Hi all,
I have a small problem to execute Asterisk Commands in Asterisk
Manager using PHP.
I am able to run all Asterisk Manager command but the problem is
comming with asterisk command.
here is the code i am trying to run.
?php
$socket = fsockopen(localhost,5038, $errno, $errstr, $timeout);
Wolfgang Borgon wrote:
A RAW file I created after converting from MP3 and WAV, sounded raspy.
Does anyone have any tips for creating the best quality voice recordings?
Generally you'd use a good-quality microphone for your recordings. The adage
Garbage in = garbage out couldn't be
We've been trying Unison (http://www.cis.upenn.edu/~bcpierce/unison/) on a 1 minute cron job. There are some theoretical issues but it has been great so far. We use it to synch prompts as well as messages.
SimonOn 12/27/05, BILL GITONGA [EMAIL PROTECTED] wrote:
What is the best method of storing
Could anybody please help me with problem..
Outbound calls work fine, however inbound calls ring the
phone, then answering the call, the service provider doesnt receive the picked
up message from asterisk.
We have narrowed it down to an incorrect checksum in the
packets being sent
Hello All,
Have anybody test ISP BILLING SYSTEM ?
http://ibs.sourceforge.net/index.html
Regards
Harry
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In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Some users with Blackberry's cant play .wav files, is there a way to save
the voicemail to save as another format like mp3?
-Kerry
In voicemail.conf edit this line.
[general]
format=wav49|gsm|wav
P.S.
Please stop replaying to mesage.
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
Did you set the timezone correctly?
All my phones are 1 hour behind because I still have to
login to all phones to alter daylight saving time setting.
It isn't just time, he misses the year! :)))
Acording to Budgetone now is 1900. ;))
Please stop replaying to mesage. If you plan to open thread do so by
writing mail to this address
asterisk-users@lists.digium.com
--
Tomislav Parcina
[EMAIL PROTECTED]
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In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I am aware of the possibility to add the option t or T to dial, so #33
transfers the call to extension 33.
It needs to be deined in feautres.conf file. So when you dial #1 you'll
hear transfer and than you enter extension.
Is there any
I'd agree with you on the quality of the audio. But, unless they've
changed a lot in the last year, the 1204's sip support and mgmt
leaves a lot to be desired.
For a box that has very poor reviews, it sure is great
to use a box that you can throw in the closet and
On 18:06, Tue 27 Dec 05, Bud Bach wrote:
But, if the agents don't log out for some reason, they will still be logged
in the next time the queue opens even if they aren't there right?
yes.
What you can do is 2 things:
* you can set the autologoff time in agents.conf. This can
give you some
What I can doing execute extension (es.200) from CLI? Thanks
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Brian Capouch wrote:
They don't perform as well as the expensive Ciscos and Polycoms, but
many of us are using them in a variety of circumstances quite happily.
I have 4 of them in a small office (GXP2000) running 1.0.12 and they're
just fine for our purposes. As Brian said, YMMV. For our
I've been working with asterisk with public ip but since I change this
and put asterisk behind NAT, get this error when my hard phones try to
register
Dec 28 11:43:33 NOTICE[8716]: chan_sip.c:10817 handle_request_register:
Registration from 'sip:[EMAIL PROTECTED]:5060' failed for
On Tue, 2005-12-27 at 22:39 +0100, Armin Schindler wrote:
On Tue, 27 Dec 2005, Dave Cotton wrote:
On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:
It looks like the call is signaled on both ports !?
On another installation in France I'm also getting this, but with 2
Fritz!
Please stop replaying to mesage. If you plan to open thread do so by
writing mail to this address
asterisk-users@lists.digium.com
--
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[EMAIL PROTECTED]
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Dave Cotton a écrit :
On Tue, 2005-12-27 at 22:39 +0100, Armin Schindler wrote:
On Tue, 27 Dec 2005, Dave Cotton wrote:
On Tue, 2005-12-27 at 19:27 +0100, Armin Schindler wrote:
It looks like the call is signaled on both ports !?
