[Asterisk-Users] High latency from Europe, 500-800ms.
We're using a 7940 from Europe, connecting to a US Asterisk server, and it works great. We setup a local Asterisk server in Europe, had the 7940 connect to it, and used IAX2/GSM to connect to the US. It is choppy using all CODECS, and I am curious if there are any recommendations on getting this to work well? I'd rather not have the phones connect directly to the US. Thanks. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High latency from Europe, 500-800ms.
Thanks for info, but this didn't seem to help. Just to clarify, we are in south-eastern Europe, and obtaining the Net via satellite. As stated, the 7940 on the same network, connecting to the US, it works great, but when we have the 7940 connect to the local Asterisk server, which then connects to the US, it is very choppy. Maybe the Cisco's have inherently better CODEC support? Maybe the one additional hop with the local Asterisk server is making the difference? Bill >There is a misprint in the IAX config file..an extract from my file is >as follows: > >jitterbuffer=yes >dropcount=4 >maxjitterbuffer=500 >maxexcessbuffer=300 > >You should enable jitterbuffer (disabled by default) and there is a typo > in 'maxexcessbuffer' in the default files. > >I am in the UK and the above config fixed my choppy audio to the US. I >was able to run a-law across the atlantic. Sounded fantastic! > >Rgds >Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] High latency from Europe, 500-800ms.
Maybe I'm not articulating myself well. The 7940 on the same network in Europe *works great*, no problems, sound is perfect, even with the higher latency. If I take that 7940 and have it connect to a *local* Asterisk server, which connects to the states, it sucks. The 7940 though, connecting directly to the states, works great. Bill >Not all sat connections are one way. But the issue with sat connections >is *drumroll* latency! >As the signal is beeing relayed over the sattelite this will cause >latency. Also if the sat service is not >providing enough downstream it's bad too. > >I would definately look into getting your network straighend out first. >There are many factors. >Is your connection shared? What speeds? > >Let say it like that if you have people on your local lan using bandwith >or running peer 2 peer >filesharing stuff this will take away your upstream speed. Do some tests. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MySQL, CDR with MySQL
I'm preparing to roll out Asterisk for the voicemail portion of my VOIP network. This week I downloaded a fresh version from CVS of Asterisk and installed the following MySQL 4.1.7 RPMs directly from Mysql.orgFor some reason after I enable MySQL for CDR and Voicemail in the cdr_mysql.conf and voicemail.conf I don't get any MySQL functionality at all. It almost seems as though MySQL support isn't even being compiled into Asterisk. I found somewhere that the Z Library was required and that is already installed. Can someone clue me in? MySQL-client-4.1.7-0.i386.rpm MySQL-devel-4.1.7-0.i386.rpm MySQL-server-4.1.7-0.i386.rpm MySQL-shared-4.1.7-0.i386.rpm MySQL-shared-compat-4.1.7-0.i386.rpm Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions.conf using TO field and not INVITEfield of SIP header
I believe I found the answer to my question and it seems to be working
properly. I can use the TO field of the SIP header the way I need to by
using the following style in my extensions.conf file.
exten => ${RDNIS},1,Wait,1
exten => ${RDNIS},2,Voicemail(${RDNIS})
exten => ${RDNIS},3,Hangup()
Bill
- Original Message -
From: VCI Help Desk
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thursday, December 09, 2004 11:09 AM
Subject: [Asterisk-Users] extensions.conf using TO field and not INVITEfield
of SIP header
Is there a way to get the extensions.conf file to use the To: field of a
SIP header instead of the INVITE field? If so, can you point me in the right
direction?
Bill
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Re: [Asterisk-Users] MySQL
I've got it working better. I started using a different user with a password instead of using root with no password and I see it trying in the /var/log/mysql.log file. I also removed the old dbuser,dbpass, etc.. lines from the voicemail.conf file. SELECT * FROM users WHERE mailbox = '540' AND context = 'default' Now I don't see where this 'default' context is coming from because I have that set to 'from-sip' in my extentions.conf and my sip.conf files. Any ideas where that is? extconfig.conf -voicemail => mysql,asterisk,users (I renamed the table to "users" and created it as such) res_mysql.conf [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = test dbport = 3306 dbsock = /var/lib/mysql/mysql.sock The res_config_mysq.so is installed in the modules directory with the others. What exactly is this new "RealTime" stuff? Is this an existing package that's being integrated into Asterisk or something made from scratch? Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 10, 2004 10:02 AM Subject: Re: [Asterisk-Users] MySQL Post your extconfig.conf. Do you have /usr/lib/asterisk/modules/res_config_mysql.so installed? Do you have /etc/asterisk/res_mysql.conf? -Matthew - Original Message - From: "Bill" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, December 10, 2004 8:14 AM Subject: Re: [Asterisk-Users] MySQL > Matthew, > > I followed these instructions this morning and something about it's not > working. I was using the voicemail.conf before and I could login and > everything worked ok but now I can't login at all. One thing that did get > fixed was the CDR records when I did the "make install" from the > asterisk-addons folder. > > In my /etc/my.conf file I have "log=/var/log/mysqld.log" so I can watch > what MySQL does. So far it never tries to do anything with the voicemail. > That's how I noticed the CDR records started working. > > Any ideas? > > Bill > > > > > - Original Message - > From: Matthew Boehm > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Thursday, December 09, 2004 5:29 PM > Subject: Re: [Asterisk-Users] MySQL > > > Sure. (I really need to write a wiki on this.) > > You have two choices here before we start. You can use RealTime one of 2 > ways: ODBC or direct MySQL. Currently these are the only two supported > methods. > > Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm > going to instruct on how to use/install it. > > The RealTime MySQL driver can be found inside asterisk-addons. Just do the > standard make, make install. > > Now copy asterisk-addons/configs/res_mysql.conf.sample to > /etc/asterisk/res_mysql.conf (or whereever your conf dir is). > > Edit the res_mysql.conf to your liking. > > Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime > config stuff. If you want voicemail, add this line: > > voicemail => mysql,asterisk,voicemail_users > > This basically says "Please use the RealTime MySQL driver, the database > asterisk and the table voicemail_users and bind that to the voicemail > family". You can change to your liking: > > voicemail => ,, > > Now go into your mysql server and make the following table: > > CREATE TABLE `voicemail_users` ( > `uniqueid` int(11) NOT NULL auto_increment, > `customer_id` int(11) NOT NULL default '0', > `context` varchar(50) NOT NULL default '', > `mailbox` int(5) NOT NULL default '0', > `password` int(4) NOT NULL default '0', > `fullname` varchar(50) NOT NULL default '', > `email` varchar(50) NOT NULL default '', > `pager` varchar(50) NOT NULL default '', > `options` varchar(100) NOT NULL default '', > `stamp` timestamp(14) NOT NULL, > PRIMARY KEY (`uniqueid`) > ) TYPE=MyISAM; > > Put in some rows. Restart asterisk and it should work. Please let me know if > it works/doesn't work. > > -Matthew > > - Original Message - > From: "VCI Help Desk" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 09, 2004 4:44 PM > Subject: [Asterisk-Users] MySQL > > > > Does anyone have any instructions for setting up MySQL with the
Re: [Asterisk-Users] MySQL
I am using the CVS from 2 days ago. The file is there in asterisk-addons. It may not be used but until you sent those instructions a few hours ago there wasn't much of anything to explain how to use the new stuff. I have been using the documentation that I see everywhere that refers to using VoiceMail with MySQL. If it's outdated then those old pages need to be corrected/removed after you make the new documentation this weekend. Any ideas where the "default" context may be coming from in the MySQL statement mentioned below? The only context I have specified in sip.conf and extensions.conf is called "from-sip". Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 10, 2004 12:13 PM Subject: Re: [Asterisk-Users] MySQL If you are using mysql-vm-routines.h then you are NOT using RealTIme and therefor you can ignore anything I've sent cause I was under the impression that you were using most recent CVS and not 1.0 stable. -Matthew ----- Original Message - From: "Bill" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, December 10, 2004 11:06 AM Subject: Re: [Asterisk-Users] MySQL > I've got it working better. I started using a different user with a > password instead of using root with no password and I see it trying in the > /var/log/mysql.log file. I also removed the old dbuser,dbpass, etc.. lines > from the voicemail.conf file. > > SELECT * FROM users WHERE mailbox = '540' AND context = 'default' > > Now I don't see where this 'default' context is coming from because I > have that set to 'from-sip' in my extentions.conf and my sip.conf files. > Any ideas where that is? > > extconfig.conf -voicemail => mysql,asterisk,users (I renamed the > table to "users" and created it as such) > > res_mysql.conf > [general] > dbhost = 127.0.0.1 > dbname = asterisk > dbuser = asterisk > dbpass = test > dbport = 3306 > dbsock = /var/lib/mysql/mysql.sock > > The res_config_mysq.so is installed in the modules directory with the > others. > > What exactly is this new "RealTime" stuff? Is this an existing package > that's being integrated into Asterisk or something made from scratch? > > Bill > > > > > > - Original Message - > From: Matthew Boehm > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Friday, December 10, 2004 10:02 AM > Subject: Re: [Asterisk-Users] MySQL > > > Post your extconfig.conf. Do you have > /usr/lib/asterisk/modules/res_config_mysql.so installed? Do you have > /etc/asterisk/res_mysql.conf? > > -Matthew > - Original Message - > From: "Bill" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Friday, December 10, 2004 8:14 AM > Subject: Re: [Asterisk-Users] MySQL > > > > Matthew, > > > > I followed these instructions this morning and something about it's > not > > working. I was using the voicemail.conf before and I could login and > > everything worked ok but now I can't login at all. One thing that did get > > fixed was the CDR records when I did the "make install" from the > > asterisk-addons folder. > > > > In my /etc/my.conf file I have "log=/var/log/mysqld.log" so I can > watch > > what MySQL does. So far it never tries to do anything with the voicemail. > > That's how I noticed the CDR records started working. > > > > Any ideas? > > > > Bill > > > > > > > > > > - Original Message - > > From: Matthew Boehm > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Sent: Thursday, December 09, 2004 5:29 PM > > Subject: Re: [Asterisk-Users] MySQL > > > > > > Sure. (I really need to write a wiki on this.) > > > > You have two choices here before we start. You can use RealTime one of 2 > > ways: ODBC or direct MySQL. Currently these are the only two supported > > methods. > > > > Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm > > going to instruct on how to use/install it. > > > > The RealTime MySQL driver can be found inside asterisk-addons. Just do the > > standard make, make install. > > > > Now copy asterisk-addons/configs/res_mysql.co
Re: [Asterisk-Users] MySQL
The only references I have to the context in voicemail.conf is pretty
much blank. Most of this file is untouched so far since I am trying to use
the MySQL. I have the following in voicemail.conf but the mailboxes are
commented out so I can test the MySQL. Am I supposed to have a reference to
a context using the MySQL somehow in this file? If so, how?
