Hi
thanks for your answer,
for dtmfmode, all sip account have dtmfmode=rfc2833 ;=)
that's don't change
bye
Gordon Henderson a écrit :
On Fri, 9 Feb 2007, Noc Phibee wrote:
Hi
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don't work's (*8EXTEN)
this is my kernel:::
*
:/usr/src/zaptel-1.4# uname -r
2.4.27-3-386
also when i type: make clear te rebuild i got errors
**
pbx:/usr/src/zaptel-1.4# make clean
make[1]: Entering directory `/usr/src/zaptel-1.4/menuselect'
rm -f menuselect *.o
make[1]: Leaving
On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java
On 7 Feb 2007, at 16:33, Jim Duda wrote:
Tim,
What sort of 'poor' quality are we talking about - when folks
complain what words do they use?
On the other end, folks complain that the voice drops out. Words
are lost. It's very frustrating to communicate.
Which codec(s) are you using?
From: younss azzayani [EMAIL PROTECTED]
Date: Fri, 9 Feb 2007 08:51:14 +
this is my kernel:::
*
:/usr/src/zaptel-1.4# uname -r
2.4.27-3-386
also when i type: make clear te rebuild i got errors
**
...
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]:
On Fri, Feb 09, 2007 at 08:51:14AM +, younss azzayani wrote:
this is my kernel:::
*
:/usr/src/zaptel-1.4# uname -r
2.4.27-3-386
also when i type: make clear te rebuild i got errors
**
pbx:/usr/src/zaptel-1.4# make clean
make[1]: Entering directory
recently i've upgraded asterisk from 1.2.4 to 1.4
All works fine but i'me experencing some instability on misdn channels.
In the last week i've experienced twice some problems with misdn (I am
using mISDN-1_0_4)
dmesg output:
mISDN_rdata: rport queue overflow 256/256 [addr:52020501
On Fri, Feb 09, 2007 at 08:51:14AM +, younss azzayani wrote:
this is my kernel:::
*
:/usr/src/zaptel-1.4# uname -r
2.4.27-3-386
also when i type: make clear te rebuild i got errors
**
pbx:/usr/src/zaptel-1.4# make clean
[snip]
make[1]: Leaving
On Fri, Feb 09, 2007 at 01:34:58AM -0800, Yuan LIU wrote:
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No such file or directory. Stop.
Kernel 2.4 header will not help you. As mentioned, you need full kernel
source with 2.4.
My experince
On 7 Feb 2007, at 20:04, Rob Schall wrote:
Here's an interesting issue we're facing...
We would like users to be able to use softphones from home/work and to
use their same extensions they do at work.
The first step of getting the phones to log in as their same
extensions
as work is easy
ci$co phones are definitively not good choice if you would like to use
with anything other than callmanager as signaling server (especially
true for new models 7911/41/61/70)
Michelle Dupuis wrote:
We used Aastra's for a good while, but gave up on them (and switched
to Cisco). Aastra's
On 9 Feb 2007, at 04:31, JR Richardson wrote:
Hi All,
I'm very interested in real world experience of double digit number of
users sustaining good quality audio in a single meetme conference.
Personally, I have seen 23 users in one conf room, all coming in SIP,
ULAW. Server is 3.2GHz proc,
ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729
On Thursday 08 February 2007 19:00, Vicky wrote:
config problem . what pbx does ip_pb2 runs ? ( is it asterisk ? ) in peer
definition try allowing all codecs .. ( gsm , ulaw,alaw,ilbc )
On 08/02/07, Florea Igor [EMAIL PROTECTED]
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
CLI sip show registry
HostUsername Refresh State
iinettrunk:5060 [EMAIL PROTECTED] 3584 Request Sent
sip.pennytel.com:5060 N 280 Registered
Yes, I have
why don't think to sugarcrm, it has an asterisk package, so you
benefit of asterisk sugarcrm at the same time
Younss AZZAYANI
Junior IT Manager
Robinson Network
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asterisk-users mailing list
To
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
You then ask the telco to include Advice of Charge (AOC) in your ISDN setup.
The AOC then is included somewhere in the Asterisk CDR, but I don't have
direct experience of this. You can then get appropriate software to issue
bills to
Because it just works.
On 2/8/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote:
This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO ports and the host
PBX deosn't give CPC.
Make sure that your NIC and your X100 are not using the same interrupt.
If they are, they will be competing for interrupts and they both will loose.
