Brian May wrote:
Hello,
Based on my reading and understanding of the documentation, in
extensions.conf all I need is:
exten = _5XXX,1,Dial(IAX2/ivt/${EXTEN})
As asterisk will look up the rest of the configuration in iax.conf:
--- cut ---
[ivt]
username=microcomaustralia
type=friend
Quoting Matt Riddell [EMAIL PROTECTED]:
Is there a way of converting the play tone to a gsm file which can be played
using the A option?
Obelix wrote:
Is it possible to get Asterisk to issue a Playtones when an outgoing call
is
answered? The examples indicate what happens when an
Colin Anderson wrote:
Onboard LAN with an un-movable IRQ would mess that up good
Only if you had just one pci slot.
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free
Nir Simionovich - CTO wrote:
Over the course of the past 3 years, I've used the following boards with
high success rates: Intel, Tyan, GigaByte, HP/Compaq and some IBM machines.
I also integrated on some Asus and SuperMicro, but I wouldn't call those a
tier-1 installation, as they were
nr k wrote:
hi generally we describe the bandwidth in kilobits per
second only.
Cool, just checking, it seemed pretty low.
According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should
be able to do 4 calls with g729.
--
Cheers,
Matt Riddell
Anton Krall wrote:
I'll need to run a test, Ill remove busydetect from zapata.conf tomorrow and
see if the line drops after a hangup, if so, then we should be set, if not,
then we are in trouble?
Unless you can change the PBX's cadences to not be the same as your disconnect
cadences...
--
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Matteo
===
Matteo Piazza, Junior Researcher
CREATE-NET
Via Solteri, 38 - 38100 Trento - Italy
email: [EMAIL PROTECTED]
Tel: +39-0461-408400ext:308
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an example).
You can then copy the file into /var/spool/asterisk/outgoing to make the call.
--
Obelix wrote:
Quoting Matt Riddell [EMAIL PROTECTED]:
Is there a way of converting the play tone to a gsm file which can be played
using the A option?
Sure, if you send me the dtmf tones you need and I'll mail you some gsm files.
--
Cheers,
Matt Riddell
Hi,
Just wondering whether anyone has done fax relaying or
pass-through using Asterisk T.38
Please let me know your thoughts as I need to come up with a
fax server using Asterisk with T.38 possible?
Cheers!
Lilantha
___
Peter Dean wrote:
I have now been successful in getting the notification lights working.
Then asterisk extensions hint required a reference to the extension
being monitored and the extension monitoring the call status.
i.e. _226,hint,SIP/226SIP/101
So with this change the asterisk hint
On Wed, 2005-11-09 at 21:32 +1300, Matt Riddell wrote:
nr k wrote:
hi generally we describe the bandwidth in kilobits per
second only.
Cool, just checking, it seemed pretty low.
According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should
be able to do 4 calls with
[EMAIL PROTECTED] wrote:
Hi List
I’m getting this notification from my one and only SNOM 360 every time a
number button is pushed.
I know that it’s only a notification, but it really irritates me. Is it
anything I can/should do anything about ??
Not really. We do not support
On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote:
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an example).
You can then copy the
Does asterisk support RFC3265 ?
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here!
Actually someone asked for you config details.
--
Cheers,
Matt Riddell
___
Hi,
I want to build a PRI pass-through with a Cisco 2600, with two VWIC E1
cards, is this possible ? and do i need any other modules except for the
E1 modules ?
What i want to do is connect the asterisk to the PRI through the Cisco
router, and let my legacy PBX utilize some of the PRI
On Wed, 2005-11-09 at 01:16 -0800, trixter aka Bret McDanel wrote:
Bandwidth is a tricky issue. You have your IP + UDP + RTP + whatever
headers (iax2 combines stuff so potentially that skews this a bit) but
something most often forgotten is link layer framing.
I should have added to this that
trixter aka Bret McDanel wrote:
On Wed, 2005-11-09 at 21:59 +1300, Matt Riddell wrote:
Matteo Piazza wrote:
If I use meetme conference room, can I invite another user during a
conversation?
In which way?
Use a .call file (search for sample.call in the asterisk source directory for
an
You can set up two *. One that will only interact with your VoIP provider and
another that will be POST gateway and will run on 5060. Connect them with IAX2.