[from-sip]
;0060 => ,Test Mailbox,[EMAIL PROTECTED]
Bill
- Original Message -
From: Matthew Boehm
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, December 10, 2004 1:45 PM
Subject: Re: [Asterisk-Users] MySQL
> Any ideas where the "default" context may be coming from in the MySQL
> statement mentioned below? The only context I have specified in sip.conf
and
> extensions.conf is called "from-sip".
Yea, what is the context stored in your voicemail.conf file?
Matthew
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Re: [Asterisk-Users] MySQL
Thanks for the help so far.
You mean this line in the app_voicemail.c file?
var = ast_load_realtime("voicemail", "mailbox", mailbox, "context",
retval->context, NULL);
I assume this is the CVS version you are referring to? If so, this
version also has the "mysql-vm-routines.h" in asterisk-addons.
This is my extconfig.conf file. Most of it is defaulted.
;
; Static configuration files:
;
; file.conf => driver,database[,table]
;
; maps a particular configuration file to the given
; database driver, database and table (or uses the
; name of the file as the table if not specified)
;
;uncomment to load queues.conf via the odbc engine.
;
;queues.conf => odbc,asterisk,ast_config
;
; Realtime configuration engine
;
; maps a particular family of realtime
; configuration to a given database driver,
; database and table (or uses the name of
; the family if the table is not specified
;
;iaxfriends => odbc,asterisk
;sipfriends => odbc,asterisk
voicemail => mysql,asterisk,users
;extensions => odbc,asterisk
Bill
- Original Message -
From: Matthew Boehm
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, December 10, 2004 4:10 PM
Subject: Re: [Asterisk-Users] MySQL
Now I am totally lost. Do you want to use old vm-routines or do you want to
use RealTime? If RealTime, then you need to be using most recent CVS. Verify
this by searching app_voicemail for the phrase "ast_load_realtime". If you
find it, good. If not. Update.
Now, send me your extconfig,conf. Are you trying to use the MySQL RealTime
driver inside asterisk-addons or are you using ODBC=>MySQL?
-Matthew
- Original Message -
From: "Bill" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Friday, December 10, 2004 1:59 PM
Subject: Re: [Asterisk-Users] MySQL
> The only references I have to the context in voicemail.conf is pretty
> much blank. Most of this file is untouched so far since I am trying to use
> the MySQL. I have the following in voicemail.conf but the mailboxes are
> commented out so I can test the MySQL. Am I supposed to have a reference
to
> a context using the MySQL somehow in this file? If so, how?
>
> [from-sip]
> ;0060 => ,Test Mailbox,[EMAIL PROTECTED]
>
> Bill
>
>
>
>
> - Original Message -
> From: Matthew Boehm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Friday, December 10, 2004 1:45 PM
> Subject: Re: [Asterisk-Users] MySQL
>
>
> > Any ideas where the "default" context may be coming from in the
MySQL
> > statement mentioned below? The only context I have specified in sip.conf
> and
> > extensions.conf is called "from-sip".
>
> Yea, what is the context stored in your voicemail.conf file?
>
> Matthew
>
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[Asterisk-Users] Voicemail and MySQL
I have Asterisk talking to MySQL using Realtime but for some reason I keep getting the wrong context used when Realtime makes the MySQL call. I can see this in my /var/log/mysql.log file. Because of this I can't login to VoicemailMain from my X-Ten phone. I can login if I statically configure the voicemail user in voicemail.conf but I prefer the MySQL. SELECT * FROM users WHERE mailbox = '0063' AND context = 'default' In my sip.conf file I have the default settings except the default context is set. I removed all the example SIP configs further down the config. [general] context=from-sip ;Default context for incoming calls In my extensions.conf file I have the following. All example extension configs have been removed. [from-sip] exten => 8500,1,VoicemailMain exten => 8500,n,Hangup What am I doing wrong? Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
Matthew, I followed these instructions this morning and something about it's not working. I was using the voicemail.conf before and I could login and everything worked ok but now I can't login at all. One thing that did get fixed was the CDR records when I did the "make install" from the asterisk-addons folder. In my /etc/my.conf file I have "log=/var/log/mysqld.log" so I can watch what MySQL does. So far it never tries to do anything with the voicemail. That's how I noticed the CDR records started working. Any ideas? Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 09, 2004 5:29 PM Subject: Re: [Asterisk-Users] MySQL Sure. (I really need to write a wiki on this.) You have two choices here before we start. You can use RealTime one of 2 ways: ODBC or direct MySQL. Currently these are the only two supported methods. Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm going to instruct on how to use/install it. The RealTime MySQL driver can be found inside asterisk-addons. Just do the standard make, make install. Now copy asterisk-addons/configs/res_mysql.conf.sample to /etc/asterisk/res_mysql.conf (or whereever your conf dir is). Edit the res_mysql.conf to your liking. Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime config stuff. If you want voicemail, add this line: voicemail => mysql,asterisk,voicemail_users This basically says "Please use the RealTime MySQL driver, the database asterisk and the table voicemail_users and bind that to the voicemail family". You can change to your liking: voicemail => ,, Now go into your mysql server and make the following table: CREATE TABLE `voicemail_users` ( `uniqueid` int(11) NOT NULL auto_increment, `customer_id` int(11) NOT NULL default '0', `context` varchar(50) NOT NULL default '', `mailbox` int(5) NOT NULL default '0', `password` int(4) NOT NULL default '0', `fullname` varchar(50) NOT NULL default '', `email` varchar(50) NOT NULL default '', `pager` varchar(50) NOT NULL default '', `options` varchar(100) NOT NULL default '', `stamp` timestamp(14) NOT NULL, PRIMARY KEY (`uniqueid`) ) TYPE=MyISAM; Put in some rows. Restart asterisk and it should work. Please let me know if it works/doesn't work. -Matthew - Original Message - From: "VCI Help Desk" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 09, 2004 4:44 PM Subject: [Asterisk-Users] MySQL > Does anyone have any instructions for setting up MySQL with the latest > CVS? I upgraded from an older version this week and none of the MySQL works > now and I believe it's due to the newer Realtime Architecture. I can't find > any instructions that explain it very well anywhere. Any help would be > appreciated. > > Bill > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
I just noticed something regarding my last post about the context issue with the MySQL and Voicemail. In mysql-vm-routines.h it looks like the context value in the SQL statement is hardcoded so no matter what my context is it'll never work unless I change my context to 'default'. Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 10, 2004 10:02 AM Subject: Re: [Asterisk-Users] MySQL Post your extconfig.conf. Do you have /usr/lib/asterisk/modules/res_config_mysql.so installed? Do you have /etc/asterisk/res_mysql.conf? -Matthew - Original Message ----- From: "Bill" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, December 10, 2004 8:14 AM Subject: Re: [Asterisk-Users] MySQL > Matthew, > > I followed these instructions this morning and something about it's not > working. I was using the voicemail.conf before and I could login and > everything worked ok but now I can't login at all. One thing that did get > fixed was the CDR records when I did the "make install" from the > asterisk-addons folder. > > In my /etc/my.conf file I have "log=/var/log/mysqld.log" so I can watch > what MySQL does. So far it never tries to do anything with the voicemail. > That's how I noticed the CDR records started working. > > Any ideas? > > Bill > > > > > - Original Message - > From: Matthew Boehm > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Thursday, December 09, 2004 5:29 PM > Subject: Re: [Asterisk-Users] MySQL > > > Sure. (I really need to write a wiki on this.) > > You have two choices here before we start. You can use RealTime one of 2 > ways: ODBC or direct MySQL. Currently these are the only two supported > methods. > > Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm > going to instruct on how to use/install it. > > The RealTime MySQL driver can be found inside asterisk-addons. Just do the > standard make, make install. > > Now copy asterisk-addons/configs/res_mysql.conf.sample to > /etc/asterisk/res_mysql.conf (or whereever your conf dir is). > > Edit the res_mysql.conf to your liking. > > Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime > config stuff. If you want voicemail, add this line: > > voicemail => mysql,asterisk,voicemail_users > > This basically says "Please use the RealTime MySQL driver, the database > asterisk and the table voicemail_users and bind that to the voicemail > family". You can change to your liking: > > voicemail => ,, > > Now go into your mysql server and make the following table: > > CREATE TABLE `voicemail_users` ( > `uniqueid` int(11) NOT NULL auto_increment, > `customer_id` int(11) NOT NULL default '0', > `context` varchar(50) NOT NULL default '', > `mailbox` int(5) NOT NULL default '0', > `password` int(4) NOT NULL default '0', > `fullname` varchar(50) NOT NULL default '', > `email` varchar(50) NOT NULL default '', > `pager` varchar(50) NOT NULL default '', > `options` varchar(100) NOT NULL default '', > `stamp` timestamp(14) NOT NULL, > PRIMARY KEY (`uniqueid`) > ) TYPE=MyISAM; > > Put in some rows. Restart asterisk and it should work. Please let me know if > it works/doesn't work. > > -Matthew > > - Original Message - > From: "VCI Help Desk" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 09, 2004 4:44 PM > Subject: [Asterisk-Users] MySQL > > > > Does anyone have any instructions for setting up MySQL with the latest > > CVS? I upgraded from an older version this week and none of the MySQL > works > > now and I believe it's due to the newer Realtime Architecture. I can't > find > > any instructions that explain it very well anywhere. Any help would be > > appreciated. > > > > Bill > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-
Re: [Asterisk-Users] MySQL - mistake in previous post
