SIP will get you no RTP, meaning it only works with SIP headers.
Asterisks CPU usage is mainly coming from RTP handling.
We glued something together that will work for RTP too, you can download
it from:
http://www.astertest.com/forum/viewtopic.php?t=4
As the moment it only seems to work for non
I recommend to deactivate the current jitter buffer and wait till a new
one is ready.
Joachim.
Stuart Elvish wrote:
Hi,
I was wondering if anyone else has a similar setup and can suggest
settings for the jitterbuffer:
I have a client with an ADSL connection at site A B with site A
being dedicated
Hello,
We had same problem in other african country. We could resolve it through using
IAX Bridge in Asterisk since it only uses one port of yoru choice.
For your solution, you need:
1-) Scan outgoing / incoming open ports by your ISP;
2-) If there remains many open ports, you may still run
Howard Lowndes wrote:
Apologies. I meant You do have a context called Fax...?
I don't. And it's working absolutely fine.
I have SpanDSP installed, faxdetect=both and the FaxReceive macro is
shamelessly lifted from http://www.voip-info.org/wiki-Asterisk+fax
--
Adrian Chapman
Director
Trivas Ltd
Michael Welter wrote:
Changing the order of things in extensions.conf around a smidge got it
all working nicely :-
[inbound-from-pstn]
include = default
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Playback(thank-you-for-calling-please-wait-a-moment)
exten = fax,1,Macro(faxreceive)
exten =
Hello all. I'm having an odd problem getting * and sipgate to work
together. From Sipgate support I have gotten this repsonse to my query:
=
Your Asterisk is registering incorrectly with our servers. It registers
like this: sip:[EMAIL PROTECTED]:5076
The s should be your SIP ID.
Hello
Everyone!
I just want
to make sure if such a mess could work for sip channel:
In sip.conf:
;
register =
some_md5_checksum@host
;
;
[host]
hostname=some_address
auth=md5
Greets
Tomek
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hi
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
roy
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Hello all,
iwould liketo configure TDM11B with
Asterisk, if any one have the configuration steps please provide me
it.
Thanks in advance
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No, is default for snom phones.
Sven
On Tuesday 08 February 2005 08:37, Altus Snyman wrote:
Did you try 00
That is what it is on the 220
On Tue, 2005-02-08 at 09:36, Paradise Dove wrote:
what is the password for Administrator in the softphone?
On Tue, 8 Feb 2005 08:01:07
Ok,
When I tried newer version of zaptel, libpri, asterisk, it didn't work. My
spans get down and RED. Then, I tried to go back to previous version. No
way.
Now, my E1 are up again, thanks for all your advice.
I don't know which one solve my problem, so, here's what I've done.
To get back to my
Can anybody help me in sipp for load testing on
asterisk?
How to use sipp with asterisk??
Thanks RegardsRitesh
Jalan
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To
That's all very well, but what do you do if you only have SIP extensions and
IAX trunk - no Zaptel card.
Will Fax detection still work at all?
Thanks
Mike
- Original Message -
From: Adrian Chapman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Last week I asked how to register the Microsoft Real_time Communications
Client with Asterisk. No replys came, however I managed to figure it out
myself. I thought I'd just post the solution for anyone else in the
future wanting to do the same.
Regards
Jerry
CString gXMLProfile = \
Thankyou so much Chris and Roger, I really appreciate your response
and suggestions
good luck :))
kind regards
Siju
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Are this currently working with CVS-HEAD?
I've got an X100P-clone, and I've patched the zaptel drivers.
But the Asterisk patches seems to be there.
But I can't make it receive Caller-ID!
Btw, by doing a cvs checkout asterisk, the HEAD-version will be
I saw at least three details in your config, which could result in problems.
