Help me with SIP account settings

[shobhit29]

Hi All,
I have configured the SIP account details as per the manual but it is showing unrechable on ASTERISK CLI i am using 2.05 version.

[dlasry]

Take a look at the thread below. It helped big time when setting up my SIP carriers.

http://www.eflo.net/VICIDIALforum/viewtopic.php?t=7449

[shobhit29]

Is it for 2.05 version as per my knowledge we need not to configure sip.conf in this version can you please guide me for the same[/b]

[dlasry]

I too am using 2.05 and these extra steps were required. Vicidial creates the carrier entries that you enter, but does not seem to associate the specific trunk group to the dialplan entries in question (extensions.conf). This seems to be for SIP trunks specifically. So the only file you would need to modify is extensions.conf

I suggest you try the instructions in the thread.

[shobhit29]

thanks a lot for this support , i will now try this aswell and will inform the status .. many thanks ..

Regards
Shobhit Anand

[shobhit29]

I have tried that aswell but when i dial a number it say sorry this is not a vailid extension .. please help.

[dlasry]

Are you able to dial between two extensions?

If not, i suggest you rebuild the extensions according to Matt's instructions.

If yes, please show us the output of your SIP debug in CLI.

Type in Asterisk -r and then
SIP debug peer [TRUNKName] or SIP debug peer [extension number]. The first will show you the output at your trunk level, and the second at your extension level.

[shobhit29]

This is i get when is put extension command.

vici*CLI> SIP debug peer extension 1



vici*CLI>
<-- SIP read from 64.132.217.195:5060:
SIP/2.0 200 OK
CSeq: 104 REGISTER
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK5421ff4b
From: <sip:crosslink@64.132.217.195>;tag=as2a84fe25
Call-ID: 21df4f98139f3b892acab8f265a55b52@127.0.0.1
To: <sip:crosslink@64.132.217.195>;tag=151021091656
Contact: <sip:s@192.168.1.2>;expires=600
Expires: 600
Content-Length: 0


--- (9 headers 0 lines) ---
Scheduling destruction of call '21df4f98139f3b892acab8f265a55b52@127.0.0.1' in 32000 ms
Oct 15 18:21:56 NOTICE[2379]: chan_sip.c:10043 handle_response_register: Outbound Registration: Expiry for 64.132.217.195 is 600 sec (Scheduling reregistration in 585 s)
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.132.217.195:5060:
OPTIONS sip:64.132.217.195 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK33e306f7;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as2e16000d
To: <sip:64.132.217.195>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 42aecaa61d0402133b45f2ac18f27978@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 15 Oct 2009 22:22:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
vici*CLI>
<-- SIP read from 64.132.217.195:5060:
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK33e306f7
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as2e16000d
Call-ID: 42aecaa61d0402133b45f2ac18f27978@192.168.1.2
To: <sip:64.132.217.195>;tag=151022091613
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '42aecaa61d0402133b45f2ac18f27978@192.168.1.2'
== Manager 'sendcron' logged off from 127.0.0.1
Destroying call '21df4f98139f3b892acab8f265a55b52@127.0.0.1'
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
12 headers, 0 lines
Reliably Transmitting (NAT) to 64.132.217.195:5060:
OPTIONS sip:64.132.217.195 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK400d59d1;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as71d7ec9a
To: <sip:64.132.217.195>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 023b51683ef6147a402f239e3245312a@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 15 Oct 2009 22:23:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
vici*CLI>
<-- SIP read from 64.132.217.195:5060:

SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK400d59d1
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as71d7ec9a
Call-ID: 023b51683ef6147a402f239e3245312a@192.168.1.2
To: <sip:64.132.217.195>;tag=151023091614
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '023b51683ef6147a402f239e3245312a@192.168.1.2'
== Manager 'sendcron' logged off from 127.0.0.1

[shobhit29]

This is i get when i run sip debug peer trunk_1


vici*CLI> sip debug peer trunk_1
SIP Debugging Enabled for IP: 64.132.217.195:5060
vici*CLI>
<-- SIP read from 64.132.217.195:5060:
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK36dbeb9b
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as18df2f54
Call-ID: 457336343656081713a39d5e27fe4143@192.168.1.2
To: <sip:64.132.217.195>;tag=151029091616
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '457336343656081713a39d5e27fe4143@192.168.1.2'