Belay that. I was looking at the wrong SQL statement. Bill - Original Message - From: Bill To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 10, 2004 11:10 AM Subject: Re: [Asterisk-Users] MySQL I just noticed something regarding my last post about the context issue with the MySQL and Voicemail. In mysql-vm-routines.h it looks like the context value in the SQL statement is hardcoded so no matter what my context is it'll never work unless I change my context to 'default'. Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 10, 2004 10:02 AM Subject: Re: [Asterisk-Users] MySQL Post your extconfig.conf. Do you have /usr/lib/asterisk/modules/res_config_mysql.so installed? Do you have /etc/asterisk/res_mysql.conf? -Matthew - Original Message ----- From: "Bill" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Friday, December 10, 2004 8:14 AM Subject: Re: [Asterisk-Users] MySQL > Matthew, > > I followed these instructions this morning and something about it's not > working. I was using the voicemail.conf before and I could login and > everything worked ok but now I can't login at all. One thing that did get > fixed was the CDR records when I did the "make install" from the > asterisk-addons folder. > > In my /etc/my.conf file I have "log=/var/log/mysqld.log" so I can watch > what MySQL does. So far it never tries to do anything with the voicemail. > That's how I noticed the CDR records started working. > > Any ideas? > > Bill > > > > > - Original Message - > From: Matthew Boehm > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Thursday, December 09, 2004 5:29 PM > Subject: Re: [Asterisk-Users] MySQL > > > Sure. (I really need to write a wiki on this.) > > You have two choices here before we start. You can use RealTime one of 2 > ways: ODBC or direct MySQL. Currently these are the only two supported > methods. > > Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm > going to instruct on how to use/install it. > > The RealTime MySQL driver can be found inside asterisk-addons. Just do the > standard make, make install. > > Now copy asterisk-addons/configs/res_mysql.conf.sample to > /etc/asterisk/res_mysql.conf (or whereever your conf dir is). > > Edit the res_mysql.conf to your liking. > > Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime > config stuff. If you want voicemail, add this line: > > voicemail => mysql,asterisk,voicemail_users > > This basically says "Please use the RealTime MySQL driver, the database > asterisk and the table voicemail_users and bind that to the voicemail > family". You can change to your liking: > > voicemail => ,, > > Now go into your mysql server and make the following table: > > CREATE TABLE `voicemail_users` ( > `uniqueid` int(11) NOT NULL auto_increment, > `customer_id` int(11) NOT NULL default '0', > `context` varchar(50) NOT NULL default '', > `mailbox` int(5) NOT NULL default '0', > `password` int(4) NOT NULL default '0', > `fullname` varchar(50) NOT NULL default '', > `email` varchar(50) NOT NULL default '', > `pager` varchar(50) NOT NULL default '', > `options` varchar(100) NOT NULL default '', > `stamp` timestamp(14) NOT NULL, > PRIMARY KEY (`uniqueid`) > ) TYPE=MyISAM; > > Put in some rows. Restart asterisk and it should work. Please let me know if > it works/doesn't work. > > -Matthew > > - Original Message - > From: "VCI Help Desk" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Thursday, December 09, 2004 4:44 PM > Subject: [Asterisk-Users] MySQL > > > > Does anyone have any instructions for setting up MySQL with the latest > > CVS? I upgraded from an older version this week and none of the MySQL > works > > now and I believe it's due to the newer Realtime Architecture. I can't > find > > any instructions that explain it very well anywhere. Any help would be > > appreciated. > > > > Bill > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBS
Re: [Asterisk-Users] MySQL
Same here. I've deleted and re-installed asterisk a few times and the RealTime voicemail never works. The best I've gotten is the MySQL query to execute with the wrong context. When I use "cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds" to download the latest version the "res_mysql.conf.sample" isn't even there. I made it from scratch but it still doesn't work. If that file isn't there what else is missing? Bill - Original Message - From: Greg - Cirelle Enterprises To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, December 12, 2004 12:50 PM Subject: Re: [Asterisk-Users] MySQL At 06:29 PM 12/9/04, you wrote: >Sure. (I really need to write a wiki on this.) > >You have two choices here before we start. You can use RealTime one of 2 >ways: ODBC or direct MySQL. Currently these are the only two supported >methods. > >Since I don't use ODBC and as the author of the MySQL RealTime driver, I'm >going to instruct on how to use/install it. > >The RealTime MySQL driver can be found inside asterisk-addons. Just do the >standard make, make install. > >Now copy asterisk-addons/configs/res_mysql.conf.sample to >/etc/asterisk/res_mysql.conf (or whereever your conf dir is). > >Edit the res_mysql.conf to your liking. > >Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime >config stuff. If you want voicemail, add this line: > >voicemail => mysql,asterisk,voicemail_users No such file res_mysql.conf only cdr_mysql_conf.sample Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MySQL
Ok, I believe the misunderstanding involves the use of CVS itself. When I talk about CVS I am referring to using the CVS method of downloading Asterisk vice FTP'ing a GZ file. I was not aware that you were referring to a version named "CVS". Are there any others besides CVS and STABLE. When someone downloads using "cvs checkout -r v1-0 " what version is that, CVS or stable? Bill - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, December 13, 2004 9:49 AM Subject: Re: [Asterisk-Users] MySQL Even though you can...why would you? You can't use some things that are in CVS addons with STABLE asterisk. res_config_mysql.c and res_mysql.conf are part of the CVS version of asterisk. This means that you cannot use them with STABLE. If you want RealTime functionality you HAVE to upgrade your entire asterisk code to CVS. -Matthew - Original Message - From: "VCI Help Desk" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, December 13, 2004 9:21 AM Subject: Re: [Asterisk-Users] MySQL > What's the proper way to download a STABLE version of asterisk and > asterisk-addons from CVS? I keep finding documentation that says two > different ways of download it. > > Now that I've downloaded the asterisk-addons that has the > "res_mysql.conf.sample" it won't compile. If I cd to asterisk-addons and do > a "make clean; make" I get the following. This used to work fine before. > > res_config_mysql.c: In function `load_module': > res_config_mysql.c:467: error: structure has no member named `static_func' > res_config_mysql.c:468: error: structure has no member named `realtime_func' > res_config_mysql.c:469: error: structure has no member named `update_func' > res_config_mysql.c:470: error: structure has no member named > `realtime_multi_func' > make: *** [res_config_mysql.o] Error 1 > rm app_saycountpl.o > > The "mysql-vm-routines.h" is still there as well. I thought that file > was removed. > > Bill > > > > > - Original Message - > From: Matthew Boehm > To: Asterisk Users Mailing List - Non-Commercial Discussion > Sent: Monday, December 13, 2004 9:09 AM > Subject: Re: [Asterisk-Users] MySQL > > > You are missing the fact that RealTime is not 1-0, its CVS. 'Thats' why > res_mysql.conf isn't even there. > > -Matthew > > - Original Message - > From: "Bill" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[EMAIL PROTECTED]> > Sent: Monday, December 13, 2004 8:32 AM > Subject: Re: [Asterisk-Users] MySQL > > > > Same here. I've deleted and re-installed asterisk a few times and the > > RealTime voicemail never works. The best I've gotten is the MySQL query to > > execute with the wrong context. When I use "cvs checkout -r v1-0 zaptel > > libpri asterisk asterisk-addons asterisk-sounds" to download the latest > > version the "res_mysql.conf.sample" isn't even there. I made it from > scratch > > but it still doesn't work. If that file isn't there what else is missing? > > > > Bill > > > > > > > > > > > > - Original Message - > > From: Greg - Cirelle Enterprises > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Sent: Sunday, December 12, 2004 12:50 PM > > Subject: Re: [Asterisk-Users] MySQL > > > > > > At 06:29 PM 12/9/04, you wrote: > > >Sure. (I really need to write a wiki on this.) > > > > > >You have two choices here before we start. You can use RealTime one of 2 > > >ways: ODBC or direct MySQL. Currently these are the only two supported > > >methods. > > > > > >Since I don't use ODBC and as the author of the MySQL RealTime driver, > I'm > > >going to instruct on how to use/install it. > > > > > >The RealTime MySQL driver can be found inside asterisk-addons. Just do > the > > >standard make, make install. > > > > > >Now copy asterisk-addons/configs/res_mysql.conf.sample to > > >/etc/asterisk/res_mysql.conf (or whereever your conf dir is). > > > > > >Edit the res_mysql.conf to your liking. > > > > > >Now edit /etc/asterisk/extconfig.conf. Down at the bottom is the RealTime > > >config stuff. If you want voicemail, add this line: > > >
[Asterisk-Users] Voicemail error message
I've been working on my voicemail server and recently starting getting
this message. Other than change the server name itself I've been making
modifications to the extensions.conf file but everything still functions
properly. Any idea what this is? I've looked at the app_queue.c file but I
didn't get anything out of it.
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/xx/INBOX/msg format:
wav49, 0x87548c0
-- x=1, open writing:
/var/spool/asterisk/voicemail/default/xx/INBOX/msg format: wav,
0x8755088
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en')
Dec 15 09:59:04 WARNING[2509]: app_queue.c:340 changethread: Can't change
device with no technology!
Dec 15 09:59:04 WARNING[2509]: app_queue.c:340 changethread: Can't change
device with no technology!
Dec 15 09:59:04 WARNING[2509]: app_queue.c:340 changethread: Can't change
device with no technology!
-- Executing Hangup("SIP/vci.net-0874c210", "") in new stack
== Spawn extension (from-sip, 8500, 107) exited non-zero on
'SIP/vci.net-0874c210'
Bill
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[Asterisk-Users] Fw: pbx.c:1279 pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 1)
I have added a sip user in sip.conf.
user name is 819,context is c819.
and I have added the follows rows in
extension.conf. like
[c819]
exten => 1,1,Answer
exten => 1,2,SetVal(voicemail=${exten})
exten => 1,3,Dial(SIP/${voicemail})
It appear a error message(pbx.c:1279
pbx_extension_helper: No application 'SetVal' for extension (c819, 1, 2))
when i dial 1 from 819.
