In article [EMAIL PROTECTED],
Jon Webster [EMAIL PROTECTED] wrote:
I'm running 2.4.5 and app_meetme never plays conf-hasleft or
conf-hasjoined with user names. I looked at app_meetme.c, but couldn't
determine the cause. Any suggestions are greatly appreciated.
exten = 600,1,MeetMe(600|i) I
HI Group! I have strange problem. Since I started to use H323 with my VoIP
provider when I dial the person that is currently busy, I can't hear busy tone
on my handset. What could be the problem? What should I look for? How is this
exactly called (because I even don't know what to look for).
Hello,
I am trying to set up a slim Asterisk. Blind transfer works, and attended
transfer always plays pbx-transfer, then I dial the extension and it
plays pbx-invalid. Here is my modules.conf :
; * Resources *
load = res_features.so
load = res_adsi.so
load = res_monitor.so
load =
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: 8. ozujak 2006 23:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: ON DEMAND call Recording
I would find two possibilities:
Hasn't anybody ever come accross these changes before?
Jason
Jason Frisch wrote:
Can someone help me with upgrading to the lastest version. I am using the
same sip.conf file, but the headers have changed and registration fails.
Has something change in the conf file that would cause this?
Best of luck :-D
I would be interested in your progress on this.
I am having very little problem in convincing ppl to upgrade their
multiple
BRI cricuits for a single pri. The cost difference between a te110
(or a
Sangoma A101) MORE than covers the difference from the customer stand
I'm very new to Asterisk, I'm tracing the Asterisk code,
i'm feeling difficulty in understanding the code, so please tell me
where i can get the documentation of the code and,
design and architecture of the code.
___
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James Harper wrote:
One use for the multi BRI card though, especially one that can do NT
mode, is that you can use it to trunk to a legacy BRI PBX, which is why
I'm still interested in finding one for use in Australia.
I'm using the Eicon Diva Server V-4BRI (~$2,200 each). They are awesome.
Hi I have installed Festival on the same box as asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing text to speech from the
command line or via a file.
Asterisk answers the call but there is no speech. I can see no errors in the
Olle E Johansson wrote:
8 mar 2006 kl. 17.07 skrev Fernando Lujan:
There is a realtime PostgreSQL driver for testing in the bug tracker,
please test it!
http://bugs.digium.com/view.php?id=5637
Hi Olle,
I want something read for production. Is there such driver, even using
the mysql
already tried it and it didnt work. Could there be any
other files that may have been messed with that is
causing this ?
Dovid
--- Mark Edwards [EMAIL PROTECTED] wrote:
Try dtmfmode=info and see if that works.
Mark
-Original Message-
From: Dovid Bender [mailto:[EMAIL
Hi,
As I am having problems with MoH and have
tried everything to solve it and nothing worked, I was thinking maybe the
timing source, i.e. ztdummy, is not working properly and that is what is
causing problem. Is there any test to check whether ztdummy is working properly?
Thanks,
Hi,
I have installed a fax machine on a HT 486 ATA in
my office, and it works perfectly, to send and receive faxes.
When I install the same ATA on a fax machine at
home (behind a NAT, in case it matters) faxes are received correctly, but I
cannot send.
Asterisk keeps showing a message
Hello
I have a rather limit hardware where i try to run
Asterisk, 1 GHz VIA C3. Everything works fine except music playback on MOH.
I have encoded the music in same codac as as the
voicestream (ulaw), ulaw is used end to end. MOH fades in and out with varius
volume, and is very choppy.
Waldo Rubinstein wrote:
Can anyone recommend a company that does professional Asterisk
recordings for things like IVR, greetings, MOH, announcements, etc?
