21 feb 2010 kl. 16.14 skrev voipas:
Hello,
I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup:
Reason: q.850;cause=17
No, you will have to change the code. I think there's a patch in the bug
tracker. Go search on issues.asterisk.org.
We do add a similar
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
19 feb 2010 kl. 10.22 skrev Randy R:
On Fri, Feb 19, 2010 at 8:54 AM, Olle E. Johansson o...@edvina.net wrote:
You propably have a type=friend where the user part matches before you even
hit the peer part, where the insecure configuration parameter matches. There
is a confusion here
19 feb 2010 kl. 11.47 skrev --[ UxBoD ]--:
exten == _988XXX.,1,Dial(DAHDI/g1/${EXTEN},20)
UxBoD - you really have to read the security advisory before sending out such
examples on the mailing list. Please go to http://www.asterisk.org now.
Have a nice weekend!
Thanks,
/O
--
17 feb 2010 kl. 19.12 skrev Joseph:
Does the sort order matter in sip.conf file?
I know sort order might effect:
allow=ulaw
allow=alaw
but does it matter where I place: insecure=invite ?
The reason I'm asking is that I've loaded almost two identical (sip.conf and
extension.conf)
17 feb 2010 kl. 23.15 skrev Michelle Dupuis:
Is it possible to just send an event from one Asterisk server to another?
(Perhaps some custom event that I could define?) Or would that break the SIP
protocol/handling in asterisk?
I think this discussion would be easier if you told us what you
While we continue discussing all possible solutions to this and build an
expanding knowledgebase, I would like to repeat myself and kindly ask everyone
that blogs, twitters, talks and teaches about Asterisk to please spread the
word and the links. Later today, there will be an official Asterisk
17 feb 2010 kl. 11.13 skrev Mian Asif:
Hi,
when call is Hangup, Asterisk send X-Asterisk-HangupCauseCode in Bye packet.
i want to remove Asterisk keyword from this string
X-Asterisk-HangupCauseCode.
please tell how i can remove Asterisk from above string at call hangup time.
You need to
17 feb 2010 kl. 12.37 skrev Håkon Nessjøen:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized
17 feb 2010 kl. 14.00 skrev Tzafrir Cohen:
On Wed, Feb 17, 2010 at 12:37:33PM +0100, Håkon Nessjøen wrote:
Only a warning, and doesn't seem to do anything bad.
But I can't seem to figure out what these warnings mean?
-- Requested transfer capability: 0x00 - SPEECH
[Feb 17 12:33:03]
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate a call:
Action: originate
Priority: 1
Context:
17 feb 2010 kl. 16.32 skrev Olle E. Johansson:
17 feb 2010 kl. 16.00 skrev James Northcott / Chief Systems:
Hi,
I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now
having a problem with Originate and chan_local.
I'm using the following Manager API action to originate
16 feb 2010 kl. 08.54 skrev Olivier:
Hi,
Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
TR-069 (see http://en.wikipedia.org/wiki/TR-069).
Has someone experienced with TR-069 ?
What do you think of this protocol set ?
And the SIP forum is about to release
16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
Yes but in any case you can enter all of the strings that reasonably match
- even if you have variable-length
16 feb 2010 kl. 10.40 skrev Alexandru Oniciuc:
Hello list,
debugging SIP, I found many empty lines like:
--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 ---
-
The IP address above corresponds to one of my accounts, which
is behind a
15 feb 2010 kl. 09.33 skrev Lenz Emilitri:
Or one could simply rewrite to:
[incoming-from-voip]
exten = XXX,1,Dial(${ext...@incoming-from-voip-old)
exten = ,1,Dial(${ext...@incoming-from-voip-old)
exten = X,1,Dial(${ext...@incoming-from-voip-old)
exten =
15 feb 2010 kl. 10.00 skrev Randy R:
On Mon, Feb 15, 2010 at 9:51 AM, Olle E. Johansson o...@edvina.net wrote:
To avoid extensive rewriting and fix the current issue.