--
--
Steven
http://www.glimasoutheast.org
Yuan LIU [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
I'm greatly surprised when
The network terminator installed by the Telco in Romania works the same
way: it has two analog outputs and two digital (S0) outputs. I've also
got a full TDM400 card with 3 FXS and one FXO, but I gave them up gladly
for a proper ISDN card (I'm using a Diva Eicon Server) - and I don't do
On Fri, 2007-02-09 at 09:21 +, Tim Panton wrote:
On 8 Feb 2007, at 12:33, Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 09:41:30AM +, Tim Panton wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and
On Fri, Feb 09, 2007 at 01:30:02PM +1100, Klaverstyn, David C wrote:
Yes, I have also since put that in and I get the error:
Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling
And if I put in rxwink I get this error:
Feb 8 19:24:30 WARNING[4022]:
On Thu, 2007-02-08 at 16:48 -0800, Yuan LIU wrote:
From: Stefan Wintermeyer [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 21:56:11 +0100
Am 08.02.2007 um 18:39 schrieb Forrest Beck:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and
On Fri, Feb 09, 2007 at 07:22:27AM -0500, C F wrote:
Because it just works.
On 2/8/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Feb 08, 2007 at 01:38:30PM -0500, C F wrote:
This device can solve many problems, and is a must for most
applications where asterisk is connected using FXO
1. We just dial the extension directly and have speed dials setup for
the first 6 parked positions. We don't use *8 at all.
2. Change the config on the phones under Account to Send DTMF via RTP
(RFC2833)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to handle
trailing # when it comes along. For example, this simplistic example
ignores trailing #'s:
Hello John,
I'm not sure - but when tou try to define a context for testq queue with:
context=testing
it is useless. From what I know you could not have such an option inside
a queue.
Did you find any documentation specifying a context for a queue?
Best regards,
## nini @
Greetings List,
I am a newbie and first time mailer so bear with me. I have 2
questions.
1. recording: I have an Meridian Option 11 hooked to my Asterisk box via a
PRI with QSIG signalling. I have set up an access code of 8 in the option
11 to access the PRi to the Asterisk Box. Is there a
Anyone got any experiences of good quality VoIP conferencing phones?
I've used Polycom analogue units in the past, and I see that they have a
SIP version (the IP4000) - but it is better/worse/as good as an analogue
version?
(ie. would I be better off with an analogue version into a TDM card
I upgraded my Asterisk system to version 1.2.14 to check if the sound
quality issues I was having with Chanspy in 1.2.7 remained. I'm still
getting them, and I'm honestly out of ideas except from RTFS.
The called party sounds normally fine, but it's impossible to hear the
caller. Sometimes, when
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Gordon Henderson
Sent: Friday, February 09, 2007 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Conferencing Phones ...
Anyone got any experiences
Hey,
anyone know if it's possible to receive faxes through a Junghanns bristuff
quadbri card?
In germany, currently I have faxes coming in on DID line into QuadBRI and
then passing to Digium TDM400 (analog) and into faxmachine. But the
reliability of TDM card is spotty, so I want to maybe just
Leo Ann Boon wrote:
Klaverstyn, David C wrote:
Yes, I have also since put that in and I get the error:
Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
signalling
And if I put in rxwink I get this error:
Feb 8 19:24:30 WARNING[4022]: chan_zap.c:10874 setup_zap: Ignoring
[EMAIL PROTECTED] schrieb am 09.02.2007 16:12:57:
Hey,
anyone know if it's possible to receive faxes through a Junghanns
bristuff
quadbri card?
In germany, currently I have faxes coming in on DID line into QuadBRI
and
then passing to Digium TDM400 (analog) and into faxmachine. But the
Spam detection software, running on the system placebo, has
identified this incoming email as possible spam. The original message
has been attached to this so you can view it (if it isn't spam) or label
similar future email. If you have any questions, see
[EMAIL PROTECTED] for details.
Content
Hi Chris,
Am 09.02.2007 um 16:12 schrieb Chris Earle:
anyone know if it's possible to receive faxes through a Junghanns
bristuff
quadbri card?
In germany
So you can read a German documentation?
, currently I have faxes coming in on DID line into QuadBRI and
then passing to Digium TDM400
Hi all, excuse this doll question, but can´t remember or find where
I used to check this list on the web, email is becoming unmanageable
along with my regular mail.
can anyone provide me withe the link to check the list´s threads under
web?:-[
Hello,
I've installed two Digium TDM2400 cards on my server. One has 24FXS
and the other has 16 FXS and 4 FXO. They are both connected to power.