On another installation in France I'm also getting this,
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says...
4)
Is it possible to make a routing, as follows
Dial 8 go to Internet Call
Dial 9 go to TelCo. Call
Read this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf
5)
How do I change the time zone for
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I am thinking to develop one.
Thank you! (in advance :))
--
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[EMAIL PROTECTED]
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To
Armin,
Please find attached the log file for 4 x test calls:
1. Call to Orange GSM mobile phone (effectively ISDN all the way) - works as
expected
2. Call to BT analogue phone (we still have a lot of analogue in the UK) -
no ringing problem (stuck in Session Progress)
3. Call to CW
Have a look at
/var/log/asterisk/queue_log
It has to be enabled on logger.conf (queue_log=yes
on the [genera] section).
- Original Message -
From:
Hall, Eric M.
To: asterisk-users@lists.digium.com
Sent: Tuesday, December 27, 2005 6:25
PM
Subject:
I am using Red Hat 9, but I don't think this changes the procedure
- Original Message -
From: Carlos Alperin [EMAIL PROTECTED]
To: 'Dov Bigio' [EMAIL PROTECTED]; 'Asterisk Users Mailing
List -Non-Commercial Discussion' asterisk-users@lists.digium.com
Sent: Tuesday, December 27, 2005 8:24
Jonathan,
many thanks for your reply. The adapter has firmware version 3.1.3(GWa).
Does that version have problems with diconnect tones? Would you
recommend I should upgrade? Can you give me some reasons or point me to
resources (apart from google) where I can research further? What would
I am setting up a phone system using [EMAIL PROTECTED], version 1.5. It runs
Asterisk 1.0.9 built by [EMAIL PROTECTED] on a i686 running Linux
(Asterisk info). I had some bigger problems:
In AmpPortal / Setup/ Extensions: When I added new SIP devices and then
looked at the resulting sip.conf
Stay away from Grandstream and AddPac. These are some of the companies
with undereducated software developers that have problems with
understanding written english, mainly the SIP RFC documents. I learned
this the hard way, wasting half a year with helping them fix problems
which shouldn't be
Go with SPA-3000. While it's much more awkward to maintain, they're rock
stable and provide the features they advertise for. I'd also add AddPac
VoiceFinder series as being not 100% asterisk compatible, expensive and
not worth your time (learned this the hard way). It took me 6 months to
I found some more information on UK settings with sipura-3000 on
http://www.voip-info.org/wiki/view/Sipura+3000
They point to a document with UK specific settings for spa-3000. This
document is at
http://www.provu.co.uk/pdf/sipura/sipura_uk_regional_settings.pdf
Hope this helps others.
My only experience is with their Budgetone 102. You basically get you
pay for.
I have since purchased a pair of Aastra 480i. Much much better. I am
going to put the Budgetone on ebay, no point dealing with all the
hassle. The main issue for me was actually not sofware but rather the
design of
Hmmm...
I feel that this is a little unfair towards GrandStream and other like
vendors. Any vendor on the market has issues with their firmware, I can
list many:
Sipura/LinkSys SPA 841 (Latest firmware):
1. Phone doesn't re-register upon network loss
2. Phone firware becomes stalled, without
Peter,
I'm using the firmware 3.1.5(GWb) and was wondering if your suggestions
would be of any benefit to me. Incidentally, I've never had an issue
upgrading or downgrading the firmware in 2 spa-3000s, I just had to make
sure the unit had only just been powered up when initiating the upgrade.
Hi,
I have not been able to find anything about persistent agents in any
wiki? Where does this command go and what is its syntax?
Thanks
Michiel van Baak wrote:
On 18:06, Tue 27 Dec 05, Bud Bach wrote:
But, if the agents don't log out for some reason, they will still be logged
in the
It is set in the queues.conf file.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Chuck Bunn
Sent: Wednesday, December 28, 2005 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Automatic logoff
Hi Ryan:
Christmas intervened!
Got it working. It turned out not to be the ww that did it, but the
toneduration parameter in the zapata.conf file.
Setting
toneduration=200
did the trick.
