Hi,Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling that my Asterisk server doesn't register with Teliax. So, the final conclusion is "My Asterisk server doesn't register with Teliax". Here I am
I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:Hi,Thank you for your response. As you said, I executed the command sip show registry. But, its not showing anything. Teliax people are also telling that
It wasn't any help. It doesn't give any reference to order of trunks, etc in
sip.conf. I'm still looking for the post, Rich Adamson made reference too...
On Friday 11 August 2006 18:24, Fran Oliveira wrote:
see http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
perhaps it can help
Hi, My user name is : rudy.pandya Thank you.Sharon Lim [EMAIL PROTECTED] wrote: I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi,Thank you for your response. As you said, I executed the command "sip
Hi,
queue announcements works when we use music on hold in the queue, but if
we use ringback e.g. queue(myqueue|r|) the announcments and hold
time are not working, it seems that * is not even trying to read the
queueu-announcment files. Is this by design, or is there a work around?
Hi,
I'm trying to figure out how to cancel a call before the other side answers.
It looks like I can do this by issueing a HangUp Action using the ChannelId
that asterisk initiates the call on. Unfortunately, I don't know of a way to
associate the NewChannel Event which contains the
Hi Friends,We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not
is there something wrong with ur syntax at exten = _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) exten = _1XX,1,DIAL(SIP/teliax,${EXTEN},30,tr)
On 8/14/06, Crazy Boy
[EMAIL PROTECTED] wrote:
Hi, My user name is : rudy.pandya Thank you.Sharon Lim
[EMAIL PROTECTED] wrote: I am not
Janahan Vivekanandan wrote:
I know that I can use the ActionID to accomplish this once I receive
the OriginateSuccess event, but I need to be able to cancel the call
before it is answered(I'm pretty sure OriginateSuccess is only sent
after the call is answered, correct me if I'm wrong...:-)
In article [EMAIL PROTECTED],
Janahan Vivekanandan [EMAIL PROTECTED] wrote:
Hi,
I'm trying to figure out how to cancel a call before the other side answers.
It looks like I can do this by issueing a HangUp Action using the ChannelId
that asterisk
initiates the call on. Unfortunately, I
In article [EMAIL PROTECTED], Stefan Reuter [EMAIL PROTECTED] wrote:
Janahan Vivekanandan wrote:
I know that I can use the ActionID to accomplish this once I receive
the OriginateSuccess event, but I need to be able to cancel the call
before it is answered(I'm pretty sure OriginateSuccess
Anybody has first-hand experience with any (or both) of these options?
Are there any other possibilites that I'm missing? Some other
foneBRIDGE-like product I still haven't heard of?
Thanks in advance
Another option would be to get a carrier grade VoIP - PRI gateway. I
have an Audiocodes 4
Roberto,
thanks for your feedback. It is probably more appropriate to continu
this discussion in the Asterisk-Users mailing list.
We have changed the 'From:' address to [EMAIL PROTECTED] instead
of [EMAIL PROTECTED] ... hopefully this will get less often stuck in
people's spam filters.
Crazy Boy wrote:
Hi,
Thank you for your response. As you said, I executed the command sip
show registry. But, its not showing anything. Teliax people are also
telling that my Asterisk server doesn't register with Teliax. So, the
final conclusion is My Asterisk server doesn't register with
Shaun Hofer wrote:
It wasn't any help. It doesn't give any reference to order of trunks, etc in
sip.conf. I'm still looking for the post, Rich Adamson made reference too...
On Friday 11 August 2006 18:24, Fran Oliveira wrote:
see http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
Crazy Boy wrote:
Hi Friends,
We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I
have connected my PSTN line directly to first port. I am making outgoing
calls and receiving incoming calls successfully through my Asterisk. The
problem is: When I am receiving a call from
Hi,
my test-bed is :
sipphone -- Asterisk PBX -- PSTN -- Cell Phone
sipphone was able to setup a connection to Cell Phone. When sipphone
hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up,
sipphone was not able to hang up.
Could it be that Asterisk was not able to recognise
I have to say that I'm experiencing the same issues, using the latest SVN
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chan Kwang
Mien
Sent: Monday, August 14, 2006 8:26 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problems with
Chan Kwang Mien wrote:
Hi,
my test-bed is :
sipphone -- Asterisk PBX -- PSTN -- Cell Phone
sipphone was able to setup a connection to Cell Phone. When sipphone
hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up,
sipphone was not able to hang up.