That works in countries where you have fixed-length numbers. Unfortunately,
not every dialplan works that way, so that can't
15 feb 2010 kl. 17.36 skrev Steve Edwards:
On Mon, 15 Feb 2010, Amit Patkar | Avhan Technologies Pvt. Ltd. wrote:
I want at least 480 concurrent PSTN-IP calls.
0) Cross-posting is a no-no.
1) Not a -dev question. If you ever have any doubt a question belonging on
-dev, it doesn't.
15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere:
Playing around with the Grandstream GXV3140.
I'm interested in having the video voicemail clips emailed in a format
that might be opened by Windows Media Player or even Quicktime. Have been
googling around a lot and have tried various bits
- is education. I doubt it
will take too long to see script kiddies exploiting this.
I can not agree more!
Thank you for the feedback.
Regards,
/Olle
On Sat, Feb 13, 2010 at 6:04 PM, Olle E. Johansson o...@edvina.net wrote:
Friends,
Last week, Hans Petter Selansky alerted us
14 feb 2010 kl. 21.04 skrev Steve Edwards:
On Sun, 14 Feb 2010, Kyle Kienapfel wrote:
strip_ampersands(${EXTEN})?
(sip.conf)
[general]
allow-characters= all
disallow-characters =
[example-did-provider]
allow-characters
12 feb 2010 kl. 16.43 skrev Klaus Darilion:
Am 11.02.2010 21:09, schrieb Olle E. Johansson:
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found
13 feb 2010 kl. 16.57 skrev JR Richardson:
Hi All,
I read some discussions about the new SIP authentication methods for
1.6.X branches and possible addition of new type of user, type=trunk.
I'm wondering about the disposition about this. Will it be added?
Not to the 1.6 branches, but we
Friends,
Last week, Hans Petter Selansky alerted us of a potential security issue in all
releases of Asterisk. In fact, it doesn't involve the code, but the most common
way to construct dialplans. If you have something like this in your Asterisk,
you need to update your dialplans:
11 feb 2010 kl. 08.49 skrev Ron Arts:
Op 11-02-10 03:42, sean darcy schreef:
Kevin P. Fleming wrote:
sean darcy wrote:
I found out that the [globals] section in extensions.conf is ignored if
an #include 'd file has a [globals] section. Is this intended?
In this particular case, the
11 feb 2010 kl. 13.30 skrev Klaus Darilion:
Am 11.02.2010 11:21, schrieb Armin Schindler:
Hello,
using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.
I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.
My rtp.conf has
8 feb 2010 kl. 08.37 skrev Steve Totaro:
On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson o...@edvina.net wrote:
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673
8 feb 2010 kl. 11.26 skrev Tzafrir Cohen:
On Mon, Feb 08, 2010 at 11:03:19AM +0100, Olle E. Johansson wrote:
You will have to recompile it with the DONT_OPTIMIZE variable set so
that the core dump actually has meaningful symbols.
Doing so hurts your performance (and also slightly changes
8 feb 2010 kl. 12.29 skrev Klaus Darilion:
Hi!
IIRC there was an announcement some time ago that it is possible now to
make conferences without the need for DAHDI anymore - but I can not
remember the name of this feature anymore, and google didn't solved my
problem.
Thus, any
7 feb 2010 kl. 15.09 skrev Per Jessen:
Thomas Winter wrote:
Hi,
my Asterisk on debian lenny died after 80 days.
server kernel: [7572666.186852] asterisk[3673]:
segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l
ibpthread-2.7.so[7f3b8e903000+16000]
Anything what can be done to
5 feb 2010 kl. 09.28 skrev --[ UxBoD ]--:
- Randy R randulo2...@gmail.com wrote:
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net
wrote:
What I have seen on my asterisk box when I had a up/down adsl line
was
that the asterisk box couldn't do dns resolution and would
5 feb 2010 kl. 10.37 skrev Randy R:
Why not run a internal DNS with forwarders to your ISP ? That way Asterisk
can still resolve itself and hosts internally.
See above:
you need a local
resolver, like a caching BIND server, on the same host.
Nice, but still, it ruins the all in one
5 feb 2010 kl. 10.36 skrev Philipp von Klitzing:
Hi!
OpenVPN by default uses UDP, but can be configured to use TCP.
