this error
On Mon, Aug 9, 2010 at 11:21 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Subject:* Re: [asterisk-users] SIP response 500 Server Internal Error
Hi,
I
Hi,
I have problem in initiating an dial out call with SIP response 500 Server
Internal Error
The sip debug as
== Using SIP RTP CoS mark 5
Audio is at 113.253.226.92 port 18284
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Subject: Re: [asterisk-users] SIP response 500 Server Internal Error
Hi,
I have problem in initiating an dial out call with SIP response 500
Server Internal Error
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *asterisk asterisk
*Subject:* Re: [asterisk-users] SIP response 500 Server Internal Error
Hi,
I have problem in initiating an dial out call with SIP response 500
Server Internal Error
The sip debug as
snip
--- SIP read from UDP
On Tue, Jul 6, 2010 at 11:08 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to
- Original Message -
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients.
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that we are
unable to URI dial our clients. We run a multi-tenant server and have set
sip.conf to forward calls to a public context based on incoming domain name.
This was all working before but not it is complaining of a loop
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a public context based on
incoming domain name. This was all working before
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found that
we are unable to URI dial our clients. We run a multi-tenant server
and have set sip.conf to forward calls to a
- Original Message -
On Mon, Jul 5, 2010 at 4:20 AM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
- Original Message -
Hi,
We have tried upgrading from 1.6.1.14 to 1.6.2.9 and have found
that
we are unable to URI dial our clients. We run a multi-tenant server
and have
dear all,
what is the meaning of this
*Got SIP response 603 Declined back from XXX.XXX.XXX.XXX*
is it asterisk related issue , because sometimes my outgoing calls working
fine , and in a day for 2 to 3 hours it gives me this
my provider says its all fine there any one know meaning of this
DHAVAL INDRODIYA wrote:
dear all,
what is the meaning of this
*Got SIP response 603 Declined back from XXX.XXX.XXX.XXX*
is it asterisk related issue , because sometimes my outgoing calls
working fine , and in a day for 2 to 3 hours it gives me this
my provider says its all fine
thanks Alex,
thanks for your reply,
is there any changes needed for resolving this issue , in sip.conf or need
to change any dial parameter
i currently Use IP-to-IP Dialing with option 'm' , also sometimes i got 503
Service Unavailable,
is there any eay to resolve it , if anything then please
The problem is with your provider, unless there is something wrong
with about 1/4th of your calls - i.e. the destination is unroutable by
that provider.
DHAVAL INDRODIYA wrote:
thanks Alex,
thanks for your reply,
is there any changes needed for resolving this issue , in sip.conf or
Hello all,
I have being trying to replicate the following call scenario with my
Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html
http://www.tech-invite.com/Ti-sip-service-8.html
I have a situation that I have to notify the calling party that the call is
being forwarded to
Marco Cordeiro schrieb:
I have being trying to replicate the following call scenario with my
Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html
http://www.tech-invite.com/Ti-sip-service-8.html
I have a situation that I have to notify the calling party that the call is
being
Philipp Kempgen schrieb:
Marco Cordeiro schrieb:
I have a situation that I have to notify the calling party that the call is
being forwarded to another number. So far, in the tests that I made, calling
from a SIP extension to another SIP extension with the forwarding activated,
I get only
Hi,
I've a probleme since few weeks that I don't be able to solve.
I use Thomson ST2030 phone and I've an error when I want to do an
attended transfer with the soft key.
The receiver of the transfer return an : Got SIP response 400 Bad
Request back from 192.168.2.13
The direct transfer with
Hi,
Johansson Olle E wrote:
Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by
checking HANGUPCAUSE instead of DIALSTATUS and you will get many more
details.
Great, that's all I need:
It gives me more ways to analyse the different reason for the hangup and
I can use the
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
-- Got SIP response 480 Do Not Disturb back from 192.168.0.34
--
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
-- Got SIP response 480 Do
Johansson Olle E schreef:
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in new stack
-- Called user3
--
15 apr 2008 kl. 13.38 skrev Ron Arts:
Johansson Olle E schreef:
15 apr 2008 kl. 12.06 skrev Stefan Guenther:
Hello,
I have switched on DND on a SNOM 360. When I call this phone, I get
the
following output:
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8,
SIP/user3|20|tr) in
The only solution that I found for this is to use Asterisk 1.4 with
devstate backport
(http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/)
and use the hints and to determine if it's inuse (or any other status)
before the dialing - in order to generate a proper reply. I
In what amount of time does 100 Trying message have to be
sent to asterisk? I see asterisk retransmitting the INVITE
message multiple times before receiving the 100 Trying message.
The INVITEs are retransmitted based on a timer T1, which starts at a default
of 500 ms and then exponentially
I need to know how fast a sip device needs to respond
to an INVITE sip message from asterisk before asterisk
retransmits the INVITE message again.