If your line doesn't
In my case the 911 goes to the Police departments dispatchers, I have to to call the main office number and make sure it is a good time to test then I can call right back to get the read out on the screen. This is great since they want me to send them some sort of information about who called from
please, can somebody tell us, if currently used jitterbuffer
implementations (iax or sip w/ jb patch) are really working/usefull if
jitter is frequently changing between 10-1000ms (on cdma connection)?
I have really big problems with using jitterbuffer between two asterisks:
- with iax2, I
Hello,
I'm working in a small call center, but with special requirements. We
currently have a couple of clients, all of them have
specific phone numbers configured in our system, so when we get a call
for a specific client we take down the information via a webpage
then it sent via email to
lo there,
i am running a python agi script that gets a DTMF number from the user
and passes it back to the script. It works fine with numbers, but if they enter a star (*), it doesn't want to play.
Is there a difference in how this is handled?
here is the snippit:
def getNumber (sound,
- Original Message -
From: Rushowr
[mailto:[EMAIL PROTECTED]
To: [EMAIL PROTECTED], 'Asterisk Users
Mailing List - Non-Commercial Discussion'
[mailto:[EMAIL PROTECTED]
Sent: Mon, 14 Aug 2006 09:28:29
-0300
Subject: RE: [asterisk-users] Problems with Hangup
I have to say that I'm
My PSTN termination is through a provider, with a SIP connection between
myself and their systems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp
Sent: Monday, August 14, 2006 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial
On Mon, Aug 14, 2006 at 10:31:30AM -0400, shawn bright wrote:
lo there,
i am running a python agi script that gets a DTMF number from the user
and passes it back to the script.
Any reason you're not using Read for this?
It works fine with numbers, but if they
enter a star (*), it doesn't
Hi everybody
My name is Jose Manuel Cortes, i'm from Colombia and im working in a asterisk
implementation for my thesis. The initial system was a pbx and a LAN separated,
now with the asterisk server the system is:
before
Telco1 ---PBXTelco2
now
Juha,
I am running the same version of Cisco 7970 SIP firmware and having the same
problem with periodic 400 Bad Request responses from it when Asterisk
sends MWI updates for a voicemail box...
-- Got SIP response 400 Bad Request back from 192.168.144.187
-- Got SIP response 400 Bad
Forgot to say running with Asterisk 1.2.10 mainline code on RedHat FC5 box.
Other Cisco 7960 (SIP 7.5) and 7912 phones (SIP ver 1.3.1?) around the house
work as expected including MWI...
- Original Message -
From: Michael J. Tubby G8TIC [EMAIL PROTECTED]
To: Asterisk Users Mailing
yep, the problem was in the regular _expression_ (only looks for a number 0-9)
i added a couple of lines that see if its not a number, but is a * it will return that.
thanks, not an asterisk issue, but a python issue.
shawnOn 8/14/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Aug 14, 2006 at
- Original Message -
From: Rushowr
[mailto:[EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion' [mailto:[EMAIL PROTECTED]
Sent:
Mon, 14 Aug 2006 11:51:23 -0300
Subject: RE: [asterisk-users] Problems with
Hangup
My PSTN termination is through a provider, with a
Well I solved the problem, by just making it one macro, not a macro
inside another one.
[macro-record]
exten = s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$
{TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m})
exten = s,3,setcallerid(${ARG2})
exten = s,4,dial(${ARG3})
exten =
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the system I'm working with needs the separate macros. I'll
update the list if anything gets worked out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Attilla De
Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the system I'm working with needs the separate macros. I'll
update the list if anything gets worked out.
pbx-1*CLI show application gosub
pbx-1*CLI
-= Info about application 'Gosub' =-
On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote:
Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a
solution, but
unfortunately the system I'm working with needs the separate
macros. I'll
update the list if anything gets worked out.
pbx-1*CLI show
I struggled with one provider for a long time until finally realizing my username on their site was not my username that I was supposed to be using in sip configuration. Make sure you are using the right username and password. However, it would seem that you would not be able to make an outgoing
Is there any way of specifying a directory to load tftp files from
instead of from the root tftp directory when booting a cisco 7960 phone ?