--
Tomislav Parcina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)393447
e-mail: tparcina#lama.hr
http://www.lama.hr
The only pointer I got is a $50/hr Mark phillip
offered.
I can make VOIP calls between my Asterisk server and
my
VOIP provider using sip channel without a problem. But
when I attempt to make a call using IAX, the call get
accepted and then get a hangup message:
This is the message I get when I
Alex, thanks so much, that was it - I don't know how I missed it. I guess I
was looking for more complicated reasons :-).
Cheers,
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alexander O. Lopez
Sent: Tuesday, November 08,
Hello Matt,
In fact I look for messaging an presence between sip
phones .
http://www.voip-forum.com/news.php?p=184c=1
I use polycom ip phone with presence (rfc3265) and IM
(SIMPLE).
Do you you think the job of Joshua Colp could help me
to use presence/IM with asterisk ?
Regards
Harry
Somebody would be interested in a such project ?
Harry
--- Kristof Hardy [EMAIL PROTECTED] a
écrit :
harry gaillac wrote:
Is it possible to add a frontend groupware with
All is possible, you're only limited by your
imagination. (always wanted
to say this :p)
I'm not sure there's a(n
Hi all,
I am having an issue with individual access vs simultaneous access.
If I set a card for individual access, make a call with that card the counter
goes to 1. If the call complets normally shouldnt the counter reset to 0?
Second call tells me that card is already in use.
simultaneous
Hello List,
I'm glad to announce that we have released the first version of
QueueMetrics that supports MySQL storage of queue_log data. It is still
experimental, so if you run such a setup and would like to give it a try,
you are welcome. The MySQL adapter should adapt to any existing table
harry gaillac wrote:
Hello Matt,
In fact I look for messaging an presence between sip
phones .
http://www.voip-forum.com/news.php?p=184c=1
Should work with current CVS HEAD version.
I use polycom ip phone with presence (rfc3265) and IM
(SIMPLE).
Do you you think the job of Joshua Colp
chawki hammoud wrote:
The only pointer I got is a $50/hr Mark phillip
offered.
Put notransfer=yes in the iax.conf section for that account.
Then try adding trunk=yes or trunk=no (try both), and if you use trunk=yes
make sure there is a (t) by the peer in iax2 show peers.
--
Cheers,
Matt
harry gaillac wrote:
Somebody would be interested in a such project ?
I think quite a few people do this kind of thing in house - it's kinda one of
those personal preferences things.
Does anyone want to make one that fits everyone's setup? I don't know. But I
don't have enough time at the
Hi,
I want to use an SPA-3000 connect my dutch kpn PSTN line to and from the
Asterisk VOIP network.
Dialing in (via my kpn pstn line) is functioning oke, with a great sound
quality.
Dialing out to the pstn line produces after a verry short ring a busy signal.
If I connect the pstn to my
i can fix that please contact me off list, i have setup now that same
as yours and i encountered that problem.
On 11/9/05, chawki hammoud [EMAIL PROTECTED] wrote:
The only pointer I got is a $50/hr Mark phillip
offered.
I can make VOIP calls between my Asterisk server and
my
VOIP provider
Hi all;
Ihave on my zapata.conf overlap=yes.
In my extension i have:
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
I want to let user more than 5s to dial, i want to let him 3s by digits.
Can you help me!!!
Thanks
//vador
___
--Bandwidth and Colocation
On Wed, 2005-11-09 at 23:00 +1300, Matt Riddell wrote:
trixter aka Bret McDanel wrote:
For anyone doing this it may not always be /var/spool/asterisk/outgoing,
especially with non linux installs. check your asterisk.conf file for
astspooldir. It should be in that directiory/outgoing :)
Hi all, weird problem, this seems to happen without any rhyme nor reason
yesterday from /var/log/asterisk/full
Nov 8 18:07:02 VERBOSE[3270]: -- Executing
BackGround(IAX2/[EMAIL PROTECTED]/4, crim/main-menu) in new stack
Nov 8 18:07:02 VERBOSE[3270]: -- Playing 'crim/main-menu'
the asterisk's answer !
