i've seen that opening a socket on the asterisk server i can originate
a call from one extension to another in a specific context.
Is it possible to transfer an existing call from the extension ...
SIP/xxx to another extension in a specific context?
thanks
On 03/17/06 17:13 Matt Riddell [NZ] said the following:
Is there anyway, to delete a message, but still have some sort of
incremental counter for the message id?
Not that I am aware of. At least unless you script something yourself.
alternatively, the OP can amend the email sent out with
On Tuesday, March 21, 2006 9:36 PM Douglas Garstang wrote:
I tried that earlier today... found it somewhere online... This is
what I get...
[EMAIL PROTECTED] mp3]# sox -V fpm-calm-river.mp3 -t au -r 8000 -U -b
-c 1 fpm-calm-river.ulaw resample -ql sox: resample opts: Kaiser
window, cutoff
Hi group. I have huge problem. My pickup exten #8 isn't working.
This is what I have configured.
pbx*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 #8
In sip.conf I have
callgroup=2
pickupgroup=2
For
Hi
ok then just add the same in to sip.conf
and same config made change in Extension.conf for outboud routing
ram
On 3/22/06, Mohammad Salaque [EMAIL PROTECTED] wrote:
Dear Ram .u miss something . as i told u . my provider didn't give me anyusername /passwd.
they just give me IP address .as
On 03/22/06 16:34 Tomislav Parèina said the following:
The person that is trying to pick up exten gets busy signal (Cisco 7905)
or error message Number does not exist. Call rejected: 404 Not Found
(SJ phone - softphone).
I'm running * 1.2.5 and phones that are trying to pick extension are
Hello,
I'm sure you can use the Asterisk as an IP PBX.
Good luck
Madhawa
Matt wrote:
Hi list,
Does anyone know the legalities of connecting an Asterisk box to the
PSTN in Manila or where I can find this info out? I know it is
illegal in some countries.
thanks
-Matt
OK, Asterisk Newbie
I've read TFOT and the Asterisk handbook and lurked, but my skills are a bit
poor so perhaps someone could post a dialplan fragment to help me
Brief details
[EMAIL PROTECTED] 2.6 installed on a miniITX system
Digium 400 card with 3 FXO modules
3FXS interfaces by Iaxy
On 3/6/06, Colin Anderson [EMAIL PROTECTED] wrote:
I was always puzzled by posts to the list about people having problems
getting hints to work on a Snom, since I always seem to have no problem
making it work. That is, until today when I tried to get a sidecar to work.
All I could do was get a
On 3/21/06, Nabeel Jafferali [EMAIL PROTECTED] wrote:
try SetCallerId or set callerid=name (xxx)xxx- in sip.conf or
iax.conf (depending on what you are using)
I am not using SIP or IAX2 clients. As mentioned in the original email, this
is from PRI to PRI.
I could use SetCallerID, but
Hi,
I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-)
Any pointers would be appreciated
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EuroISDN uses uLaw, so Asterisk does as well, because it
doesn't need to do transcoding then...
Sure? At my knowledge, in Europe aLaw is always used.
Am I wrong?
Thanks again?
Mimmus
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Title: Asterisk perms in manager.conf
Hi,
can someone sched a light what exactly mean the read write permissions in manager.conf?
[public]
secret = private
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.255.0.0
read = system,call,log,verbose,command,agent,user
write =
nik600 wrote:
Is it possible to transfer an existing call from the extension ...
SIP/xxx to another extension in a specific context?
you can do this with the redirect action:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
=Stefan
signature.asc
Description:
On 3/21/06, Mimmus [EMAIL PROTECTED] wrote:
Hi,
at my Asterisk box, I have a few of IAX2 phones (configured with
alaw/ulaw/gsm codecs, in this order) and a PRI E1 line.
In iax.conf I hav:
disallow=all
allow=alaw
allow=ulaw
allow=gsm
During some incoming call, I read at console:
On Wed, Mar 22, 2006 at 05:54:27AM -0500, David Hajek wrote:
[public]
secret = private
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.255.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
Lets say I want some users to use dial through
How is the device you are calling using to IAX configured?