Thanks,
Waldo
___
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asterisk-biz
Try Allison at theivrvoice.com. She is the voice of Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, March 08, 2006 11:06 PM
To: Commercial and Business-Oriented Asterisk Discussion
Cc: Asterisk Users Mailing
I tried doing the same things as you to make Festival
work with Asterisk,
but I had a small problem with Festival only prducing
the sound if the text
was tess than 14 characters
So I used the other approach and used the text2wave
utility instead (I saw
on some postings that people
Release notes of ruby-agi-1.1.2
March 07, 2006
In this release bug # 3722 has been fixed
Details of this can be found at
http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477
http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477
Feedback,
Darren Wright wrote:
Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.
Look for ANY of the 257* series...
Just ebay for t1 echo
-D
Do *not* forget the Orion. We have 3 in place now working beautifully.
Can't speak on Darren's problems, but our units
Title: digium certification for Europe
Im little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series.
What about TE110 or TDM400P? Can someone confirm that?
Thanks,
David
In extension.conf i read this:
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
I tried to define:
exten = _50,1,Dial(...)
exten = _5!,1,Dial(...)
If i dial 50, due to asterisk reordering _5! is
I am using the odbc set up with postgres right now and it works fine.
http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL
has most of the info to get you running. As for meetme, I took the
app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still
testing it, but it
So has anybody tried installing the new SIP version?
It seems nobody has had luck with the 7970 and it's new SIP image and the description for the 7940/60 specifically says for CCM
v5.0.
_
Mobilcom
http://www.mobilcom.net
___
Anyone have any information on the performance impact of using
qualify=yes for hundreds (500ish) of SIP UAs?
I have seen tidbits on qualifyspreading=yes, but not enough to
understand what it does. I assume lessens the peak load of qualify sip
options queries?
Thx!
Qualify=yes
This might be a better question for the dev list, but I don't think they
want to be bothered by my silly questions. Does anyone know when we can
expect to see a jitter buffer for SIP channels?
I know they've been working on a generic jitter buffer since around last
summer, just wondering if
Mailing List wrote:
So has anybody tried installing the new SIP version?
It seems nobody has had luck with the 7970 and it's new SIP image and
the description for the 7940/60 specifically says for CCM v5.0.
I'm planning on trying early this afternoon. (EST)
Doug
--
Ben Franklin quote:
Hii have enabled on demand recording on my asterisk server.i want to have voice playing when i start recording saying recording has started and when i press *1 to stop the recording it should play recording stopped .
is that possible with the latest version [EMAIL PROTECTED] 2.6 . or is there
Hi all,
I'm planning to test my two [EMAIL PROTECTED] one is 1.5 and another is 2.5
Does any one got already Astertest - asterisk stress testing tool working one?
I've red Asterisk Guru, http://www.asteriskguru.com/tutorials/astertest.html
and after all the tutorial still remaining questions
Sean Cook wrote:
I am using the odbc set up with postgres right now and it works fine.
http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL
has most of the info to get you running. As for meetme, I took the
app_cbmysql stuff for webmeetme and rewrote it for postgres. I am still
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi everybody!
Does anyone know what is the exactly modulation that
the Digium TDM400P works? DTMF or FSK?
If anyone know where to get a good material about it,
please let me know.
Thanks for any help.
Regards,
Filipe Mordhorst
Brazil-SC
smime.p7s
Description:
Hello,
I have an E1 and the possibility to use different
caller ids in this E1, so, before a Dial, I always have a SetCallerIDNum("User",
number).
When I check the CDR, the originator of the calls
appears to be this "number" I set in the caller id, but not the actual user that
originated
Thanks for all of your replies!
I was thinking the server was a little overboard, but I want this to
last and also be expandable. We might be adding users/offices within
the next year so I wanted to plan ahead.
The DSL speed at the remote office is 1.5 to 6.0 Down and 384 to 608 up.
The DSL
Can
you be more specific? All digital cards (regardless of manufacture) use the same
modulation, it is a standard and you can probably google it.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Filipe
MordhorstSent: Thursday, March 09,
So has anybody tried installing the new SIP version?
