2008/10/6 Tarek Sawah <[EMAIL PROTECTED]>
> i haven't facedthse tpe of problems you mentioned with mysql.. but there is
> one thing that you need to edit the sip.conf iax.conf or you can use the
> sample ones in the samples folder..
> other than that.. i've been with trixbox for over three years n
Steve Anness wrote:
> I posted earlier in the day about needed help with IAX trunking. I did
> some more reading and made some more changes.
>
> Here is what I have thus far:
>
> Iax.conf on one server
>
> [general]
> bindport = 4569
> bindaddr = 0.0.0.0
> disallow=all
> allow=ula
I have a new Fedora 9 firewall I am setting up in front of an Asterisk 1.4
box. I ported over all of my iptables rules..but now have a strange
problem: SOMETIMES, the audio is only 1-way (i.e. and RTP path problem).
Can someone offer a tip here? Since I have conntrack_sip loaded on the
firew
The problem is that the Linux SIP ALG is not RTP-aware and doesn't NAT
the RTP. If that's changed, it would have to be in the last one or two
kernel releases.
Your solution is OpenSER (Kamailio/OpenSIPS) + nathelper + mediaproxy or
rtpproxy.
OCG Technical Support wrote:
> I have a new Fedora
Tilghman Lesher wrote:
> On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
>
>> Wesley Haut wrote:
>>
>>> Yell at me if you will, but I hate func_realtime - it's not very
>>> usable nor is it change-friendly (update your database and your
>>> dialplan completely breaks).
>>>
>>
Hi,
What does retransmitting NAT means? I have a client that uses SPA 942,
and his phone sometimes cannot be called. i did a sip sebug and i keep
on seeing retransmitting NAT.
on the realtime it shows that it is registered, so when i try to call it
, asterisk thinks it is still online so it tr
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten
I am tinkering with a new router, a Linksys wrt610n dual-band, etc. But
the when I connect it, the softphones(x-lite) on the computers don't even
register. After a couple of hours of hassle, I found out that if I dmz the
router to the computer I am using, the softphone starts to work. Problem
I have a customer that wants to use meetme but they want to have the users
record their name so it is played to the other people on the conference. Is
there an easy way to do this?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 E
Yep - trixbox offers this so the answer is yes.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Carlos Chavez
> Sent: We
Carlos,
Druid Open Source Edition supports this free and truly open source.
http://www.voiceroute.org/druidose/screenshots/conference_room
Ming
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Carlos Chavez
>> Sent: Wednesday, 8
Carlos Chavez wrote:
> I have a customer that wants to use meetme but they want to have the users
> record their name so it is played to the other people on the conference. Is
> there an easy way to do this?
>
> --
> Carlos Chavez
> Director de Tecnología
> Telecomunicaciones Abiertas de Méxic
Hi,
Can anybody point me to an online resource that will assist with
interpreting Asterisk log files?
I note that a similar question was asked in this forum some time ago
(http://lists.digium.com/pipermail/asterisk-users/2007-June/189793.html),
which doesn't appear to have received any responses.
Maybe you can write your own patch that will allow this based on the
useragent somehow mapping it to 2nd peer based on the useragent? But
this feature is not there now.
What will happen when host=dynamic is the last registration will be
the one used, so if you have two SIP devices trying to regist
Tilghman Lesher wrote:
>> Can someone suggest the best way to deal with this without resoring to a
>> highly repetitive/iterative dialplan?
>>
> Leif and I discussed something like this at Astricon 2008, and we came up with
> this patch:
> http://bugs.digium.com/view.php?id=13632
Nice! For t
Hi Jared,
That's what i thought as well, every time it registers it should be added.
but every time i call it it just fills up my screen (sorry i pasted the
logs on this mail), like this, and the time looks the same.
[Oct 8 15:27:28] -- Added extension '103100' priority 1 to
sipregcontex
Is there an install script or step-by-step instruction somewhere on
whaty is needed to migrate from zaptel to dahdi?
I read the document that digium published which nicely states some of
the differences between zaptel and dahdi but i was looking more for
something like step-by-step instructions.
--- On Tue, 10/7/08, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> I am not sure if it is possible to somehow invoke a function
> to pick
> up the call via dialplan, if it is a combination of that
> function and
> DISA should do what you need.
Thanks!
I could configure the ATAs to "auto-dial"
Greetings list,
I've been updating some of my configs to prepare for a move to 1.4 at some
point in the near future, and added call-limit=99 to the SIP devices which need
to be represented via BLF.