Since I'm relative new to asterisk, take my tips with care.
register = XXX:[EMAIL PROTECTED]
should be register = XXX:[EMAIL PROTECTED]/XXX
fromuse=XXX
should be fromuser=XXX
auth=md5 ; I'm not shure
hi
how can I tune SIP jitter? is it possible today in asterisk?
ryo
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Thanks for your reply. I had tried the register line as you have it
before and it fails. When I call my UK number from sipgate it's just a
busy signal. If I remove my SIP-ID as I have in my current configs the
calls go through fine. Outgoing calls another matter. It seems to lock
up my
i am working on asterisk. i am using fedora core 2 on
my asterisk mechine. when i was working on stable
version my voicemailmenu was working well. i can
lissten to menu and send dtmf to control menu now i
have compiled CVS version of asterisk. now when i
configure my voicemail for any extension
hi
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
roy
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Any idea when that is likely to be ready?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of joachim
Sent: Tuesday, February 08, 2005 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] jitterbuffers - suggested
Hi,
is it and how is it possible to live
monitor (barge - in) a SIP to SIP call without
any Zap Interface? I am using asterisk
1.0.5 with chan_capi from Junghanns
and SIP clients. I was looking for chan_spy
application but it seems to be
no longer available.
Bye,
I am having problems transferring calls from one sip extension to
another - the extensions use various phones hardware/software.
From what I can tell I should just be able to press # and then dial an
extension to blind xfer a call right? How do I do attended xfer?
Either the phones (for this
Hello,
is it and how is it possible to live monitor (barge - in) a SIP to SIP call
without
any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns
and SIP clients. I was looking for chan_spy application but it seems to be
no longer available.
You can do something like
On Tue, 8 Feb 2005, Daniel Nyström wrote:
Are this currently working with CVS-HEAD?
I've got an X100P-clone, and I've patched the zaptel drivers.
But the Asterisk patches seems to be there.
But I can't make it receive Caller-ID!
The X100P is unsuited for use with the Swedish PSTN for several
On Tue, 8 Feb 2005, Roy Sigurd Karlsbakk wrote:
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams.
What asterisk version
I know we had a problem with one of the cvs
We couldn't use the transfer buttons,but # worked
What about the Dail(SIP/111,12,tT) in your extensions.conf
On Tue, 2005-02-08 at 13:50, Mark Benson wrote:
I am having problems transferring calls from one sip extension to
Yes, that would work - but I have no
Zap and therefor no meetme - or is there
a way to start meetme with SIP interfaces
only ?
[EMAIL PROTECTED] schrieb am
08.02.2005 08:53:06:
Hello,
is it and how is it possible to live monitor (barge - in) a SIP
to SIP call
without
any Zap
On 08/02/2005 19:23 [EMAIL PROTECTED] said the following:
and SIP clients. I was looking for chan_spy application but it seems to be
no longer available.
oddly, ChanSpy seems to be removed from mantis. any idea why this was done ?
--
Regards, /\_/\ All dogs go to
On Tue, 8 Feb 2005, Sven Lohmann wrote:
Yes, that would work - but I have no Zap and therefor no meetme - or is
there
a way to start meetme with SIP interfaces only ?
Use ztdummy or zaprtc. All that is needed is the zaptel timing. Another
option may be to use app_conference (use google).
I've asked the same question at least three times, but no-one has replied.
Surely the person who removed the bug must know ;)
Julian.
Dinesh Nair wrote:
On 08/02/2005 19:23 [EMAIL PROTECTED] said the following:
and SIP clients. I was looking for chan_spy application but it seems
to be
no longer
I am one of these unhappy people using
the wrong USB chip and building my own
kernel (RTC is activated) is no option
due to company policies.
[EMAIL PROTECTED] schrieb am
08.02.2005 13:11:44:
On Tue, 8 Feb 2005, Sven Lohmann wrote:
Yes, that would work - but I have no Zap and therefor no
Good day all
I have a asterisk installation,1.0.3, and spandsp.
I got asterisk working,I edited the make file myself.
Now when I receive a fax I only get half a page or nothing
any Ideas why
Please let me know
Altus
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On February 8, 2005 03:31 am, joachim wrote:
I recommend to deactivate the current jitter buffer and wait till a new
one is ready.