Thanks for the help, hope this tip helps someone else later on.
Happy New Year!
Roger
[EMAIL PROTECTED]
Why don't you send this to the offender instead of the list?
From: Tomislav Parcina [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Why no sound from festival?
Date: Wed, 28 Dec 2005 12:07:52 +0100
To: asterisk-users@lists.digium.com
Please stop replaying to
I think the unfairness stems from Grandstreams generally being people's
first IP phone - it seems like a cheap entry point to try things out.
They then falsely assume everything else has to be better, especially if
it has a higher price tag. Wrong. The standard for VoIP phones is total
crap.
I am about to sent some Sipura 2002 ATAs out to a call center. I want to
use the dual line capability of the units, but I realize that the second
channel will not be able to use G729 simultaneously. What do you think
would the best option be for that channel?
--
Chris Mason
NetConcepts
(264)
Thanks very much! I'll definitely sook at these resources as well if
other problems come up.
I also found some more info on sipura setup un the UK, see
http://lists.digium.com/pipermail/asterisk-users/2005-December/140037.html
Thank you very much again for your comments. Good to hear that you
On Tuesday 27 December 2005 21:52, Erick Baum wrote:
that, there is now a bad echo if one of the GXP users turns their volume up
too high, the other party can hear an echo. If the GXP user turns their
I'm afraid you're going to find this with pretty much *every* phone. Normal
POTS phones
Hi,
Anybody have some experience and did some testing
with ipVolution E1/T1 cards?
goran
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Hi,
When I add 'persistentmembers=no' in queues.conf and reload I get a
message in the message log file saying unknown keyword
'persistentmembers'. I got the syntax from
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes
section.
Thanks
Alexander Lopez wrote:
It is
Just a small word of caution on the spa3k...
It too has issues with handling echo cancellation on long loops. Or,
maybe I should say on loops with long echo tails. Sipura/Linksys tech
support has suggested downgrading from the current 3.1.7 firmware to
3.1.3a to improve the issue. I've not heard
I am about to sent some Sipura 2002 ATAs out to a call center. I want to
use the dual line capability of the units, but I realize that the second
channel will not be able to use G729 simultaneously. What do you think
would the best option be for that channel?
You might double check that.
Chuck Bunn wrote:
When I add 'persistentmembers=no' in queues.conf and reload I get a
message in the message log file saying unknown keyword
'persistentmembers'. I got the syntax from
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes
section.
You haven't told us what
It's possible to register oh323 with gnugk ?
Any one knows one good oh323 how to?
Regards,
--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
SIP : [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
www : http://www.telconet.net
On this page http://www.voip-info.org/wiki-Asterisk+record+calls there
is Example by Mojo. I have done everything he said and I have sox
package installed.
[EMAIL PROTECTED] recordings]# sox -help
sox: Version 12.17.7
...
When I open this web page http://10.0.0.26/recordings/index.php I get
Sorry... the question is related with ooh323
It's possible to register ooh323 with gnugk ?
Any on knows one good ooh323 how to?
On Wed, 2005-12-28 at 09:48 -0500, Guillermo Salas M wrote:
It's possible to register oh323 with gnugk ?
Any one knows one good oh323 how to?
Regards,
--
Hi,
Oh sorry I am using asterisk 1.2.1
Thanks
Kevin P. Fleming wrote:
Chuck Bunn wrote:
When I add 'persistentmembers=no' in queues.conf and reload I get a
message in the message log file saying unknown keyword
'persistentmembers'. I got the syntax from
hi all i use asdterisk in my company with Flash Panel Operator to know
who is talking or ringing. But i dont know any web application to know
who is online or offline. any body know any webapp for that ?
--
thanks.
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In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
Do I have to do something more? Does it work for anybody else?
Is there any other way to combine in and out soundfile when I use
automon option?
My error. Everything works fine... Sorry.
