Displaying 20 results from an estimated 74 matches for "handle_response_invite".
2011 Sep 13
0
WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468)
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 s...
2008 Dec 03
0
chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE 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 the...
2015 Aug 06
3
Asterisk uses "Anonymous", but why?
...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
&g...
2009 Sep 08
1
SIP Error
...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)
-- Execut...
2007 Jun 25
2
callback and bridge problem
...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-...
2008 Jan 11
5
Congestion/Forbidden issue with new carrier
...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)
-- Executi...
2015 Aug 06
2
Asterisk uses "Anonymous", but why?
...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 A...
2015 Aug 06
4
Asterisk uses "Anonymous", but why?
...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.
Ric...
2020 Jun 11
3
Forbidden call
...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
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2015 Aug 05
2
Asterisk uses "Anonymous", but why?
...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
2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
...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&quo...
2005 Sep 30
1
Not Authenticate
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 t...
2006 Mar 18
0
T38 Passthrough testing -- unknown media type error
...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 s...
2006 Oct 31
1
wrong password on authentication for INVITE
...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.
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2007 Apr 19
1
Failed to authenticate on INVITE
...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
2007 Aug 02
0
callback and bridge problem
...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
[...
2009 Jan 26
1
Dial weirdness
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
2010 Mar 04
1
InterPBX communication using SIP
...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 &...
2012 Jul 29
0
just did sched_add waitid Warnings 1.8.14.1
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
2006 Jun 29
0
Asterisk with Sipbroker calling / routing problem
...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/2.0 407...