search for: handle_request_invit

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extens...

Hi, Why are we getting message in the asterisk [Sep 10 12:55:23] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>; tag=2f498fbd [Sep 10 12:55:24] NOTICE[15043]: chan_sip.c:23647 handle_request_invite: Failed to authenticate device 601<sip:601 at 111.118.185.107>;tag=209a8aa9 Regards Deepak Bhatia -------------- next part ------------...

...ome (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '6442032987219' rejected because extension not found. [2011-12-28 22:53:44] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '7442032987216' rejected because extension not found. [2011-12-28 22:53:46] NOTICE[...

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at this point call fails, it is not being passed through to asterisk. I'm using Linksys 3102, PSTN answer delay is set to 3sec. to allow for caller ID to pass through. When I decrease timing to...

Hi Now, my Cisco AS5300 sent call to my asterisk, but two problems: When i call the phone number, i have: [Oct 28 06:01:16] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. [Oct 28 06:01:18] NOTICE[12813]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '0426000000' rejected because extension not found. (0426000000 = my phone number) F...

...tup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet ! sip show channels shows the call setup with ulaw. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

...ne. I have added the sip peers of Voxbone Belgium1 like this in the sip.conf [81.201.82.39] host=dynamic type=friend insecure=very context=your_context canreinvite=no qualify=no deny=0.0.0.0/0.0.0.0 permit=81.201.82.39/255.255.255.255 but unfortunately when I receive a call I got this nice error: handle_request_invite: Failed to authenticate user "075XXXXXXXX" <sip:075XXXXXXXX at voxbone.com>;tag=76596. I am in doubt now because with the insecure=very, I must receive any incoming calls from from voxbone (81.201.82.39) without any problems. Do you know how to fix this please?

Greetings all, I have been seeing a lot of [Jan 2 16:36:31] NOTICE[7519]: chan_sip.c:23149 handle_request_invite: Sending fake auth rejection for device 100<sip:100 at 108.161.145.18>;tag=2e921697 in my logs lately. Is there a way to automatically ban IP address from attackers within asterisk ? Thank you

i create inbound number but i calling and send this error: [Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite: Call from '101396_procall' to extension '8888246463' rejected because extension not found. but the extensin existed -- Bayardo S?nchez Garc?a Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxi Support E-mail: bayardo.sanchez at gmail.com Linux...

Good morning, I've been having some problems getting the SPA-3102 working properly in India. Specific problem is that calls from the Asterisk server out the FXS port is failing. When trying to make calls, I'm getting this message: [Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774 handle_request_invite: Call from '' to extension '66200' rejected because extension not found. I believe everythign is set up correctly because when I was doing tests here in the US everything worked perfectly before sending the unit to India. Now that its there, I'm having some troubles. If there...

Are these attempts to scam SIP calls through my Asterisk server: [May 13 22:50:41] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '084312297134' rejected because extension not found. [May 14 13:36:35] NOTICE[30888]: chan_sip.c:17295 handle_request_invite: Call from '' to extension '0114312297136' rejected because extension not found. [May 15 12:20:29] NOTICE[3420...

Hi I have a problems with a new Asterisk Server, when i want call, i have: [Nov 14 09:12:38] NOTICE[31992]: chan_sip.c:18160 handle_request_invite: Call from 'PHISIP000001' to extension '00420225352184' rejected because extension not found. but into my extensions.conf: exten => _00420X.,1,Set(CDR(CodeTier)=CZE) exten => _00420X.,2,Dial(SIP/${EXTEN}@AS5350,180,rt) exten => _00420X.,3,Hangup...

Hi All, I see this kind of attack on our Asterisk Server, do you know how to block that IP? [Sep 4 07:41:06] NOTICE[7375]: chan_sip.c:23375 handle_request_invite: Call from '' (213.136.81.166:9306) to extension '34422' rejected because extension not found in context 'default'. Thanks in advance, -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-user...

Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register => 8001187e0:password at sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing...

...Similar messages from those threads (1 line each): -- Executing [123456 at from-sip-external:1] NoOp("SIP/mypbx.com-00000751", "Received incoming SIP connection from unknown peer to 123456") in new stack Aug 7 23:32:03 mypbx asterisk[3686]: NOTICE[27307]: chan_sip.c:18047 in handle_request_invite: Failed to authenticate user <sip:165411 at mypbx.com>;tag=1660ec63 Aug 8 00:03:50 mypbx asterisk[3686]: NOTICE[27307]: chan_sip.c:18044 in handle_request_invite: Sending fake auth rejection for user <sip:165499 at mypbx.com>;tag=e6786d03 NOTICE[2578]: chan_sip.c:21250 handle_requ...

...s working configured and working OK with my Asterisk but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I can only dial 3-digit extension. Anything longer than 3-digits is cut off, example I dial extension 1000: [Apr 29 10:03:30] NOTICE[3817][C-000000e9]: chan_sip.c:25902 handle_request_invite: Call from '54' (10.0.0.115:5060) to extension '100' rejected because extension not found in context 'internal'. My dial plan is working OK as when I register Linsys/Sipura with asterisk "context=internal" and I dial any number and dial plan is working OK. It se...

...from-trunk:1] Wait("SIP/wsol-00820870", "1") in new stack -- Executing [6035467131 at from-trunk:2] Dial("SIP/wsol-00820870", "SIP/4610") in new stack -- Called 4610 -- SIP/4610-00838160 is ringing [Dec 11 12:18:38] NOTICE[2624]: chan_sip.c:13753 handle_request_invite: Call from '' to extension '6035467131' rejected because extension not found. -- SIP/4610-00838160 answered SIP/wsol-00820870 Really destroying SIP dialog '08c6697823d4542917eaaf607babc786 at 10.100.0.1' Method: NOTIFY -- Executing [6035467131 at from-trunk:1] Wait(...

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8 my caller ID is not working WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515 -- Joseph

...d extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and extension.conf) insecure=invite is not taking effect. I get: "... username mismatch, have <4>, digest has <pstn-4444> handle_request_invite: Failed to authenticate user "KMIEC J" Someone have mentioned that sort order in sip.conf might effect the way it works. -- Joseph

Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an "404 not found" error on their side. What can be the problem? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106,...