Hi guys this is my Ist mail on this group, I am running asterisk with CentOS
4.4 machine. When i initiate a call then error message apears. calling
Number is provided to Asterisk by the php application. Error message appears
like this
Got SIP response 500 Internal Server Error back from
On Wed, 10 Oct 2007 12:54:42 -0500, Russell Bryant wrote:
I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc.
Another proposal has been using 1.5 to indicate that it is a release
candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the
release candidates for the
On Wed, 10 Oct 2007, Raúl Gómez C. wrote:
Hi list,
I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year
2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache),
768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb
NIC for server.
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more
information
X-ECN Telecoms-MailScanner: Found to be clean
X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED]
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My third try, humph!
Yusuf wrote:
Hi,
I am trying to understand the RTCP stats in
Hi,
is there a way to turn of SIP METHOD OPTIONS in asterisk?
I have a sip pbx which ignore Sip Option Messages from a unknown user.
Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip
server expects From: [EMAIL PROTECTED] server domain].
So i have to turn off
Hi List;
I am trying to find a link to see the polycom IP
Phones that work with Asterisk, but not able to find
until now.
I checked this link, but did not find any thing
related to Polycom IP Phones:
http://www.voip-info.org/wiki/view/Asterisk+phones
So any advise where I can find a link to
Ok, so i made the terminal screen wider, but during the call nothing changes:
( # = Audio Level * = Max Audio Hit )
(RX)
(TX)
###*
Rx: 10736 (10736) Tx: 0 (0)
What could be the
bilal ghayyad wrote:
I checked this link, but did not find any thing
related to Polycom IP Phones:
http://www.voip-info.org/wiki/view/Asterisk+phones
So any advise where I can find a link to see the IP
Phones of Polycom and its configurations?
Regards
Bilal
If you type polycom in
Totally agree *IF* the SIP elements behind your router/firewall have real
IP addresses and you are not using NAT in your router.
With NAT scenarios, I prefer to have a copy of Asterisk running on
firewall/NAT router so it at least has one public IP address to make
various SIP games a little
Hi List;
Any one can advise me to a good link to see and buy
Polycom IP Phones?
Also, if I need support (in case the Phone was damaged
and need to replace, so the warantee), so which web
can provide that? I do not need to buy from one and he
is not responsible for support.
Regards
Bilal
Hi,
I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from
1000 to 4
If I calling asterisk with GTALK in english everything is ok, however, some
of my friends with the italian version of gtalk they cannot have the audio.
Is it a bug? Or I did some mistake
hi all i have a TE110P connected to my PBX when i try to call a
extension number in other location 3525 the asterisk give me a error
-- User entered '3525'
-- Executing [EMAIL PROTECTED]:4] GotoIf(Zap/31-1, 0?6:5) in new
stack
-- Goto (lacnicuy,450,5)
-- Executing [EMAIL
www.telephonydepot.com has good prices. Never needed their support so I
can't comment
www.voipsupply.com a bit more expensive than above. Great support
- Original Message -
From: bilal ghayyad [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, October 11, 2007 12:08
Hello
Has someone used the OpenVox A400P01 (ie. a supposedly
Digium-compatible A400P board with a single FXO module
www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully?
I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a
more recent PC with an Asrock
On Wednesday 10 October 2007 12:54:42 Russell Bryant wrote:
I have been having discussions with various members of the development
community in regards to changes to the way we manage open source Asterisk
releases. The changes that we eventually decide on will determine how we
manage the 1.6
If I calling asterisk with GTALK in english everything is ok, however, some
of my friends with the italian version of gtalk they cannot have the audio.
Audio problems might be experienced with older Gtalk clients. Version
1.0.0.104 is reported to work.
The following resources may help you :
On Thu, Oct 11, 2007 at 08:47:52AM -0500, Tilghman Lesher wrote:
One of the problems with this traditional approach is that it's not obvious
unless you know what rc means. In the case of someone new to software
development, I want them never to assume that 1.6.0-rc2 means 1.6.0
plus
Tilghman Lesher wrote:
This method is no less obvious than rc1 for the untrained and ensures that
people who do not wish to become guinea pigs will remain out of that arena
(i.e. if they only choose the version that sorts to the bottom of the
directory, they will always be running a release).