It seems nobody has had luck with the 7970 and it's new SIP image and the
description for the 7940/60 specifically says for CCM
v5.0.
Just downloaded it after your email and got it working on the first try.
Give me a few minutes to write up
Steven [EMAIL PROTECTED] wrote:
Hi I have installed Festival on the same box as asterisk and followed the
instructions to integrate it with asterisk.
Festival seems to work fine on its own performing text to speech from the
command line or via a file.
Asterisk answers the call but there is no
If you have many old voicemail messages, to get to the most recent one,
you have to keep hitting 6 until you reach the last one. It would be
better if you could hit 4 from the first message to get to the last
message and/or have a digit that takes you the first and last messages
respectively.
Yes you do need unixODBC before you compile asterisk. Once you have
installed unixODBC , asterisk will compile and offer you the following
modules:
cdr_odbc.so
res_config_odbc.so
res_odbc.so
res_odbc.conf and cdr_odbc.conf are the related config files...
Sean
On Thu, 2006-03-09 at 11:57
hey, how i do to do that with php agi's?
Este Mensaje Esta Hecho 100% con Electrones Reciclados
- Original Message -
From: Adam Robins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, March 09, 2006 10:04
That is what the accountcode field is for, you can set
a unique accountcode for each devcice if you want to.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dov
BigioSent: Thursday, March 09, 2006 10:05 AMTo:
asterisk-users@lists.digium.comSubject:
9 mar 2006 kl. 15.26 skrev Adam Moffett:
This might be a better question for the dev list, but I don't think
they want to be bothered by my silly questions. Does anyone know
when we can expect to see a jitter buffer for SIP channels?
I know they've been working on a generic jitter buffer
Doug Lytle wrote:
Matt wrote:
Tellabs looks a little too up-scale for what I need :). $1k for a
single port orion unit might be worth considering for really stubborn
installs though.
Is there a way to display the time of the 7960 running firmware 7.4? Im
unable to find any information.
Add the following to SIPDefault.cnf or SIPMAC.cnf:
sntp_server: time.nrc.ca
sntp_mode: unicast
time_zone: EST
You should of course change your NTP server and/or time zone.
Nabeel
exten = _50,1,Dial(...)
exten = _5!,1,Dial(...)
Remove the _ from the first line.
Nabeel
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
So has anybody tried installing the new SIP version?
It seems nobody has had luck with the 7970 and it's new SIP image
and the description for the 7940/60 specifically says for CCM
v5.0.
Just downloaded it after your email and got it working on the first
try. Give me a few minutes to
Hi List,
Merlin Magix hardware v02
I'm trying to get asterisk to act as a voicemail server for a lucent
merlin magix PBX that we purchased used. We have 4 FXO channels between
the two PBXs on a Sangoma A200 card. The 770 dialgroup is working
properly, in that calls to 770 are answered by
I have installed asterisk @
home 2.6. I am using a Telasip VOIP account. When I make outbound or
inbound calls the calls seem to connect and then get hung up. I was
wondering if there was something that I am misisng. I have tried several
different sip.conf configurations. Here is what they
Hello,
my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1
fxs ), 1 phone, 1 softphone
I'm in France
When someone from PSTN calls and hangs up before the call is answered,
internal extension keeps ringing until timeout occurs. PSTN line keeps
busy. Hangup detection
The DSL speed at the remote office is 1.5 to 6.0 Down and 384 to 608 up.
The DSL does have a static IP address and it's pretty rock solid in
regards to stability.
Curious, why the huge range in numbers? I have 1.5mb/s down and
512kb/s up, it's always been that. Or do you mean you have
Hi thanks for the help .vocie mail problem has been fixed but the delay is still there i have changed Interdigit Long Timer =2 and Interdigit short Timer=1thanksGiridhar Bandi
On 3/8/06, Filipe Mordhorst
[EMAIL PROTECTED] wrote:
You're almost right.