//
Connected to Asterisk 1.2.0-beta2 currently running on
serveur1 (pid = 2729)
Remote UNIX connection
Verbosity is at least 3
Nov 9 11:48:21 WARNING[2926]: chan_sip.c:7251
receive_message: Received message to
sip:[EMAIL
On Wed, 2005-11-09 at 23:39 +1300, Matt Riddell wrote:
harry gaillac wrote:
Somebody would be interested in a such project ?
I think quite a few people do this kind of thing in house - it's kinda one of
those personal preferences things.
Does anyone want to make one that fits everyone's
bails napisał(a):
Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3 (format
ulaw): Permission denied
^
Any Ideas?
Maybe this is a problem with permisions to this file?
--
Best regards,
Bartosz Piec
___
Hi!
I read in the archive a lot of problems using the Dell 1850 servers and
digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has tried
the Dell Poweredge 850 series and can report some experiences?
btw: Does somebody knows why there are problems with 1850 but not with
2850
You don't have this problem when using AstBill to manage Asterisk.
We are doing call forwarding from the database to single or multiple extensions.
As the Dial command is managed from the MySQL database we ignore voicemail forwarding when ringing multiple extensions.Are
Sounds like a good deal to me. If you want free answers don't sound so irritated that you haven't got a reply in $0 time. :)RobOn 11/9/05, chawki hammoud
[EMAIL PROTECTED] wrote:The only pointer I got is a $50/hr Mark phillip
offered.I can make VOIP calls between my Asterisk server andmyVOIP
The very problem is that DELL in the small one, block the IRQ. And this
can make conflict to the cards.
Bruno.
Klaus Darilion wrote:
Hi!
I read in the archive a lot of problems using the Dell 1850 servers
and digium TE4xxP and TE2xxP hardware. I wonder if ever anybody has
tried the Dell
Hi all,
We use asterisk as a local pbx and we connect to a pstn/sip provider for
calls to pstn.
Since the messages on asterisk are on gsm format, we need gsm, but to call
pstn, we need g729 or g723.
How can we enable both codecs to be able to call pstn and hearing voicemail
messages for
i think gsm you mention is gsm sound files not gsm codecs.
On 11/9/05, Olivier Taylor [EMAIL PROTECTED] wrote:
Hi all,
We use asterisk as a local pbx and we connect to a pstn/sip provider for
calls to pstn.
Since the messages on asterisk are on gsm format, we need gsm, but to call
pstn, we
Right,
I must suppose I need gsm codec to hear gsm files, I miss?
olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Angelito
Manansala
Envoyé : mercredi 9 novembre 2005 12:28
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re:
John Fraser a écrit :
Hi all,
I am having an issue with individual access vs simultaneous access.
If I set a card for individual access, make a call with that card the counter
goes to 1. If the call complets normally shouldnt the counter reset to 0?
Second call tells me that card is already
You simply need to have g729/g723 codecs. Asterisk comes with gsm by
default.
Regards,
Sahil Gupta
VoiceValley
On Wed, 9 Nov 2005, Olivier Taylor wrote:
Right,
I must suppose I need gsm codec to hear gsm files, I miss?
olivier
-Message d'origine-
De : [EMAIL PROTECTED]
On Wed, Nov 09, 2005 at 11:43:34AM +1100, Mark Edwards wrote:
Hey Waldo.
AFAIK there is quite a lot of scope for tuning the compression of speex
- just as there is for mp3. I have no doubt that if you tune complexity,
quality and bitrate parameters you will be able to get that filesize
down
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent
to use gsm for voicemail and g729 for outbound calls?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Tzafrir Cohen
Envoyé : mercredi 9 novembre 2005 12:35
À :
I can't open your on-line demo. (9. 11. 2005. at 12:49
GMT+2)
Tomislav
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
snacktimeSent: 3. studeni 2005 2:22To: Asterisk
Users Mailing List - Non-Commercial Discussion; Commercial and
Business-Oriented Asterisk
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent
to use gsm for voicemail and g729 for outbound calls?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Sahil Gupta
Envoyé : mercredi 9 novembre 2005 12:33
À : Asterisk
Some parts of it, yes.
On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote:
Does asterisk support RFC3265 ?
Harry
--- Matt Riddell [EMAIL PROTECTED] a écrit :
harry gaillac wrote:
nobody has an answer here!