Hello,
Up to a while ago I thought that the released versions are checkpoints of
the trunk versions; however, now I understand they are not, as I see
differences between the two trains. So, what is the relation between them?
Examples for differences:
- When the language is different than
Good point Gordon, but I have 2 spare drives (of line), the server has 2
(redundant) PSU, one of this brand new, the fans has already failed and has
bee replaced, so there are brand new too.
I'm not sure if a server has another component that is prone to fail, so any
advise/suggestion is welcome.
Hi. I'd like for my sound files to be exposed through http.
You know, the ones located in var/lib/asterisk/sounds.
This is probably an apache thing i'd have to configure or is
accessible through some asterisk http routing?
1. how one would configure this?
2. what are the security costs of doing
On Thu, 2007-10-11 at 03:08 -0700, bilal ghayyad wrote:
Hi List;
Any one can advise me to a good link to see and buy
Polycom IP Phones?
Also, if I need support (in case the Phone was damaged
and need to replace, so the warantee), so which web
can provide that? I do not need to buy from
On 10/11/07, Dominic Son wrote:
Hi. I'd like for my sound files to be exposed through http.
You know, the ones located in var/lib/asterisk/sounds.
This is probably an apache thing i'd have to configure or is
accessible through some asterisk http routing?
1. how one would configure this?
On Thu, 2007-10-11 at 15:07 +0200, Vincent wrote:
Hello
Has someone used the OpenVox A400P01 (ie. a supposedly
Digium-compatible A400P board with a single FXO module
www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully?
I've put it in an older PC with a Gigabyte GA-7ZX
Philipp Kempgen wrote:
Don Kelly wrote:
http://www.sandman.com/autodial.html
These phones look like the ones we had in Germany
20 years ago. ;-P
Hey, don't knock it, Phillipp :) -- I'm as big a fan of German
technology as anybody, but these phones are amazing pieces of
engineering.
Hello all, i've configured a TDM400P card but some calls hangs up and when
i take the phone to do a call y hear someone that callme. How is the way to
check the line before to do a call?. Other thing, is there a way to use Dial
application without ring the phone if the line is busy or
I have a customer that recently started having a problem with their
Call Center SIP extensions. The problem is that after some time the
caller will hear a triple tone (beep, beep, beep), a 5 second pause,
another triple tone and then the call will be dropped. This usually
happens between
Nick Richardson wrote:
Hi all,
I recently purchased a Junghanns ISDNguard and to my horror I found out:
- Junghanns technical support is non-existant
- I can't use it without recompiling Asterisk with res_watchdog
Let me know if you get any response on this bounty.
Cheers,
Stephen Bosch
On Thu, 11 Oct 2007, Raúl Gómez C. wrote:
Good point Gordon, but I have 2 spare drives (of line), the server has 2
(redundant) PSU, one of this brand new, the fans has already failed and
has bee replaced, so there are brand new too.
I'm not sure if a server has another component that is
If you are worried about it affecting asterisk, you could copy them to another
web server.
--
--
Steven
http://www.glimasoutheast.org
Dominic Son [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
Hi. I'd like for my sound files to be exposed through http.
You know, the ones
Have anyone maided like 200 simultaneous connections to asterisk AMI
(manager). ??
How many connections can I made without problems ?
I’m using a Quad core DELL poweredge machine.
Roberto Fernandes Lopes
Diretor Presidente
Dialtech Telecom. e Sistemas Ltda.
(11) 6986-8886
No
I'm in Venezuela, and I have buyed over 5K$ to htt://www.voipsupply.com,
excellent service and they sell warranty extensions for any product!
On 10/11/07, Patrick [EMAIL PROTECTED] wrote:
You did not say were you are located so here's a suggestion for a US
company that sells Polycom via the
Use the astmanproxy and move the load elsewhere. (If you just want to
passively listen to messages, your box is about 100 times faster than
you need :)
Zoa
Roberto wrote:
Have anyone maided like 200 simultaneous connections to asterisk AMI
(manager). ??