It is a legacy PBX formerly connected to a PRI line.
If the remote end does not support, or is not configured to use
aLaw, then uLaw is your second choice, and the protocol will
fall back to that.
No because now I disabled uLaw
On Tuesday 21 March 2006 19:18, Douglas Garstang wrote:
The point is that if you do have to, then you shouldn't lose any data.
In a production environment, the last thing you want to do is affect
customers. Given that Asterisk is supposed to be carrier-grade, I'd have
thought this was a given.
On Wednesday 22 March 2006 05:26, Steve Davies wrote:
Another hint for getting hints working, although this only relates
to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is
that status changes are not notified for channels where there is a
hyphen '-' in the channel name, so
Greetings All,
I'm about to embark on what will hopefully be a wonderous journey. We will be slowly moving towards a full Asterisk solution, but along the way, I have to put up with the Legend system for a little while longer.
So far, I've only been able to look at this from a theory
Hi,
Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on
certain extensions.
I have usecallingpres=yes in zapata.conf, and am using
SetCallerPres(prohib) in my dialplan prior to the Dial command. No
matter what I set SetCallerPres to the CID is still displayed.
Is there something
Hi.
Im trying to get a Beronet QuadBRI card work with
bristuff drivers. Though qozap module loads right, all card spans are in
deactivated status. Im quite sure my configuration is correct and using
a single BRI card instead of the quadBRI the status is active and I can place
and receive
Barry Flanagan wrote:
Hi,
Using * 1.2.5 with a euro_isdn PRI I need to hide the callerID on
certain extensions.
I have usecallingpres=yes in zapata.conf, and am using
SetCallerPres(prohib) in my dialplan prior to the Dial command. No
matter what I set SetCallerPres to the CID is still
I would stick with POTS over using the internet. With
the internet there are more points of failure (IMO).
As far as people cutting the wires we have both. We
have a reg. pots line and a prepaid cell phone
connected in for backup.
--- Lists [EMAIL PROTECTED] wrote:
Thanks Bob for your response.
I have a rule in my outlook to delete any email that
in the subject it says asterisk users mailing list
traffic. Get my drift ? This topic has been around
for a long time as others have mentioned and people
keep replying. This useless topic alone adds several
messages to my inbox daily.
C F wrote:
Polycoms are not the best if you want a phone that works behind NAT.
Do you mean in general? Or only if you are trying to interconnect
multiple offices?
Are Polycoms fine for just one office, if the entire office is behind a
NAT device, and the phones are only being used for
Lists wrote:
Has anyone successfully
hooked up ADT or Brinks home alarm
system to say a analog port or SIPURA through Asterisk?
I tried to to do with with another alarm system and gave up. The tones
are too high frequency to work.
--
Chris Mason
NetConcepts
(264)
I agree. Gmail and mailing lists go together as well as Coke and ice. I moved all my mailing lists over to Gmail and it's great. It keeps lists out of my main email and the search is so good, you think it's Google :-)
On 3/20/06, Brian Roy [EMAIL PROTECTED] wrote:
On 3/19/06, James Harper
[EMAIL
Lists wrote:
I am hoping the alarm companies adopt quicker to the internet.
I don't see that happening. Internet reliability is not going to be
sufficient for alarms. PSTN lines, for all their issues, don't fail, and
alarm systems can sense the dial tone and alert if it is missing.
I would
It's a type of shoe you can get at any Macys
On 3/22/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
2. If you receive it because you have an 800 number, you are not
allowed to use it for anything else (read marketing) but billing.
Can
Use a Macro, or you could try the local channel
On 3/21/06, Jason Frisch [EMAIL PROTECTED] wrote:
Hi everybody.
I have been searching and trying for an answer, but no luck, so here I go..
Is there anyway to execute multiple commands on a single priority in
extensions.conf?
eg:
exten =
Hi,
Without separate incoming and outgoing context you could not secure your
system from an outside caller using your system to dial a long distance
number.
