Sip trunk not registering

telecomtechnician

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#1
Hi elastix community

My VOIP provider send me the following configuration file for my elastix server:

SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
maxexpirey=60 ; Max length of incoming registration we allow
defaultexpirey=180 ; Default length of incoming/outoing registration

disallow=all ; Disallow all codecs
allow=g729

dtmfmode=rfc2833


register => 771XXXXXXX:123456789:771XXXXXXX@ssc01.payedsip.com/771XXXXXXX

[ssc01.payedsip.com]
type=friend
disallow=all
allow=g729
host=ssc01.payedsip.com
username=771XXXXXXX
fromuser=771XXXXXXX
secret=123456789
context=sip-in
canreinvite=no
nat=yes
qualify=yes
insecure=very



extensions.conf:

[sip-in]
exten => 771XXXXXXX,1,Dial(SIP/100) ; donde 100 es una extension interna de la pbx



How can I actually do this in elastix?,

I tried to configure this in TRUNK, ADD SIP TRUNK but it does not work.
My provider says that I have something missing, that is why I can not register.

Waiting for your comments

David Medina
 

dicko

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#2

dingoland

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#3
Hi,

Try to delete the second 771xxxxx (just before the @) in the string registration (as username and authname are the same) and add fromdomain=ssc01.payedsip.com in the peer details of your trunk.

Regards
Greg
 

ramoncio

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#4
Hi David,

If you haven't solved it yet, could you paste some sip debug messages?
I have one provider that when I set qualify=yes doesn't work, but removing this line it works great.
 

telecomtechnician

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#5
Hi Ramoncio, Here are my sip debug message

I hope you can help me

David Medina




Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 105 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Retransmitting #5 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 105 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Reliably Transmitting (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #2 (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #3 (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #6 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK709c0ca5;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as0983d179
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 105 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Retransmitting #4 (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK60745335;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as3806bbf9
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '10ddcb5b4ae3a03d561ba8282720ae05@190.206.213.117' Method: OPTIONS
> doing dnsmgr_lookup for 'ssc01.comvoz.com'
REGISTER 10 headers, 0 lines
Reliably Transmitting (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 106 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Really destroying SIP dialog '3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3' Method: REGISTER
Retransmitting #1 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 106 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Retransmitting #2 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 106 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Retransmitting #3 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 106 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0


---
Reliably Transmitting (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK61ffa6db;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as4d500399
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 0337e27052ddc61e060ddaa90a5dddc8@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 216.22.81.36:5060:
OPTIONS sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK61ffa6db;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@190.206.213.117>;tag=as4d500399
To: <sip:ssc01.comvoz.com>
Contact: <sip:Unknown@190.206.213.117>
Call-ID: 0337e27052ddc61e060ddaa90a5dddc8@190.206.213.117
CSeq: 102 OPTIONS
User-Agent: Elastix
Date: Fri, 20 Aug 2010 14:26:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #4 (NAT) to 216.22.81.36:5060:
REGISTER sip:ssc01.comvoz.com SIP/2.0
Via: SIP/2.0/UDP 190.206.213.117:5060;branch=z9hG4bK5101745f;rport
Max-Forwards: 70
From: <sip:7711800883@ssc01.comvoz.com>;tag=as23b213e2
To: <sip:7711800883@ssc01.comvoz.com>
Call-ID: 3f1d60c5016d260b74cc79af1e0637c2@192.168.0.3
CSeq: 106 REGISTER
User-Agent: Elastix
Expires: 180
Contact: <sip:7711800883@190.206.213.117>
Content-Length: 0
 

telecomtechnician

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#6
Hi Greg

Thank you for your help, but it did not work. I did not mention (I am doing it now) that there was a moment when the trunks registered with no problem. At nights I turned off the voip server, and the next day I turned it on again, there was days when there was no trunk registration, other days the trunks registered with no problem. But from a month ago, the problem is permanent, no trunk registration.

Waiting for your comments

David Medina
 

telecomtechnician

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#7
Re:Sip trunk not registering (SOLVED)

Hi elastix community

The problem was solved in the following way:

My router´s state tables had to be flushed. Once it was done, (resetting the router to default), the trunks registered with the sip provider with no problem.

Last night I got into the elastix IRC chat, and somebody gave me the solution. Definetely, network acknowledge is very important.

I have to be aware of this, because I turn off all my telecom equipment at night. Or maybe I might not turn off the router and internet modem.

Thanks everybody

David Medina
 

alfreluis

Joined
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Messages
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#8
hi David, u know...I have the same debug figure (retransmitting...etc) and my sip trunk cannot registering in to my sip VoIP provider...

What's the router you have reset? is it the external router, right?

Thank you very much
Alfred
 

jmvelazquezmx

Joined
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#9
Trunk not answering

Hi all:

I am newbie in elastix, after installing release 1.6.5.2, added my sip extensions (both hardware and sip), was able to comunicate among them; however, when trying to make outbound calls I keep receiving the same message "all our circuitries are busy at this moment. I am using Digium 2 ports. Any body have any ideas?

Thanks in advance
 

trymes

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#10
Re: Trunk not answering

JM: Two things:

1.) You should create a new topic with your question, ad it is not related to this topic, and you'll likely get a better response if you start your own thread.
2.) It sounds as if you have not created an outbound route to tell your system how to send calls out to the PSTN. Configure an outbound route or verify the settings on your existing one and see if that helps.

Tom
 

jmvelazquezmx

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#11
Re: Trunk not answering

Thanks for your response. Outbound route is the one that comes by default. No chenges were made to this section. I will open this new topic.
 

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