Hello,
How good is :SPA 841 form SIPURA.
Thanks
Varun
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Hi All:
I am trying to compile asterisk with oh323 but I can't
compile it. I am using instruction provided at http://www.oinko.net/astrecipes/index.php?from=1q=astrecipes/compiling+asterisk+with+oh323.
The compile error I am getting is as follows. Quite a few other people
Hi,
On Sat, Aug 06, 2005 at 09:05:51AM -0400, Zachary Whitley wrote:
On Sat, 2005-08-06 at 16:14 +0600, Madhawa Jayanath wrote:
Kumara Jayaweera wrote:
Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any
success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during
I have two 300s and 4 500s. The 300s talk the same language, but have a
lousy screen. The other thing to consider is, while it does have the
'monitor only' speaker, the volume is horrible. Cranked up to its highest
setting, you can't hear voicemail with ANY background sound. Go for the 501
Does anyone know how to connect the planet VIP-152T
phone to asterisk because I can't get it to work at
all. What configuration is needed? I have a
[EMAIL PROTECTED] setup and works perfect with the soft
phones but not with the planet... I 'm a newby so help
me pleaseee
[EMAIL PROTECTED] wrote:
This is a similar idea to LCR (least cost routing) on normal pbx
systems.
Any advice would be nice, since I'm sure those users who use asterisk
for more commercial purposes have figured our a way to do this...
Jump to the LCR section on this page:
On Sat, Aug 06, 2005 at 08:50:58AM +0200, Christian Stredicke wrote:
Please take a look at http://www.snom.com/howto40.html. We tried to make
the upgrade procedure as smooth as possible, if you are having problems
please tell us and we will try to make it more simple. For example, if
you have
I have installed a HFC-S card in nt mode according to the documentation
on voip-info.org and it works quite well except for two problems:
1. When I pick up the phone I no not get a dial tone indicating that I
can start dialing but asterisk seems to jump automatially to extension s.
2. When I
On Sun, August 7, 2005 2:07 am, [EMAIL PROTECTED] said:
How good is :SPA 841 form SIPURA.
Not good if voice quality is a requirement. Talking on the handset sounds
to the caller and callee like you're on one of those really old
speakerphones that clips the beginning of each phrase after a
In you sip.conf what if you change:
register = 7771::[EMAIL PROTECTED]/7771
to
register = 7771:[EMAIL PROTECTED]/7771
PB
Jenna Cole wrote:
im using iptel.org SER proxy.
the proxy is working without authentication.
the problem is that the Asterisk is not sending a
REGISTER sip message.
Does this explain why they would not register, or do I have to worry that
there is some new setting which caused the problem? I do not want to go
through the pain of upgrading the customer's phones (probably with a site
visit) only to find that I have to downgrade them and go though it again
I too have been having inbound dtmf problems with VP Connect using
iax2 for inbound. When I switched to sip, and added the
relaxdtmf=yes, all 10 inbound test calls I did seemed to work fine for
dtmf. I'm going to leave my config set up to use sip for inbound VP
Connect calls for a while and see
Hello,
I'm a newbie trying to figure out how to install an ISDN Card.
I downloaded the latest Asterisk iso and did install it (actually version
1.3).
ISDN Card details:
Eicon Diva Server BRI-2M/-2F
I did listen something that capi 2.0 is needed and some drivers etc. but I
can't figure out how
I noticed another user on this had the same problem that i'm having.
Through the webpage I can't connect to database I get unavailable
database message. In my http logs I get this:
[Sun Aug 07 00:13:38 2005] [error] [client 172.25.25.30] DBI
connect('database=astcc ; your astcc database
There are already some bug reports at bugzilla.atrpms.net on
enhancements and bugs in the packages, see
Hello all:
I need help configuring * to register with Net2phone using the
credentials provided with an Innomedia MTA 3328-2r fxs device. In using
ethereal I see where the user agent string includes the MAC address of
the device. Net2phone also is using MD5 authentication.