Here is an example outgoing context that restricts who can call long
distance. If a SIP phone does not belong to the 'longdistance'
On Wed, Mar 22, 2006 at 01:04:44PM +0100, Francesco Angi wrote:
Hi.
I'm trying to get a Beronet QuadBRI card work with bristuff drivers.
Though qozap module loads right, all card spans are in deactivated
status. I'm quite sure my configuration is correct and using a single
BRI card instead
Hi,
I have two asterisks connected via IAX2 trunk. The first * use dial
prefix 2XX, the second one 3XX.
Calls routing works OK.
But I don't know how to get dialtone of remote asterisk pbx.
I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of
asterisk #1 after dialing 2.
I
Karlos wrote:
Hi,
I have two asterisks connected via IAX2 trunk. The first * use dial
prefix 2XX, the second one 3XX.
Calls routing works OK.
But I don't know how to get dialtone of remote asterisk pbx.
I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone
of asterisk #1 after
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls
I get a double ring tone (UK style + US style). I also have a DECT phone
Hello all!
For several months now weve been experiencing a
really strange problem with sound which best can be explained as
choppy/stuttery, and with a touch of echo on top. Basically, parts of a
conversation might be choppy, but often combined with some echo as well. The
sound problem
21 mar 2006 kl. 19.07 skrev Douglas Garstang:
Ready to scream here..
No one is surprised ;-)
1. After 6 months with Asterisk I'm STILL trying to understand the
difference between a SIP user, friend and peer.
* A friend is a peer object and a user object. It's just a
configuration
Karlos,
Sounds like you want ignorepat = 2 (or 3) in the context that holds
the dial patterns. This will continue the dialtone after you dial 2 or
3 in your dialplan.
IE:
[system-2]
ignorepat = 3
exten = _3XX,s,1,Dial(IAX2/system-2/${EXTEN})
Jason
Karlos wrote:
Hi,
I have two
A secondary issue may be insurance. In a domestic situation, if you receive a
discount for having an alarm installed, you may find that the insurance
discount is only valid if the alarm is installed over POTS, and usually by
hardwiring.
This is for actuarial reasons, that is to say the
On 3/22/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Wednesday 22 March 2006 05:26, Steve Davies wrote:
Another hint for getting hints working, although this only relates
to older 1.0.x versions of Asterisk (It is already fixed in 1.2.x) is
that status changes are not notified for
On Wednesday 22 March 2006 08:34, Olle E Johansson wrote:
4. WHY then does a reload clear this list? Doesn't this list come
from the astdb file?
I explained this in the bug tracker.
Reload clears everything but registered static peers, these are re-
configured from astdb.
We do not
Quoting Chris Mason (Lists) [EMAIL PROTECTED]:
Lists wrote:
I am hoping the alarm companies adopt quicker to the internet.
I don't see that happening. Internet reliability is not going to be
sufficient for alarms. PSTN lines, for all their issues, don't fail, and
alarm systems can sense
I use asterisk 1.2.5 and h323 that comes with addons 1.2.1.
Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and
tries to make attendant transfer to person B (local SIP phone). They speak.
Then A hangs up. Call form h323 trunk doesn't get to person B.
This is what I
Quoting [EMAIL PROTECTED]:
A secondary issue may be insurance. In a domestic situation, if you receive a
discount for having
an alarm installed, you may find that the insurance discount is only valid if
the alarm is
installed over POTS, and usually by hardwiring.
This is for actuarial
Hadley Rich wrote:
Hi all,
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't
answer calls, for unknown reasons. I gave up
Ciao Francesco,
in data mercoledì 22 marzo 2006, alle ore 13.04, hai scritto:
On Beronet installation manual I read that Beronet and Junghanns cardsare identical in their construction but Junghanns made bristuff so that only their cards can work with their drivers. In the same document and
Hi,
I was wondering how other people have approached this problem?
* Customer Calls Business
* Placed in a Queue
* Logged in Agent answers call
* Call Ends
Now the big question; at the end of each call, I need to:
* enter some info in about the call such as Customer Number, Wrap up code
The ignorepat statement allows me hear dialtone of my pbx, not remote;
I'd like to dial some number (for example 5) and get dialtone of
remote asterisk.