So what's the configuration on the Snom? Can I change it?
Google is your friend:
http://wiki.snom.com/Networking/VPN
So what you're saying is that you have full access
4 feb 2010 kl. 19.42 skrev Steve Totaro:
On Thu, Feb 4, 2010 at 11:30 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
- Ken D'Ambrosio k...@jots.org wrote:
It's just come to my attention that newer phones from both Snom and
Grandstream support OpenVPN. Is this a new trend or something?
4 feb 2010 kl. 21.54 skrev Alex Balashov:
On 02/04/2010 03:48 PM, Doug Lytle wrote:
OpenVPN by default uses UDP, but can be configured to use TCP.
So, under UDP, there should be no issues with retransmits.
It does have a primitive built-in backward acknowledgment mechanism even
for
5 feb 2010 kl. 06.49 skrev Anthony Messina:
On Thursday 04 February 2010 23:22:27 Alex Samad wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed like
3 feb 2010 kl. 08.11 skrev Alex Balashov:
On 02/03/2010 02:03 AM, Olle E. Johansson wrote:
2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
Hello all,
Does asterisk accept uri tel: instead of sip: ?
No, but I think it would be a good addition.
Why? Just curious.
Well, adding
2 feb 2010 kl. 11.20 skrev BERGANZ Francois:
Hello all,
Does asterisk accept uri tel: instead of sip: ?
No, but I think it would be a good addition.
/O
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30 jan 2010 kl. 23.40 skrev Michiel van Baak:
On 14:29, Sat 30 Jan 10, Olle E. Johansson wrote:
Friends,
Before the Christmas holidays, I did send this letter and did not get a lot
of response, but some. Since then, I've been able to get interest from a few
parties that are willing
29 jan 2010 kl. 17.20 skrev Kristian Kielhofner:
On Fri, Jan 29, 2010 at 10:31 AM, Kevin P. Fleming kpflem...@digium.com
wrote:
Well, that's the problem, and it's the reason why 603 is so commonly
used. This is a situation where the current request has failed, but
there is no indication
years...
With IPv6 greetings!
/Olle
Vidarebefordrat brev:
Från: Olle E. Johansson o...@edvina.net
Datum: 17 december 2009 09.39.40 CET
Till: Asterisk Non-Commercial Discussion Users Mailing List -
asterisk-users@lists.digium.com
Ämne: [asterisk-users] Asterisk IPv6 update - we need an update
Agree that the 603 is wrong. It hasn't caused me issues but I see where it
could. And it goes against what I have been teaching in my classes, which is
irritating ;-)
In Asterisk, it's only used when we have no other hangup cause - and is
propably an indication that there is a code path that
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich
Inviato: giovedì 28 gennaio 2010 21:41
A: asterisk-users@lists.digium.com
Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short
I would very much
-users
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27 jan 2010 kl. 11.47 skrev Administrator TOOTAI:
Hi,
we had an attack on a server and we don't understand how it was
possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL,
network 188.161.128.0/18
Hacked account had following setup:
[111]
type=friend
username=111
26 jan 2010 kl. 16.48 skrev Örn Arnarson:
Hi guys,
I am wondering (and have been unable to find out thus far) whether Asterisk
sets some special channel variables or something when a call is transfered
with the REFER method.
Basically, I'm trying to figure out if it is possible to
:
http://lists.digium.com/mailman/listinfo/asterisk-users
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asterisk-users
Hello!
I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria
or a recent Eyebeam on my mac. I know it used to work, but right now I get 100
trying and nothing else from the softphone.
Anyone that knows what's going on here?
Thanks,
/O
--
15 jan 2010 kl. 08.23 skrev Yuji Kondo:
I have two questions for Asterisk feature.
1. Can Asterisk support presence feature ?
Asterisk is a telephony PBX and supports presence subscriptions for extension
states - if a phone line is busy or not, over a few different SIP presence
12 jan 2010 kl. 20.56 skrev David Gibbons:
snip
'w' is really only supported on channels where digit-by-digit dialing is
the norm, which generally means analog trunks (or digital trunks using
CAS signaling).
/snip
Thanks Kevin, that's what I figured (though not quite so concisely)...