Unfortunately some of the FXS module fail to initialize and I find
following messages in the logs (the rest of the FXS modules work
well). Could
There is a link provided at the bottom of almost every message that will
get you close to where you want to be
Give it a try.
John Novack
MF wrote:
Hi all, excuse this doll question, but can?t remember or find
where I used to check this list on the web, email is becoming
unmanageable
you don't have to connect the power connecter to TDM if you are using
FXO, it's used with FXS to generate a signal to phones
2007/2/9, MBIT Technologies [EMAIL PROTECTED]:
Hi David
Also make sure the power connector is also connected to the board.
Regards
Mark Brooker
T: 02 4959 8670
M:
Hi. I'm currently setting up a particular conference: 3 members (a,b,c),
a can speak with b and c, b and c can speak only with a and not between
them.
I found my possible solution with paging/intercom using option d
(full-duplex), but I need to make ringing the phone in intercom.
Now I set
check your sip.conf and make sure it has allow=ulaw and allow=alaw line (
you can even remove gsm to test it it works fine or not )
On 09/02/07, Florea Igor [EMAIL PROTECTED] wrote:
ip_pbx2 is not asterisk, it knowk only PCMU,PCMA,g723,g729
On Thursday 08 February 2007 19:00, Vicky wrote:
Hi,
I download the last svn and I also look around but I cannot find the source,
I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919
any one can help me out.
thx
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The point is to use more than one port, I think the only way is to use the
redirect from iptables
On 2/6/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Ciao,
just change port value in sip.conf.
Giorgio
Il Neofita wrote:
Hi,
I was wondering if it is possible to set asterisk in order to
We use the Polycom soundstation 2W plugged into an iaxy...works very
well...
Gregory P. Scasny
Golden Technologies, Inc.
http://www.golden-tech.com
[EMAIL PROTECTED]
219-462-7200 - Ph.
574-233-1300 - Ph.
(866) 806-7127 - Toll Free
219-462-7257 - Fax.
-Original Message-
From: [EMAIL
What are the specific dependencies that Asterisk has on databases? Some
hi-perf data is stored in BDB, CDRs are in a relational DB like MySQL.
Is there a list of specific dependencies by specific modules on specific
tables? A complete list, so switching from the default DB can drop the
old
This isn't included in the trunk for the moment.
You have to use the patch to get chan_cellphone.
Regards,
Tristan Mahé
Il Neofita a écrit :
Hi,
I download the last svn and I also look around but I cannot find the
source, I only found the patch
The Asterisk development team is pleased to announce the release of
Zaptel 1.2.13.
This release contains a large number of bug fixes, an important
performance improvement for most Digium cards, and support for new
Digium hardware and some significant improvements in the XPP driver for
Xorcom's
The Asterisk development team is pleased to announce the release of
Asterisk 1.2.15.
This release contains a large number of bug fixes, and some significant
improvements:
* Support for Zaptel-based transcoder hardware, initially the Digium
TC400B 92/96 channel transcoder.
* Handling of
From: Steve Murphy [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 07:11:50 -0700
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to
handle
trailing # when it
I start the patch and automatically created the file. But now on the menu I
cannot select chan_cellphone
I launched ./bootstrap.sh
and after ./configure
in my /usr/include/bluetooth I have the header
but I cannot select the option
any idea?
On 2/9/07, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
I
Ok that worked for normal transfers. Now here is another situation. When we try
to transfer a call directly to voicemail it plays the voicemail message but we
can't transfer the call. The only way I could get it to work was to do a
conference and then drop out of that conference.
My dial plan
On Friday 09 February 2007 11:50, [EMAIL PROTECTED]
wrote:
Anyone got any experiences of good quality VoIP conferencing phones?
I've used Polycom analogue units in the past, and I see that they have a
SIP version (the IP4000) - but it is better/worse/as good as an analogue
version?
(ie.
