About two weeks ago there was a thread about asterisk suddenly dying - I
posted a response that the same happens to my asterisk about once a
month, sometimes more.
Someone suggested using 'safe_asterisk' (and get hold of a core dump)
which sounds like a good idea, but one thing I can't figure is
On 22/02/10 16:18, --[ UxBoD ]-- wrote:
Hi,
looking for your valued input on suitable suggestions for high quality VoIP
DECT phones. I am having real issues with my Snom M3s and Asterisk 1.6 and
looking to a new manufacturer.
Another vote for the Siemens Gigaset range. Been using the
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News) alansli...@gmail.com wrote:
Another vote for the Siemens Gigaset range. Been using the S685IP almost
since the day it was released here in the UK. Nice handsets, great voice
quality, but as others have said the UI can be a bit slow.
Alan, don't
On Mon, 22 Feb 2010, Gordon Henderson wrote:
On Mon, 22 Feb 2010, --[ UxBoD ]-- wrote:
Hi,
looking for your valued input on suitable suggestions for high quality
VoIP DECT phones. I am having real issues with my Snom M3s and Asterisk
1.6 and looking to a new manufacturer.
Siemens
Hello,
worst aspect is that - if SIP clients do not have such a timeout, and in
that case if killing an asterisk and to start it up again -
so it is nothing to do with this asterisk timeout.
Regards,
On 23 February 2010 08:44, Olle E. Johansson o...@edvina.net wrote:
23 feb 2010 kl. 01.47
On 23/02/10 08:38, Randy R wrote:
On Tue, Feb 23, 2010 at 9:23 AM, Alan Lord (News)alansli...@gmail.com
wrote:
Another vote for the Siemens Gigaset range. Been using the S685IP almost
since the day it was released here in the UK. Nice handsets, great voice
quality, but as others have said
- Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
looking for your valued input on suitable suggestions for high
quality
VoIP DECT phones. I am having real issues with my Snom M3s and
Asterisk
1.6 and looking to a new manufacturer.
Define high quality.
Hello,
I am relative new to Asterisk and we want the following:
ExternalCall--UserPBX--DialOutNormal
| ^
V |
Asterisk
| ^
V |
Application
We have the above configuration and we would like to tell
Thank you Steve, that's a good idea.
If I use a global variable like
-- IF GLB 2 GLB = 0
dial(iax2/isp${GLB}/${EXTEN})
-- GLB = GLB +1
I believe this could cause a race condition if two calls are sent to
the carrier at the same time?
--
On 23 Feb 2010, at 09:58, Bert Mengerink wrote:
I know we need a trusted relation between the UserPBX and the
Asterisk, but what command do we need to instruct the UserPBX to set
the original call from ExternalCall through to DialOutNormal?
Ask whoever made 'UserPBX'?
S
--
Hi Steve,
UserPBX could be any brand PBX, like Ericson, Avaya, etc. Or even
another asterisk.
Therefor I added the provision, that the UserPBX should support such a
strategy. Is there a general SIP command to provide the action we want?
Kind regards,
Bert
-Original Message-
From:
Hi All,
We have encountering issue that IAX enable voice gateways not registering
with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29
Before that IAX works very well.
If any one have similar issue and solution for that let me know.
--
Thanks Regards,
Vidura Senadeera,
Sri Lanka.
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote:
About two weeks ago there was a thread about asterisk suddenly dying - I
posted a response that the same happens to my asterisk about once a
month, sometimes more.
Someone suggested using 'safe_asterisk' (and get hold of a core dump)
On Mon, Feb 22, 2010 at 11:23:29PM +, Gordon Henderson wrote:
On Mon, 22 Feb 2010, Roderick A. Anderson wrote:
Gordon Henderson wrote:
Interesting thread recently about virtual servers...
I'm thinking of doing something similar - right now looking at Containers
(lxc) rather than
On Tue, Feb 23, 2010 at 10:50 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
High quality to me means well built, reliable, good protocol support and
above all a responsive manufacturer.
Incidentally, I've dropped two of the S675IP handsets on the hardwood
floor a few times, still working fine.
Tzafrir Cohen wrote:
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote:
About two weeks ago there was a thread about asterisk suddenly dying
- I posted a response that the same happens to my asterisk about once
a month, sometimes more.
Someone suggested using 'safe_asterisk' (and
Hi Group,
Can anybody explain me in detail how the codec translation happens on
asterisk side when 2 endpoints have different codecs?
Thanking you in advance.