The version of asterisk is 1.0.3
Please help me. Thank a lot.
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[Asterisk-Users] asterisk router problem
How can found dynamic dialplan? in extensions.conf [default] exten => 111,1,DBget(aaa=111/forwarding);It can be 2 to 9 begins. exten => 111,2, exten => _[2].,1,Dial(SIP/[EMAIL PROTECTED]) ;AA exten => _[3].,1,Dial(SIP/[EMAIL PROTECTED]) ;BB ... ,how transfer to correct router. namely,it will goto auto AA if it begin 2, else goto BB if it begin 3. Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context of transfer
How set the context of Transfer function? There are 2 context in extensions.conf. [con1] exten => _0.,1,Dial(SIP/[EMAIL PROTECTED]) [con2] exten => 812,1,Transfer(001345566);How can use the dialplan of context con1? Thanks! Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context of transfer
How set the context of Transfer function? There are 2 context in extensions.conf. [con1] exten => _0.,1,Dial(SIP/[EMAIL PROTECTED]) [con2] exten => 812,1,Transfer(001345566);How can use the dialplan of context con1? Thanks! Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] inquery auto monitor in 1.0.3
I want to use auto monitor function in version 1.0.3 . I have put the options 'wW' to Dial application. but it do nothing when pressing *1 in call. How can auto monitor in 1.0.3? Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How connect 2 extension by AGI
I want to connect 2 extension by AGI. like auto dial out. How can i do? Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How set language in Auto-dial out
I have set 2 extensions. 820 and 821。 The default language is fr。 and I have created the following call file: Channel: SIP/820 MaxRetries: 2 RetryTime: 30 WaitTime: 30 Context: c820 Extension: 821 Priority: 2 The 820 hear the english greeting when 821 on thephone。Normal,It will hear fr greeting when 820 call 821。 How can change the language to fr in auto dial out? Thanks! Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail & "Couldn't read username" error
Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason when I call *99 from a SIP extension I am prompted for a Mailbox number but then nothing happens. It used to prompt for a password. I've un-installed the MySQL/Voicemail support and it still doesn't work. From the Asterisk console I get the following message when I hangup when I am NOT prompted for a password. Aug 24 12:34:36 WARNING[319509]: app_voicemail.c:3568 vm_execmain: Couldn't read username Any ideas? I can see on several lists that this is a common problem but I haven't found the answer yet. Bill Dunn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail & "Couldn't read username" error
I got this problem fixed by putting the following in the sip.conf file. dtmfmode=inband Bill Dunn - Original Message - From: Bill To: [EMAIL PROTECTED] Sent: Tuesday, August 24, 2004 12:49 PM Subject: Voicemail & "Couldn't read username" error Hi, I have Asterisk running with the VoiceMail. Using the latest CVS. I have my extensions.conf setup so that if a SIP caller dials *99 the VoicemailMain() as follows: exten => *99,1,Wait(1) exten => *99,2,VoicemailMain() A couple days ago I installed the MySQL/Voicemail support described at http://www.voip-info.org/wiki-Asterisk+voicemail+database Now for some reason when I call *99 from a SIP extension I am prompted for a Mailbox number but then nothing happens. It used to prompt for a password. I've un-installed the MySQL/Voicemail support and it still doesn't work. From the Asterisk console I get the following message when I hangup when I am NOT prompted for a password. Aug 24 12:34:36 WARNING[319509]: app_voicemail.c:3568 vm_execmain: Couldn't read username Any ideas? I can see on several lists that this is a common problem but I haven't found the answer yet. Bill Dunn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Gizmo from outside firewall
I've beaten myself bloody dealing with this one... No luck so far. In summary, incoming calls from Gizmo establish, but neither get nor send sound. Outbound calls to Gizmo work fine (well a bit choppy but work) My thought is that the SIP connection is being made fine, but the RTP is getting stopped / blocked / misdone somewhere. Here is the thing: Asterisk 2.5 on Linux (No hardware cards yet) X-Lite softphones on a few machines Gizmo clients and Gizmo accounts on the internet Gizmo client on the localnet PF firewall New to asterisk Okay - here are things that work and what I have tried: Works: If I call a Gizmo user outside the network from an XLite SIP phone inside the network it works. Works: If I call a Gizmo user inside the network from an XLite phone inside the network it works. NOT WORK: If I have asterisk register with gizmo and a gizmo person outside the network calls me, they get connected - but no sound either way. NOT WORK: If I have gizmo inside my network and I dial to my asterisk connected gizmo line it connects, but no sound. I logged all dropped packets at the firewall and am not blocking anything (I was at first dropping some incoming UDP in the 9000-2 range, but that has been fixed. The only thing I have not been able to do is to try to have an external xlite phone connect in and work. I think this would rest the blame on the firewall or gizmo... The only thing that seems weird is that is only happens when Gizmo originates the call. I can see the prompts and stuff playing on the CLI, but nothing gets sent to the other end. Also, if I answer a call, sound goes neither way. I've tried a bunch of things My SIP.conf has register => 1747xxx:[EMAIL PROTECTED] [gizmo-inbound] type=peer context=from-gizmo dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm nat=yes host=proxy01.sipphone.com insecure=very canreinvite=no externip=69.10.14.12 localnet=192.168.0.0/255.255.255.0 I have no idea what to check / try next... My gut instinct tells me it has to do with the firewall NAT and the RTP connection - but nothing is getting dropped or blocked, and I can dial out to them. Internally, Xlite -> asterisk works fine also. Any ideas would be immense help! Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Help with Gizmo from outside firewall
Sorry, send this part from an unregistered account > > I know this is going to a "duh" statement to a lot of people, but just > in case... when the non-audio gizmo connection rolls to voicemail, on > the cli I get: > > app.c:645 ast_play_and_record: No audio available on > SIP/proxy01.sipphone.com-x?? > > I am guessing this is since there is no RTP connection. > > Thanks > > Bill > > > > > On Wed, 15 Mar 2006 15:06:47 -0500 > Bill <[EMAIL PROTECTED]> spake: > > > > > I've beaten myself bloody dealing with this one... No luck so far. In > > summary, incoming calls from Gizmo establish, but neither get nor send > > sound. Outbound calls to Gizmo work fine (well a bit choppy but work) > > > > My thought is that the SIP connection is being made fine, but the RTP > > is getting stopped / blocked / misdone somewhere. > > > > Here is the thing: > > > > Asterisk 2.5 on Linux > > (No hardware cards yet) > > X-Lite softphones on a few machines > > Gizmo clients and Gizmo accounts on the internet > > Gizmo client on the localnet > > PF firewall > > New to asterisk > > > > Okay - here are things that work and what I have tried: > > > > Works: If I call a Gizmo user outside the network from an XLite SIP > > phone inside the network it works. > > > > Works: If I call a Gizmo user inside the network from an XLite phone > > inside the network it works. > > > > NOT WORK: If I have asterisk register with gizmo and a gizmo person > > outside the network calls me, they get connected - but no sound either > > way. > > > > NOT WORK: If I have gizmo inside my network and I dial to my asterisk > > connected gizmo line it connects, but no sound. > > > > I logged all dropped packets at the firewall and am not blocking > > anything (I was at first dropping some incoming UDP in the 9000-2 > > range, but that has been fixed. > > > > The only thing I have not been able to do is to try to have an external > > xlite phone connect in and work. I think this would rest the blame on > > the firewall or gizmo... > > > > The only thing that seems weird is that is only happens when Gizmo > > originates the call. I can see the prompts and stuff playing on the > > CLI, but nothing gets sent to the other end. Also, if I answer a call, > > sound goes neither way. > > > > > > I've tried a bunch of things > > My SIP.conf has > > > > register => 1747xxx:[EMAIL PROTECTED] > > > > [gizmo-inbound] > > type=peer > > context=from-gizmo > > dtmfmode=rfc2833 > > disallow=all > > allow=ulaw > > allow=alaw > > allow=ilbc > > allow=gsm > > nat=yes > > host=proxy01.sipphone.com > > insecure=very > > canreinvite=no > > externip=69.10.14.12 > > localnet=192.168.0.0/255.255.255.0 > > > > I have no idea what to check / try next... My gut instinct tells me it > > has to do with the firewall NAT and the RTP connection - but nothing is > > getting dropped or blocked, and I can dial out to them. > > > > Internally, Xlite -> asterisk works fine also. > > > > Any ideas would be immense help! > > > > > > Bill > > > > > > > > > > > > > > > > > > > > > -- > > Bill Chmura > Director of Internet Technology > Explosivo ITG > Wolcott, CT > > p: 860.621.8693 > e: [EMAIL PROTECTED] > w. http://www.explosivo.com -- Bill Chmura Director of Internet Technology Explosivo ITG Wolcott, CT p: 860.621.8693 e: [EMAIL PROTECTED] w. http://www.explosivo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Gizmo from outside firewall <- update
Well, I got off site today with my notebook and an x-lite install. I was able to connect into to the system and hear things, etc... But since the phone connects ahead, this may be a different thing than an incoming gizmo call eh? If someone could even point me in the direction to look, I would be greatful! On Wed, 15 Mar 2006 15:06:47 -0500 Bill <[EMAIL PROTECTED]> spake: > > I've beaten myself bloody dealing with this one... No luck so far. In > summary, incoming calls from Gizmo establish, but neither get nor send > sound. Outbound calls to Gizmo work fine (well a bit choppy but work) > > My thought is that the SIP connection is being made fine, but the RTP > is getting stopped / blocked / misdone somewhere. > > Here is the thing: > > Asterisk 2.5 on Linux > (No hardware cards yet) > X-Lite softphones on a few machines > Gizmo clients and Gizmo accounts on the internet > Gizmo client on the localnet > PF firewall > New to asterisk > > Okay - here are things that work and what I have tried: > > Works: If I call a Gizmo user outside the network from an XLite SIP > phone inside the network it works. > > Works: If I call a Gizmo user inside the network from an XLite phone > inside the network it works. > > NOT WORK: If I have asterisk register with gizmo and a gizmo person > outside the network calls me, they get connected - but no sound either > way. > > NOT WORK: If I have gizmo inside my network and I dial to my asterisk > connected gizmo line it connects, but no sound. > > I logged all dropped packets at the firewall and am not blocking > anything (I was at first dropping some incoming UDP in the 9000-2 > range, but that has been fixed. > > The only thing I have not been able to do is to try to have an external > xlite phone connect in and work. I think this would rest the blame on > the firewall or gizmo... > > The only thing that seems weird is that is only happens when Gizmo > originates the call. I can see the prompts and stuff playing on the > CLI, but nothing gets sent to the other end. Also, if I answer a call, > sound goes neither way. > > > I've tried a bunch of things > My SIP.conf has > > register => 1747xxx:[EMAIL PROTECTED] > > [gizmo-inbound] > type=peer > context=from-gizmo > dtmfmode=rfc2833 > disallow=all > allow=ulaw > allow=alaw > allow=ilbc > allow=gsm > nat=yes > host=proxy01.sipphone.com > insecure=very > canreinvite=no > externip=69.10.14.12 > localnet=192.168.0.0/255.255.255.0 > > I have no idea what to check / try next... My gut instinct tells me it > has to do with the firewall NAT and the RTP connection - but nothing is > getting dropped or blocked, and I can dial out to them. > > Internally, Xlite -> asterisk works fine also. > > Any ideas would be immense help! > > > Bill > > > > > > > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Bill Chmura Director of Internet Technology Explosivo ITG Wolcott, CT p: 860.621.8693 e: [EMAIL PROTECTED] w. http://www.explosivo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] meetme