If the mac address
Chris Coulthurst wrote:
I have two 300s and 4 500s. The 300s talk the same language, but have a
lousy screen. The other thing to consider is, while it does have the
'monitor only' speaker, the volume is horrible. Cranked up to its
highest setting, you can't hear voicemail with ANY
Hello all
We succesfully added a H323 Gateway to our
CallManager 3.0 that resides in Mexico and were/are able to make calls from
CallManager SCCP phones to the Asterisk Server phones in the U.S.; however, we
have not been able to call from Asterisk server in U.S. to CallManager phones in
Hi all,
I'm new to the forum. Oh nonewbie question
coming, I hear you all cry!
I'm playing around with [EMAIL PROTECTED] and have installed software and fiddled
around with sip and extensions files.
I have manage to make out going calls through
Sipgate using X-Lite but cannot for some
As you can see, the channels are set properly. One thing I did notice is
all of the ;; in front of the [ext] sections. Does that seem
correct? I removed them and it didn't change anything. Other files that you
would like to look at?
Thanks,
Mike
Looks a bit more
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Paul Dugas
Envoyé : dimanche 7 août 2005 16:11
À : Asterisk Mailing List
Objet : Re: [Asterisk-Users] SPA 841 form SIPURA
On Sun, August 7, 2005 2:07 am, [EMAIL PROTECTED] said:
How good is
Hello
I have created an iax exten in my iax.conf file:
[300]
type=friend
username=300
secret=***
context=default
host=dynamic
callerid=some name 300
auth=md5
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX
I've posted my config files in Adobe pdf format at
http://www.brianmccarey.com/voip/sip
http://www.brianmccarey.com/voip/extensions
http://www.brianmccarey.com/voip/trunk
I think you're either going to get complaints about the pdf files or
people are simply going to ignore your question. Is
I knew about that one. I have Silence Suppression set to NO.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED],
Jim Duda [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for
Angus Comber wrote:
I can dial from iaxComm (a soft IAX client) and that works fine. But
when I try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No channel type registered
for 'IAX'
NOTICE[22077]: app_dial.c:777 dial_exec: Unable to create channel of
type 'IAX'
Try IAX2
I was wondering if there would be any interest or support out there for
an IVR voice prompt repository, a la atrpms but for voice prompts
instead of rpms. I was thinking of something that collected the meta
data such as spoken text, gender, file size, speaker ID, language,
duration, encoding, MD5,
On 8/7/2005, Angus Comber [EMAIL PROTECTED] wrote:
Then in my extensions.conf I have:
exten = 300,1,Dial(IAX/${EXTEN},20)
exten = 300,2,Hangup
I can dial from iaxComm (a soft IAX client) and that works fine. But
when I try to dial 300 get:
WARNING[22077]: channel.c1970 ast_request: No
I think that's a good idea, something that I'd have use of atleast :)
That be if there does not exist a site like this already, but non that I
have heard of.
Johan
There are two major products that come out of Berkeley: LSD and UNIX. We
don't believe this to be a coincidence. -- Jeremy S.
[EMAIL PROTECTED] wrote:
coming through the list and I get tired of reading all the questions that
5 minutes of reading the configuration files or searching the wiki (as out
of date as it is) or even typing 'help' at the CLI prompt can remedy.
And some of us are getting tired of people
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
I knew about that one. I have Silence Suppression set to NO.
Ah, ok. Puzzling then. If you'd like to post the full budgetone config
page(s), one of us might be able to spot something.
What revision of budgetone firmware are you
I was tinkering with Asterisk and the Festival text-to-speech engine, and
wrote some short Asterisk::AGI scripts to read back live weather reports.
After that, I thought I needed something more interactive to work with...