Yuan LIU wrote:
From: Steve Murphy [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 07:11:50 -0700
On Thu, 2007-02-08 at 21:48 -0800, Yuan LIU wrote:
From: Yuan LIU [EMAIL PROTECTED]
Date: Thu, 08 Feb 2007 21:28:03 -0800
Not necessarily. You only have to program your existing context to
handle
I also encountered the problem of port 5060 being blocked by some user's isp
and redirected port 5098 to 5060 but still asterisk wasnt able to detect
hangup properly and had load of voice problems ( lot of nat involved and
softphones were being used ) so i made asterisk listen on 5098 and
1000 Hz is recommended if you use lot of meetme channels ( and maybe iax
trunking ? ) without a hardware timer .
On 08/02/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 7 Feb 2007, Mark Coccimiglio wrote:
Ok here is a real geek question,
I building my own linux kernel for my
I have a Dish 301 receiver that will not display CallerID when connected
to FXS module on TDM400. Uniden phone connected to the same FXS module
does display CallerID.
When Dish 301 receiver is connected to IAXy CallerID is displayed
properly.
Any suggestions on getting the CallerID to display
I have a telco providing DTMF inband, they say they can't provide it any
other way. This is creating headaches for me.
What is the common method for SIP DTMF? Kpml, or 2833 or inband?
My handsets don't support inband so I'm tying up some expensive
resources to convert the inband DTMF to
JABBER: gtalk_account OUTGOING: ?xml version='1.0'?stream:stream
xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client' to='
gmail.com' version='1.0'
localhost*CLI jabber show tes
JABBER: gtalk_account INCOMING: ?xml version=1.0
encoding=UTF-8?stream:stream from=gmail.com
I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure
what it's supposed to do, but I wouldn't expect it to continue processing
the dial plan.
Any pointers? Documentation locations that address hanging up would greatly
appreciated!
TIA!!
Thanks,
David Ruggles
CCNA MCSE
On Fri, 2007-02-09 at 15:31 -0500, David Ruggles wrote:
I've got an X100P that doesn't seem to be detecting hang-ups. I'm not sure
what it's supposed to do, but I wouldn't expect it to continue processing
the dial plan.
Any pointers? Documentation locations that address hanging up would
Hi guys
i have a problem with an isdn (E1) pri works fine but once or twice a
week i got ring requested on channel X then every channel get blocked
so i should restart the pbx to fix it, i try not using cdr mysql,
several linux distros and every 1.2.x asterisk version, even i try to
ask
Thanks for the conf file, but it didn't make any difference. If I hang-up
during a record it will hang the channel until I stop Asterisk.
If I hang-up during playback I get the following:
[Feb 9 16:22:06] WARNING[4005]: pbx.c:2449 __ast_pbx_run: Invalid extension
'D', but no rule 'i' in context
I've been doing some googling and I found references to using debug=1 with
wctdm to see what's actually going on. It says this will be printed to the
console. I'm running my * box headless in another room and sshing in to the
box. I can't find where the debug out (if there is any) is going. Can
On 10:12, Fri 09 Feb 07, Chris Earle wrote:
Hey,
anyone know if it's possible to receive faxes through a Junghanns bristuff
quadbri card?
In germany, currently I have faxes coming in on DID line into QuadBRI and
then passing to Digium TDM400 (analog) and into faxmachine. But the
Ciao Neofita.
I'm trying my GTalk account and I'm still having the same problem.
I've installed the gnuTLS-developer rpms and rebuilt and re-installed the
complete Asterisk package but without success.
I'm working with OpenSuse 10.2.
This is my debug info that's quite similar to what you've
Awww... This is when I feel stupid, and for the sake of others... I will
expose my shame:
Be sure you run `autoconf` after applying the patch (and making the
required changes to configure.ac)
Since it's altering configure.ac afterall, and not configure; then
of course run configure and etc.
I
From: David Ruggles [EMAIL PROTECTED]
Date: Fri, 9 Feb 2007 16:43:41 -0500
I've been doing some googling and I found references to using debug=1 with
wctdm to see what's actually going on. It says this will be printed to the
console. I'm running my * box headless in another room and sshing in to
By your post I can conclude that the console wctdm debugs to is the asterisk
console. In that case I'm not getting anything from wctdm. I'm not using the
safe_asterisk script I'm running asterisk -cvvv from the command line.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer
From: David Ruggles [EMAIL PROTECTED]
Date: Fri, 9 Feb 2007 16:23:18 -0500
Thanks for the conf file, but it didn't make any difference. If I hang-up
during a record it will hang the channel until I stop Asterisk.
