[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 -- _ -- 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
Re: [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@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 -- _ -- 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
Re: [asterisk-users] Regarding error in Asterisk dail plan:
From: viswavardhanreddy karna viswavardhanre...@gmail.com Sent: Wednesday, January 26, 2011 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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 - 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 -- _ -- 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
Re: [asterisk-users] Regarding error in Asterisk dail plan:
On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna viswavardhanre...@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 -- _ -- 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
Re: [asterisk-users] Regarding error in Asterisk dail plan:
thanks to all, but i am working for register scenario can anyone please help me when i have sent the sipp command from sipp like this ./sipp -sf reg.xml -inf users.csv -p 5060 -i 192.168.1.99 192.168.1.100 i got the error message in asterisk like this chan_sip.c:21819 handle_request_register: Registration from '105 sip:105@192.168.1.100:5060' failed for '192.168.1.99' - No matching peer found and in wire shark i got 404 not found. anyone please help me. On Wed, Jan 26, 2011 at 6:35 PM, Steve Murphy m...@parsetree.com wrote: On Wed, Jan 26, 2011 at 9:28 AM, viswavardhanreddy karna viswavardhanre...@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 -- _ -- 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 -- _ -- 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
Re: [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 sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
_
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Re: [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_betai
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@sedwards.comwrote:
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 sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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--
_
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Re: [asterisk-users] Regarding error in Asterisk dail plan:
Un-top-posting... On Wed, 26 Jan 2011, viswavardhanreddy karna wrote: Hi all, i am doing my master thesis on server perfromance On Wed, 26 Jan 2011, viswavardhanreddy karna wrote: 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 i have written sip.conf and extension.conf i got error could you plz help me by writing the sip.conf and extensions.conf Something about 'doing my master thesis' makes me think this is something you should solve with a bit more effort on your part. -- Thanks in advance, - Steve Edwards sedwa...@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