Then I had a flashback to 1996, first year university, standing in the C
O
On 8/7/2005, Kevin P. Fleming [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
coming through the list and I get tired of reading all the questions that
5 minutes of reading the configuration files or searching the wiki (as
out of date as it is) or even typing 'help' at the CLI prompt can
Here's my kernel info:
Linux asterisk.hulber.com 2.6.9-11.EL #1 Fri May 20 18:17:57 EDT 2005
i686 i686 i386 GNU/Linux
And my Asterisk version:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running
Linux on 2005-08-05 21:21:13 UTC
Kumara Jayaweera wrote:
Hi MARK,
Thanks a lot for
And card status:
asterisk*CLI zap show status
Description Alarms IRQ
bpviol CRC4
Wildcard TDM400P REV E/F Board 1 OK 0
0 0
Kumara Jayaweera wrote:
Hi MARK,
Thanks a lot for the reply. my box is Intel
Dear folks,
Actually this is my first post here, so sorry for any inconvenience. Im planning for a solution a bit larger in scale than ususal. I'm goin to use * as a PSTN gateway with E1 links and use two other 3rd party Gateways for FXO lines. I should be able to switch from every incoming
[EMAIL PROTECTED] wrote:
snip
I am not just picking on you Angus. I do tend to read almost every
message coming through the list and I get tired of reading all the questions
that 5 minutes of reading the configuration files or searching the wiki (as out
of date as it is) or even typing
Wow! Not sure what else to say. This ranks right up there with my ability to
open my garage door from asterisk...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 07, 2005 1:46 PM
To: asterisk-users@lists.digium.com
Thanks for the assistance.
I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago.
I'm interfacing with an Asterisk box on my local lan.
My sip.conf is as follows:
[100]
type=friend
context=home
callerid=Jim 100
secret=mysecret
host=dynamic
nat=no
canreinvite=yes
[EMAIL PROTECTED] wrote:
At least I used a personal pronoun... or are you speaking for Digium? 8-)
Nope, just Sunday afternoon venting... no official position should be
inferred from my comments G
Well - here on the list I have seen people still using 1.0.5. Maybe that
info is correct
[EMAIL PROTECTED] wrote:
How good is :SPA 841 form SIPURA.
I won't be ordering any more of them... One of the units we ordered had
problems with the dial pad not registering the correct key press. i.e.
when pressing the line 1 button line 2 would activate and dialing 5
would dial 5
If you find a wiki page that is incorrect, incomplete or needs any
other editing, do it! The rest of the community will be thankful for
your help.
I don't want to get in the middle of this but what wiki are we referring
to? voip-info.org/wiki-asterisk ?? I would be willing to contribute if
On 8/7/2005, John Novack [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote:
snip
I am not just picking on you Angus. I do tend to read almost every
message coming through the list and I get tired of reading all the
questions that 5 minutes of reading the configuration files or
searching the
On Sun, 2005-08-07 at 14:59 -0500, Tim Connolly wrote:
Wow! Not sure what else to say. This ranks right up there with my ability to
open my garage door from asterisk...
Sarcasm or serious? Sounds cool to me.
___
Asterisk-Users mailing list
On 8/7/2005, Zachary Whitley [EMAIL PROTECTED] wrote:
If you find a wiki page that is incorrect, incomplete or needs any
other editing, do it! The rest of the community will be thankful for
your help.
I don't want to get in the middle of this but what wiki are we
referring to?
I've re-uploaded the config files in NON
pdf
Any help welcomed.
Regards
- Original Message -
From:
Brian McCarey
To: asterisk-users@lists.digium.com
Sent: Sunday, August 07, 2005 5:55
PM
Subject: [Asterisk-Users] Configuring [EMAIL PROTECTED] for Sipgate.
Please excuse my ignorance but doesn't the VOIP/PSTN gateway
(Broadvoice,VP Connect) have to support T38 in order for an T38
supported ATA to do any good ?
Yes. BroadVox supports it. I've started a wiki page to track
providers who do, mainly because such providers are so hard to find,
and
The bounty stands at $5,500. I'm seriously considering taking a shot
at it if I can find a decent T.38 provider to test with (I'm still
hoping for reliable PAYG T.38).
It looks like a lot of very smart people have done a lot of very hard
work (t38modem, spandsp) that would go towards getting
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
Thanks for the assistance.
I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago.
I'm interfacing with an Asterisk box on my local lan.
