authentication problem to my provider

choob

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Jul 25, 2007
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#1
Hello i m now user to elastix.

I would like to configure a trunk SIP to my provider who authenticate my by my IP only

insecure=yes
host=69.xxx.XXX.xxx
type=friend


and in CLI that show UNREACHABLE when i do sip show peers

i tryed type peer, user and nothin change. I think 1 month ago i succed connect with trixboxbut for now that do the same from trixbox or elastix and the support from my provider said no support third party and they don t hae change nothing.
If someone use IPSMARX with asterisk tell me.
IPSMARX work correctly with a no asterisk system actually.

All help will be appreciate
Thanks
 

cowboy47

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#2
I am seeing the same thing on several of my providers. I have no clue other than I had to change routers.
 

cowboy47

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#3
I am seeing the same thing on several of my providers. I have no clue other than I had to change routers.
 

choob

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#4
That my debug SIP to my trunk

Can someone help me plz

Authentication to my provider is only by IP
I don t know what should i write in my trunk

Thanks


Retransmitting #1 (no NAT) to 69.90.65.245:5060:
OPTIONS sip:69.90.65.245 SIP/2.0
Via: SIP/2.0/UDP 72.55.134.51:5060;branch=z9hG4bK27c49d95;rport
From: "Unknown" <sip:Unknown@72.55.134.51>;tag=as7e88c4c3
To: <sip:69.90.65.245>
Contact: <sip:Unknown@72.55.134.51>
Call-ID: 74e6e4fa4c24ea625a3085bd2ef66f2a@72.55.134.51
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 04 Aug 2007 05:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #2 (no NAT) to 69.90.65.245:5060:
OPTIONS sip:69.90.65.245 SIP/2.0
Via: SIP/2.0/UDP 72.55.134.51:5060;branch=z9hG4bK27c49d95;rport
From: "Unknown" <sip:Unknown@72.55.134.51>;tag=as7e88c4c3
To: <sip:69.90.65.245>
Contact: <sip:Unknown@72.55.134.51>
Call-ID: 74e6e4fa4c24ea625a3085bd2ef66f2a@72.55.134.51
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 04 Aug 2007 05:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #3 (no NAT) to 69.90.65.245:5060:
OPTIONS sip:69.90.65.245 SIP/2.0
Via: SIP/2.0/UDP 72.55.134.51:5060;branch=z9hG4bK27c49d95;rport
From: "Unknown" <sip:Unknown@72.55.134.51>;tag=as7e88c4c3
To: <sip:69.90.65.245>
Contact: <sip:Unknown@72.55.134.51>
Call-ID: 74e6e4fa4c24ea625a3085bd2ef66f2a@72.55.134.51
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 04 Aug 2007 05:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (no NAT) to 69.90.65.245:5060:
OPTIONS sip:69.90.65.245 SIP/2.0
Via: SIP/2.0/UDP 72.55.134.51:5060;branch=z9hG4bK27c49d95;rport
From: "Unknown" <sip:Unknown@72.55.134.51>;tag=as7e88c4c3
To: <sip:69.90.65.245>
Contact: <sip:Unknown@72.55.134.51>
Call-ID: 74e6e4fa4c24ea625a3085bd2ef66f2a@72.55.134.51
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 04 Aug 2007 05:51:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog '74e6e4fa4c24ea625a3085bd2ef66f2a@72.55.134.51' Method: OPTIONS
elastix*CLI>
 

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