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
12 jan 2010 kl. 19.47 skrev Danny Nicholas:
Looking out for shots back on this, but Dial(SIP/X,w1234) should produce a
1/2 second delay before dialing, ww1234 a 1 second delay, etc.
Try it with 2 or 3 w's instead of 1...
I have no solution, but can only say this: a 'w' in a SIP dialstring
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest version (1.6) . Can you please let me
know what are the major benefits when upgrading from Asterisk 1.4 to Asterisk
1.6 ?
Please
My apologies for the multiple copies.
Had issues with a mailserver that somehow wasn't talking to DNS properly. Now
fixed. It behaved like Asterisk does sometimes, very poor when it can't connect
to DNS. Had power outage yesterday and I think that started it all...
Meanwhile, I tried to
13 jan 2010 kl. 09.26 skrev hadi motamedi:
On Wed, Jan 13, 2010 at 7:36 AM, Olle E. Johansson o...@edvina.net wrote:
13 jan 2010 kl. 06.56 skrev hadi motamedi:
Dear All
I have Asterisk 1.4 installed on my Debian server . I am considering
upgrading my Asterisk to the latest
11 jan 2010 kl. 12.25 skrev ahmed magdy:
Hello,
I am new in Asterisk Community, i am working on Asterisk 1.6, i need to know
how can i monitor the extension status?
when i wrote sip show peers on asterisk
Extension Domain port Status
111/111
11 jan 2010 kl. 16.23 skrev --[ UxBoD ]--:
Hi,
why would Asterisk core dump with the following test dialplan extension ?
exten = 8100,1,Answer()
exten = 8100,n,Set(CALLERID(all)=)
exten = 8100,n,PrivacyManager()
exten = 8100,n,GotoIf(${[${PRIVACYMGRSTATUS} = FAILED]}?:nocid)
exten =
Net::DNS::Async is a fire-and-forget asynchronous DNS helper. That is, the
user application adds DNS questions to the helper, and the callback will be
called at some point in the future without further intervention from the user
application. The application need not handle selects, timeouts,
8 jan 2010 kl. 08.01 skrev Tilghman Lesher:
On Thursday 07 January 2010 21:17:52 JR Richardson wrote:
On Thu, 7 Jan 2010, Tilghman Lesher wrote:
On Thu, Jan 7, 2010 at 6:27 PM, JR Richardson wrote:
problem I'm running into is if the DNS server is not responding, the
script hangs and waits
7 jan 2010 kl. 10.21 skrev Aggio Alberto:
Hi,
I have occasionally experienced the same problem too, and I suspect it was
caused by some spikes in network traffic (e.g. for an intensive file
transfer) that delayed too much SIP OPTION response, so that Asterisk marked
these devices as
7 jan 2010 kl. 12.00 skrev Olivier:
Hello,
I've read in Mantis that asterisk.conf's internal timing option could
positively impact Asterisk behaviour during faxing
(http://issues.asterisk.org/view.php?id=16374).
Before using it, I would be very pleased to read a line or two about its
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4 jan 2010 kl. 09.34 skrev Remco Barendse:
Is there any fix or workaround for the DNS problem (old standing bug that
when the box starts and domain names do not resolve quickly enough from
DNS then asterisk stops using the outgoing trunks.
I read on the list before that it is considered
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint is IP x.x.x.x port x, and I can use codecs
A,B,C. The remote endpoint responds with a 200 OK, saying my audio
stream is at IP
5 jan 2010 kl. 10.08 skrev hadi motamedi:
On Tue, Jan 5, 2010 at 8:59 AM, Olle E. Johansson o...@edvina.net wrote:
4 jan 2010 kl. 14.46 skrev Kevin P. Fleming:
hadi motamedi wrote:
Sorry . I didn't get the point clearly . In the SIP Invite message , it
says my audio endpoint
/asterisk-users
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3 jan 2010 kl. 17.47 skrev Steve Edwards:
1 jan 2010 kl. 20.04 skrev Shariq Khan:
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
On Sun, 3 Jan 2010, Olle E. Johansson wrote:
No, Asterisk only supports one port.