On Sun, 09 Apr 2006 09:12:42 -0700 Miles Scruggs <[EMAIL PROTECTED]> spake: > > >>> > I'm having issues getting meetme to work: > > Apr 9 05:26:22 WARNING[23882]: pbx.c:1688 pbx_extension_helper: No > application 'MeetMe' for extension (internal, , 2) > == Spawn extension (internal, , 2) exited non-zero on > 'SIP/mileslap-569b' > > the only thing I could find was this: > > http://72.14.203.104/search?q=cache:XirZjsPxJO8J:lists.digium.com/pipermail/asterisk-users/2004-April/044795.html+%22No+application%22+%27MeetMe%27+for+extension&hl=en&gl=us&ct=clnk&cd=4&lr=lang_en&client=firefox-a > > > > but I have the timer working (I think): > > lsmod | grep dummy > ztdummy 2608 - > > I'm really confused as to what to do next, if someone could help me > out that would be great: > > I'm using gentoo with kernel 2.6.15. asterisk has been compiled > from scratch with asterisk 1.2.5(I know not the latest) and zaptel > 1.2.5 > > Thanks > > Miles > >>> > >>> If you type "modprobe zaptel" "modprobe ztdummy" at the Linux CLI, > >>> what do you get? > >> Nothing, they were loaded before, and loaded just fine. > >> > >> lsmod Module Size Used by > >> ztdummy 2608 - > >> rtc10620 - > >> zaptel186468 - > >> crc_ccitt 1576 - > >> 3c59x 40240 - > >> ___ > >> Did you have the ztdummy and stuff compiled into the kernel before you compiled asterisk? If not, asterisk skips compiling the meetme application. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR prompts: attempts at a standard list
For the GENERAL single-word/short phrases, would it be reasonable to ask for three inflections of each? One mid-sentence (flat inflection), one end-of-sentence (falling inflection), and one for questions (rising inflection). This could make your patch-together sentences sound much more professional. Bill Jennings John Todd ([EMAIL PROTECTED]) wrote: > > I'm looking to get Allison Smith (http://www.theivrvoice.com) to > record a bunch of prompts for me. I sat down and put together a > number of phrases and words that I would expect to be strung together > GENERAL > DNS [dee en ess] > accepted > account > after the tone > an > and > at the moment > but > busy > code > complete > dash > database > dial > down > enter > error > extension > failed > goodbye > hang up > hello > host > incomplete > info > is > key > mail > message > minute > moment > not > not responding > number > or > otherwise > password > ping > please > pound > press > sales > star > thank you > that > the > time > to play > transfer > up > wait > web > your > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Question about asterisk
I wanted to know if there was any way to setup an asterisk server as a PSTN gateway? That is, I wanted the asterisk server to accept invites from any sip client and send them through its T1 or FXO cards. So far I've only been able to make asterisk accept invites from users it knows about. Is there any way to make it blindly accept all sip invites without any authentication? Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CERT advisory (SIP)
In a quick search of the list archives, I found no mention of the recent CERT advisory concerning vulnerabilities in some implementations of the SIP protocol (i.e. whether or not * users were impacted by it, and if so, to what extent and/or in what configurations), so I figured it would be worthwhile to toss the question out there ... Link: http://www.cert.org/advisories/CA-2003-06.html As someone in the early stages of investigating *'s potential usefulness for both my own needs and those of my clients - and as one who readily admits of possessing little knowledge of any but the most rudimentary aspects of telephony and CTI - I would be grateful if someone familiar with the "nuts and bolts" of * and SIP could provide a brief assessment on this point to the list. TIA! -- Bill Mullen [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Known SIP - NAT Solutions?
Matthew Farley wrote: I have recently begun experimenting with Asterisk, and have been mightily impressed by its capabilities and flexibility. I have run across one problem, however, that challenges my ability to use it as a production system. My Asterisk box has a public Internet IP, and works great with SIP (ATA 186) clients that also have public IP addresses. Unfortunately, most of the locations that I would like to put these SIP phones into are behind NAT. Calls placed from behind NAT are consistantly unsuccessful. I have read in several places that there are software solutions to this problem, though I have found no specific references to precisely what software to use, or how it should be configured to hand these calls off to Asterisk. Has anyone on the list successfully overcome this limitation? If so, any advice you might be able to provide would be greatly appreciated. Thanks! Sincerely, Matthew Farley [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users There are basically two ways of doing SIP-through-NAT. The first is to configure the firewall to forward ports 5060 and 1-10100 or whatever the IP phone uses for SIP and RTP. Then configure the IP phone and set its NAT IP address. The second way is to use a RTP Proxy. The way vonage handles SIP-through-NAT is to have their SIP Proxy modify the sdp packets for the INVITE, 183, and 200 OK messages and put the RTP Proxy's IP address and ports into the sdp portion of the message. That way each endpoint sends RTP packets to the RTP Proxy. The RTP Proxy waits for the first packet from each endpoint, then it knows which port to send the RTP packets to. There is an open source project siproxd at http://sf.net/projects/siproxd which has a basic implementation of RTP Proxy. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IPv4...NAT...etc
Jon Pounder ([EMAIL PROTECTED]) wrote: > At 06:08 PM 3/5/2003 -0600, you wrote: > >On Wed, Mar 05, 2003 at 05:45:13PM -0600, Jim Fleming wrote: > >> http://lists.digium.com/pipermail/asterisk-users/2003-March/008088.html > >> "why is there such a delay in getting ipv6 rolled out when it solves > >all these problems ?" > > > >I doubt that users will stop using NAT until ISPs stop charging > >per address for SOHO customers. > > and if ipv6 made the addresses much more freely available your isp could > afford to dish out all you could ever want on a $20/month connection. > > The ip's are not just free to the ISP, there is a cost associated with them > each year in addition to the bandwidth you use. Not to mention it makes > routing rules more complex the more ips an isp has to deal with in separate > groups. > I think the main reason ISPs want to charge more for static IPs is that with static IPs you can run servers. If you run servers, you are going to use more bandwidth (read: "expensive resources"). Ergo, they pass the expense on to you - the BUSINESS customer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Exceptionally long queue length.
We've been getting the following message from channel.c: Exceptionally long queue length queuing to Now while we still have to fix the base problem of why we're getting such long queue lengths, I'm wondering the following: a) Why do we check queue length after we've Q'd and not before? b) Why do we crash deliberately? Is there a good reason we don't simply note the condition, and then toss the frame? Thanks Bill -- Bill Leckey - Senior Software Design Engineer TPG Research and Development Ph: +61 2 62851711 Fax: +61 2 62853939 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What I need to make this work?
Hello! I just joined and am looking around for a place to start. Would this Asterisk Developers Kit be a good place to start? This is what I'd like to do. I am a wireless ISP. Several of my customers have offices in different communities that I serve. The local telephone company charges 8-10 cents a minute for calls town to town. Currently I use a Cisco ATA 186 to talk to one of my programmers in another town using the FWD gateway. Could I implement the Asterisk software and what recommended hardware to connected these users through the Asterisk head end to place calls and log their activity for charge back? I'm considering having up to 12 VOIP users, possibly 2-3 concurrent users. If I load RH 8.0 what's the minimum hardware requirements -CPU/RAM? Any other suggestions/ ideas would be appreciated. Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
This is to state that a recent message posted to asterisk mailing list by [EMAIL PROTECTED] regarding the pricing of our sample phones is NOT accurate. Grandstream Networks has NOT changed the list price for its products and samples. Our BudgeTone 100 series IP phones lists at $75 for model 101, NOT $60. Grandstream is committed to supporting the asterisk community and this message is posted for the sole purpose of correcting a misinformation regarding our product. Thanks for your attention to this matter. Grandstream Customer Support ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Correction regarding price of Grandstream Budgetone 100 series phone
Hello Wade and Asterisk users, As we are committed to supporting Asterisk community, we will not be able to answer questions related to Grandstream product through Asterisk mailing list, this is to be fair and respectful to the Asterisk community as a whole. The previous email is to clear a pricing info regarding the product because a lot users start to use that price as reference price for the phone. Should you have any questions and issues regarding Grandstream product, please send your email to [EMAIL PROTECTED] or [EMAIL PROTECTED] Thank you for your attention and interest in Grandstream product. Best regards, Grandstram Customer Support Wow! A phone manufacturer is actually monitoring this list! Nice work Grandstream. Can you tell us which phones you currently have in stock, and pricing on all models? Can you also let us know if your 1-port FXS device is shipping? Pricing? Thanks in advance, -wade > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Bill Zhang > Sent: Monday, June 09, 2003 4:14 PM > To: [EMAIL PROTECTED] > Cc: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Correction regarding price of Grandstream > Budgetone 100 series phone > > This is to state that a recent message posted to > asterisk mailing list > by [EMAIL PROTECTED] regarding the pricing of our > sample phones is NOT > accurate. Grandstream Networks has NOT changed the > list price for its > products > and samples. Our BudgeTone 100 series IP phones lists > at $75 for model 101, > NOT $60. > Grandstream is committed to supporting the asterisk > community and this > message is posted for the sole purpose of correcting a > misinformation > regarding > our product. > Thanks for your attention to this matter. > > Grandstream Customer Support > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MWI question
Using a TE410P with Zhone 24FXS channel banks to power standard analog phones I can't seem to find out if it's possible to support FSK or voltage type message waiting lamps. I don't want to use stutter dial tone because of the dramatic difference in per phone cost. TIA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Subject: [Asterisk-Users] Provisioning CO lines
I'm brand new to asterisk but not to T1s so here's my bit to contribute.