I think the only way is to use DISA or?
Karlos.
Jason Bachman wrote:
Karlos,
Sounds like you want ignorepat = 2 (or 3) in the
Dovid Bender wrote:
I personaly use VoipJet, Teliax and myPhoneCompany.
They are all great. Dont remember if teliax supported
IAX. I know that myPhoneCompany for sure dosent. They
use SIP. I did however ind that thier voice quality is
very good.
I'm sorry, but you don't remember if Teliax
I apologize, but the fix I was thinking of wasn't directly related to this.
It was in app_voicemail.c, but related to using the channel's context for
the Directory application. The fix for your issue may be indirectly
related, though. I would open a bug.
Regards,
- Brad
-Original
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
Thanks,
Ron
___
The ignorepat statement allows me hear dialtone of my pbx, not remote;
I'd like to dial some number (for example 5) and get dialtone of remote
asterisk.
I think the only way is to use DISA or ?
Karlos.
Jason Bachman wrote:
Karlos,
Sounds like you want ignorepat = 2 (or 3) in the context
Are you using which version of Asterisk?? Did you check if you are
facing the old audio bug on bridge calls that appeared ?
http://asteriskvoip.blogspot.com/2006_01_01_asteriskvoip_archive.html
Wednesday, January 25, 2006
Update: No audio - Update your Asterisk
This morning we discovered a
Agreed, but if all else fails, set a different context for that PRI,
and in that context, force the CallerID using SetCallerID before
making the onward call.
Agreed, but I like clean configs.
It is possible that the Zaptel callerid= field does not accept the
Name number format (I'd have to
Has anyone knows the best configuration for DTMF detection on Sipura SPA
2000???
Regards,
___
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Karlos wrote:
The ignorepat statement allows me hear dialtone of my pbx, not remote;
I'd like to dial some number (for example 5) and get dialtone of remote
asterisk.
I think the only way is to use DISA or ?
yes, DISA would be the way to go. Use something like
exten =
Do you want to dial an outgoing line as well as the SIP line?
Dial(SIP/${OUTGOING}/${EXTEN}) ?
I can't say obviously without more info but it sounds to me like you are
looking for the wrong solution
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
On Wed, 2006-03-22 at 09:22 -0500, Ron Joffe wrote:
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
You can
On 22/03/2006, at 10:22 PM, Ron Joffe wrote:
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
Thanks,
Ron
It's a toll free number. You can call it from anywhere and the costs of the
call go on the callee not the caller.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Wednesday, March 22, 2006 7:50 AM
To: Asterisk Users Mailing
On Wed, 2006-03-22 at 11:52 +0100, Paul Brown wrote:
Hi,
I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-)
Any pointers would be appreciated
=-o Wonder who took over his mail account :-P
Aaron
On Tue, 21 Mar 2006, Martin Joseph wrote:
On Mar 21, 2006, at 4:01 PM, Douglas Garstang wrote:
Oooo I think I am gonna poo my pants.
Using the microbrowser on a Polycom 601, I was able to get it to execute a
cgi script upon selection of
Yeah, once they re-register after the default time period, they come back.
We've got ours set to 5 minutes from phones on the network, 1 minute for
phones off the network, so if you do a reload, it usually takes about 5
minutes to get all the phones re-registered.
Aaron
On Tue, 21 Mar 2006,
Hello,
Here is part of my extensions.conf.
I set both absolute and response timeouts according to
the day context.
I wish to asterisk hangup after 60s and 10s to play or
replay the annoucement .
Asterisk doesn't jump to T extension.
How can fiox this problem ?
harry
...
[day]
exten =
were using a Manager Proxy, i guess you should programm a proxy as
well, it will simplify things
AsterisManager - Manager Proxy -- Multiple Clients of Manager
On 3/21/06, Douglas Garstang [EMAIL PROTECTED] wrote:
That's way too much Java for me. I'm lost already.