The PAP2 has some features
that
hi
i'm running asterisk since 2 weeks and sometimes it crashes reporting
some ouch ... broken pipe error
i wolud like to write a script shell that check if asterisk is
correctly started and, if not, it restart it, can i do it?
how?
i'm using asterisk 1.2.4 on slackware 10.2
thanks
- Original Message -
From: Nabeel Jafferali [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 09, 2006 10:42 AM
Subject: RE: [Asterisk-Users] 7940/60 SIP 8.2
So has anybody tried installing the new
Hi,
I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to
send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID!
I only have User ID of the SPA 3000 as caller number.
The caller number is present on the PSTN line.
I'm in france, maybe the SPA 3000 is not
Just a note:
This vendor is selling cards with local side echo cancellation. Most of
the cards that I purchased didn't have it. The 3 that I've purchased
from him did.
Two questions. One, why the need for local side echo cancellation? I
thought you could just reverse the connection and
Yeah, this is the same procedure I went through with mine, worked like a
charm, zero problems whatsoever... Anyone have any idea what if any the
new features are of this firmware?
Aaron
Nabeel Jafferali wrote:
So has anybody tried installing the new SIP version?
It seems nobody has had luck
My guess, is nat problems. Just for fun, try dialing your inbound
number from something not connected to that asterisk box, say a
cellphone. I know you're using IAX and SIP, so you'd think you
wouldn't run into a double-nat problem (nat going out, nat coming in),
but you never know. I have odd
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote:
Is there a way to display the time of the 7960 running firmware 7.4? Im
unable to find any information.
Add the following to SIPDefault.cnf or SIPMAC.cnf:
sntp_server: time.nrc.ca
sntp_mode: unicast
time_zone: EST
You should
Gavin Adams wrote:
In Asterisk 1.2.4 is love being able to recording conferences. However,
using the default variables, the files are being written to
/var/lib/asterisk/sounds instead of /var/spool/asterisk/meetme.
If I change MEETME_RECORDINGFILE variable to something different in works,
bit
You sure it's not the Zaptel hardware creating the DTMF issues? what
Digium card you using?
On 3/8/06, Darren Wright [EMAIL PROTECTED] wrote:
Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.
Look for ANY of the 257* series...
Just ebay for t1 echo
-D
Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.
Are you guys talking about SIP?
On 3/9/06, Mailing List [EMAIL PROTECTED] wrote:
- Original Message -
From: Nabeel Jafferali [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
On Thu, 2006-03-09 at 17:16 +0100, Mickaël Cissé wrote:
Hi,
I'm using a Sipura SPA 3000 to receive calls from a PSTN line and to
send them to Asterisk. But on Asterisk, I don't see the PSTN caller ID!
I only have User ID of the SPA 3000 as caller number.
The caller number is present on the
We had that problem for a while. You have to configure the ntp server
in the phone so it'll pull the time otherwise it just randomly loses it.
Aaron
Greg Oliver wrote:
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote:
Is there a way to display the time of the 7960 running firmware
the Linksys and Sipura SPA-3000 are the same, just the plastic box is
different
- Original Message -
From: John Jensen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, March 02, 2006 1:53 PM
Subject: Re: [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000
Hi
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote:
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote: Is there a way to display the time of the 7960 running firmware
7.4? Im unable to find any information. Add the following to SIPDefault.cnf or SIPMAC.cnf: sntp_server: time.nrc.ca
OK, I found it now, it's under the NONSIP link on Ciscos site.
Acording to the docs it's meant only for Cisco Call Manager, does it
work with Asterisk?
On 3/9/06, C F [EMAIL PROTECTED] wrote:
Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.
Are you guys talking about SIP?
Thanks for the response Joseph.
It ended up that Telasip needed to make a change on there end. They needed to
disable re-invites.
BTW, I wanted to give a big plug for Telasip. I thought when I called they
would simply tell me it was my problem and they did not support asterisk. This
was
Hi gentlemen :-)
I am searching a radio base GSM or DECT with high power for long range, and
the terminal units (handy).
