search for: handle_response_invit

Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP Phones has NR (Not Register) at the LCD, and it is able to receive and originate calls !! I was think if this is bug or if it is related to...

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward th...

...why? > <snip> > >> Here is the CLI command used: > >> > >> ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out application dial > >> == Using SIP RTP CoS mark 5 > >> [Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 > > handle_response_invite: Received response: "Forbidden" from > > '"Anonymous" > > <sip:<did>@69.59.234.67<http://69.59.234.67>>;tag=as69898393' > >> ubuntu*CLI> > > > > Use the AMI Originate action or a call file. You can specify a caller &...

...1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"cc101" < sip:xxxxxxxxx at 203.196.128.56 <sip%3Axxxxxxxxx at 203.196.128.56> >;tag=as09c56cf2' -- SIP/Sama203-09fbdaa0 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Execu...

...dial extensions within asterisk. The asterisk log: -- Called 06301111111 at neophonex-out -- Call on SIP/neophonex-out-081a9cc0 left from hold -- SIP/neophonex-out-081a9cc0 is making progress passing it to SIP/neophonex-081ab240 [Jun 25 16:57:07] WARNING[18232]: chan_sip.c:11839 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '44c971692552f3245aa7b4e834bdafab at sip.neophonex.hu'. Giving up. -- Call on SIP/neophonex-out-081a9cc0 left from hold -- SIP/neophonex-out-081a9cc0 answered SIP/neophonex-081ab240 -- Native bridging SIP/neophonex...

...ro("SIP/400-08280ae0", "trunkdial|SIP/trunk_1/043401xxxx"? ) in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/400-08280ae0", "SIP/trunk_1/043401xxxx") in new stack -- Called trunk_1/043401xxxx [Jan 11 14:33:16] WARNING[2439]: chan_sip.c:11918 handle_response_invite: Received response: "Forbidden" from '"400" <sip:028012xxxx at iinetphone.iinet.net.au <mailto:028012xxxx at iinetphone.iinet.net.au>>;tag=as0767eb78' -- SIP/trunk_1-08284e38 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Execut...

...t; > >>> Here is the CLI command used: > >>> > >>> ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out application dial > >>> == Using SIP RTP CoS mark 5 > >>> [Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 > >> handle_response_invite: Received response: "Forbidden" from > >> '"Anonymous" > >> > > <sip:<did>@69.59.234.67<http://69.59.234.67><http://69.59.234.67 > >>;tag=as69898393' > >>> ubuntu*CLI> > >> > >> Use the...

...ontext=from-pstn > > canreinvite=no > > > > Here is the CLI command used: > > > > ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out application dial > > == Using SIP RTP CoS mark 5 > > [Aug 5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 > handle_response_invite: Received response: "Forbidden" from '"Anonymous" > <sip:<did>@69.59.234.67>;tag=as69898393' > > ubuntu*CLI> > Use the AMI Originate action or a call file. You can specify a caller id there. You cannot specify one from the command line. Ri...

...ecs: ulaw,alaw,gsm Context: dialout Exten: callprogress Priority: 1 Timeout: 20000 Variable: SIPADDHEADER="Alert-Info: Ring Answer" ActionID: 100014 CallerID: Axis < 525 > The SIP/2012 is a IP Speaker on the computer. The error is: [Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received response: "Forbidden" Why am I getting "Forbidden" ? Its a call file on my server and the speaker is directly connected to my server. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/...

...omuser=<did> fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 context=from-pstn canreinvite=no Here is the CLI command used: ubuntu*CLI> originate SIP/732-xxx-xxxx at vonage-out application dial ? == Using SIP RTP CoS mark 5 [Aug ?5 14:16:49] WARNING[32891][C-00000006]: chan_sip.c:23160 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:<did>@69.59.234.67>;tag=as69898393' ubuntu*CLI>? Thanks for your help murthy

...w peers shows both as registered. this is the error when try and place a call from Asterisk 1 to Asterisk 2: - Executing [582 at a1:1] Dial("Console/dsp", "SIP/VoipDirect777821|60|") in new stack -- Called VoipDirect777821 [Dec 1 23:20:21] WARNING[25399]: chan_sip.c:12334 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk at 141.122.139.16>;tag=as070b02e2' -- SIP/VoipDirect777821-0876c360 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [582 at a1:2] Hangup("Console/dsp&qu...