Each local telco {be they ILEC or CLEC} is different depending on their
CO switch and the software options they've purchased for it.
In Alaska, the "break-even" for switching from POTS to T1 is about 13
trunks.
Your telco will offer "regular T1" and/or ISDN-PRI. Up here the tariffed
rate on ISDN-PRI makes it as expensive as POTS lines. We lose
callerID if we go to regular T1 but that's because the local telco hasn't
spent the money to upgrade their switch.
Best thing to do is tell your sales rep you want quotes for 10-24 trunks in
PRI-ISDN, regular T1 and POTS. This can be like pulling teeth but it's
what you need to make the best buying decision. Then you can decide
when/if it's time to jump to digital and what kind to go for.
hth
> Hi all,
>
> This is a NEWBIE question, so all you experienced types that are
> tired of stupid questions can move on...
>
> I've pretty much given up trying to do my entire phone system
> over IP (including local service), so I have to select and
> provision my local CO lines. I need about 10-12 lines which can
> be POTS lines, of course. But, I thought, why not get something
> digital and expandable like a DS1, PRI, T1 or whatever they call
> it with 23 or 24 channels of 64 kbps voice. It seems like it
> would be simpler for me to deal with this (and better quality)
> and it *should* be simpler for the phone company, too.
>
> However, while everyone can sell me POTS lines, when I ask about
> getting these in some sort of digital muxed interface, I seem to
> confuse the providers. In one case, I was able to get something
> called "channelized T1" which cost a lot and did not actually
> include the "phone" service for any of the channels, that was
> additional. So the cost to go from POTS lines to something
> digital was extreme, so much more than I can't understand why
> anyone would have T1 voice interfaces, yet all the PBXes have
> this and it seems commonly used. I must be doing this "wrong".
>
> Okay, so I need help with:
>
> 1. Understanding terminology so I can ask for the "right thing".
>
> 2. Advice on when it is reasonable to go POTS versus something
> else and what that something else is.
>
> 3. Feedback on what others are doing with 10-12 lines in the US
> that may want to expand to ~20 lines.
>
> 4. Interfacing so many POTS lines to Asterisk. I guess that
> means an FXO channel bank to T1 card? Kind of stupid to go
> digital/analog/digital in the last 100 feet.
>
> Help?
___
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Re: [Asterisk-Users] Voicemail 1 and 2 -Martin?
Martin, Your statement below is somewhat confusing. Where do you find the choice of 1 or 2? This is the latest voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats for writing Voicemail ;format=g723sf|wav49|wav format=gsm|wav49|wav ; Who the e-mail notification should appear to come from serveremail=asterisk ;[EMAIL PROTECTED] ; Should the email contain the voicemail as an attachment attach=yes ; Maximum length of a voicemail message ;maxmessage=180 ; Maximum length of greetings ;maxgreet=60 ; How many miliseconds to skip forward/back when rew/ff in message playback skipms=3000 ; ; Each mailbox is listed in the form =,,, ; if the e-mail is specified, a message will be sent when a message is ; received, to the given mailbox. If pager is specified, a message will be sent there as well. ; [default] ;1234 => 4242,Example Mailbox,[EMAIL PROTECTED] ;4300 => 3456,Ben Rigas,[EMAIL PROTECTED] ;4310 => 5432,Sales,[EMAIL PROTECTED] ;4069 => 6522,Matt Brooks,[EMAIL PROTECTED] ;4110 => 3443,Rob Flynn,[EMAIL PROTECTED] At 01:24 PM 9/12/2003 -0500, you wrote: you can copy voicemail.conf.sample to be your voicemail.conf ... Martin On Fri, 12 Sep 2003, Olle E. Johansson wrote: > Steven Critchfield wrote: > > > On Fri, 2003-09-12 at 10:34, Olle E. Johansson wrote: > >>While on the subject of Voicemail - what is the difference between > >>voicemail() and voicmail2() ? > > >>From the application stand point there is little difference, but from > > the configuration stand point there is a fair amount of difference. > > Consult the sample configs to start you on your path to deciding what > > you want. > Steven, > Thank your for responding. > > I find only one config in the sample directory - voicemail.conf.sample > and it looks the same as my voicemail.conf > - should I look in another place? > > /Olle > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * <--> FWD
Hello! There is much info using SIP for this connection type. I have a TDM400P with a regular phone connected to it. What string in extensions.conf would need to be added so I can call a FWD number as well as receive a call from FWD? I have seen some mention of using IAX. Is this necessary or can this connection be accomplished without enabling IAX? Thanks, Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A WORKING EXAMPLE
Hello! I've looked at the reference examples they are all for SIP. I have two X100p and a TDM400P. Can someone send me a working example so I can receive calls and make them. I'm stuck at first base. [I'm using standard phones - not SIP] Help please! Thanks, Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * website needs a place for
Hello! This should be a list to come find support and not get jumped on! The * website should instruct where to find information better. Often times the first response to trying to learn something is to ASK a question. I too, first found the archive list tonight. I've been on this list reading since February. Better documentation is the key and since this is a product being developed daily keeping up with the documentation is difficult. It's the new people coming in which keep this idea alive as we, who have been around tell them. What do people see when they read list mail? I see PJ trying to help and John B. who BTW, is also a VOIP reseller, jumping on people who are not changing subject lines. Education and documentation is key to making a product succeed. Possibly a * web page re-design would better educate new people coming into this list so they conform to the lists standards. Also a reminder to those who know far more than I, You too started someplace and someone answered your questions and you learned. Please, lets be considerate of others. Possibly an automated daily message could be sent to the list reminding people to change the subject line or provide a link to the archives... Helping people succeed with * helps everyone who has an interest. Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Budget Hotel PBX
I'm considering using asterisk to replace an existing PBX in a 40 room hotel and
would appreciate any comments, corrections or insight before I begin.
Only 8 PSTN connections are initially required but since the guests need dial-up
internet access in the rooms it has to be Frac-T1 as opposed to using FXO ports on
a channel bank.
IP phones are not an option strictly because of price. The analog phones must
have FSK message waiting lights instead of the cheaper voltage type since asterisk
doesn't support that.
So, a TE410P {or 400} and two Zhone 24FXS channel banks will be used. I
couldn't google up any info on what mobo but I'd like to start with a 450mhz since I
have one laying around with 64bit slots but if that's marginal I could get a dual
Athlon server board or whatever.
I'd also greatly appreciate knowing if anyone out there is actually using asterisk in
a
similar hotel application today.
TIA
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[Asterisk-Users] Outgoing call spool
I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating earpiece. Is it possible to somehow generate a ring in the earpiece until the remote phone is picked up? Bill -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
Andrew Joakimsen wrote: No, because asterisk cannot deal with the G723 codec, it can only act as a "middle man" of sorts between devices that support it. Ok, that makes sense. Could I get the ringing somehow if I changed to (say) the G711 codec? Or, is it possible that this could be done by (say) the SIP RINGING message? I believe that while the remote phone is being rung then the originating call is currently in a "call up" state, which means a SIP RINGING isn't allowed, but I guess I'm wondering if something like this might work? Thanks, Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Leckey Sent: Sunday, September 28, 2003 7:03 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Outgoing call spool I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating earpiece. Is it possible to somehow generate a ring in the earpiece until the remote phone is picked up? Bill -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing call spool
Andrew Joakimsen wrote: Post the line with the Dial() from your extensions.conf Do you mean post it here? exten => _[1-9]XX,206,dial,sip/BYEXTENSION|30| Bill Im not sure that it will work, but its worth a try. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream and voicemail waiting
I've been playing with Asterisk and Grandstream phones. I've seen a few messages here about how the phones tell you when a voicemail is waiting for you, but can't seem to get mine to do that. Can anyone tell me what I need to configure on the phones and in Asterisk? Thanks. Bill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and voicemail waiting
Hi Dave, Thanks for that. It all works now. Bill Dave Cotton wrote: On Thu, 2003-10-16 at 08:25, Bill Leckey wrote: I've been playing with Asterisk and Grandstream phones. I've seen a few messages here about how the phones tell you when a voicemail is waiting for you, but can't seem to get mine to do that. Can anyone tell me what I need to configure on the phones and in Asterisk? Thanks. In sip.conf for that phone set up mailbox=1234 or even mailbox=1234,1235,1236 and on the GS Voice Mail UserID: whatever extension you have for voicemail WFM. (tm) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip-info.org DNS seems broken
For the last few days I can not resolve voip-info.org from many DNS servers. It does resolve with some DNS servers but I suspect it may be related more to caching. Using the host command: host -a voip-info.org 130.179.16.23 Trying "voip-info.org" Using domain server: Name: 130.179.16.23 Address: 130.179.16.23#53 Aliases: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 33642 ;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 2, ADDITIONAL: 2 ;; QUESTION SECTION: ;voip-info.org. IN ANY ;; ANSWER SECTION: voip-info.org. 86318 IN NS ns2.lj.net. voip-info.org. 86318 IN NS ns1.lj.net. ;; AUTHORITY SECTION: voip-info.org. 86318 IN NS ns2.lj.net. voip-info.org. 86318 IN NS ns1.lj.net. ;; ADDITIONAL SECTION: ns2.lj.net. 3518IN A 64.65.89.226 ns1.lj.net. 138603 IN A 64.65.89.226 Received 133 bytes from 130.179.16.23#53 in 95 ms ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZT_CHANCONFIG failed on channel 2
Hi, FWIW This issue had been resolved. The fix is nothing to speak of except that maybe this post may be informative for someone out there. It turned out to be a hardware issue in the PC, after swapping the Zaptel cards to another PC, it has been up and running with no "ZT_CHANCONFIG failed on channel 2: No such device or address (6) [FAILED]" errors anymore. Unfortunately I don't have the cycles to research exactly what on the MB or NIC may have been causing this, it could be anything, it's easier just to leave the Zaptels in the new box and find something clever to do with the old one like tie it to my bumper and drag it around for a bit :) I can say this though, the box would loose network connectivity ~60 seconds after the error. An ifconfig -a showed the NIC Up and the routing table on the box was correct as well. Link lights looked fine and the CISCO catalyst port showed up/up as well. The Zaptel cards did not need to be powercycled because a "shutdown -r" worked. Regards, -bh Quoting [EMAIL PROTECTED]: > Hi, > > Thank you for the reply, actually the cards installed are a TDM400P (Single > port) and an X100P. > > I don't need to power down the PCI cards by turning of the PC, a simple > "shutdown -r now" does it. > > -bh > > Quoting Tilghman Lesher <[EMAIL PROTECTED]>: > > > On Thursday 11 December 2003 17:42, [EMAIL PROTECTED] wrote: > > > [EMAIL PROTECTED] asterisk]# service zaptel start > > > Loading zaptel framework: [ OK ] > > > Loading zaptel hardware modules: wcfxo wcusb > > > Running ztcfg: ZT_CHANCONFIG failed on channel 2: No such device > > > or address (6) [FAILED] > > > > Typically this means that the driver cannot detect the device. Since > > it is the S100U that seems to be undetected, try unplugging it, > > waiting 20 seconds, and plugging it back in. > > > > -Tilghman > > This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: time to build an open phone?