Looking at the logs this morning, I have a number of entries like this:
Oct 8 09:22:36 NOTICE[14
See the documentation for DISA, it is restricted by context. So
assuming you already have your dialplan configured securely, there are
no security implications.
Be aware that the behavior of the phones change when you dial through
DISA, you can no longer use features such as "redial." That is beca
I'm looking for some documentationof the implementation of the SRV DNS
on asterisk. I'm using asterisk 1.4.22.
1) From voip-info.org: "Currently, Asterisk only reads the first SRV
entry without bothering with priorities and weights".
IS It still true? from my first tests I can see that asterisk
...and now for something completly different
In very small situations I've been very happy with this desktop POE switch $69
http://shop1.frys.com/product/4971591
Jim
www.sigma-networks.com
- Original Message -
From: "Daniel Hazelbaker" <[EMAIL PROTECTED]>
To: "Asterisk Users M
Hi,
A theorical question :
For some SIP hardphones (or mobile handsets), when receiving a busy
response, it is possible to ask for automatic redial and your phone will
later try to call your contact.
As I'm wondering how I could provide a way to cancel those automatic
redials, I'm trying to under
Hi,
I use hints to drive the LEDs on my snom phones, something like:
exten =>
601,1,Dial(SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server&SIP/mjc_library,20,trj)
exten => 601,2,Voicemail([EMAIL PROTECTED],u)
exten => 601,102,Voicemail([EMAIL PROTECTED],u)
exten =>
601,hint,SIP/mjc_office&
This certainly seems better value than single port POE insertors at $40
each.
Michael
--Original Message Text---
From: Sigma Networks
Date: Wed, 8 Oct 2008 09:27:28 -0400 (EDT)
p { margin: 0; }...and now for something completly different
In very small situations I've been very happy with th
Chris Bagnall schrieb:
> I've been updating some of my configs to prepare for a move to 1.4 at some
> point in the near future, and added call-limit=99 to the SIP devices which
> need to be represented via BLF.
>
> Looking at the logs this morning, I have a number of entries like this:
> Oct 8
>
> Hi Jerry,
>
> Hm, okay. We had to use md5secret (instead of secret) in the sip.conf for
> our 7970's to get them to successfully register with asterisk. However, if
> you had them working before then I doubt this is the issue. You can try
> anyway though,
>
> http://www.voip-info.org/wiki/inde
Dr. Michael J. Chudobiak schrieb:
> I use hints to drive the LEDs on my snom phones, something like:
>
> exten =>
> 601,1,Dial(SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server&SIP/mjc_library,20,trj)
> exten => 601,2,Voicemail([EMAIL PROTECTED],u)
> exten => 601,102,Voicemail([EMAIL PROTEC
>> 601,hint,SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server&SIP/mjc_library
>>
>> Sometimes asterisk gets confused, though, and reports my extension as
>> in-use, even though no channels are active. Dialing something makes the
>> hint report "inactive" - the states are inverted, in other w
Dr. Michael J. Chudobiak schrieb:
>>> 601,hint,SIP/mjc_office&SIP/mjc_home&SIP/mjc_lab&SIP/mjc_server&SIP/mjc_library
>>>
>>> Sometimes asterisk gets confused, though, and reports my extension as
>>> in-use, even though no channels are active. Dialing something makes the
>>> hint report "inactive
Philipp Kempgen wrote:
>> Hmm, I'll see if that gives me any clues...
>
> Or you could try 'sip show inuse'.
Thanks, Philipp! I never noticed that command; I'm sure it will be very
handy for debugging.
- Mike
___
-- Bandwidth and Colocation Provided
On Tuesday 07 October 2008 21:58:31 Karl Fife wrote:
> So that leaves only one question:
>
> exten => ?
>
> What extension the following:
> '3129842314'
> '*989'
> '+13129842314'
>
> BUT does not match:
> 'i'
> 'james'
>
> is this possible?
I think you already described it in your original post:
On Wednesday 08 October 2008 02:20:38 Rob Hillis wrote:
> Tilghman Lesher wrote:
> >> Can someone suggest the best way to deal with this without resoring to a
> >> highly repetitive/iterative dialplan?
> >
> > Leif and I discussed something like this at Astricon 2008, and we came up
> > with this p
Hello,
maybe somebody has fastagi examples, or can advice how to do. I just want
to do a single ton connection to mysql server. Cause now I'm using AGI, and
each call creates mysql connection and so on. I just want alleviate CPU load
... Asterisk and mysql servers are on the same box, and is it a
Hi,
I need a way to have mobile phones in a queue and have to press a key when
picking up the phone (to thwart answer phones) before transfering.