-Original
On Tuesday 21 March 2006 22:19, C F wrote:
What you said before about everything being *incoming* from asterisks
point of view is true, however that just shows that we were saying the
same thing, since for the context (not asterisk DP context) of this
post we are looking at the point of view
I've extracted the asterisk-addons-1.2.2.tar.gz file and don't see a
COPYING or LICENSE file, nor is there a mention of GPL or BSD in the
README file.
asterisk-sounds is BSD licensed, so it's not entirely clear how
asterisk-addons is licensed -- though I would guess GPL. Can anyone
clarify the
On Wednesday 22 March 2006 10:21, Aaron Daniel wrote:
Yeah, once they re-register after the default time period, they come back.
We've got ours set to 5 minutes from phones on the network, 1 minute for
phones off the network, so if you do a reload, it usually takes about 5
minutes to get all
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission
It's in the NON-SIP section of the site, you'll find it on the page
somewhere under the 7970 SCCP images... They're harping that this release
is for their new CCM, so although it's SIP, it kinda sucks.
Aaron
On Wed, 22 Mar 2006, Paul Brown wrote:
Hi,
I couldn't find the 7970 SIP image on
Hi allI'm trying to dial out with a Digium X100P card set upon channel Zap/1 to local number (25921163). My call file is: (out.call)Channel: Zap/1/25921163Callerid: 25921163
MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: outgoingExtension: sPriority: 1 And my extension.conf is,[outgoing]exten
22 mar 2006 kl. 15.02 skrev Andrew Kohlsmith:
On Wednesday 22 March 2006 08:34, Olle E Johansson wrote:
4. WHY then does a reload clear this list? Doesn't this list come
from the astdb file?
I explained this in the bug tracker.
Reload clears everything but registered static peers, these
Did you download it from asterisk.org? I didn't have the latest -addons,
but I just downloaded it and it does have Copying which contains the GPLv2.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. J.
Meidlinger
Sent: Wednesday, March
Hi,
What is the best way people have found to remotely upgrade the
firmware in SPA-2002 devices?
___
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Tim Panton ha scritto:
I don't suppose you have an ethereal packet capture from a
bad call ???
Or a description of the 'badness'?
I have myself problems with iax2 sometimes, it drops a lot of packets
even if there's no apparent reason to.
For example two asterisk connected via iax2 on a
Isn't it curious that, given the delete rule, he knows this has been
discussed once again, and feels the need to add to the number of
useless messages.
In fact, since the subject keeps coming up, perhaps there really is an
issue that needs to be addressed??
The only thing constant is
Ron Wellsted ha scritto:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing calls
I get a double ring tone (UK style + US
Cell works, but it has to be analog cell (or similar). It's the
compression that's the enemy.
We're spoiled, by the way, as IT Pro's. We have RFC's and Open Source.
The alarm industry has proprietary hack-jobs and patents. It isn't that
they don't want to do IP, they just want to force their
Hello everyone,
I am working on something now that could really use a snapshots. For
those that are not familiar, basically what it involves is having a
server with httpd running automatically checkout asterisk and friends
from SVN, tar it up, and place it in a directory with the date that
On Tuesday 21 March 2006 18:12, Douglas Garstang wrote:
I had to drop realtime with sip users. If you do a reload or a restart, you
lose all the sip peer information (even with rtcachefriends=yes). That just
wasn't acceptable for us.
I just spoke to oej about this on IRC.
The only time this
http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns
says:
! wildcard, matches zero
or more characters immediately
(only Asterisk 1.2 and later, see note)
Note: The exclamation mark wildcard, which is
available only in Asterisk 1.2 and later, behaves
-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 22, 2006 8:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime / SIP Peers etc
22 mar 2006 kl. 15.02 skrev Andrew Kohlsmith:
On
LOL, I've only just begun testing Polycoms, so I can't say I've seen that
before... I think it's really a question of when you're doing the work on
the server... We do all our major server work when there's nobody on the
system so there's no effect on the users, but even then, we rarely
Sounds to me like the packets (ACKS maybe) are arriving late. Sufficiently so
that Asterisk is about to retransmit the packet. However, right at the last
minute it got the ACK from the last one and stopped the retransmission as it
found the ACK.