Actually someone asked for you config details.
--
Cheers,
Matt
Thanks Colin and Mark for your answers. I finally
manage to start up asterisk bycopying libbluetooth to
/usr/lib/.
Now the final step comes, make this bluetooth
thing works.
These are my configuration
files:
/etc/bluetooth/rfcomm.conf:
rfcomm0
{ bind yes;
device
00:0E:6D:34:BD:B1;
HI all,
Is Asterisk able to work as SIP and H.323 Gatekeeper
same time?
If it has the capability to work which i should open?
Yours suggestion will be high appriciated.
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!:
Bartosz Piec wrote:
bails napisał(a):
Nov 9 10:37:29 WARNING[2083]: Unable to open crim/main-menu.mp3
(format ulaw): Permission denied
^
Any Ideas?
Maybe this is a problem with permisions to this file?
It was indeed a permissions problem
I'm not a developper !
What do you mean Some parts of it, yes.
harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
Some parts of it, yes.
On 11/9/05, harry gaillac [EMAIL PROTECTED]
wrote:
Does asterisk support RFC3265 ?
Harry
--- Matt Riddell [EMAIL PROTECTED] a
écrit :
harry
I get this when I boot asterisk. I have a Wildcard TDM400P REV H with 1 FXO
board on it.
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/usr/local/etc/asterisk/zapata.conf': Found Nov 8 22:11:51
WARNING[24698]: chan_zap.c:935 zt_open: Unable to specify channel 1:
Operation not supported
Salut Harry,
Tu quittes Ser pour asterisk?
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : mercredi 9 novembre 2005 13:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users]
not so complicated.. use IAX trunking to share dialplans
On 11/9/05, KRTorio [EMAIL PROTECTED] wrote:
Our Setup:
In our company we run multiple asterisk servers, and agents login (using
AgentCallbackLogin) to any of these. One person, one agent number ID.
The Problem:
Dialing an agent
Hi Harry
We are doing some of it with AstBill and love to work with you to include your requirements.
We have Click to Dial from address book, SMS and Fax is not released but working with clients.
We want to intergrate AstBill with a Groupeware or CRM but want input what people will prefeer.
Hi all,
I have some problems with music on hold.
It works with the default category but not with additional ones.
When I start Asterisk, 2 mpg123 processes are started with the default
moh but none with additional ones.
Does someone already had this problem and could help me?
Thanks,
Amaury
Works well. I am running 1.0.9 stable on this with FC2 on kernel 2.6.9 The
kernel needs patching to pick up the onboard SATA (ICH7), or we use a pci
express SATA raid controller with a TE110p. The only real hassle is the
single 'standard' pci slot in it. Remote access is via SOL and the
On Wed, 9 Nov 2005, Olle E. Johansson wrote:
That is not supported yet. There is a patch in the issue tracker that
does this, but it's a proof-of-concept code. It will burden your
asterisk quite a lot if you put it to use in larger production sites.
Which issue are you refering to?
--
Hi,
Thanks for your response.
I checked the setting, and indeed it was set to yes. However, once I
change it to no and click on apply but after rebooting it's enabled
again (with all settings reverted to factory defaults, as usual).
Maxi.
2005/11/8, Rusty Dekema [EMAIL PROTECTED]:
It's
Nonetheless .. Thanks everyone for the responses! I think I have it
now! You guys are great!
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615
Poor planning on your part does not
If you unplug the ethernet cable on a Sipura SPA and then reset the
power it'll boot up in a diagnostic mode. When you pick up the phone
that's connected to it you'll get a dialtone and there are speical codes
you can dial to do various things.
Reset it to factory defaults by dialing
On 11/9/05, Craig Guy [EMAIL PROTECTED] wrote:
Works well.I am running 1.0.9 stable on this with FC2 on kernel 2.6.9Thekernel needs patching to pick up the onboard SATA (ICH7), or we use a pci
express SATA raid controller with a TE110p.
Which pci-e SATA controller are you using? The one that
Yes. I believe the Cisco phones do conferencing in the same fashion. I'm
not 100% on whether or not the SPA-841 or the new SPA-941 does it.