--
Tomislav Parcina
[EMAIL PROTECTED]
AFAIK you need to use different actions for each command.,
sending 3 commands in the same action wont work. I have no
problems to issue commands, originates etc.On 12/28/05, Code Lover [EMAIL PROTECTED] wrote:
Hi all,I have a small problem to execute Asterisk Commands in AsteriskManager using PHP.
You can use both channel as G.726/32 at the same time, or lower than 32. Is
the best solution we found.
Regards,
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Wednesday, December 28, 2005 8:44 AM
To:
Pablo Allietti wrote:
hi all i use asdterisk in my company with Flash Panel Operator to know
who is talking or ringing. But i dont know any web application to know
who is online or offline. any body know any webapp for that ?
Flash Operator Panel _is_ a web application.
I'm not sure how this is suppose to work. But I want to be able to call
people from a SIP phone and transfer them into a conference room. If I
call another extension that is a SIP phone I can hit # and then enter
the conference room number. If I call from the PSTN to the SIP extension
phone I
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote:
Pablo Allietti wrote:
hi all i use asdterisk in my company with Flash Panel Operator to know
who is talking or ringing. But i dont know any web application to know
who is online or offline. any body know any webapp for that ?
I agree, GrandStream does seem to become the poor man's VoIP solution -
making the bar for other VoIP phones very low to pass.
I believe that GrandStream have a very good chance to basically being
bought by a bigger company, like what happened to Sipura. What
would happen then would be that
Does adding the line nat=yes into your sip.conf file help?
Leah Newmark
Capalon
www.capalon.com
Message: 21
Date: Wed, 28 Dec 2005 11:48:25 +0100
From: Rafael Ledesma [EMAIL PROTECTED]
Subject: [Asterisk-Users] Wrong Password?
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL
you need to set the extensions paramters to qualify=yes or
qualify=integer and then FOP (flash operator panel) will
reflect the status of the extensions.
Pablo Allietti wrote:
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote:
Pablo Allietti wrote:
qualify=yes in both sip.conf and iax.conf, seems to highlight both the
users and trunks who are currently available in FOP
Bails
Pablo Allietti wrote:
On Wed, Dec 28, 2005 at 09:15:15AM -0600, Kevin P. Fleming wrote:
Pablo Allietti wrote:
hi all i use asdterisk in my company with Flash
The
'status' is only as goodas the frequency of the qualify periodand
you can say hello to a LOT of SIP OPTIONS messages being sent from Asterisk to
each phone.
-Original Message-From: Adrian Carter
[mailto:[EMAIL PROTECTED]Sent: Wednesday, December 28, 2005 8:36
AMTo:
My thanks to all the great people that have submitted replys. I have the
information I need now.
Bob Rawlinson
Robert Rawlinson wrote:
I acquired a Blackberry 7100T over Christmas. I had heard it will work
with * and that is what I want to do with it. But I think it needs a
SIM card to make
Having tried EVERY single product from Grandstream, I don't think it's
fair to judge Grandstream the way people do.
I'm very happy with Grandstream products.
As long as you upgrade the firmware they work fine.
In fact they sometimes handle NAT better than any other device that I've
tried
On Thu, Dec 29, 2005 at 02:36:05AM +1100, Adrian Carter wrote:
you need to set the extensions paramters to qualify=yes or
qualify=integer and then FOP (flash operator panel) will reflect the
status of the extensions.
Pablo Allietti wrote:
yep. this solve my problem Thanks!!
Diego Mariano Velo wrote:
Hi, i have a cisco 7912G with SIP firmware, its connect to the asterisk
through nat. The only problems is in the voice mailasterisk not
detect the tones, therefore i cant access to my voice mail extension.
Check the DTMF settings...
Hey everybody,
I'm trying to figure out a problem with Caller-ID info coming in from
one of our facilities. The Caller-ID name is all that comes across. I
figured out that I probably could do a database lookup against the name
and set the Caller-ID number to their extension. I'm using
Hello,
I need to test my configuration please to dial
sip:[EMAIL PROTECTED] .
Your call will be sent to a queue .
Regards
Harry
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Sorry if I am always here asking for MWI, but I do not know how to
solve this issue, I have my ATAs (Azatel 200 and Fritz!Box) that they
think that I have a message waiting.