Any particular reason why? I am using the following jitter buffer settings
with Jan 2005 CVS HEAD without any issues, and it seems to work reasonably
well:
Would it be possible to use the programmable led+keys on the Snom phones
to signal that there is an incoming call that is ringing a call group or
pickup group?
We use this on our existing PBX if for example the accounting dept. is out
for lunch but nobody can hear their phones. This way you
On Tue, Feb 08, 2005 at 01:47:57PM +0100, Remco Barende wrote:
Would it be possible to use the programmable led+keys on the Snom phones
to signal that there is an incoming call that is ringing a call group or
pickup group?
We use this on our existing PBX if for example the accounting dept.
Robert P. McKenzie schrieb:
So far Sipgate has proved to be the most problematic provider I've tried
using.. and now they have my money with no refunds and I can't use them.
Sigh.
I never had any problems with (the german) sipgate so far.
My sip.conf (only the sipgate-parts):
register =
? What's wrong with the current jitterbuffer..
-Original Message-
From: joachim [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 08, 2005 2:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] jitterbuffers - suggested settings
I recommend
On Mon, Feb 07, 2005 at 11:08:51PM -0500, Jon Radon wrote:
Instead of hijacking the thread you could just look it up. (HINT: it's
a feature in cvs)
I'm using stable rather than CVS. I did look on voip-info and I searched the
mailing list archives. If there's another place I could've looked
On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote:
hi
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what studio lite means to you. Maybe hard figures would
be
hi,
i have the problem that i'm not able to set and receive the Service
Indication (SIN) from our E1-PRI and from our ericsson BP250.
The problem is, that the Bearer Capability (BC) together with the High
Level Compatibility (HLC) and Low Level Compatibility (LLC) forms the
Service Indicator
Mike Sander wrote:
That's all very well, but what do you do if you only have SIP extensions
and IAX trunk - no Zaptel card.
Will Fax detection still work at all?
Absolutely no idea. We're working with X100s, and haven't started
looking at Fax sending yet.
--
Adrian Chapman
Director
Trivas Ltd
Apparantly the new one will do things like interpolation so that if
packets are lost it will generate new ones to fill the gap. The current
jitterbuffer doesn't do that so you get silence on packet loss. There
are a bunch of other features too that I don't remember, but that was
the most
Nicolas Bougues wrote:
On Tue, Feb 08, 2005 at 12:26:39PM +0100, Roy Sigurd Karlsbakk wrote:
hi
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what studio lite means to you. Maybe
can anyone tell me how to add extension to extension
table
i think this is the main prblem. any one to guide me.
++-+-+--+--+-+
| id | context | exten | priority | app
| appdata |
Hi to all,
I and using asterisk with following setup.
1. TDM400p card with four FXS modules,
so there are four analog phone lines on four zap channels,
My setup is working fine.
And version is like such
Asterisk CVS-v1-0-11/27/04-20:48:45
I want your guidance for the following issue.
with help
Looking for some advanced thoughts relative to exten number assignments.
We're in the planning stage for rolling out asterisk at multiple small
US telco/isp operations. Their typical voip customer has had their
pstn line for a long time and wants to keep the pstn line and number,
but add
Hello,
I have * config with one quadBRI card for PSTN ISDN lines and one
TDM400P for analogue faxes and modems. I could never get a fax or modem
to work reliably over ISDN. With bristuff 0.2.0 RC3a (* 1.0.3) modem
connection would drop after a few secs, and fax would never get through
if it had
Thanks RegardsRitesh JalanSenior Engineer - Test
AuditNet4india Ltd.703, Bhikaji Cama Bhawan11, Bhikaji Cama
PlaceNew Delhi 110066Tel: 91 (011) (26160129 - 131) (Extn
131)URL: http://www.net4india.com
- Original Message -
From: Ritesh
Jalan
To: asterisk-users@lists.digium.com
On Tue, 8 Feb 2005 10:47:40 +0100, Roy Sigurd Karlsbakk
[EMAIL PROTECTED] wrote:
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
v1-0 is the tag used for the latest changes to the stable branch.