What I've built so far is as follows but unfortunately if the mobile number
picks up the call is immediately bridged.
extensions.conf
;Announce th
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, October 08, 2008 10:13 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] cisco phones getting SIP 401 unauthorized
>
> Hi Jerry,
>
> Hm, okay. We had to use md
>
> Hi Jerry,
>
> Hmm. We had to replace our router with one that supported SIP ALG (we went
> cisco). However, since you mention all the phones are in the LAN this
> shouldn't make a difference.
>
> Does the problem go away if you go back to the old firewall?
>
> Thanks,
> Matt
>
unfortunatel
Yell at me if you will, but I hate func_realtime - it's not very usable nor
is it change-friendly (update your database and your dialplan completely
breaks).
I'm getting a new 1.6 box built out and working, and wanted to emulate the
functionality of APP_realtime somehow, so I started digging aroun
Did you check sip.conf to make sure that the port is correctly set to 5060?
Please show the output of Cli> sip show peer and the contents of
your SEP.cnf file.
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Wednesday, October 08,
Can anyone point me at a code fragment (C would be nice) that I could
use to subscribe to hints on a * box?
I'd like to write a small (hopefuly efficient) widget to show custom
device states and believe that a subscription to the hint would be the
most efficient but I'm very open to suggestions.
>
> Did you check sip.conf to make sure that the port is correctly set to 5060?
>
> Please show the output of Cli> sip show peer and the contents of
> your SEP.cnf file.
>
> Dave
>
sip.conf has :
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=X.X.
On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
> Can anyone point me at a code fragment (C would be nice) that I could
> use to subscribe to hints on a * box?
>
> I'd like to write a small (hopefuly efficient) widget to show custom
> device states and believe that a subscription to the hi
Wesley Haut wrote:
> Yell at me if you will, but I hate func_realtime - it's not very
> usable nor is it change-friendly (update your database and your
> dialplan completely breaks).
I agree completely. As it stands, the REALTIME() function is nearly
completely useless. If Asterisk had better
Ken D'Ambrosio wrote:
> Hey, all. We're rolling out VoIP, and I'm wondering about PoE
> recommendations, as we're going to have to replace our current network
> equipment. My first inclination would be to just plunk down the cash and
> do a Cisco system, but I'm relatively certain that would get
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
> Wesley Haut wrote:
> > Yell at me if you will, but I hate func_realtime - it's not very
> > usable nor is it change-friendly (update your database and your
> > dialplan completely breaks).
>
> I agree completely. As it stands, the REALTIME(
Jared Smith schrieb:
> On Wed, 2008-10-08 at 18:51 +0100, Russell Brown wrote:
>> Can anyone point me at a code fragment (C would be nice) that I could
>> use to subscribe to hints on a * box?
>>
>> I'd like to write a small (hopefuly efficient) widget to show custom
>> device states and believe t
I am very confused on IAX Trunking. I mean I understand what is does but I
am confused on how to make it work right with my Asterisk server and what
all I need to add to make it work like it should. Basically we have two
servers and want to call between the two. I have configured the iax.conf
fi
Tilghman Lesher wrote:
> On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
>
>> Wesley Haut wrote:
>>
>>> Yell at me if you will, but I hate func_realtime - it's not very
>>> usable nor is it change-friendly (update your database and your
>>> dialplan completely breaks).
>>>
>>
> Well, after very quickly making a test call it's not Vitelity. It could be
> something with your account? Might want to try opening a support ticket. If
> you want, create a sub account and e-mail me off list the username and
> password and I'll test it with my box or vice versa.
I am now able t
-- Executing [EMAIL PROTECTED]:2]
VoiceMailMain("SIP/17865221569-b6b03f60", "3523782778|s") in new stack
-- Playing 'vm-youhave' (language 'en')
app5*CLI>
<--- SIP read from 74.170.252.213:5060 --->
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.54;branch=z9hG4bKaf5e8319A2DD1
I have several branch offices, each with their own Asterisk server
(version 1.4.22.1) handling their PBX functions. All of these offices
need to talk to each other. In sip.conf I created a peer entry for each
office with a username of branch-user and a friend entry for every
branch-user with
I posted earlier in the day about needed help with IAX trunking. I did some
more reading and made some more changes.
Here is what I have thus far:
Iax.conf on one server
[general]
bindport = 4569
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes
[vvfarm]
ty
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