--SM
--
_
-- Bandwidth and Colocation Provided by
My dear friend Matt Riddell insists that the Manager only can dial 5 calls
per seconds, which I find ridiculous. Is there a way to prove him wrong and
have him lift the limit that has been plaguing the life of us users of
SineDialer and SmoothTorrque
Philip
--
But this won't help if 100 or 101 wants to call 102.
What I want is, if a call coming from a trunk 100 rings, and if the
caller wants to be transfered to 101, the transfer is denied. In other
words, 101 can't get transfered calls.
Danny Nicholas wrote:
Follow-me will most likely be your best
On Tue, 23 Feb 2010, Tzafrir Cohen wrote:
My aim is to actually use LXC as it has kernel level support (as of
2.6.29) and will be supported by most distros soon if not already.
Linux-Vserver appears to be depreciated by at least Debian, probably
Ubuntu too, but I've no idea about the world of
- Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote:
Hi!
looking for your valued input on suitable suggestions for high
quality
VoIP DECT phones. I am having real issues with my Snom M3s and
Asterisk
1.6 and looking to a new manufacturer.
Define high quality.
On Tue, Feb 23, 2010 at 12:19:31PM +, Gordon Henderson wrote:
On Tue, 23 Feb 2010, Tzafrir Cohen wrote:
But then again, lxc uses much of the work on containers done also by and
for OpenVZ. Sort of like the VMWare/Xen/KVM story all over again, with
lxc playing the role of KVM.
And
On 22 February 2010 14:07, Razza razz...@gmail.com wrote:
I'm using CentOS5.4, can anyone advise how I can make DAHDi work with
a generic HFC-S card?
On 22 February 2010 15:12, Pedro Santos pnlsan...@gmail.com wrote:
I´m using centos 4.8 server, and i don't now how integrate zaphfc with dadhi
On Tue, Feb 23, 2010 at 12:48:15PM +, Razza wrote:
On 22 February 2010 14:07, Razza razz...@gmail.com wrote:
I'm using CentOS5.4, can anyone advise how I can make DAHDi work with
a generic HFC-S card?
On 22 February 2010 15:12, Pedro Santos pnlsan...@gmail.com wrote:
I´m using centos
On 23 February 2010 12:58, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Have you managed to install those zaphfc drivers?
Those are basically the same ones from http://code.google.com/p/zaphfc/
Hi Tzafrir. I checkout out that but there were no instructions.
--
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to have the RTP stream bypass the gatewayy altogether
(directrtp/reinvite). Is this possible with these to protocols?
Thanks
Hi!
We have encountering issue that IAX enable voice gateways not
registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29
Before that IAX works very well. If any one havesimilarissue and solution
for that let me know.
Search or google for calltokenoptional.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Michelle Dupuis skrev:
We're creating a SIP gateway for a client that will take one leg of a
call in via SIP, and out the other side via H.323. To minimize load on
the gateway, we would like to have the RTP stream bypass the gatewayy
altogether
Tommy Botten Jensen wrote:
Michelle Dupuis skrev:
We're creating a SIP gateway for a client that will take one leg of a
call in via SIP, and out the other side via H.323. To minimize load on
the gateway, we would like to have the RTP stream bypass the gatewayy
altogether
On Tue, 2010-02-23 at 08:22 -0500, Michelle Dupuis wrote:
We're creating a SIP gateway for a client that will take one leg of a
call in via SIP, and out the other side via H.323. To minimize load
on the gateway, we would like to have the RTP stream bypass the
gatewayy altogether
On Tuesday 23 February 2010 05:27:55 Per Jessen wrote:
Tzafrir Cohen wrote:
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote:
About two weeks ago there was a thread about asterisk suddenly dying
- I posted a response that the same happens to my asterisk about once
a month,
On Tue, 23 Feb 2010, Alejandro Recarey wrote:
If I use a global variable like
-- IF GLB 2 GLB = 0
dial(iax2/isp${GLB}/${EXTEN})
-- GLB = GLB +1
I believe this could cause a race condition if two calls are sent to the
carrier at the same time?
Yes. If that's an issue for your carrier,
Hi,
I am registering my Asterisk boxes to a SIP provider for outgoing calls.
My outgoing dialplan context tries to dial out in sequence, starting with the
SIP provider then ISDN lines and finally analog lines.
So the idea is that if the SIP trunk fails then all calls are dialed out via
ISDN
Hi!
My outgoing dialplan context tries to dial out in sequence, starting
with the SIP provider then ISDN lines and finally analog lines.
[...]