Releases are still your best bet, but if you are monitoring
On Tue, 08 Feb 2005 10:16:46 +0200, joachim [EMAIL PROTECTED] wrote:
SIPP works for asterisk testing too, but you need the correct
commandline. What did you use ?
Perhaps you can just give us the _correct_ command line for those of
us who are unknowing? :)
Thanks,
Leif Madsen.
Yeah, I also noticed that * lacks the ability to forward calls based on
the type of call, but I have no idea whether this issue has any priority
with the development team. It is probably better to ask this question on
Asterisk-Dev mailing list.
Frank Sautter wrote:
hi,
i have the problem
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Is the channel physically being hung up before the * tone is heard?
Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't
support Kewlstart-style disconnect notification.
The sequence I hear on
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what studio lite means to you. Maybe hard figures would
be more precise.
G.722 might be interesting : 64 kbps, 7 kHz. It's not free.
Otherwise,
Hi
Try going into vi /etc/profile insert the lines in brackets.
USER=`id -un`
LOGNAME=$USER
MAIL=/var/spool/mail/$USER
MONITOR_EXEC=/usr/bin/soxmix
VPB_TONE=BUSY,P,400,100,500(insert the following line)
Try ACT P104
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philipp von
Klitzing
Sent: Monday, February 07, 2005 1:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] iax hardphone
Hi!
Is there such a beast
I have this in my extensions.conf
exten = 08700xx,1,Dial(SIP/test1SIP/test2SIP/test3,30,t)
To ring a group of internal extensions for any call coming in on that
number
And
exten = 100,1,Dial(SIP/test1,20,Trt)
exten = 100,2,Voicemail(u100)
exten = 100,3,Hangup()
exten =
On Tue, 8 Feb 2005, Frank Sautter wrote:
is there any usable documentation on the HLC or LLC octets (bytes)?
i searched etsi and was overwhelmed with the searchresults (1531). what
i need to modify libpri would be a table of possible values and where to
find the HLC and LLC fields in the
On February 8, 2005 08:44 am, David Brodbeck wrote:
The sequence I hear on the extension, when I plug in an analog phone, is
the click of the phone at the other end being hung up, followed immediately
by a * touchtone. Then there's silence until I hang up.
Hmm... I bet it has everything to
On February 8, 2005 08:14 am, Mazhar Hussain wrote:
I want your guidance for the following issue.
with help of agi scripts i am able to insert caller id number in
database of mysql now i want to pop it up via html or php page but can
any one of you let me know how can i use php scripts in agi
We have a client that wants to bond 2 DSL circuits instead of getting
a T-1 (or similar) at their office to run their VoIP traffic on. We
came across this Multihomed Gateway (MH200):
http://www.cyberpathinc.com/mh200/details.htm
Does anybody think this would work if installed at the client
I put dtmfmode=rfc2388 into the sip.conf definitions for each sip client
and now asterisk is recognising the # key press - guess it wasn't
hearing the dtmf tones...
Now blind xfer works - how do I do attended xfer? I have read posts
about it being in the CVS version - I am running the 1.0.3
-- Forwarded message --
From: Carlos Gabriel Drach [EMAIL PROTECTED]
Date: Tue, 8 Feb 2005 11:20:01 -0300
Subject: Re: [Asterisk-Users] Record() cut off after 40 sec
To: Steven Critchfield [EMAIL PROTECTED]
On Mon, 07 Feb 2005 15:35:46 -0600, Steven Critchfield
[EMAIL PROTECTED]
I have a question regarding to OS platform.
As I see on Wiki -s homepage there are many type of linux version.And in some of them there are reported errors (regarding to asterisk ) for exemole in rad hat .