Julian
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Any reason that you can't set variables before you use Gosub, then
access them in the subroutine?
Attilla De Groot wrote:
On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote:
Rushowr wrote:
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the
On 2006-08-13 06:23:29 -0700, Barry Fawthrop [EMAIL PROTECTED] said:
Thanks Dovid
I have port forwarding enabled on the linksys router ports 5060 and
1-2. I was wondering if I should also enable DMZ to the
internal IP address of the phone ?
No. This would mean all ports attempted
On Sun, 2006-08-13 at 11:28 +0200, Attilla De Groot wrote:
On Aug 13, 2006, at 11:22 AM, Dovid Bender wrote:
Please include what you send to the macro from your extensions.conf
so we can see what you are sending down to the macro.
[...]
Sorry, didn't thought it was relevant, since the
Has there been any progress on getting Call Parking to work with Linksys
SPA-942 phones and Asterisk? I am willing to assist, if there are people
working on this already. I have done a little research on this, and it
looks like there are people asking for it, just haven't found anyone
*doing*
Hi there!
I'm having lots of problems with an Asterisk used by a customer.
Got hundreds (yes hundreds, about 3-4 per minute) of this messages
every hour:
WARNING[12685] channel.c: Avoided initial deadlock for '0x96dee78', 10 retries!
The hex number changes with every message. The warning
At 02:14 AM 8/14/2006, you wrote:
We have installed Asterisk with Digium 04B card (4 FXO ports). Now,
I have connected my PSTN line directly to first port. I am making
outgoing calls and receiving incoming calls successfully through my
Asterisk. The problem is: When I am receiving a call from
Thank you Mark. I've went from The number you are dialing is not in service, please check the number and dial again to a fast busy tone...I think I'm getting closer..-- Anything else, let me know.
-Dominic Sonwww.DominicSon.comOn 8/12/06, Mark Phillips [EMAIL PROTECTED]
wrote:Sounds to me like
Anyone have a script or example to do this? I want to run the script
every so often and then drop a call file into the outgoing folder and
call multiple numbers. I figured I would ask before wasting my time
re-inventing the wheel.
Also, I dont want to setup nagios or any other system right
In order to bring the ESCAUX net.PBX solution another step further, we
have added a new configuration template called 'SOHO analog'. This
template auto-configures all your Cisco, Polycom, Swissvoice and Thomson
IP Phones and connects your asterisk server to the PSTN network via a
Linksys
I'm looking for a VOIP provider in Panama that will support outging DIDs
and SIP or preferably IAX.
Can anyone help?
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Frank,Try posting to the -biz list with this query. This list is Non-Commercial Discussion only.AlexOn 8/14/06, Frank Tarczynski
[EMAIL PROTECTED] wrote:I'm looking for a VOIP provider in Panama that will support outging DIDs
and SIP or preferably IAX.Can anyone
On 140806, 15:35, Steve Totaro wrote:
Anyone have a script or example to do this? I want to run the script
every so often and then drop a call file into the outgoing folder and
call multiple numbers. I figured I would ask before wasting my time
re-inventing the wheel.
Also, I dont want
- Original Message -
From: Martin Joseph [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, August 14, 2006 2:14 PM
Subject: [asterisk-users] Re: SIP Connection Problems
On 2006-08-13 06:23:29 -0700, Barry Fawthrop [EMAIL PROTECTED]
said:
Thanks Dovid
I have
Greeting Everyone,
I don't have access to Asterisk box right now or I'd check this myself...
If my client phone uses g.711 (alaw) and my outbound trunk leaving
asterisk uses g.711 (ulaw), will asterisk have to transcode? If so is
the processing overhead much?
regards,
Dave
Here's an interesting situation; please let me know if you have any insight on it.I am wanting to setup Asterisk with spandsp and Asterfax on a Dapper Drake box. Have any of you had any success with this? Any idea if the Synaptic packages will allow for easy installation and setup?
Thanks,David
I've got a strange issue with SNOM's and Asterisk v.1.2.10
[EMAIL PROTECTED] ~]# asterisk -rx show version
Asterisk 1.2.10 built by root @ comp on a i686 running Linux on
2006-07-24 23:42:12 UTC
Verbosity is at least 10
Core debug is at least 1
My SNOM's are a mixture of 360's and 320's.