How many connections can I
Carlos Chavez wrote:
I have a customer that recently started having a problem with their
Call Center SIP extensions. The problem is that after some time the
caller will hear a triple tone (beep, beep, beep), a 5 second pause,
another triple tone and then the call will be dropped. This
Andreas Bayer wrote:
is there a way to turn of SIP METHOD OPTIONS in asterisk?
I have a sip pbx which ignore Sip Option Messages from a unknown user.
Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip
server expects From: [EMAIL PROTECTED] server domain].
So i
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening,
I have something strange, when I add an ALERT_INFO variable to a ring group,
the invite generated contains 2 lines with Alert-Info and my phones return a
400 Bad Request...
I've checked in my config files, there is only one line with
Péter Tóth wrote:
Ok, so i made the terminal screen wider, but during the call nothing changes:
( # = Audio Level * = Max Audio Hit )
(RX)
(TX)
###*
Rx: 10736 (10736) Tx: 0 (0)
bilal ghayyad wrote:
Hi List;
Any one can advise me to a good link to see and buy
Polycom IP Phones?
I've been using http://tritechcoa.com/ and they are always very prompt
about email support, I've never had to send anything back to them though.
Hi list,
I'm now considering to buy a new server for an Asterisk installation, since
I've been kindly
advisedhttp://lists.digium.com/pipermail/asterisk-users/2007-October/198146.htmlnot
to use an old server for a mission critical app...
Well, playing around in Dell's, HP's and IBM's online
On Thu, Oct 11, 2007 at 08:47:52AM -0500, Tilghman Lesher wrote:
One of the problems with this traditional approach is that it's not obvious
unless you know what rc means. In the case of someone new to software
development, I want them never to assume that 1.6.0-rc2 means 1.6.0
plus something
On Thu, Oct 11, 2007 at 04:21:09PM +0200, Tzafrir Cohen wrote:
Anyway, following that logic, go for 1.5.99-rc2 ?
Please don't.
That parses as the second release candidate for 1.5.99.
Really.
To everyone.
I'm not much for .99 in the first place, but you get one or the other;
not both.
At this point I was wondering if Asterisk gets real benefits on systems
with several cores (up to 8 in Dell PE2950) for a system that will handle
up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog
phones/fax (Sangoma A400D PCI card).
I suppose that yes. Asterisk uses
Yehavi,
The release branches (1.2, 1.4) were at one time trunk. When it was decided
to release 1.4, for example, it was branched off from trunk as the
1.4branch. New functionality continued to be added to trunk after
that. Once
the release branches are created, they are feature-frozen and only
On 10/11/07, Raúl Gómez C. [EMAIL PROTECTED] wrote:
At this point I was wondering if Asterisk gets real benefits on systems with
several cores (up to 8 in Dell PE2950) for a system that will handle up to
35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax
(Sangoma
I had the same problem with 45 polycom 601 phones in the same page group.
It was just like you describe it and I got the same answer from polycom.
What I did to go around that was add a second line key with a different
extension number on each phone and then create the page group with the
second
At 23:41 10/10/2007, Luki wrote:
Is it possible to configure a PAP2 to
auto-answer for either paging or intercom?
No. You cannot force the connected device (phone) to auto-answer.
Imagine you have a plain old phone attached to it, who's going to lift
the receiver?
Yep, I guess even if
I need to process a number of lines of code in the dialplan before answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate this?
Thanks in Advance,
Vic
Erik Anderson wrote:
For this load level (even with high-load transcoding), a multi-core
machine certainly would not be needed. That said, it certainly
wouldn't hurt anything to add on extra cores, especially if they're
free ;-)
Raul,
The points concerning overall load are valid, but I agree
On 10/11/07, James FitzGibbon [EMAIL PROTECTED] wrote:
What you do in between is up to you. Many people use something like
Wait(2) to give a comfort ring, since PRI-connected incoming calls can
often be set up nearly instantaneously. You'd want to limit the time
obviously, and have proper
On 10/11/07, Victor [EMAIL PROTECTED] wrote:
I need to process a number of lines of code in the dialplan before
answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate
this?
You start
Doug wrote:
At 23:41 10/10/2007, Luki wrote:
Is it possible to configure a PAP2 to
auto-answer for either paging or intercom?
No. You cannot force the connected device (phone) to auto-answer.