My sip.conf is as follows:
[100]
type=friend
context=home
On Sun, August 7, 2005 1:15 pm, Thierry Wehr said:
This is not true
You have to switch to last firmware and/or disable silent suppression
I believe Thierry is not alone in having success with these units. I
cannot explain it but my guess is that there are some inconsistencies in
their hardware
On Sunday 07 August 2005 14:45, [EMAIL PROTECTED] wrote:
Now Zork is back! Listen as the eerie voice of Festival takes you into
the Underground Empire, and marvel as you explore this world with your
dial pad, unlocking the secrets within!
Haha! Hehe, very cool. READ ALL ABOUT IT! How old
Hi,
My asterisk server is behind firewall and i am trying
to connect to FWD. i hv configured as mentioned in
this link
http://www.freeworlddialup.com/advanced/iax. i am able
to register my server with FWD. But when i dial
393612, i always get 'No one is available to answer
this time, try again
Hi all!
I got information that NT1 devices with analog ports (ab ports like on
Siemens Santis or Intracom NetMod or etc.), can be used like ISDN BRI/Analog
converters (i.e. for connecting analog phones or faxes to a HFC based ISDN
cards in NT mode).
Yesterday, I tried to do so with Intracom
I've attached my zip file. Thanks for the help.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED]
wrote:
Thanks for the assistance.
I'm running version 1.0.6.7 of the software, tftp updated a few weeks
It appears that incoming calls (IAX) through voicepulse are being
rejected... anyone else experiencing this?
-Trent Tuggle
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
http://bugs.digium.com/view.php?id=4631
FYI
I am experiencing the same, but due to lack of cooperation from ITSP
am not able to proceed with debugging it.
Feel free to pursue...
regards,
mark
On 8/8/05, Justin Richards [EMAIL PROTECTED] wrote:
I too have been having inbound dtmf problems
Today the front page of http://www.voip-info.org/ was taken out by a
spammer. It also seem the history page for http://www.voip-info.org/
was also nuked. I've restored the best I could using google cache, but
still missing some information.
Who is an admin on http://www.voip-info.org/ and
Jim Duda wrote:
I've attached my zip file. Thanks for the help.
Jim
Tony Mountifield [EMAIL PROTECTED] wrote in message
Try using the IP address of the server instead of the name.
Doug
___
Asterisk-Users mailing list
Adam Megacz wrote:
The bounty stands at $5,500. I'm seriously considering taking a shot
at it if I can find a decent T.38 provider to test with (I'm still
hoping for reliable PAYG T.38).
It looks like a lot of very smart people have done a lot of very hard
work (t38modem, spandsp) that would
Steve Underwood wrote:
produce anything more than a botch for it. A couple of people have said
they are reworking sip.c to make the addition of new codecs, transports,
etc. and their renegotiation function smoothly. I haven't seen any
results so far. I did only minimal work on sip.c in the
Thank you very much
kumara
- Original Message -
From: MF Hulber [EMAIL PROTECTED]
To: Kumara Jayaweera [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Sunday, August 07, 2005 10:24 PM
Subject: Re: [Asterisk-Users] Does
I have a NY 212 packet8 service if you would like to work with me to set
this up on my [EMAIL PROTECTED] service, I'm happy to test this with you.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Megacz
Sent: Sunday,
Correction.
i hv port forwarded udp 4569 and 5060.
-B
--- Balaji NJL [EMAIL PROTECTED] wrote:
Hi,
My asterisk server is behind firewall and i am
trying
to connect to FWD. i hv configured as mentioned in
this link
http://www.freeworlddialup.com/advanced/iax. i am
able
to register my
Balaji NJL wrote:
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
It's working fine..
== No one is available to answer at this time
-- Executing Goto(SIP/200-d345,
s-NOANSWER|1)
in new stack
..but you're not answering the phone, or it's offline.
Try
hi,
thank you vary much for the updates, i jsut got the latest from cvs and the
error is fixed, however, i got this new error, when running make,
/usr/local/sparc-sun-solaris2.8/bin/ld: cannot find -lncurses
collect2: ld returned 1 exit status
make: *** [asterisk] Error 1
pls advise.
much
64 matches
Mail list logo