Anyone knows how to solve this issue?
Thank you
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I am an Asterisk newbie and don't have any
telecom experience. I do know some about Linux
and Windows as a sysadmin of Windows servers.
I need to know what hardware to buy to replace a
broken PBX.
I have currently:
-CBeyond as my carrier
-16 port Cisco router with analog termination
(not sure
I just have upgraded from Asterisk 1.0.7 to 1.2.1 and im having problems with my AGI script that takes care about
routing the calls. It worked perfectly for the last year with 1.0.7,
now is getting stuck when the is launched. I have agi debug enabled and
this is the output:
-- Launched
I am an Asterisk newbie and don't have any
telecom experience. I do know some about Linux
and Windows as a sysadmin of Windows servers.
I need to know what hardware to buy to replace a
broken PBX.
I have currently:
-CBeyond as my carrier
-16 port Cisco router with analog termination
(not sure
I am an Asterisk newbie and don't have any
telecom experience. I do know some about Linux
and Windows as a sysadmin of Windows servers.
I need to know what hardware to buy to replace a
broken PBX.
I have currently:
-CBeyond as my carrier
-16 port Cisco router with analog termination
(not sure
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr ser91623 cdr)
exten = _00.,3,MYSQL(Query resultid ${connid} select\
Hi all,
It sas been a while since I have been on the mailing list but am really hoping
someone can help me.
Using Dell SC1420, Fedora 4, Asterisk 1.2.1 and a TE210P I cant seem to get the
card to configure itself properly for CCS, it always appears as ESF in the
/var/log/messages file,
John - You might consider getting a T1 and splitting it using some of the
pipe for your voice traffic and some for your data traffic. You can set up
a VLAN on your internal network for your phones if you want to migrate to
SIP phones and Asterisk, or you could implement a channel bank out to
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid
I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I
think the
problem is in the PRI signalization.
I can see the zap hangup messages when trying to call a disconnected number.
.
-- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack
-- Called
Remarque : message transféré en pièce jointe.
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exceptionnels pour appeler la France et l'international.
Anyone know what version of Asterisk is the most stable running Real-time
queues and agents ?
I am setting up a 200 phone call center and the first test run caused the
system to crash 3 time in 3 days with only about 100 calls an hour.
I used the same build that I have used in prior stable
I use 1.0.9 and 1.0.10
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
--
[EMAIL PROTECTED]
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Sending the message 3 times within 7 minutes wont get you responses
any faster than just sending it once.
Welcome to asterisk. Put your seat belts on and get ready for a few
weeks of reading, testing, and caffeine. Here are my recommendations
(assuming you want to set it up alone, otherwise just
In 1.0.x the command is SetCIDNum
http://www.voip-info.org/wiki-asterisk+cmd+setcidnum
On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I use 1.0.9 and 1.0.10
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
--
[EMAIL PROTECTED]
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
Do you know of anyway to set it up through AMP, so it works with all calls?
Michael Sampson
Information Systems Manager
Customer Contact
Look at this post:
http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html
On 12/28/05, Doug Lytle [EMAIL PROTECTED] wrote:
Hey everybody,
I'm trying to figure out a problem with Caller-ID info coming in from
one of our facilities. The Caller-ID name is all that comes
John,
I just switched from an old Merlin system myself! (haven't looked back).
I was in the same situation as you (Windows server sysadmin with minimal
*nix experience).
Here's the setup I have currently in production:
-Custom built server: Gigabyte motherboard with AMD Sempron 2800+ (I
Will this help:
http://lists.digium.com/pipermail/asterisk-users/2005-December/140074.html
On 12/27/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Saturday 24 December 2005 16:40, Kevin P. Fleming wrote:
Interestingly, some systems I manage also began exhibiting this behavior
in the past
hello,
Is this so difficult to call an ip phone towards
another via sip ?
Does ser and asterisk projects are dedicated to the
telephony or mail servers ?