ACES - Asterisk Communications Endpoint System
{the following could be used by any IP-PBX but the name pays homage to Mark Spencer
and friends who
cannot be lauded enough for their fine work}
As you read this it will be obvious I am not a professional engineer but I do have
enough knowledge
to be fairly certain what I'm proposing is feasible from not only an engineering, but
production
cost and perhaps most importantly, marketing standpoint.
An open phone is a great idea but as soon as you "get physical" you add a quantity
issue that
doesn't exist in software. Multiply this for keypads, handsets, bells, etc. etc. etc.
and you have
a lot of work but more importantly NO ONE has built a phone that can simultaneously be
brain-dead
simple to operate for one person yet offer the advanced user whatever functionality
they might
want. You will never solve that issue as long as you have a keypad of any kind.
So you end up with what started this open-phone thread in the first place... a
plethora of IP,
analog or digital phones with a dizzying array (or lack thereof) of bells and whistles
all trying
to achieve a balance between quality, ease of use and functionality which will sell
enough units to
make their manufacturing and distribution profitable. In this environment you will
always have at
the low end manufacturers competing on price and inevitably that results in quality
issues. Right
now it's Grandstream but next year it'll be someone else at a $30 price point and the
same issues
will apply all over again.
I've never seen stats, but it's probably a safe assumption that the majority of IP
phones are
sitting next to a PC and the additional expense has been incurred because "people want
a phone that
looks and works like a phone". That's certainly been my experience far outweighing
any technical
issues with quality or reliability of a PC-softphone. In every market I can think of
with the
possible exception of hospitality I think ACES could be successfully sold a
substantial number of
times even though it does not "look like a phone" because it affords a much better way
to resolve
the conflict between ease of use and functionality. For the unconvinced, a more
elaborate version
could include the obligatory keypad and cosmetic plastic but I would submit that the
ability to
pick up a handset and place a call by saying "call Pat" alone would "sell" most
potential customers
on learning how to operate a two position switch on a device that doesn't have a
conventional
keypad. At it's simplest, to use the phone you need to know that position A is used
to hangup and
dial by saying "dial 1-800-555-1212" (or whatever number you want called) and position
b is used to
talk.
ACES has three components and for simplicity of description I won't go into VERY cool
extensions to
these components for conferencing and/or duplication of the typical 2,3 or 4 line
analog phone
features. It also assumes a LAN environment again only for simplicity of initial
description.
There's no reason that an ACES Call control server couldn't support multiple,
geographically
dispersed Asterisk servers.
The heart of this concept is use of text-to-speech to replace keypad functions. I
cannot emphasize
enough how acutely aware I am of the HUGE resistance users will have to buying
something without a
keypad but bear with me and I hope you'll agree that this has enough "sex appeal" to
overcome this
historically undefeated resistance. Each "phone is two complete analog/IP circuits
defined as:
Talk - a subset of what Asterisk uses now not requiring any of the control functions
TTSControl - moving control functions currently handled by DTMF over to a
text-to-speech engine
located on ACES component 3 described below. The TTS engine would be capable of
translating most
peoples voices when they speak the word "call" and the ten digits required to place a
call. The
"phones"(ACES component 2 described below) would simultaneously be user-specific so
individual
users could train their personal library to recognize them when they are "logged in"
at that phone
to place calls by saying "call Pat", etc. etc. etc. and of course to receive calls.
ACES Component 1
EM unit-Ear and Mouth piece, this is a headset or handset with a two position switch
and a 4
conductor jack that plugs into the IP unit(ACES component 2). FOr prototyping two
typical monaural
PC headsets into a 2.5mm switchbox would do fine. Switch position one connects the
1st mike and
earpiece to the 2 "talk" pins on the Talk/TTSControl port on the IP unit and Switch
position two
connects the 2nd mike and earpiece to the 2 "ttsControl" pins on the Talk/TTSControl
port on the IP
unit. Obviously production handsets/headsets would have only one earpiece/mike with
the switch
changing the connection from one pair of pins to the other.
ACES Component 2
IP unit - a black box containing 5 physical inte
[Asterisk-Users] Re: time to build an open phone?
> > I've never seen stats, but it's probably a safe assumption that the > > majority of IP phones are sitting next to a PC and the additional > > expense has been incurred because "people want a phone that looks and > > works like a phone". That's certainly been my experience far > > outweighing any technical issues with quality or reliability of a > > PC-softphone. In every market I can think of with the possible > > exception of hospitality I think ACES could be successfully sold a > > substantial number of times even though it does not "look like a phone" > > because it affords a much better way to resolve the conflict between > > ease of use and functionality. For the unconvinced, a more elaborate > > version could include the obligatory keypad and cosmetic plastic but I > > would submit that the ability to pick up a handset and place a call by > > saying "call Pat" alone would "sell" most potential customers on > > learning how to operate a two position switch on a device that doesn't > > have a conventional keypad. At it's simplest, to use the phone you need > > to know that position A is used to hangup and dial by saying "dial > > 1-800-555-1212" (or whatever number you want called) and position b is > > used to talk. > > Soft phones are only as reliable as the host OS. It would be extremely > hard to explain to a user that they need to upgrade their PC or close apps > so their call quality can stay at the expected level. This is especially > true if you are wanting to do Speech Recognition. Which by the way, you > make that mistake many times in this post, you are wanting speech > recognition to determine what the person on the phone says, not text to > speech where the computer could read to the user. Speech recognition uses > significant resources to be accurate. In the long run you only shift cost > from your add on to the PC. Then you have to support whatever OS is on the > desktop, not a good idea. The reason for people wanting a real hardware > phone on the desk next to the PC is that they understand that computers > crash, have virus problems, have upgrade incompatibilities and any number > of other instabilities that can render their workstation down for a day or > more. These people must still be able to use the phone no matter the > condition of the machine on the desk. Many peoples jobs can still be > preformed when the PC is either non functional or not functioning > optimally. > > Take my mothers job for a option, she routes freight for her company. If > her computer was to become inoperable for a period of time, she usually > has a hour or more of paperwork she can complete on the phone with her > customers and freight companies. She could probably use a VoIP phone, but > not one tied to the stability of her computer. I'm sure this is true with > many other jobs. I can also tell you that my mothers windows computer > crashes several times a day, and some of the calls she makes requires her > to be on hold for 10-20 minutes. If she was to experience a crash in that > wait period, it would basically waste the time she had been on hold. > Sounds like we're arguing the same thing for different reasons. For whatever reason PC-softphones are not a viable option. I totally agree with that statement. > So try to remember that we wish to bring efficiencies to the > worker/person using our devices not new roadblocks. It probably doesn't look like it, but I tried to keep the initial comments low so I didn't go into detail on exactly how it would work but I am certain that the standard phone functions will all be at least as easy and as fast as any analog, digital or IP system I've seen so far and a dramatic improvement over most. > > > The heart of this concept is use of text-to-speech to replace keypad > > functions. I cannot emphasize enough how acutely aware I am of the HUGE > > resistance users will have to buying something without a keypad but bear > > with me and I hope you'll agree that this has enough "sex appeal" to > > overcome this historically undefeated resistance. Each "phone is two > > complete analog/IP circuits defined as: Talk - a subset of what Asterisk > > uses now not requiring any of the control functions TTSControl - moving > > control functions currently handled by DTMF over to a text-to-speech > > engine located on ACES component 3 described below. The TTS engine > > would be capable of translating most peoples voices when they speak the > > word "call" and the ten digits required to place a call. The > > "phones"(ACES component 2 described below) would simultaneously be > > user-specific so individual users could train their personal library to > > recognize them when they are "logged in" at that phone to place calls by > > saying "call Pat", etc. etc. etc. and of course to receive calls. > > Speech recognition would be less helpful than a computerized rollodex with > click to call functionality. A home user may have a short enough list of > p
[Asterisk-Users] Re: Nauti miles
I might as well add to the offtopic thread... why are natuical miles longer than "regular" miles? Andrew A nautical mile is 1 minute of latitude.