This equipment must be connected to a T1 port from an Asterisk.
The number of simultaneous channels must be 7 to 10.
Do you know a manufacturer with nice equipments at
The image is located in the non-sip section, go figure. They're harping
that this is for their new sip ccm...
Aaron
C F wrote:
Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.
Are you guys talking about SIP?
On 3/9/06, Mailing List [EMAIL PROTECTED] wrote:
-
Hi all,
I setup a new asterisk machine and all is working fine for maybe 10 days
Today there was the problem that nobody can hear the music during you are
waiting in a/the queue/s. Only silence was the answer.
Then I want to shutdown asterisk with stop now and nothing happends
After killing
No, I did not install Festival, but I saw that the text2wave module is
in the usr/bin directory.
I'm running RH Ent 2.4 kernel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: Thursday, March 09, 2006 10:17 AM
To:
This should get you where you need to go as long as you have a login:
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-7900ser
On 3/9/06, C F [EMAIL PROTECTED] wrote:
Why can't I find an 8.2 on Ciscos site? the latest I can find is 7.5.Are you guys talking about SIP?On 3/9/06, Mailing List [EMAIL
This issue has been fixed in SIP firmware 7.5
Omar A. Sabek
On 3/9/06, Nathan Bowyer [EMAIL PROTECTED] wrote:
On 3/9/06, Greg Oliver [EMAIL PROTECTED] wrote:
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote:
Is there a way to display the time of the 7960 running firmware 7.4? Im
I made a small change to apps/app_voicemail.c to permit circular
navigation when listening to messages. If you are at the first message,
and press 4, it takes you to the last message. If you are already at
the last message and press 6, it takes you to the first message. I
did a quick test
Hi allI have installed AAH 2.6created extension,and created Trunkcreated outbound routingiam able to make calls outand configured incoming, also working finewith the extension
I have problem hereI ahve extension sitting in same network where the AAH installedMy provider support canreinvite=yeswhen
Do you have a timeout set somewhere? Try Set(TIMEOUT(digit)=3), and/or
Set(TIMEOUT(response)=5)
On 3/9/06, Darren Ellis [EMAIL PROTECTED] wrote:
Hi List,
Merlin Magix hardware v02
I'm trying to get asterisk to act as a voicemail server for a lucent
merlin magix PBX that we purchased used.
Does that mean that since CCM supports SIP, Cisco will just make sure
that their SIP images work with CCM?
On 3/9/06, Aaron Daniel [EMAIL PROTECTED] wrote:
The image is located in the non-sip section, go figure. They're harping
that this is for their new sip ccm...
Aaron
C F wrote:
Why
Thanks. Its an
analog card and I really didnt find anything with good explanations (at
least no for me).
The problem is that the
people who give me support for my actual PABX, asked me the standard tone signaling.
Im trying to get
in my actual PABX from asterisk through the PABX fxo
I haven't been through everything line by line but I did notice a new Security
Configuration where you can set an Encrypt Key
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Ladies and Gentlemen, this is way, way, way off topic at this point.
Douglas's point was raised and a valid counter point was offered, let's
please just move on.
No amount of additional discussion is going to add this feature into
Asterisk. If this is a deal breaker for you, Douglas, you are
Please contact Globetel Communications http://www.globetel.net/ +1 954 241 0590. they have division that handles DECT solution that can interoperate with asterisk
On 3/9/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi gentlemen:-)I am searching a radio base GSM or DECT with high power for long
Joseph Tanner wrote:
Just a note:
Two questions. One, why the need for local side echo cancellation? I
thought you could just reverse the connection and it would now disable
echo in the opposite direction? Just curious, I don't have a T1, and
this is just based on what I've read.
I
I have an AMD64 x2 and running asterisk 1.2.4
When I call in to the dialplan all I have is:
exten = 11,1,Playback(demo-congrats)
exten = 11,n,Hangup
The console is showing the demo-congrats playing
but no audio.