Thanks
__
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On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote:
I need to know how fast a sip device needs to respond
to an INVITE sip message from asterisk before asterisk
retransmits the INVITE message again.
Thanks
Snip ---
7.2.1 INVITE received
When an INVITE request is received by the
In what amount of time does 100 Trying message have
to be sent to asterisk? I see asterisk retransmitting
the INVITE message multiple times before receiving the
100 Trying message.
--- David Boyd [EMAIL PROTECTED] wrote:
On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote:
I need to know how
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
Look at:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to another via SIP and on
the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from
xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
These messages also
Well, but isn't lines that begin with -- on the same verbosity level?
So lowering the verbosity would in this case mean that you also stop
displaying the dialplan execution steps. I have a similar problem
regarding the -- SIP Seeding peer from astdb messages. I get a lot of
these, so I tried
22 feb 2007 kl. 11.19 skrev Torbjörn Abrahamsson:
Well, but isn't lines that begin with -- on the same verbosity
level? So lowering the verbosity would in this case mean that you
also stop displaying the dialplan execution steps. I have a similar
problem regarding the -- SIP Seeding peer
I do need MWI notifcation, just not on this particulary trunk. Is there
anyway to to turn off MWI on a particular trunk or can it only be done
globally?
On 2/22/07, Olle E Johansson [EMAIL PROTECTED] wrote:
22 feb 2007 kl. 08.24 skrev Davy Chan:
**I have one Asterisk box registering to
Agreed, but your response to the OP said to lower the verbosity, and I
commented that it might not be possible, due to then seeing no dialplan
execution... :)
How about the seeding messages then? Will you move these to a debug
level? Or do a bug need to be filed in Mantis?
// Tobbe
Olle E
22 feb 2007 kl. 11.36 skrev Eric Bishop:
I do need MWI notifcation, just not on this particulary trunk. Is
there anyway to to turn off MWI on a particular trunk or can it
only be done globally?
You enable it per device in sip.conf - that's the only way.
/O
22 feb 2007 kl. 11.54 skrev Torbjörn Abrahamsson:
Agreed, but your response to the OP said to lower the verbosity,
and I commented that it might not be possible, due to then seeing
no dialplan execution... :)
Well if you want that level of detail during the execution, these
error messages
OK, nice. Any chance of it finding its way into 1.2-branch?
I agree in some extent that one get a lot of information when looking at
the dialplan execution, but the difference is that this is usefull
information. Looking at the dialplan pass by is not made easier by
having the seeding
**I do need MWI notifcation, just not on this particulary trunk. Is there
**anyway to to turn off MWI on a particular trunk or can it only be done
**globally?
**
**On 2/22/07, Olle E Johansson [EMAIL PROTECTED] wrote:
**
** Why enable MWI notification when you don't need it?
I guess my
I have one Asterisk box registering to another via SIP and on the registar
console I keep getting:
-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx
Anyone know how to turn off this feature?
___
--Bandwidth and Colocation provided
**I have one Asterisk box registering to another via SIP and on the registar
**console I keep getting:
**
**-- Got SIP response 603 Declined (no dialog) back from xxx.xxx.xxx.xx
**
**Anyone know how to turn off this feature?
Look at:
Surely there must be a simpler way than patching the Asterisk code? After
all this is Asterisk-to-Asterisk registration not some third party
softswitch idiosyncrasy. Would setting up fake voicemail boxes help?
On 2/22/07, Davy Chan [EMAIL PROTECTED] wrote:
**I have one Asterisk box
-users] SIP response 482 Loop Detected
I have a Cisco Call Manager - and need to use the IVR Feature from
Asterisk.
My extension is 400 and I am calling 558 on Asterisk In my
extension.conf I have these lines :
exten = 558,1,Answer
exten = 558,2,Playback(message.wav)
exten = 558,3,Dial(SIP
-users] SIP response 482 Loop Detected
I have a Cisco Call Manager - and need to use the IVR Feature from
Asterisk.
My extension is 400 and I am calling 558 on Asterisk In my
extension.conf I have these lines :
exten = 558,1,Answer
exten = 558,2,Playback(message.wav)
exten = 558,3,Dial(SIP
I have a Cisco Call Manager - and need to use the IVR Feature from
Asterisk.
My extension is 400 and I am calling 558 on Asterisk In my
extension.conf I have these lines :
exten = 558,1,Answer
exten = 558,2,Playback(message.wav)
exten = 558,3,Dial(SIP/[EMAIL PROTECTED])
When I call 558
Hi everyone,
This is a message I am getting on the Asterisk console, 192.168.1.80
refers to my Nokia E61, any ideas what this means?
-- Got SIP response 400 Bad Request back from 192.168.1.80
Thanks.