You can configure OpenSER/Kamailio/OpenSIPS
23 dec 2009 kl. 19.52 skrev Ira:
Someone posted a message suggesting someone try sendtext() and so I
thought I'd see if it was useful. Much searching through help at the
CLI has failed to find any help for sendtext, but I did find that:
core show function vmcount fails but:
core show
23 dec 2009 kl. 16.00 skrev didier.cuffaut:
I apologize for my poor English.
So, i don't really understand 'how to' realize thus
When you use the cmd Dial and want to get $ from caller channel to callee (or
callee channel from caller), which way is the right way ?
If you
24 dec 2009 kl. 08.18 skrev listu...@spamomania.co.uk:
Hi,
How would I go about troubleshooting this:
[Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
retries exceeded on transmission a50346a4-bfdc3...@192.168.1.95 for
seqno 101 (Critical Response) -- See
24 dec 2009 kl. 10.30 skrev Shelvananda, Ramananda Arkalgud:
Hi All,
Is someone implemented Tel uri support in the latest asterisk ? If yes, can
you guys share some info on it
No.
But I am very interested in why you ask? Do you have devices that support Tel:
uri's? DO you have an
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23 dec 2009 kl. 06.17 skrev prasha...@digilink.in:
Hi,
How asterisk distinguish whether the re-invite is for codec change or for a
session refresh? I know that it checks the session version and decides the
same. But even if session version is different from the initial invite and
23 dec 2009 kl. 08.53 skrev jonas kellens:
Can I define the realm on a per peer basis ??
Can I define a realm to be used for one peer and another realm for another
peer in sip.conf ??
I have an ITSP that I need to authenticate with a realm that they set. But
this realm is not valuable
23 dec 2009 kl. 10.16 skrev Zhang Shukun:
hi, all
when i run make menuselect, it say
Terminal must be at least 80 x 21.
menuselect changes NOT saved!
in the bottom message, what's wrong?
Terminal must be at least 80x21
You need a terminal window that handles at least 80 characters
23 dec 2009 kl. 11.25 skrev David Cunningham:
Shukun,
It tells you No such file or directory. Is the file in your modules
directory?
Actually, to be more specific. The module cdr_radius.so exists, but can't bind
to the radius library libradiusclient-ng.so.2.
Check LD_LIBRARY_PATH
/O
Dear Asterisk community,
Yesterday, Russell Bryant finally made up his mind and confirmed on the
asterisk-dev mailing list that the next release of Asterisk will be 1.8, which
will also be a Long Term Support (LTS) release. This also means that the 1.4 is
now officially classed as a LTS
21 dec 2009 kl. 09.34 skrev Olivier:
2009/12/21 Olle E. Johansson o...@edvina.net
21 dec 2009 kl. 00.04 skrev Olivier:
Hi,
I'm banging my head over this.
Usually, I'm using a SIP hardphone feature called Call Pickup Starcode to
enhance BLF with Directed Call Pickup
21 dec 2009 kl. 12.00 skrev jonas kellens:
My SIP-provider sends my a SIP-invite like this :
INVITE sip:329298y...@80.xx.xx.69:5060 SIP/2.0
Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
Max-Forwards: 70
From:
20 dec 2009 kl. 09.10 skrev Magnus Benngård:
Hi!
Probably me that cannot read the manual...
I am trying to get all Keys that belongs to a certain Family
from the manager interface. Can just get single values for example:
Action: DBGet
Family: CFIM
Key: 0317998975
While discussing
21 dec 2009 kl. 15.11 skrev Danny Nicholas:
You can do virtually any command with the manager command object; the trick
is getting the syntax down. There's a decent example on voip-info.org that I
used to set up mine.
Well, that's one way, but you have to be careful, because the CLI output
21 dec 2009 kl. 00.04 skrev Olivier:
Hi,
I'm banging my head over this.
Usually, I'm using a SIP hardphone feature called Call Pickup Starcode to
enhance BLF with Directed Call Pickup :
basically, SIP hardphone (here a Thomson ST2030S) is configured to send an
INVITE message whenever
Friends,
At the first Astricon I was very happy to see Marc Blanchet as one of the
attendees. I knew he was one of the IPv6 gurus and wanted someone to show some
interest in Asterisk and IPv6.