Can you tell me what is the best linux paltform ,( version ), which is supported by digiroom
at first it was not answering (there was complete
silence after 200 Ok and ACK). i dont know what was
the reason. but now it is answering me(asking for
mailbox then password). but the problem that is is not
authenticating me to check mailbox i have defined
mailbox and 1234 password (it is
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
On February 8, 2005 08:44 am, David Brodbeck wrote:
The sequence I hear on the extension, when I plug in an
analog phone, is
the click of the phone at the other end being hung up,
followed immediately
by a *
-Original Message-
Good day all.I get the warning message on my system,this is for a snom
220,it repeats this message a few times,please help Feb 8 09:29:26
WARNING[1093445952]: chan_sip.c:683 retrans_pkt: Maximum retries
exceeded on call [EMAIL PROTECTED] for seqno 105
(Non-critical
I know Q931 cards are supported, does anybody know
how to
go about supporting DASS II ?
Thanks
Stephen
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Hey gang,
I'm trying to work out all possible scenarios using SER Asterisk in our
upcomming deployment. The example scenario is 50 different customers, all
with different numbers of SIP UAs. All UAs would register with SER; This
will help keep any inter-office conversations off our bandwidth
On February 8, 2005 09:28 am, David Brodbeck wrote:
What puzzles me is it works fine if I dial *, but if I hang up instead and
the PBX sends *, Asterisk doesn't seem to get it.
With you listening in on the same physical 2-wire that the PBX uses and you
send *, does Asterisk see it? If you're
Which linux is prefereable ? for asterisk ?
As long as you know how to rebuild your kernel, how to install modules, and
how to manage basic unix security, the best Linux for Asterisk is the one
you're most comfortable with.
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-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
On February 8, 2005 09:28 am, David Brodbeck wrote:
What puzzles me is it works fine if I dial *, but if I hang
up instead and
the PBX sends *, Asterisk doesn't seem to get it.
With you listening in on the
Does the order in which you allow codecs matter? cuz i've found that
somethings work better if you allow them in a particular order.
Alot of warnings and errors can also be eliminated.
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On Tue, Feb 08, 2005 at 02:58:01PM +0100, Roy Sigurd Karlsbakk wrote:
are there any codecs around that allows high quality as in studio
lite? it may consume high bandwidth, and hopefully allow some packet
loss.
I'm not sure what studio lite means to you. Maybe hard figures would
be more
On Tue, 2005-02-08 at 06:27 -0600, Rich Adamson wrote:
Looking for some advanced thoughts relative to exten number assignments.
We're in the planning stage for rolling out asterisk at multiple small
US telco/isp operations. Their typical voip customer has had their
pstn line for a long
Okay, the problem appears to be that I'm tone deaf. ;)
I finally thought to turn on debugging on the channel. The PBX is sending
D, not *. The programmer of the previous voice mail system (whose
configuration I was cribbing from) seems to have made the same mistake.
On February 8, 2005 09:48 am, David Brodbeck wrote:
With you listening in on the same physical 2-wire that the
PBX uses and you
send *, does Asterisk see it? If you're on a call from the
PBX to Asterisk and dial * from the PBX phone, does * see it?
Yes, in both cases.
How short is the
Pedro,
My understanding is that this will not allow for any balancing on any
connections once they are established. Any connection on the first line
that is already established will continue to stay on that line/ip
address until the connection is dropped and a new one is established.
It would
I got the called-name lookup going using php: http://muware.com/asterisk
If you want to pop up additional details, you'll need a client
application to notify a computer near the extension -- this is possible,
but will require quite a bit more work.
-Original Message-
From: Mazhar
Checkout http://www.voip-info.org/wiki-NVBackgroundDetect
I haven't had a chance to try it yet, but supposedly it works on SIP,
ZAP, and IAX.
On Tue, 8 Feb 2005 21:26:28 +1100, Mike Sander
[EMAIL PROTECTED] wrote:
That's all very well, but what do you do if you only have SIP extensions and
Hi guys,
do you know if it's possible to handle more than 1 call per card
with astcc ?
Thank you.