___
--Bandwidth and Colocation provided by
[Jan 23 19:56:44] -- Got SIP response 300 Multiple choice back
from 194.120.0.201
[Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to
'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED]'
(thanks to SIP/voipstunt-5c8c)
[Jan 23 19:56:44] NOTICE[3439]:
Hello,
I saw that the error:
SIP response 484 Address Incomplete
is converted into
DIALSTATUS = NOANSWER
HANGUPCAUSE = 16 (NORMAL_CLEARING)
shouldn't it be something like
HANGUPCAUSE = 1 (UNALLOCATED)
HANGUPCAUSE = 28 (INVALID_NUMBER_FORMAT)
or another cause, other than NORMAL ???
Hi all
I am using Asterisk 1.2.0 on debian sarge, and linphone as client.
I get this message at regular interval:
-- Got SIP response 481 Subcription Does Not Exist back from
192.168.0.117
..117 is me.
Everything seems to work fine, however. Does anyone has a clue?
Is there a SIP client for
On Sun, Nov 20, 2005 at 10:07:39AM +0100, Fred Blaise wrote:
Hi all
I am using Asterisk 1.2.0 on debian sarge, and linphone as client.
I get this message at regular interval:
-- Got SIP response 481 Subcription Does Not Exist back from
192.168.0.117
..117 is me.
Everything seems to
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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google
On Mar 22, 2005, at 12:59, Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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Asterisk-Users@lists.digium.com
rfc3261
http://www.faqs.org/rfcs/rfc3261.html
Ronald Wiplinger wrote:
Where can I get a list of all possible SIP ... response numbers and
their meaning?
bye
Ronald
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A9.com: http://a9.com/SIP%20response%20numbers%20
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: Tuesday, March 22, 2005 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP
Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get:
Got SIP response 302 Moved Temporarily
When forwarding the call to other SIP server.
This is a bug:
http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html
---
Jan Baggen - [EMAIL PROTECTED]
IP2 Internet BV /
]
Sent: Monday, December 06, 2004 7:01 AM
Subject: [Asterisk-Users] SIP response 302 Moved Temporarily
Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get:
Got SIP response 302 Moved Temporarily
When forwarding the call to other SIP server.
This is a bug:
http://lists.digium.com
Hi all,
if you turn on Anonymous Call Block on a Cisco 7960, the phone rejects
incoming calls that have Anonymous as callerid with
-- Got SIP response 488 Not Acceptable Here back from 192.168.1.1
== No one is available to answer at this time
Normaly, the next priority after the dial-command
I have a dlink dvg-1120s voip-router. I can make calls out from the router,
but when calling the router I got
-- Executing Dial(SIP/2010-b437, SIP/2021|30|r) in new stack
-- Called 2021
-- Got SIP response 404 Not Found back from 62.79.78.74
-- SIP/2021-473b is circuit-busy
What
Hi!
I am getting the following error message:
Got SIP response 403 That is ugly -- use From=id next
time (OB) back from 195.37.77.101
I'm not quite sure what that means. Does anybody know
what I might have done wrong?
Here is my configuration:
sip.conf
register = account:[EMAIL
Hey,
--- Philipp von Klitzing
[EMAIL PROTECTED] wrote:
Hi!
I am getting the following error message:
Got SIP response 403 That is ugly -- use From=id
next
time (OB) back from 195.37.77.101
I'm not quite sure what that means. Does anybody
know
what I might have done wrong?
It means that the username in From and the username in digest
credentials are different.
The reason for this test is that we do not want our users to pretend
that they are somebody else. Without this test it would be possible to
put [EMAIL PROTECTED] in From and all phones will display it,
hi. read my mail 'bout 487 response.
I wrote a patch to fix that in chan_sip .
It's good for a occasional fix, until
mark updates chan_sip to handle retransmissions.
matteo.
Il mar, 2003-03-18 alle 16:58, Christoph Frei ha scritto:
Hello Guys,
i'm using a Cisco 7960 and a Cisco 2610 with a
Hi,
Asterisk is not sending the CANCEL when you hang up back to the gateway
that Matches the ORIGNAL INVITE Request.
It looks like it was fixed with the last cvs release.
Dave
On Tue, 2003-03-18 at 10:58, Christoph Frei wrote:
Hello Guys,
i'm using a Cisco 7960 and a Cisco 2610 with a
His 481 issue is not the same as your 487 issue.
Mark
On 18 Mar 2003, Brancaleoni Matteo wrote:
hi. read my mail 'bout 487 response.
I wrote a patch to fix that in chan_sip .
It's good for a occasional fix, until
mark updates chan_sip to handle retransmissions.
matteo.
Il mar,
You could turn off reinvite.
Mark
On Thu, 6 Mar 2003, Eric Wieling wrote:
I'm getting the following message:
-- Executing Dial(SIP/2111-b825, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/2111-0bd5 answered SIP/2111-b825
-- Attempting native bridge
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