viswavardhanreddy karna
2011-Jan-26 16:28 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
Hi all,
i am doing my master thesis on server perfromance evaluation i am
using asterisk as sip proxy server and sipp tool as traffic generator...
i have run basic testing of asterisk like as shown in website
http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp
when i have copied sip.conf and extensions.conf from the site and run the
client and server i am getting error like this
NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to
extension 'service' rejected because extension not found in context
'default'
i dont know y this is coming its really troubling me a
lot...................................
please any one send me some xml, dial plan and sip.conf files for
registering and for inviting. I have been trying for this a lot if any one
help me i would be more thankful .....
BR
viswavardhanreddy
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Paul Belanger
2011-Jan-26 16:54 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
On 11-01-26 11:28 AM, viswavardhanreddy karna wrote:> Hi all, > i am doing my master thesis on server perfromance evaluation i am > using asterisk as sip proxy server and sipp tool as traffic generator... >Asterisk is not a SIP Proxy, it is a B2BUA.> NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to > extension 'service' rejected because extension not found in context > 'default' >You can troubleshoot dialplan issues pretty easy with: *CLI> dialplan show service at default The message states, exten => service,1,blah() is missing from the [default] context. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org
Bryant Zimmerman
2011-Jan-26 16:55 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
----------------------------------------
From: "viswavardhanreddy karna" <viswavardhanreddy at
gmail.com>
Sent: Wednesday, January 26, 2011 11:29 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] Regarding error in Asterisk dail plan:
Hi all, i am doing my master thesis on server perfromance
evaluation i am using asterisk as sip proxy server and sipp tool as traffic
generator...
i have run basic testing of asterisk like as shown in website
http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_usi
ng_SIPp
when i have copied sip.conf and extensions.conf from the site and run the
client and server i am getting error like this
NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to
extension 'service' rejected because extension not found in context
'default'
i dont know y this is coming its really troubling me a
lot...................................
please any one send me some xml, dial plan and sip.conf files for
registering and for inviting. I have been trying for this a lot if any one
help me i would be more thankful .....
BR viswavardhanreddy
----------------------------------------------------------------------------
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-------------------------------------
viswavardhanreddy
Your inbound request is not being sent with any target context or it is not
matching the ip found in your sip peers. This causes the default context to
trying and handle the call and you don't have anthing in it that can
complete the call.
The three options are
1 if you are doing registration make sure that the sending device is
specifiying a context. (It does not look like you are based on your link)
2 make sure that the sending ip matches your peer account or change the
peer account to friend (also change your peers to use insecure=port,invite
and see if that helps)
3 add a universal handler to the [default] contect to direct the call
to your test contects (exten => _.X,1,Goto(test,s,1)
One of these ideas may help you if I am understanding your issue.
Bryant
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Steve Murphy
2011-Jan-26 17:35 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna < viswavardhanreddy at gmail.com> wrote:> Hi all, > i am doing my master thesis on server perfromance evaluation i am > using asterisk as sip proxy server and sipp tool as traffic generator... > > i have run basic testing of asterisk like as shown in website > http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp > > > when i have copied sip.conf and extensions.conf from the site and run the > client and server i am getting error like this > > NOTICE[2715]: chan_sip.c:20314 handle_request_invite: Call from '' to > extension 'service' rejected because extension not found in context > 'default' > > i dont know y this is coming its really troubling me a > lot................................... > > >viswavardhanreddy-- I'm sorry, I'm a bit tight on time, I haven't read your link. But I did some performance testing of Asterisk some years ago, and wrote a doc about it and it's part of the source tree of Asterisk (At least in 1.6 ). See doc/chan_sip-perf-testing.txt There I show how I tied sipp and asterisk together. It might not at all help you, might not be your approach at all, but it might give you some ideas. Best of luck! murf -- Steve Murphy -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110126/fca31c3f/attachment.htm>
Steve Edwards
2011-Jan-26 17:53 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
On Wed, 26 Jan 2011, Bryant Zimmerman wrote:> ????3 add a universal handler to the [default] contect to direct the call to your test contects (exten => > _.X,1,Goto(test,s,1)exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}]) exten = _!.,n, goto(test,s,1) -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000
viswavardhanreddy karna
2011-Jan-26 18:15 UTC
[asterisk-users] Regarding error in Asterisk dail plan:
Hi edwards, i have taken register.xml and csv file from this site http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta <http://wiki.opencsta.org/index.php/SIPp_3.1_Registering_on_Asterisk_1.6.1_beta>i have written sip.conf and extension.conf i got error could you plz help me by writing the sip.conf and extensions.conf plz send me some file regarding this............ On Wed, Jan 26, 2011 at 6:53 PM, Steve Edwards <asterisk.org at sedwards.com>wrote:> On Wed, 26 Jan 2011, Bryant Zimmerman wrote: > > 3 add a universal handler to the [default] contect to direct the call >> to your test contects (exten => >> _.X,1,Goto(test,s,1) >> > > exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}]) > exten = _!.,n, goto(test,s,1) > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110126/c7e41359/attachment.htm>