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-Original Message-
From: David Brodbeck [mailto:[EMAIL PROTECTED]
Okay, the problem appears to be that I'm tone deaf. ;)
I finally thought to turn on debugging on the channel. The
PBX is sending
D, not *. The programmer of the previous voice mail system (whose
configuration I
We have a client that wants to bond 2 DSL circuits instead of getting
a T-1 (or similar) at their office to run their VoIP traffic on. We
came across this Multihomed Gateway (MH200):
http://www.cyberpathinc.com/mh200/details.htm
Does anybody think this would work if installed at the client
It's probably too late for me to say I don't want to sound like a
jerk. :-P It was late and I get frustrated when people don't use
available resources. I apologize. Anyways, a quick search of
google..
http://www.google.com/search?q=asterisk%20n%20priority
pulls up
.
Specifically, X is not a digit, you must either use for no
interuptions permitted or use 0123456789 for all digits available to
interupt.
I also 'discovered' that you cannot send a sequence of commands to
asterisk without
reading the results between each command submission. Similar to the
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call
is
mapped to [EMAIL
Noah,
Thanks for your input on this. I am not sure if it handles incomng
connections or not - will have to check. I don't think it will work
either - worth a shot to ask though.
Thanks!
- Pedro
On Tue, 8 Feb 2005 10:26:48 -0500, Noah Miller [EMAIL PROTECTED] wrote:
We have a client that
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of thieumS
Sent: Tuesday, February 08, 2005 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ASTCC simultenous calls per card
Hi guys,
do you know if it's
I had problems as well. It was do to my sip.conf and extension.conf
Here are my conf files.
sip.conf
[general]
port=5060 ; Port to bind to
bindaddr=0.0.0.0 ; Address to bind SIP channel to
context=default ; Default context for incoming calls
Sounds like maybe you don't have either Postgres installed or PHP
confirgured to use it.
If you use RPMs, check for something in the php-pgsql family (%yum
install php-pgsql)
As a warning, you will also need to enable PHP globals in your php config.
Hope that helps,
J
On Tue, 8 Feb 2005
hi
I've been trying to fax digium this agreement for a month or so now
Any chance they can fix their fax?
roy
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Steve Blair writes
I can redirect and relay calls to numerous destinations via
SER but because the Octel needs an SMDI interface for mailbox
identification I am stuck, none of the solutions thus far support
SMDI-SIP munging.
I just started thinking about the possibility of using Asterisk
with a
I have just setup music on hold by downloading and installing mpg123 r
Now I have music on hold but it sounds terrible - clipping, buzzing,
digital distortion, and its too loud (which probably isn't helping) and
I'm just running it thru the 'default' line in music onhold.conf line
default =
I am looking for termination of numbers in the 479 area code. I would like
to either port them through my * box or direct SIP connection from the
customer. I am in need of over 100 DID's. Anyone know of anyone that has
this service besides Vonage or Packet8?
---
Kelly D Griffin
Network
how can I tune SIP jitter? is it possible today in asterisk?
I assume you are asking for how to alleviate the effects of jitter on
the
RTP audio streams initated by SIP? Asterisk currently only has a jitter
buffer for IAX, not for RTP streams. There are pland for the next
generation jitter buffer
is the v1-0 CVS branch supposed to be stable as in STABLE, or should
one use releases?
v1-0 is the tag used for the latest changes to the stable branch.
Releases are still your best bet, but if you are monitoring the CVS
mailing list for commits to v1-0 stable, then you may see a patch go
in that
On Tue, 8 Feb 2005 11:56:18 +0200, Yousri Farouk [EMAIL PROTECTED] wrote:
Hello all,
i would like to configure TDM11B with Asterisk, if any one have the
configuration steps please provide me it.
Thanks in advance
Have you tried looking at Digium's site??
Yes, We offer that stuff we can get numbers in most U.S area's
Contact us
800-508-1251
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kelly
Griffin
Sent: Tuesday, February 08, 2005 11:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
The stable tree from cvs includes any patches since release that was
also commited for the v1-0 tag since some issues were found after the
release but not major enough for a new tar ball release.
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