Can't configure sip trunk

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Can't configure sip trunk

Postby webgurru » Sun May 10, 2009 9:44 am

Hi,
I have installed vicidialnow successfully. I could not configure SIP trunk. I tried these settings throug admin section aswell as through shell using vi editor

etc/asterisk/sip.conf file

[trunk_1]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=<username>
secret=<password>
host=<your_sip_provider>
dtmfmode=inband
qualify=1000
fromuser=<username>

and in the file etc/asterisk/extensions.conf

SIPtrunk=SIP/trunk_1

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten => _91NXXNXXXXXX,3,Hangup

I am in UK and my when I register my SIP provider with my soft/sip phone, I can dial all UK number without dialing international and country codes. To dial a number I just dial like 01214348989 for local and 07888889899 for mobiles numbers. I configured EYBEAM SIP phone it is logged in and I loaded about 20 numbers in the leads. I logged into agent and my sip phone rings telling me I am only one logged in. When I dial any number from lead my soft phone message "I am sorry that's not a valid extension, please try again" I am sure there is something wrong in SIP configuration. Please help me to get this work.

Best regards,
Webgurru
webgurru
 
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Location: United Kingdom

Postby marcin » Sun May 10, 2009 2:21 pm

You have to match your dial plan to numbers you a re dialing:
for 01214348989 you dialplan entry should something like this:
_90XXXXXXXXX but you need to dial 9 as prefix

and 9 should be add as prefix to your compaign

or _0NXXXXXXXX to dial without prefix 9
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Joined: Wed Mar 18, 2009 8:00 am

Postby webgurru » Sun May 10, 2009 2:41 pm

Hi Marcin,

Thanks for your quick reply. I did this change but no luck. Could you please look my lead file to import.

"PHONE CODE","PHONE NUMBER","FIRST NAME","LAST NAME","ADDRESS1","CITY","STATE","POSTAL CODE","COUNTRY CODE "
"1","1214862327","Home","BT","Test address 1","Test City 1","CA","12345","44"
"1","1216604706","Home","SIP","Test address 2","Test City 2","CA","12345","44"
"1","1375382403","Azam","Zulf","Test address 3","Test City 3","CA","12345","44"
"1","2085012835","Zulf","Home","Test address 4","Test City 4","CA","12345","44"
"1","1215555555","Test","Test","Test","Test","CA","12345","44"
"1","1214444444","Test","Test","Test","Test","CA","12345","44"
"1","1213333333","Test","Test","Test","Test","CA","12345","44"
"1","1215555678","Test","Test","Test","Test","CA","12345","44"
"1","1213456789","Test","Test","Test","Test","CA","12345","44"
"1","1215456876","Test","Test","Test","Test","CA","12345","44"
"1","1217646754","Test","Test","Test","Test","CA","12345","44"

Is there anything wrong with this CSV file to import.

Best regards,
Webgurru
webgurru
 
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Joined: Thu May 07, 2009 11:10 am
Location: United Kingdom

Postby mflorell » Mon May 11, 2009 4:57 am

a Phone code of 1 is North America, and those don't look like north american numbers. If you are dialing in the UK you need to use 44 or your GMT offsets will not be correct.
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Postby webgurru » Mon May 11, 2009 10:46 am

Hi,

Is there anyway to check if my SIPTRUNK is successfully registered?
Best regards,

Webgurru
webgurru
 
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Joined: Thu May 07, 2009 11:10 am
Location: United Kingdom

Postby mflorell » Mon May 11, 2009 11:08 pm

sip show registry
sip show peers
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Postby gardo » Tue May 12, 2009 1:14 am

It can take up to a minute for the changes you made on the web admin page to take effect. Login to your server and access the Asterisk CLI. Check if the changes you made were committed by running as Matt said "sip show registry" and "sip show peers".
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Postby webgurru » Tue May 12, 2009 4:29 am

Hi Mflorell,

I am sorry for this but from where I can see SIP registry and SIP peers. In Admin area under Admin->Carriers, it only show carriers list with Active Y or N.

Best regards,
webgurru
 
Posts: 147
Joined: Thu May 07, 2009 11:10 am
Location: United Kingdom

Postby webgurru » Tue May 12, 2009 7:52 am

Hi,

I manage to see SIP registry and peers. On asterisk CLI, I found a problem it's continuously trying to connect my SIP provider and displaying message Registration for 'my account provider name' timed out, trying again. I changed my SIP provider, I have three provider but with everyone asterisk shows same message. I can ping my SIP provider from server window. I can logon and make calls from all three provider using my IP and Soft phone with same setting as used in vicidialnow. Could anyone help me out this why my sip provider registration fails?

Best regards,
webgurru
 
Posts: 147
Joined: Thu May 07, 2009 11:10 am
Location: United Kingdom

Postby gzpxyj » Tue Jun 02, 2009 7:02 pm

You did not mention that you have an registration string in place. Usually registration for the trunk is not a problem.
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Postby kddacraker » Fri Jun 05, 2009 2:10 am

Hi,

I have register the sip account,

but calls are not going through....

My dial plan entry is :

exten => _0NXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0NXXXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten => _0NXXXXXXXX,3,Hangup

I want to dial US with prefix " 1 "

is the dial plan correct or I am making some mistake?

I am getting 404 error when i tried to call form eyebeam.

I am not able to get any kind of error message on CLI...


any suggestion ?
kddacraker
 
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Joined: Fri Feb 15, 2008 5:18 am
Location: Nasik India

Postby kddacraker » Fri Jun 05, 2009 3:59 am

Hello...

Now I am getting this message on my CLI

Jun 5 10:27:10 NOTICE[26487]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/cc101-b7b0cab8", "") in new stack
== Spawn extension (default, 12127773456, 3) exited non-zero on 'SIP/cc101-b7b0cab8'
-- Executing DeadAGI("SIP/cc101-b7b0cab8", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
kddacraker
 
Posts: 71
Joined: Fri Feb 15, 2008 5:18 am
Location: Nasik India

Postby gzpxyj » Sat Jun 06, 2009 8:04 am

kddacraker wrote:Hi,

I have register the sip account,

but calls are not going through....

My dial plan entry is :

exten => _0NXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _0NXXXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},55,o)
exten => _0NXXXXXXXX,3,Hangup

I want to dial US with prefix " 1 "

is the dial plan correct or I am making some mistake?

I am getting 404 error when i tried to call form eyebeam.

I am not able to get any kind of error message on CLI...


any suggestion ?

First, you need to make sure your sip is registered through asterisk cli. From your message, I am not sure you actually get the sig registered with your provider from your vicidial.
Second, you did not tell which country you were dialing from, US to US? or some country to US. That differ how the dial plan should be. If you are dialing from US to US, your normal string should be 1+area code + 7 digits phone number. So your dial plan should be _91NXXNXXXXXX. 9 is added for vicidial. If from other country to US, you should dial prefix to get out of your country + 1 + area code + 7 digits phone number. You can figure out the dial plan from US to US plan.
Hope this help.
gzpxyj
 
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Joined: Sun Mar 22, 2009 7:56 am


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