asterisk users - May 2011 - 40sec between dial execution and sending SIP request

Dear
I have a small pbx with asterisk 1.6.2.16.
I have a funny problem, there is exactly 40sec between dial execution and
sending first invite packet on sip.
do you have any idea where the problem is ?

Best regards

-- 
Pezhman Lali
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On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali <lopl at lopl.net> wrote:
> Dear
> I have a small pbx with asterisk 1.6.2.16.
> I have a funny problem, there is exactly 40sec between dial execution and
> sending first invite packet on sip.
> do you have any idea where the problem is ?
>
Check the dial timeout on your phone itself.  What model phone do you have?

-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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thanks,
this delay is occurred   on asterisk server, between dial execution and
"CALLED ....."


On Mon, May 9, 2011 at 7:12 PM, Warren Selby <wcselby at selbytech.com>
wrote:
> On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali <lopl at lopl.net>
wrote:
>
>> Dear
>> I have a small pbx with asterisk 1.6.2.16.
>> I have a funny problem, there is exactly 40sec between dial execution
and
>> sending first invite packet on sip.
>> do you have any idea where the problem is ?
>>
>
> Check the dial timeout on your phone itself.  What model phone do you have?
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.com
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Pezhman Lali
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Show us the cli trace of the delay. 

Thanks,
--Warren Selby, dCAP

On May 10, 2011, at 2:18 AM, Pezhman Lali <lopl at lopl.net> wrote:
> thanks,
> this delay is occurred   on asterisk server, between dial execution and
"CALLED ....."
> 
> 
> On Mon, May 9, 2011 at 7:12 PM, Warren Selby <wcselby at
selbytech.com> wrote:
> On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali <lopl at lopl.net>
wrote:
> Dear
> I have a small pbx with asterisk 1.6.2.16.
> I have a funny problem, there is exactly 40sec between dial execution and
sending first invite packet on sip.
> do you have any idea where the problem is ?
> 
> Check the dial timeout on your phone itself.  What model phone do you have?
> 
> -- 
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.com
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> -- 
> Pezhman Lali
> 
> 
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Dear Pezhman Lali,

Just below lines add in you sip.conf, after this if you get same problem do
2nd step, that is run on your server update_server_ip command this for your
database is not matching the your current IP..

externip=abc.net.org(if your server access remotly that URL add here if not
leave that line )
localnet=10.10.10.0/255.255.255.0(your local ip address series)

On Mon, May 9, 2011 at 5:56 PM, Pezhman Lali <lopl at lopl.net> wrote:
> Dear
> I have a small pbx with asterisk 1.6.2.16.
> I have a funny problem, there is exactly 40sec between dial execution and
> sending first invite packet on sip.
> do you have any idea where the problem is ?
>
> Best regards
>
> --
> Pezhman Lali
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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Dears

thanks for your all helps.
with assigning ip to the sip.conf and disabling srv_lookup the delay was
removed.
thanks again

best

On Tue, May 10, 2011 at 12:31 PM, mahesh katta <maheshkatta at
flexydial.com>wrote:
> Dear Pezhman Lali,
>
> Just below lines add in you sip.conf, after this if you get same problem do
> 2nd step, that is run on your server update_server_ip command this for your
> database is not matching the your current IP..
>
> externip=abc.net.org(if your server access remotly that URL add here if
> not leave that line )
> localnet=10.10.10.0/255.255.255.0(your local ip address series)
>
> On Mon, May 9, 2011 at 5:56 PM, Pezhman Lali <lopl at lopl.net>
wrote:
>
>> Dear
>> I have a small pbx with asterisk 1.6.2.16.
>> I have a funny problem, there is exactly 40sec between dial execution
and
>> sending first invite packet on sip.
>> do you have any idea where the problem is ?
>>
>> Best regards
>>
>> --
>> Pezhman Lali
>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Best Regards,
>
> Mahesh Katta
> *BUZZ**WORKS* Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E) Mumbai 400069
> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Web http://www.buzzworks.com
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
Pezhman Lali
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