The SPA-941 does conferencing and it works exactly like the transfer.
a soft button you hit twice, Conf once to dial the invited 3rd party
and once to
On Tuesday 08 November 2005 18:20, George Pajari wrote:
To make a long story short, according to Intel Dealer Technical Support
(we became Intel dealers in order to get answers to our questions) there
is no Intel motherboard that permits the IRQs to be configured uniquely.
They are all
On Wednesday 09 November 2005 03:29, Matt Riddell wrote:
Colin Anderson wrote:
Onboard LAN with an un-movable IRQ would mess that up good
Only if you had just one pci slot.
With 1U systems that is often all you get.
-A.
___
--Bandwidth and
You don't need to be a developer to understand my statement.
The current chan_sip does support some of the behaviors and methods
described in RFC3265 to support the presence functionality that is
currently part of Asterisk and chan_sip.
Does this help?
On 11/9/05, harry gaillac [EMAIL
I followed your steps to the letter but after resetting to factory defaults
unfortunately it still doesn't record the configuration changes I do.
2005/11/9, Adam Moffett [EMAIL PROTECTED]:
If you unplug the ethernet cable on a Sipura SPA and then reset the
power it'll boot up in a diagnostic
bails wrote:
Hi all, weird problem, this seems to happen without any rhyme nor reason
yesterday from /var/log/asterisk/full
Nov 8 18:07:02 VERBOSE[3270]: -- Executing
BackGround(IAX2/[EMAIL PROTECTED]/4, crim/main-menu) in new stack
Nov 8 18:07:02 VERBOSE[3270]: -- Playing
Olivier Taylor wrote:
User-agents have g729, g723.1 and gsm, isn't it possible to force user-agent
to use gsm for voicemail and g729 for outbound calls?
No.
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Thank you very much for trying it for me, Dave. I really appreciate it.
Paul
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Rahn
Sent: Tuesday, November 08, 2005 8:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Andrew Kohlsmith wrote:
On Tuesday 08 November 2005 18:20, George Pajari wrote:
To make a long story short, according to Intel Dealer Technical Support
(we became Intel dealers in order to get answers to our questions) there
is no Intel motherboard that permits the IRQs to be configured
Well then you have me stumped. There's a Sipura forum at voxilla.com here:
http://voxilla.com/forum-viewforum-f-14.html
Maybe someone there will know more about it.
I followed your steps to the letter but after resetting to factory defaults
unfortunately it still doesn't record the
Hi all,
I need help with
Asterisk. Recently I setup an Asterisk with TE210 and spandsp to check the fax
capabilities of Asterisk + Spandsp. The two E1 are connected back to back. I
created a script that opens a channel from the first E1 and calls a channel in
the second in order to send a
A nasty screech. That's what callers here sometimes when they dial into
my FXO port from the PSTN. But usually, it works OK.
Is this common?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
I have install Festival in Asterisk, but I don't
listen to anything, but Asterisk show this message
Parsing '/etc/asterisk/cdr_custom.conf': Found
-- Executing Answer(SIP/101-35e3, ) in new
stack
-- Executing Festival(SIP/101-35e3, Hello
asterisk user| how are you today?) in new stack
Hi,
I'm trying to send some DTMF dialtones (for an extension on the other end).
My call is done from a Zap channel, to Asterisk, throught an IAX provider,
to a PSTN line in some university.
The phone number I am trying to reach is 555-555- exten 1234.
What I did is an
Exten =
I just installed FXO module in an older TDM400 card in port 1 and had
problems. Moved it to port 2 and everything is fine now.
- Dustin -
Bill Michaelson wrote:
A nasty screech. That's what callers here sometimes when they dial
into my FXO port from the PSTN. But usually, it works OK.
Is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
my topology is:
CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services
I need to connect my phones registered on CME to ISP Services using
g729 codec.
Well, on cisco I set the codec preference with a voice class:
voice
Hi George,
I run an Intel D865GBF Desktop board with Digium's TDM400P with 4 FXOs
just fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of George
Pajari
Sent: Tuesday, November 08, 2005 6:20 PM
To: Asterisk Users Mailing List - Non-Commercial
On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote:
18: 1204255212 IO-APIC-level wctdm
19: 1198491079 IO-APIC-level t1xxp
22: 1198502476 IO-APIC-level wcte11xp
Holy shit and you've got three Digium cards in there... all on their own IRQ.