Imagine you have a plain old phone attached to it, who's going to lift
the receiver?
Hi All,
I have done some research on Asterisk and I would like to try it in my
office. Here's what I'm looking at for my system:
AsteriskNow running one of two ways:
1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB
ram)
2) on a P4 2.4Ghz with 768mb RAM
I'm looking at
Victor wrote:
I need to process a number of lines of code in the dialplan before answering a
call. Can standard ring back tones be played to the caller while this is
happening prior to answering the call. Which commands would facilitate this?
I strongly doubt those lines are going to take up
Just dont answer it till the processing is done. No debate is needed
for this. I do this millions of times per month.
/b
On Oct 11, 2007, at 2:56 PM, Eric \ManxPower\ Wieling [EMAIL PROTECTED]
wrote:
Victor wrote:
I need to process a number of lines of code in the dialplan before
Joseph Begumisa wrote:
I had the same problem with 45 polycom 601 phones in the same page
group. It was just like you describe it and I got the same answer from
polycom. What I did to go around that was add a second line key with a
different extension number on each phone
Brian West wrote:
Just dont answer it till the processing is done. No debate is needed
for this. I do this millions of times per month.
Yes, this is one of those things too simple to be obvious. Like Brian
said, just do your processing and THEN Answer() -- Generally, the
caller will
Ex Vito wrote:
2. On the remaining locations we have a problem
which I have been studying and trying to address...
Faxing over IP.
Could the 'remote' locations make do without a fax machine proper? We
have sheet-fed pdf scanners here, drop the document in and hit the
Am I correct in understanding that if the call comes in g729 and it is ended in
g729 ( by the provider ) , asterisk does only bridging, therefore using very
few CPU ressources ?
Am I correct in understanding that this bridging means that calls ( rtp )
pass from one provider to another,
On Thursday 11 October 2007 12:45:45 Jay R. Ashworth wrote:
People who know little should not be *trying* to interpret version
numbers; they should be using what a packager, a website, or a
knowledgeable other source *tells* them to use.
This I disagree with, fundamentally. People should be
Thank you
I need to wait the international version of gtalk
On 10/11/07, Philippe Sultan [EMAIL PROTECTED] wrote:
If I calling asterisk with GTALK in english everything is ok, however,
some
of my friends with the italian version of gtalk they cannot have the
audio.
Audio problems might
That is brought to you by the sip reinvite, in short yes, unless you set
canreinvite = no to either side of that.
Apa Minerala wrote:
Am I correct in understanding that if the call comes in g729 and it is
ended in g729 ( by the provider ) , asterisk does only bridging,
therefore using very
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I installed my super-duper new TE412P card today, without remembering to
check the
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I've added the auto-answer header in my dialplan, and it works great.
However, there is
Hi Gerald,
Well we have 2 APC UPSes in the server room, so each power supply will be
connected to one UPS, and the UPSes are connected to (a transfer system of)
an auxiliary power generator that start in less than a minute after a
blackout. The server will have RAID5, of SAS disc
But thanks for
Julian Lyndon-Smith wrote:
I am *really* sorry about hijacking this thread, but the only way I can
post to the -user list is by replying to another thread. (btw, this is
getting really annoying. Please, Digium, sort the filters out!)
I installed my super-duper new TE412P card today,
Mojo with Horan Company, LLC wrote:
Ex Vito wrote:
2. On the remaining locations we have a problem
which I have been studying and trying to address...
Faxing over IP.
Could the 'remote' locations make do without a fax machine proper? We
have sheet-fed pdf
Meeting Start: 10/13/2007 - 11:30am
Hello all Twin Cities Asterisk Users,
It's time once again to have another meeting.
We'll talk about what's new with the Digium product line, discuss
everything you n/ever wanted to know about dtmf and how each channel is
configured and interoperates as
Hi, all. My company is setting up a branch office in Germany, and I'm
very interested in a VoIP provider over thataway. However, I'd need a few
things:
- Reliability. Can't have my branch office's DID's just going down. A
company with a proven track record would be very, very good.
- English.
Hello Sean,
Does this clear things up?
Yes! Thanks!
__Yehavi:
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HOw to call queues in asterisk ?
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