J. Oquendo wrote:
I've got a strange issue with SNOM's and Asterisk v.1.2.10
[EMAIL PROTECTED] ~]# asterisk -rx show version
Asterisk 1.2.10 built by root @ comp on a i686 running Linux on
2006-07-24 23:42:12 UTC
Verbosity is at least 10
Core debug is at least 1
My SNOM's are a mixture of
Is there a manger
command that will reload these two configs, something like extensions reload, so
it doesn't drop calls in progress.
Jordan Novak
Senior TelecommunicationsEngineer
Logistics Health
Inc.
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I think it's just "reload"On Aug 14, 2006, at 3:23 PM, Jordan Novak wrote: Is there a manger command that will reload these two configs, something like extensions reload, so it doesn't drop calls in progress. Jordan Novak Senior Telecommunications Engineer Logistics Health
Are you using mixmonitor ?
Alyed Tzompa a écrit :
Hi there!
I'm having lots of problems with an Asterisk used by a customer. Got
hundreds (yes hundreds, about 3-4 per minute) of this messages every hour:
WARNING[12685] channel.c: Avoided initial deadlock for '0x96dee78', 10
retries!
The
Ok,
I test the Asterisk time using SayUnixTime and see the following:
-- Executing SayUnixTime(SIP/1000-0822ec80, ||ABdY 'digits/at'
IMp) in new stack
-- Playing 'digits/day-1' (language 'fr')
-- Playing 'digits/mon-7' (language 'fr')
-- Playing 'digits/14' (language 'fr')
--
I know I could do that, and I do for some instances, but the biggest point
of writing the macros (at least in my case) is to reduce typing etc...
Murf and I have uncovered a bug relating to this. Macros can call macros
just fine pre-dial, but once there's been a hangup, we've discovered that
Hi,
Is there a API or framework available to write solution in the Java programming language for Asterisk?
Functionality:
* Manage Asterisk calls include call forward etc.
* Receive events if there is for example an incoming call etc.
* Manage Asterisk extent ions etc.
* etc...
E.g. I
Anyone using pyAst?
There's absolutely no docs.
It doesn't seem to
work anyway...
foo =
asterisk.manager.Manager()foo.connect("pbx1.xxx.com",5038)foo.login("callout","password")res
= foo.command("Show Channels")print res
this
yields:
[14:[EMAIL PROTECTED]:~]# ./test1.py Follows
I have a Cisco 7961 up and running quite nicely with Asterisk utilizing
Asterisk 1.2.10 and Cisco's SIP image SIP41.8-0-2SR1S on the 7961 phones.
All my 'hints' are properly defined and I have a speeddial extension setup
on the 7961 of one of the other phones, however I do not see and presence or
On 8/14/06, Jordan Novak [EMAIL PROTECTED] wrote:
Is there a manger command that will reload these two configs, something like
extensions reload, so it doesn't drop calls in progress.
reload app_queue.so
reload chan_agent.so
Doing these reloads should not drop calls in progress. If it
Hi,
I found:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
Is there more such examples or tutorials available that show me how to control Asterisk using PHP?
I want to be able to have full control over Asterisk using PHP.
For example:
* Execute PHP code when there's an incoming
Sorry but I don't follow you. Where in the configuration is there a
dkey_retrieve option? I've never seen it and I checked preferences,
function keys, speed dials, etc. Can't find this option.
If you mean line configuration, this is how I have it set up
Configuration Line 1
Login Information:
Yep, it's called Asterisk-Java. http://asterisk-java.sourceforge.net/It's a java API for the manager interface.Alex
On 8/14/06, Lennie De Villiers [EMAIL PROTECTED] wrote:
Hi,
Is there a API or framework available to write solution in the Java programming language for Asterisk?
Hi,
I found:
http://www.voip-info.org/wiki/view/Asterisk+AGI+php
Is there more such examples or tutorials available that show me how to control Asterisk using PHP?
I want to be able to have full control over Asterisk using PHP.