-A.
Do a debug voip ccapi on the CME and look through it. It will have
detailed codec negotiations, etc in it.
-Greg
On Wed, 2005-11-09 at 16:10 +0100, Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
my topology is:
CME (Cisco) -- [sip trunk] -- Asterisk --
asterisk183 napisał(a):
therefore don't show error.
Test the Festival server console (festival --server). I had permision
denied for localhost.localdomain. You must change it in festival.smd
file (maybe the name is a bit different).
--
Best regards,
Bartosz Piec
Andrew Kohlsmith wrote:
On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote:
18: 1204255212 IO-APIC-level wctdm
19: 1198491079 IO-APIC-level t1xxp
22: 1198502476 IO-APIC-level wcte11xp
Holy shit and you've got three Digium cards in there... all on their own IRQ.
This will not work. The PRI uses a single D channel for signalling.
It can only go to one PBX, either * or legacy. Yes the cisco can map
DS0 between the E1, but I believe you need the VWIC-MFT series to do
so (may be wrong on that) but that will definately break the PRI.
Either run the PRI
On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote:
Andrew Kohlsmith wrote:
On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote:
18: 1204255212 IO-APIC-level wctdm
19: 1198491079 IO-APIC-level t1xxp
22: 1198502476 IO-APIC-level wcte11xp
No. APIC was in 2.4 as well, but you need an Intel CPU in there (I
think) in order to be able to take advantage of it. AMD's don't have
this option available.
On 11/9/05, Pete Barnwell [EMAIL PROTECTED] wrote:
On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote:
Andrew Kohlsmith
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On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote:
Do a debug voip ccapi on the CME and look through it. It will have
detailed codec negotiations, etc in it.
thanks for your answer, Greg.
Could you help me?
Pete Barnwell wrote:
On Wed, 2005-11-09 at 09:37 -0600, Eric ManxPower Wieling wrote:
Andrew Kohlsmith wrote:
On Wednesday 09 November 2005 09:34, Eric ManxPower Wieling wrote:
18: 1204255212 IO-APIC-level wctdm
19: 1198491079 IO-APIC-level t1xxp
22: 1198502476 IO-APIC-level
harry gaillac a écrit :
Hello,
Is it possible to add a frontend groupware with
asterisk in order to Provide send receive fax to mail,
sms to mail, voice messages .
Asterisk or openpbx could be the server of the unified
messagerie .
click to dial contact in address book ,...
[Shameless plug]
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I've forgotten my dial-peer config:
dial-peer voice 500 voip
description ext
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.17.10
dtmf-relay rtp-nte
no vad
192.168.17.10 is *, .1 is CME.
Post up your dial-peer 500 config as well. It is doing codec 0x2
(g.711Alaw) from the get go.
Also post relevant config for the phone from asterisk and dialplan entry
used.
-Greg
On Wed, 2005-11-09 at 17:08 +0100, Andrea Riela wrote:
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On Nov
I am running an Asterisk server (which has gone from 1.x to 1.2b2 at the
moment) that has 3 X100P cards and around 10 SIP phones in my office and I
have a problem when I want to redirect my desk phone to my cell phone.
I have a Polycom 600 phone on my desk (I have also tried this with
Just put codec g729(whatever version you need) in your dialpeer.
I do not see what the voice-class codec 1 is without that section.
-Greg
On Wed, 2005-11-09 at 17:17 +0100, Andrea Riela wrote:
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I've forgotten my dial-peer config:
dial-peer
take for example a phantom SIP/400b from a previos phone config, without
restarting * how can I purge only 400b?
testserver*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
400c/400c (Unspecified)D 0Unmonitored
Ive been playing with Asterisk for a few weeks and its
working great.
I have a question about getting multi-line receptionist phones
working.
I was thinking about getting one of these expansion ports:
That's a call to pstn
Callee and caller have 9729 but asterisk (astlinux and soekris) tell me that
there is no match and give me an error :(
Any idea?
Kind regards,
Olivier
9 headers, 11 lines
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 82.146.123.246:38098
Nevermind I found a note about Hint
which can be used for this purpose.
Bill
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs
Sent: Wednesday, November 09, 2005
11:52 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject:
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