Well, he did not only get interested in it, but started coding on it. The
results have been
point for exploration.
Cheers,
/O
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11 dec 2009 kl. 23.21 skrev John Taylor:
I have multiple trunks to the same ITSP. Incoming calls to any trunk
go to the last incoming label defined in those trunks' contexts in
sip.conf.
My ITSP insists on insecure=very in the trunk context; is this the cause?
This is an effect of the
Hello,
Several folks working with Kamailio, SIP Router, SER, OpenIMSCore,
SEMS and Asterisk are in Berlin next week, so we think of having a
dinner (or beer) meeting Thursday, 19:00, Nov 12, 2009. If happens
that you are around and want to join, please send me an email to make
sure you
14 sep 2009 kl. 12.05 skrev Stanisław Pitucha:
2009/9/9 Stanisław Pitucha s...@gradwell.net:
I've got different customers that may use the same asterisk. Each
user
can blind-transfer a call to whatever place they want. But of course
the transferring side should be billed for it.
What can
10 sep 2009 kl. 12.33 skrev DHAVAL INDRODIYA:
hello
I would like to take value RTPAUDIOQOS channel variable on DAHDI /
IAX Channel...
DAHDI doesn't use the Realtime Transport Protocol, RTP.
/O
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10 sep 2009 kl. 17.35 skrev Alex Balashov:
Andrew Stewart wrote:
Figured out the problem. There is an inspect sip command in our
global policy map on our Cisco ASA firewall. That was fixing the
CALL-ID. Took it out and all is working now.
Ah, yes. Those ALGs (or other = Layer 5
You can also use our jabber/xmpp integration and send an Instant
message to the user/desktop before you place the call with dial(). Or
do it in the dial() macro as soon as someone answers.
/O
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8 sep 2009 kl. 10.17 skrev jonas kellens:
Erik,
I have placed everything in features.conf in comment ( ; ). Still
when I run show features, I get this :
clarkconnect*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup
8 sep 2009 kl. 15.40 skrev Benny Amorsen:
I see a lot of these on an otherwise idle Asterisk 1.6.0.15:
Extension Changed 773[Hints] new state Ringing for Notify User
792-00041327d17e-1. Then a little while later it changes to InUse or
Idle, completely randomly. It happens for many different
5 sep 2009 kl. 20.02 skrev Jens Wolf:
Hi,
i have try to send IM from Client A (Ekiga) to Client B (Ekiga).
Realtime text is sent in the RTP stream. What you're sending is a SIP
message. Those are two different things.
Asterisk does not support SIP messages outside of a call - so test
5 sep 2009 kl. 01.05 skrev Matt Riddell:
On 4/09/09 6:22 PM, Olle E. Johansson wrote:
4 sep 2009 kl. 00.44 skrev Matt Riddell:
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR
5 sep 2009 kl. 04.58 skrev Jai Rangi:
Hello,
I have a issue between asterisk and windows based VoIP system
(Client).
Vendor SIP Server -- My asterisk -- Client
Here is ethereal trace between asterisk and client.
1 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE
- 192.168.3.222 SIP/SDP Status: 200 OK,
On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson o...@edvina.net
wrote:
5 sep 2009 kl. 04.58 skrev Jai Rangi:
Hello,
I have a issue between asterisk and windows based VoIP system
(Client).
Vendor SIP Server -- My asterisk -- Client
Here
3 sep 2009 kl. 11.40 skrev Michiel van Baak:
On 14:24, Thu 03 Sep 09, ABBAS SHAKEEL wrote:
Hello
Previous context :- After Looking up sip and IAX2 that require
configuration at router level which may cause some problems like
connection
break etc... so i left them . and start
4 sep 2009 kl. 00.44 skrev Matt Riddell:
On 4/09/09 10:41 AM, Doug Lytle wrote:
Todd Routhier wrote:
Trying to do something like this in the sip.conf under my incoming
provider profiles:
setvar=CDR(accountcode)=${EXTEN}
Set(CDR(accountcode)=${EXTEN})
Nah he's trying to do it in
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