If I hang-up during playback I get the following:
[Feb 9 16:22:06] WARNING[4005]:
Ioan Indreias wrote:
Hello John,
I'm not sure - but when tou try to define a context for testq queue with:
context=testing
it is useless. From what I know you could not have such an option
inside a queue.
Did you find any documentation specifying a context for a queue?
Best regards,
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk Box
Incoming Line: Can I have in the dial Plan
[incoming]
exten =
Ken Williams wrote:
i have two problems with my Grandstream GXP2000 :
1- When i wan pickup a call, that's don't work's (*8EXTEN)
and when i test whit Softphone, i have a error too, he say me
[EMAIL PROTECTED] not found ..
in features.conf, i have:
*8 doesn't take an
On Fri, 2007-02-09 at 18:35 -0500, Barry Fawthrop wrote:
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk Box
From: Barry Fawthrop [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 18:35:43 -0500
Hi All
Curious will this work
Std. PSTN line ---x-- X100p
|
-- Fax Machine
Using a standard home phone pstn line with a splitter connecting a fax
machine and X100 Asterisk
Hi
I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.
The problem happens:
- With both software and hardware phones.
- With
I have found a site that list the following (no date in the post, so
it may be old):
since all transcoding and calls still go through one core in asterisk,
it doesn't make sense to buy a multi-core or hyperthreaded system that
will only slow you down
Does that still applies in asterisk
I recently read about the following new technologies from Digium. Has
anyone tried the new HPEC or knows when it will be available?
TDM800P and HPEC
The TDM800P is an 8-port analog telephony interface card, so it fills the
gap between Digium's 4-port and 24-port cards. Analog phones and POTS
On Fri, Feb 09, 2007 at 02:45:17PM -0800, Yuan LIU wrote:
From: David Ruggles [EMAIL PROTECTED]
Date: Fri, 9 Feb 2007 16:43:41 -0500
I've been doing some googling and I found references to using debug=1 with
wctdm to see what's actually going on. It says this will be printed to the
console.
Hi all!
First off all, sorry for my bad english.
I have a setup where some of the users have several extensions(work,
home, mobile etc). Therefore i have made a ring group for each of the
users with more than one extension. The ring group is set up to use
ring all.
What i want is that no mather
i saw the same problem and here is a thread where i mentioned how i fixed
it..
http://lists.digium.com/pipermail/asterisk-users/2006-November/171783.html
look for my previous mails in this thread sometime september-november 2006 .
btw, i can't get asterisk to work with google talk yet.
Thanks Guys
I already have the fax machine a brother all-in-one Printer, scanner, fax.
I realize the s,3, answers the line
But How can I get s,2, to detect if it is a fax and take it from there
without answering?
Or can someone explain what make an incoming goto exten = s,..
Erick Perez wrote:
I have found a site that list the following (no date in the post, so
it may be old):
since all transcoding and calls still go through one core in asterisk,
it doesn't make sense to buy a multi-core or hyperthreaded system that
will only slow you down
Does that still applies
From: Barry Fawthrop [EMAIL PROTECTED]
Date: Fri, 09 Feb 2007 21:49:17 -0500
Thanks Guys
I already have the fax machine a brother all-in-one Printer, scanner, fax.
I realize the s,3, answers the line
But How can I get s,2, to detect if it is a fax and take it from there
without answering?
It
Hi Ango -
Does any can give me some example to setup call parking and call
transfer of a call? In my understanding, call parking and call transfer should
be like
something below. Am I right?
Call parking:
caller A - callee B
callee B park her call
callee B get back her call in another phone
I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.
Any suggestions?
I have questions:
1) what version of 1.2?
2) Anything come
Noah,
Thanks for you reply. I have a problem in call parking as following.
scenario 1
1.Caller A - callee B
2.Callee B answered
3.callee B dial # to park the call and hear transfer
4.callee B dial 700 to park the call
5.callee B hang up and caller A hear 701
Why caller A hear the call parked
Noah Miller wrote:
I am using Asterisk 1.2 and for the life of me, I am unable to transfer
outbound calls (eg calls I initiate from sip extensions). When I press
#, nothing happens. Inbound calls transfer fine, but only once per call.
Any suggestions?
I have questions:
1) what version of
I have a problem with asterisk-1.2.13, where it retries SIP INVITEs
too quickly. It happens when qualify is on, and the server it tries to
reach is only 1ms away according to qualify.
The time between the first SIP INVITE and the 7th (last) is then only
64ms, and that can be too short for the
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