For example:
* Execute PHP code when there's an incoming
I guess you didnt even write in google asterisk java before sending a question to the list right?http://asterisk-java.sourceforge.net/
On 8/14/06, Lennie De Villiers [EMAIL PROTECTED] wrote:
Hi,
Is there a API or framework available to write solution in the Java programming
On 8/11/06, Ralph Liebessohn [EMAIL PROTECTED] wrote:
On 7/24/06, Thomas Laurids Pedersen [EMAIL PROTECTED] wrote:
I have the same card, but in my zaptel.conf I have the following linespan=1,1,0,hdb3,crc4as you can see from the status your line is down.BR Thomas Lincoln Zuljewic
Silva
Hello
We (Intervoice Solutions Company, http://www.ivsol.net/) are about to release as free open source, a PHP router daemon that does just that, but requires a patch to asterisk called MAGI. Contact me off-list if you are interested.
I call it free open source since i havent had the details about the
Hi Dovid,
I can't see how to easily do that in [EMAIL PROTECTED] :-(
Any ideas?
Paul
- Original Message -
From: Dovid Bender [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 13, 2006 3:44 PM
Subject:
I'm trying to provision some spa-942 phones via TFTP. The phones get
their address from a dhcp server which sends it option 66 (address of
the tftp server). After spending some time with the phones and even
breaking down to sniff traffic from the phones I see that they are not
requesting their
Julian Lyndon-Smith wrote:
Is there any way of specifying a directory to load tftp files from
instead of from the root tftp directory when booting a cisco 7960 phone ?
SIPDefault.cnf:
# TFTP Phone Specific Configuration File Directory
tftp_cfg_dir: ./7960/ ; Example: ./sip_phone/
Andre Courchesne - Consultant wrote:
[EMAIL PROTECTED] tmp]# date
Mon Aug 14 16:44:15 EDT 2006
The Linux command line time is connect, but not Asterisk...
just guessing... not sure:
date -u
is showing what?
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Hi,
did anyone try do load-testing on asterisk,for sip channel calls?
I want to have a rough estimate about - how many calls, an asterisk server, running on say dual 240 opteron with 1 GB memory, can handle?
Also how much internet bandwidth does a typical call requires? I heard around 20Kbps with
I thought I'd bounce
this around here before I opened a bug.
PhoneA makes a
call to phone B.Phone B is still registered, but is physically turned
off.
Asterisk takes the
INVITE message from phoneA.
Now,
1) It sends RINGING
back to phone A before it has even sent an INVITE to phone B.
Hi,
What would you suggest I use?
I'm a software developer with 5 years experienceand can basically use anything.
Kind Regards,
Lennie De Villiers
---Original Message---
From: Rushowr
Date: 2006/08/14 11:37:40 PM
To: 'Lennie De Villiers'; 'Asterisk Users Mailing List -
Jeremiah Millay wrote:
I'm trying to provision some spa-942 phones via TFTP. The phones get
their address from a dhcp server which sends it option 66 (address of
the tftp server). After spending some time with the phones and even
breaking down to sniff traffic from the phones I see that they
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Mon, 14 Aug 2006 19:28:11 -0300
Subject: [asterisk-users] Sending INVITE to
an unavailable phone - Bug?
I thought I'd bounce
i couldn't find any docs on it either, found some scripts that just use pythonto communicate with asterisk. Thats kinda why i have been eying ruby. However, i did get some stuff to work from some examples on the net, if you like i can send them to you.
-shawnOn 8/14/06, Douglas Garstang [EMAIL
I am having a weird issue with my zap channel (Digium TDM01B).
Randomly it appears that the POTS line is not seeing all of the digits passed. We
have to dial a 1 and the area code to call most numbers here, and we get the
error that we need to dial a 1 and the area code when dialing this
Lennie: Tomorrow in the morning I will be talking with the people that will decide when to make the release and how handle this. I have your email, you will have news from me soon :)Regards
On 8/14/06, Lennie De Villiers [EMAIL PROTECTED] wrote:
Hi,
What would you suggest I use?
I'm
I do it all the time thru http. are you sure tftp thru DHCP option 66
is supported?
I usualy have to log in at least once for each phone into the web
interface and add the url.
What does your DHCP option 66 contain?
Of course the last question only deserves an answer if the first one is yes.
date -u
shows:
[EMAIL PROTECTED] tmp]# date -u
Mon Aug 14 23:15:08 UTC 2006
[EMAIL PROTECTED] tmp]# date
Mon Aug 14 19:15:13 EDT 2006
Ok, so Asterisk uses UTC...