When the DSL is down I get:
sip show registry:
HostUsername Refresh State
Reg. Time
Ex-girlfriend is another answer to your query.
Set it up like this
Exten = 101,1,Verbose(let's call ext 101)
Exten = 101/100,n,Dial(SIP/101,20,KkTt)
Exten = 101/102,n,Dial(SIP/101,20,KkTt)
Exten = 101,n,Playback(cant-dial-it)
Exten = 101,n,hangup
-Original Message-
From:
--- On Tue, 2/23/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Look at qualify= for sip.conf, and consider to extend your
diaplan for a
better routing decision with a snippet like this:
exten = _00.,n,Set(VOIPCHECK=0)
exten = _00.,n,NoOp(-- ${PEERCHECK1}
Also, why are you saying your name is Philip?
On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote:
My dear friend Matt Riddell insists that the Manager only can dial 5
calls per seconds, which I find ridiculous. Is there a way to prove
him wrong and have him lift the limit that has
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
On 24/02/2010, at 12:59 AM, CDR vene...@gmail.com wrote:
My dear friend Matt Riddell insists that the Manager only can dial 5
calls per
We're doing a project that requires H.323 to an Avaya. Does anyone have
experience to share on which H.323 driver to use in asterisk 1.6? Is the
diference between h323 and ooh323 still worth the extra effort? (We've only
installed h323 under 1.4)
If you have setup/config experience with this
Hi All,
We have encountering issue that IAX enable voice gateways not
registering with asterisk after upgrade from asterisk 1.4.18.1 - 1.4.29
Before that IAX works very well.
If any one have similar issue and solution for that let me know.
Check for ERROR[] chan_iax2.c: Call rejected,
So you're saying that you could at least theoretically push more than 5 CPS
through, you just would get a lot of no account code responses? Reading
the SmoothTorrque Wiki, I could see where a user might want to process more
than 300 CPM (5*60), but if I'm going to spend the money for over 300
On Tue, Feb 23, 2010 at 09:18:36AM +0100, Per Jessen wrote:
About two weeks ago there was a thread about asterisk suddenly dying
- I posted a response that the same happens to my asterisk about once
a month, sometimes more.
Someone suggested using 'safe_asterisk' (and get hold of a core
23 feb 2010 kl. 20.18 skrev Matt Riddell:
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
Where's the bug report id?
I haven't heard about this limit. I don't know what it is, but we should
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Olle E. Johansson skrev:
23 feb 2010 kl. 20.18 skrev Matt Riddell:
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
Where's the bug report id?
Yeah, the problem's not the origination.
The problem is that calls originated asyn with accountcodes show up in
show channels concise without details.
Pretty simple to test with sipp and core show channels concise.
I assume it's because the call origination happens at a faster rate
than
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
Matt Riddell skrev:
Yeah, the problem's not the origination.
The problem is that calls originated asyn with accountcodes show up in
show channels concise without details.
Pretty simple to test with sipp and core show channels concise.
I
Yeah, so at say 10 calls per second originated from the manager with
async on, you'd likely have about a thousand channels.
Then if you type show channels concise you'll see about 20% of the
calls are missing accountcode, destination etc.
I wrote some code to just repeat this test over and
On Tue, Feb 23, 2010 at 8:22 AM, Michelle Dupuis supp...@ocg.ca wrote:
We're creating a SIP gateway for a client that will take one leg of a call
in via SIP, and out the other side via H.323. To minimize load on the
gateway, we would like to have the RTP stream bypass the gatewayy altogether
Hi -
I have a Macro that contains a GoTo. The documentation indicates:
If you GoTo out of the Macro context, the Macro will terminate and control
will return at the location refered to by the Goto.
I thought I might convert the Macro to a GoSub routine, but the
documentation doesn't mention
On Tuesday 23 February 2010 21:35:39 hugolivude wrote:
Hi -
I have a Macro that contains a GoTo. The documentation indicates:
If you GoTo out of the Macro context, the Macro will terminate and control
will return at the location refered to by the Goto.
I thought I might convert the Macro
Message: 18
Date: Tue, 23 Feb 2010 15:02:24 +0100
From: Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Subject: Re: [asterisk-users] IAX devices not registering after
upgrade to asterisk 1.4.29
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Tue, Feb 23, 2010 at 7:50 PM, Danny Nicholas da...@debsinc.com wrote:
What I want is, if a call coming from a trunk 100 rings, and if the
caller wants to be transfered to 101, the transfer is denied. In other
words, 101 can't get transfered calls.
WHat about using featuresmap to replace the
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