem-router and as a private address.

I think about a NAT problem but I don't know how to resolve it.

I have included some debug and configuration.

 

 

 

The trunk is registered as shown by "sip show registry" command:

Host                                        Username       Refresh   State


sip.myprovider.fr:5060             08704412XX         105      Registered


 

It appears "UNREACHABLE" when I execute "sip show peers"

Name/username    Host            Dyn Nat ACL     Mask
Port     Status    

cmm_sip/0870441  213.186.61.81        N           255.255.255.255
5060     UNREACHABLE

 

I have configured "sip_additionnal.conf" with parameters send by my provider
:

register=08704412XX:5496 at sip. myprovider.fr/08704412XX

[cmm_sip]

type=friend

username=08704412XX

secret=5496

context=default

host=sip. myprovider.fr

permit=sip. myprovider.fr

qualify=yes

disallow=all

allow=g729

 

I also include a copy of  "debug sip peers" :

Sip read: 

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 193.252.35.74:5060;branch=z9hG4bK14ed2d36

From: "Unknown" <sip:Unknown at 193.252.35.74>;tag=as740c103b

To: <sip:213.186.61.81>;tag=as051709c0

Call-ID: 3c0ac3642407d9d02898fcb3215e78cc at 193.252.35.74

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY

Contact: <sip:213.186.61.81>

Accept: application/sdp

Content-Length: 0

 

 

11 headers, 0 lines

Destroying call '3c0ac3642407d9d02898fcb3215e78cc at 193.252.35.74'

Retransmitting #3 (no NAT):

OPTIONS sip:213.186.61.81 SIP/2.0

Via: SIP/2.0/UDP 192.168.8.251:5060;branch=z9hG4bK14ed2d36

From: "Unknown" <sip:Unknown at 192.168.8.251>;tag=as740c103b

To: <sip:213.186.61.81>

Contact: <sip:Unknown at 192.168.8.251>

Call-ID: 3c0ac3642407d9d02898fcb3215e78cc at 192.168.8.251

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Date: Fri, 02 Dec 2005 16:17:26 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Content-Length: 0

 

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