Harry
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C F wrote:
Look at this post:
http://lists.digium.com/pipermail/asterisk-users/2005-December/139952.html
Actually, I don't think the caller ID number is being sent in my
situation, I am wondering what I can't manually set it.
Thanks for the reply!
Doug
The Grandstream certainly has issues, but it seems most of the SIP phones
do. I continue to have excellent results with the Aastra 9133i. The latest
firmware (1.3) supports busy lamps with Asterisk 1.2.x. I think that dollar
for dollar, it is a fine phone and works better than most. Again,
is there an mp3 format for voicemail? what's the difference between wav49 and wav?On 12/28/05, Tomislav Parcina [EMAIL PROTECTED]
wrote:In article [EMAIL PROTECTED],
[EMAIL PROTECTED] says... Some users with Blackberry's cant play .wav files, is there a way to save the voicemail to save as
Merry Christmas List,
Any body with experience on the GSM-gatewas that
Cyber-telecom.net sell?
The thing keeps on asking for a PASS and ...
pretty much that's all.
Help anyone?
benchev
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Téléchargez sur
Rich Adamson wrote:
I am about to sent some Sipura 2002 ATAs out to a call center. I want to
use the dual line capability of the units, but I realize that the second
channel will not be able to use G729 simultaneously. What do you think
would the best option be for that channel?
You might
On Wed, December 28, 2005 16:38, bails said:
qualify=yes in both sip.conf and iax.conf, seems to highlight both the
users and trunks who are currently available in FOP
Bails
Note that some IAX clients do not seem to like qualify=yes. I use DIAX,
and when I use Qualify=yes, it becomes
I have the follwoing setup:
Asterisk SVN-tag-1.2.1-r7367
6 Polycom 500 Sip version 1.5.x
4 Sipura SPA3000 (not sure what build) (FXO port)
All on flat single network, no NAT, and no gateways to reach each other.
Sometimes (happens around 3 times a day, but sometimes far more
often), while on the
I am testing [EMAIL PROTECTED] V2.2
I want to interface with our PBX via a FXO card (TDM400P). I have one extension hooked up right now, and I can call into the Asterisk system from both a PBX connected phone, or through a DID number, but I can't dial from an IP phone out to our PBX system or out
Here is a reference cdr:
29.
2005-12-28 13:02:01
Zap/1-1...
8103970196
81039701968103970196
5100
ANSWERED
78
30.
2005-12-28 12:59:54
Zap/1-1...
8104590192
81045901928104590192
5128
ANSWERED
23
31.
2005-12-28
I was looking at the asterisk in spanish webpage. The register form is
giving timeouts.
On 9/19/05, Sergio Serrano [EMAIL PROTECTED] wrote:
Try in www.asterisk-es.org
-Mensaje original-
De: Sebastian Milioto [mailto:[EMAIL PROTECTED]
Enviado el: lunes, 19 de septiembre de 2005
I have the follwoing setup:
Asterisk SVN-tag-1.2.1-r7367
6 Polycom 500 Sip version 1.5.x
4 Sipura SPA3000 (not sure what build) (FXO port)
All on flat single network, no NAT, and no gateways to reach each other.
Sometimes (happens around 3 times a day, but sometimes far more
often), while
Hello everyone,
Im having an outbound calling issue with our SIP
phones. When one call is made to the PSTN another person trying to call receives
a 404 error on the SIP phone. If we call the PSTN using SIP phone A and also
calling from SIP phone B to SIP phone C everything works. The
For somereason I think it's the polycom, which means I need logging
for the Polycom and not the spa.
On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote:
I have the follwoing setup:
Asterisk SVN-tag-1.2.1-r7367
6 Polycom 500 Sip version 1.5.x
4 Sipura SPA3000 (not sure what build) (FXO
In any case I'm trying to figure out if maybe someone else has seen
this problem. Or if they know what it might be.
On 12/28/05, C F [EMAIL PROTECTED] wrote:
For somereason I think it's the polycom, which means I need logging
for the Polycom and not the spa.
On 12/28/05, Rich Adamson [EMAIL
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