I can call phone to phone and hear audio just fine.
Is there an issue with 64
Ditto on our installation of an Orion solution here in South Africa! works like a charm.CheersRobOn 09/03/06, Mike Clark
[EMAIL PROTECTED] wrote:Darren Wright wrote:Forget the orion.lots of DTMF problemstech support is not
Terribly well spoken.Look for ANY of the 257* series...Just ebay
Try to change the Short
Timer field back to the default value. If this doesnt help
either, use the pound key from your telephone key pad right after the last
digit is pressed, this will make the pap2 start the send
procedure.
Interdigit long timer is
the right field to change for the
On 3/9/06, Adam Robins [EMAIL PROTECTED] wrote:
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
I think your
Yeah, I noticed that too, and there's now five call managers in the
configuration you can set... not really sure how that'll help us in the
asterisk community ;) but guess we'll find out soon enough... As far as
I can tell, there's really no benefit to having this, other than maybe a
few bug
I just started looking at the differences between 8.2 and 7.5.
In addition to the new Security Configuration, there is also
compatibility added for provisioning the phone to CCM. The firmware
appears to be working fine.
Also, the copywright date range has been changed to 2000-2006.
Omar A.
PascalHere in South Africa we encountered a simialr problem and wrote a patch that has been incorporated into Asterisk 1.2.x , what we do here is:add this to your zapata.conf
For Cape Town:
busydetect=yes
busycount=4
busypattern=500,500
callprogress=no
For Johannesburg:
I figured it out. It should read:
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 8000 -o
/tmp/1141915933.wav
The 8 was missing in front of the 000'.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, March 09, 2006 12:04
Hi
I have 2 port T1 card in asterisk server. I am facing
following problem,
Span 1 is connected to
em circuit with wink-start
Span 2 is connectecd to
em circuit with feature group d
Is I activate the span two then I faced the following
problem.
1 call come on channel 26
Hello
Some questions about codecs..
What's the difference between the this codecs?
Which is used by asterisk?
Thanks
Juan Salas
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Hello all,
I have a client interested in GnuDialer. My question
is: Is there anyone on this list who has been using GnuDialer and I was
wondering if you would be willing to share your experiences with it.
Thank You
Adam
___
It looks like there is lots of discussion already going on about it at
http://bugs.digium.com/view.php?id=6457
On 3/7/06, Matt Riddell [NZ] [EMAIL PROTECTED] wrote:
The only catalyst to getting it fixed will be if someone posts a bug
entry with full details on bugs.digium.com
If you do, post
Hi all,
Do anyone have experience www.Chinaroby.com VOIP
phones?
I am very interestedfor models:PY-60 and PB-35
Phones.
Good or bad
experience with sip and IAX2, please comment.
I did not find any
comment on google
Regards
Darko
Sundek
eLink
Group
Kotor-Montenegro
Hi Pascal !
France is not more difficult than other country.
This is one of my channels behind France Telecom :
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
I believe they've done that the entire time. I've never known them to be real
supportive of competing third party solutions.
_
Mobilcom
http://www.mobilcom.net
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Quoting Mailing List [EMAIL PROTECTED]:
I believe they've done that the entire time. I've never known them to be real
supportive of
competing third party solutions.
They support third-party partners such as Broadsoft.
This
On Thu, 2006-03-09 at 13:11 -0600, Shane Young wrote:
Quoting Mailing List [EMAIL PROTECTED]:
I believe they've done that the entire time. I've never known them to be
real supportive of
competing third party solutions.
They support third-party partners such as Broadsoft.
Broadsoft is
Is there some way I can follow this list from a newsgroup??
Is this the same as the gmane group gmane.comp.telephony.pbx.asterisk.user
??
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From: [EMAIL PROTECTED] on behalf of ram
Sent: Thu 3/9/2006 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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