Why Asterisk show this message? What I can do? Sep 30 15:45:18 NOTICE[3608]: chan_sip.c:9096 handle_response_invite: Failed to authenticate on INVITE to '"100" <sip:100@192.168.100.17>;tag=as413bd6a8' -- SIP/sip.uni.it-df15 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Thanks!! ___________________________________ Yahoo! Messenger: chiamate gratuite in...

...g in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk and is then bridged to the SPA-2100 we see the following messages: Native bridging SIP/sn2400a-0952 and SIP/6046285076-cabc WARNING[562]: chan_sip.c:10290 handle_response_invite: Strange... The other side of the bridge don't have udptl struct WARNING[562]: chan_sip.c:10290 handle_response_invite: Strange... The other side of the bridge don't have udptl struct WARNING[562]: chan_sip.c:3884 process_sdp: Unknown SDP media type in offer: image 16412 udptl t38 Any...

...o I can check it against another one. The PBX that fails, fails with any extension number. Replacing the phone with a SPA has no problems. I don't understand how it can work on one server but not another. The error that is occurring is: Nov 1 14:31:42 WARNING[32190]: chan_sip.c:9720 handle_response_invite: Forbidden - wrong password on authentication for INVITE to '"DavidR (Perth)" sip:2030@192.168.24.251>;tag=as11bbecc0' Please help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments...

...oblem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes failed to make call with the following error. Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite: Failed to authenticate on INVITE to '"9002" <sip:90002@10.0.0.22>;tag=as2ff0c493' Any solution to let them call each others? ango

...tried, but the call hung up again. I'd be greatful for more ideas of solving the problem. Fresh logs when hanging up, from asterisk console: -- SIP/neophonex99-out-08213ac8 is making progress passing it to SIP/neophonex57-out-081e8a78 [Aug 2 21:54:51] WARNING[24739]: chan_sip.c:11948 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '29312b1d2fc464560ff9ef7747c614d2 at sip.neophonex.hu'. Giving up. -- SIP/neophonex99-out-08213ac8 answered SIP/neophonex57-out-081e8a78 -- Native bridging SIP/neophonex57-out-081e8a78 and SIP/neophonex99-out-08213ac8...

I'm seeing this response to SIP calls originated in the following manner: Dial(SIP/${EXTEN}&SIP/{$DID},30,r) handle_response_invite: Re-invite to non-existing call leg on other UA. The response is from the second part of the dial. What exactly does it mean and how can I fix it? Thanks in advance Bruce

...context=[default] deny=0.0.0.0/0.0.0.0 permit=172.16.200.176/255.255.255.128 and i get the following in pbx1: -- Executing [18021 at default:1] Dial("SIP/8029-b7413678", "SIP/pbx2/8021||TWw") in new stack -- Called pbx2/8021 [Mar 4 16:49:13] WARNING[3392]: chan_sip.c:12679 handle_response_invite: Received response: "Forbidden" from '"Khalid Touati" < sip:8029 at 172.16.200.176 <sip%3A8029 at 172.16.200.176>>;tag=as1dcf5ff2' -- SIP/pbx2-09cf4468 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel...

I'm getting the following warning with SOME calls on 1.8.14.1 Is it a cause for concern? Is there a way to fix it? I can't tell for sure if it is impacting calls or not. WARNING[27796]: chan_sip.c:20497 handle_response_invite: just did sched_add waitid(4077) for sip_reinvite_retry for dialog 358b7af85cf24cf0609cb0195b273935@[ip removed] in handle_response_invite

...he "phone numbers you can call" page -) but fails for some others. For example I've got a friend who's at freephonie so to call him, I would dial *759608xxxxxxxx (7596 being freephonie.net). When I do that, I get the following error: Jun 29 10:27:21 NOTICE[7916]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '<sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe' And here's a snippet of what I get from 'sip debug': -------------------------------------------------------------------------- <-- SIP read from 24.196.79.163:5060: SIP/2.0 40...