On 3/22/06, Dan Littlejohn [EMAIL PROTECTED]
I can't tell you exactly what it means, but you can make it go away by
not logging debug information.
look in /etc/asterisk/logger.conf
Mar 21 15:56:25 DEBUG[18402] chan_sip.c: Stopping retransmission on
'[EMAIL PROTECTED]' of Request 102: Match
Found
Mar 21 15:56:25 DEBUG[18402] chan_sip.c:
On 3/22/06, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 21 March 2006 22:19, C F wrote:
What you said before about everything being *incoming* from asterisks
point of view is true, however that just shows that we were saying the
same thing, since for the context (not asterisk DP
Sheeju .R.Alex wrote:
Before I pick the target phone(25921163), the channel
is answered and completes the extensions. i,e omly if
I pick the phone immediately I can listen to some part
of playback (even If I donot pick the phone the
channel is answered).
Asterisk seems to assume that the
I'm not sure how to answer your question about where I downloaded it. The
web page I used was...
http://www.asterisk.org/download
...but the download link for Addons Version 1.2.2 on the right side points
to...
http://ftp.digium.com/pub/asterisk/asterisk-addons-1.2.2.tar.gz
On Wednesday 22 March 2006 10:51, John Novack wrote:
In fact, since the subject keeps coming up, perhaps there really is an
issue that needs to be addressed??
It has been addressed. There are a host of forum sites specific to Asterisk,
and even more (such as tek-tips) which include Asterisk
set debug 0
Disable debug logging in logger.conf
Debug logs is not an indication for an error, it's just information
for developers.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/* Bootcamp in
Europe in April - register today
Hi there
I am using an IAX2 softphone built from the IaxClient library dialing into
Meetme conferences. The IaxClient seems to use silence suppression, and not
sure if this can be disabled. The client works fine through most routers,
but for some it disconnects the client after about 5 minutes
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wed, 22 Mar 2006, Simone Cittadini wrote:
Ron Wellsted ha scritto:
This is slowly driving me nuts!
I have several Cisco 7960s with SIP 8.2/7.5 fw connecting to Asterisk
1.2.5 driving a TE110P on a BT EuroISDN PRI line. On all outgoing
I'm trying to make a call through 2 asterisk servers, and the call
simply hangs up after one ring.
The path is like this:
ATA1 - (SIP) - server1 - (IAX) - server2 - (SIP) - ATA2
The two ATAs are registered to their respective asterisk servers and can
make and recieve calls to local extensions
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 22, 2006 8:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?
On Tuesday 21 March 2006 18:12, Douglas Garstang wrote:
I had
I'm not sure I'm talking about the same thing, but if this is true, it may
clear a few things up.
Firstly, from what I've been told my Kevin Fleming on this list, and by calling
Digium directly, support for multiple Asterisk systems accessing the same MySQL
database for sip user/peer
If you're so worried about downtime, why not have the phones re-register
at a higher interval? When I'm not screwing around with settings on the
servers and if I don't upgrade/update the system often, asterisk will stay
up and running for well over a month without a reload (probably longer,
LOL, but it's such an interesting discussion :-P
On Wed, 22 Mar 2006, Dovid Bender wrote:
I have a rule in my outlook to delete any email that
in the subject it says asterisk users mailing list
traffic. Get my drift ? This topic has been around
for a long time as others have mentioned and
Arrgh.
I just made a call with Asterisk to extension 2944093. That extension exists in
astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database
query...
SELECT * FROM ast_sip_users WHERE name = '2944093'
Uhm... Why?
Doug
___
What are these error messages?
Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Failed to read gains: Invalid
argument
Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Failed to read gains: Invalid
argument
Mar 22 17:41:10 DEBUG[3592] chan_zap.c: Updated conferencing on 1, with 0
conference users
Mar 22 17:41:10
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