Date: Mon, 14 Aug 2006 19:19:24 -0300
From: Hermann Wecke [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Asterisk time not the
Urban wrote:
Hi,
queue announcements works when we use music on hold in the queue, but
if we use ringback e.g. queue(myqueue|r|) the announcments and
hold time are not working, it seems that * is not even trying to read
the queueu-announcment files. Is this by design, or is there a work
Ok, what's the deal with qualify in sip.conf. The docs on the voip wiki at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify
state that it can take either yes, no, of a number which represents how long in
milliseconds between polling. I set it to 1000, (ie qualify=1000), did a
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Mon, 14 Aug 2006 20:24:35 -0300
Subject: [asterisk-users] SIP Qualify
Ok, what's the deal with qualify in sip.conf. The docs
AGI+PHP would be a good
place to do all of this. However, be aware that interpreted code such as PHP
incurs a performance hit and may not be suitable for very large installations,
in addition to the issue of passing call control away from Asterisk in general.
(ref: "Asterisk Performance",
On Mon, Aug 14, 2006 at 03:02:45PM -0500, David R. wrote:
Here's an interesting situation; please let me know if you have any insight
on it.
I am wanting to setup Asterisk with spandsp and Asterfax on a Dapper Drake
box. Have any of you had any success with this? Any idea if the Synaptic
Kai Fürstenberg and Patrick, who has assigned you the policemen duty on this list? If you don't have manners to talk politely, better keep quite. (if I'll have to talk to you your way, you'll not like it)
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Hi,
It doesn't seem that anybody is moderating digium's mailing lists, thats why some uncivilized people with no manners to talk keep making this list dirty. Recently I've noticed increase in irresponsibly typed and rudly answered messages. If there are moderators here, they should stop it and
After much playing and getting nowhere, I was on the phone to the guys
from www.voipshop.com.au and mentioned that the pri dropout problem was
occuring and if they had any solutions.
Immediately they mentioned something that causes a problem in australia.
On longdistance phone calls (sometimes)
There is no Debian package currently available for AsterFax. We do have
plans to make a full apt based install available for Debian similar to the
yum based one we now have for CentOS/RedHat.
You have a couple of options in the mean time:
1. Install AsterFax manually using the installation
Hi Kevin
Yes you shouldn't need the busy detect in there for digital systems.
Busydetect is only really useful on analog systems which can't detect
incoming hangup.
Regards
Mark Brooker
T: 02 4959 8670
M: 0415 846 865
F: 02 4950 5609
E: [EMAIL PROTECTED]
W: http://www.mbit.com.au
Hi,
Is there any variable set that would indicate the reason why an call
initiated by a call-file hit the failed extension?
Thanks,
Andre
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Hi:
I am configuring an Asterisk server with all inbound only lines on a PRI
using a T100 card. The card works and Asterisk sees all the lines. My
problem is that Asterisk refuses to answer any of the lines unless there
is an extension defined for it in extensions.conf. I want all
In that case, if I set qualify=1000, and it still polls every 60s, then how can
it consider it unreachable at 1000ms?
Doug.
-Original Message-
From: Joshua Colp [mailto:[EMAIL PROTECTED]
Sent: Mon 8/14/2006 1:42 PM
To: Asterisk Users Mailing List -
Does anyone have a listing on file/directories that asterisk needs
ownership of to run as a user other than root?
I know about the major items --- /etc/asterisk, /var/spool/asterisk/,
/var/lib/asterisk, etc... Anyone have a script to fix all the
directories?
Thanks in advance.
FB
If you're gonna top post...so am I.
I think you misunderstand what qualify is/does. It appears that you believe
that qualify=1000 means that it'll send out a qualify packet every 1000ms.
This isn't an unreasonable assumption, but it is wrong. The qualify=1000 means
that Asterisk will wait
On Monday 14 August 2006 21:18, Rich Adamson wrote:
Shaun Hofer wrote:
It wasn't any help. It doesn't give any reference to order of trunks, etc
in sip.conf. I'm still looking for the post, Rich Adamson made reference
too...
On Friday 11 August 2006 18:24, Fran Oliveira wrote:
see
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