[Asterisk-Users] sound of static removed by hitting flash button
When making calls users are hearing static on the phone. If they hit the flash button once, the static is removed and they can continue with their call. This problem occurs even if they are just checking voicemail. We are using Digium X100P and TDM400 cards under asterisk v 0.7.1. Any thoughts? Thanks, Bill
Re: [Asterisk-Users] Re: Voicepulse
Atually with the root servers dropping their domain name announcement "nothing" would have helped. Well, except for hard codeing the IP rather than using fqdn in the config. Or making a static entry in the local hosts file ( both have it's issues) I prefer to use IP rather than fqdns when possible. But that can introduce other problems if the host system decides to move you to another host machine by just changing the DNS name. Using fqdns in mission critical applications is not a good idea IMHO, it just adds another layer of something that can go wrong. Just my $.02 worth ;) -b Quoting Chris Albertson <[EMAIL PROTECTED]>: > > --- Steve Sobol <[EMAIL PROTECTED]> wrote: > > Matt Lawson wrote: > > > > > I was just about to write the same thing. It says "busy". Is is > > REALLY > > > busy or is something else wrong? > > > > > > This on the heels of switch-1.nufone.net being missing out of DNS. > > > > > > We have customers that expect their VOIP to work. Is there anybody > > > > > that's reliable? > > I've been doing some testing and so far I'm not 100% impressed > by the VOIP services I've seen. They provide a good service but > my local phone company and AT&T longdistance service is more > reliable. > > But this is not to say _you_ can't built a reliable VOIP based > system. Get _two_ providers and set up your dial plan in > extensions.conf to "fail over" if one service fails to > connect to dial via the next one and finally if both fail > use pstn. your users will see a system the "just works". > > About Nufone's problem. I bet they'll start thinking about > getting a backup DNS service and maybe geographic deversity. > A company should be able to even stay on the air if there is a > server room fire using techniques like round robin DNS and > West cost and East coast servers run by different, unrelated > hosting companies. > > > > = > Chris Albertson > Home: 310-376-1029 [EMAIL PROTECTED] > Cell: 310-990-7550 > Office: 310-336-5189 [EMAIL PROTECTED] > KG6OMK > > __ > Do you Yahoo!? > Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes > http://hotjobs.sweepstakes.yahoo.com/signingbonus > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by The CCIS.net MailScanner, and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by the Bugs.Hamel.Net MailScanner, > and appears to be clean. > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7960 Phone disconnects when dialing using speaker
Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in advance, -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 7960 Phone disconnects when dialing using speaker
Quoting Brian West <[EMAIL PROTECTED]>: > Works fine here.. got two of em. > > bkw Hmpf! I donno whats wrong then, both phones do the same thing. So you can keep the headset in the cradle, hit the 'speaker' button, dial a call and it doesn't disconect ? I wonder, are you using an xml dial plan or anything on you phones ? Thanks -bh > > On Fri, 16 Jan 2004, Bill Hamel wrote: > > > Hi, > > > > Just got some CISCO 7960 phones. They seem to work great except if I make > any > > SIP call using the speaker phone (leaving the hand set in the cradle)the > call > > will disconnect in about 5 or so seconds. If I pick up the hand set and > make a > > call, it's fine. > > > > Has anyone else run into this ? Any solution ? > > > > The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. > > > > Thank you in advance, > > -bh > > > > -- > > > > > > > > This message was sent using IMP, the Internet Messaging Program. > > This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent timeout then Dial() ?
Hello, I have agents / queues working to the extent that agents can login, logout and I can send a caller into the queue and the logged in agent's phones will ring. Maybe I've spent to much time googleing and reading and my eyes are crossing now, but what I am trying to do is this but cannot find any reference to it. 1. Xfer the caller into the Queue... If Noone is logged into the queue, the caller will be directed to a PSTN number instead (or extension, same thing) 2. Xfer the caller into the Queue... Agents are logged in, but the call times out for whatever reason, I would then like to have it go to an extension as in above 3. When say 6PM rolls around and all agents are gone I would like to automagically log them out just incase they forgot to. I will be happy with an answer for 1 and 2 - I can always use a big stick for #3 :) I did find a reference to adding a member "local" in queues.conf eg: member => local/[EMAIL PROTECTED],10 And have a context in extensions.conf like this [timeout] exten => s,1,Wait,1 exten => s,2,Answer exten => s,3,Playback(transferring_you_offsite) exten => s,4,Dial,IAX2/office/[EMAIL PROTECTED] Even with the metric of '10' to try and give the "local member" less preference it will give logged in agents like half a ring and then xfer to the "timeout" context right away. Any help, pointers would be greatly appreciated. Many thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a way to # of agents logged into a queue ?
Hi, Looking around I can't seem to find a way to show the number of agents currently logged into a queue and if possible who they are. Is there a way to do this ? Thanks -b -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?
Hi Chris, This sounds what I am looking for, many thanks ! Also, I do not see an attachment, the patch that is :) I dont know if the list strips attachments, perhaps send it to my email address [EMAIL PROTECTED] Thanks again, -bh Quoting "C. Maj" <[EMAIL PROTECTED]>: > I attached a patch I've been using to show the # of agents > (members) and callers on a per queue basis. It adds a new > manager command, "AgentQueues". It returns on the manager > interface the following for each queue: > > Queue: queuename > Agents: # > Callers: # > > There's another manager command, "QueueStatus", that might be > what your are looking for. There's also "Queues" but that > is a PITA to parse. Fine if you just want to display it in > a text widget or something. > > --Chris > > > -- > > Chris Maj > Pronunciation Guide: Maj == May > Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146 > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by The CCIS.net MailScanner, and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by the Bugs.Hamel.Net MailScanner, > and appears to be clean. > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] He really doesn't care
I'm new to this all, and had never heard of NuFone until someone raised the question of whether they were in trouble. This was a net positive for NuFone because it made me a aware of their existence. The next question in my mind was about their service, which I had to evaluate based on second hand information provided by others' comments, which were varied. Then I got first hand info, and this sums it up well... Our network and services speak for themselves. If they don't like my attitude after they publicly flame us they can find another provider, I really don't care. Jeremy McNamara
Re: [Asterisk-Users] re: help with voicepulse connect IAX2
Curious what your iax.conf looks like. Also FWIW - if you are connecting directly to VoicePulse with a SIP phone, wouldn't that mean that you have a SIP account and not an IAX account ? -b Quoting yair hakak <[EMAIL PROTECTED]>: > hello, > after playing with an asterisk configuration for voip for a few weeks i'm > trying to get outbound dialing with voicepulse going - i've cut down the > asterisk to a very minimal install (1 SIP client) to try to localize the > problem. The SIP client works fine (SIP and * on the same NAT) and could > access the demo from samples before i removed it, and can call itself - so > > i am pretty convinced the SIP setup is OK. > > This is the error message: > Jan 29 12:21:54 NOTICE[262161]: app_dial.c:527 dial_exec: Unable to create > channel of type 'IAX2' > when i try to call the PSTN from the SIP device. > i've tried the wiki, the handbook, the voicepulse site, and all sorts of > other sites, and nothing helps. i also downloaded and compiled the code > today (jan 29) and that didn't help either. if anyone has ideas i would be > eternally grateful - this is driving me crazy. > > thanks- > yair > > p.s. i am using the right login and password; not the ones from the website, > > and i know my account at voicepulse works because i can connect direct > through a SIP client. it seems to be a specifically IAX2 problem. > > here are my files > > sip.conf > ; > ; SIP Configuration for Asterisk > ; > [general] > port = 5060 ; Port to bind to > disallow=all; Disallow all codecs > allow=ulaw ; Allow codecs in order of preference > allow=gsm > > [yairphone] > type=friend > insecure=no > username=yairphone > secret=yairphone > host=dynamic > dtmfmode=inband > callerID = "Yair Hakak" > nat=true > > extensions.conf > [general] > ; > static=yes > writeprotect=no > > > [default] > > exten => _1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],20) > exten => 8665,1,Dial(SIP/yairphone,20) > > iax.conf > [general] > port=5036 > disallow=all > allow=ulaw > > jitterbuffer=no > > [voicepulse] > context = VPWS > secret=mypassword > auth=md5 > type=friend > host=gw5.voicepulse.com > > _ > The new MSN 8: smart spam protection and 2 months FREE* > http://join.msn.com/?page=features/junkmail > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by The CCIS.net MailScanner, and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by the Bugs.Hamel.Net MailScanner, > and appears to be clean. > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Shot in the dark here ... Do you have: canreinvite=no Set in sip.conf for the SIP phones in question ? Ciao, -b Quoting Steve Foy <[EMAIL PROTECTED]>: > Hi, > > I've got a fairly working Asterisk setup, with a few minor glitches, one of > which is very very irritating. > > Sometimes, during a call, the remote end just drops off. We're using > software > SIP phones (SJPhone) connecting to * then out through analogue lines with > X100P cards. > > There is nothing in the logs and nothing on the console, the call just seems > to 'go away'! > > Can anyone shed any light on this? This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Try adding it to the phones involved so it looks like this: ; Shirley [100] type=friend username=xxx secret=xxx host=dynamic dtmfmode=rfc2833 callerid="Shirley O'Neill" <100> context=internal [EMAIL PROTECTED] qualify=yes canreinvite=no -b Quoting Steve Foy <[EMAIL PROTECTED]>: > Bill, > > On Fri, Jan 30, 2004 at 08:19:51AM -0500, Bill Hamel wrote: > > Shot in the dark here ... > > > > Do you have: > > > > canreinvite=no > > > > Set in sip.conf for the SIP phones in question ? > > No, I don't. > > All I have in sip.conf is the general stuff like: > >[general] >port = 5060 ; Port to bind to (SIP is 5060) >bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) > >allow=all >allow=GSM >allow=G729 >allow=iLBC >allow=SpeeX; Allow all codecs >allow=ulaw > > and then about 10 friends like this: > >; Shirley >[100] >type=friend >username=xxx >secret=xxx >host=dynamic >dtmfmode=rfc2833 >callerid="Shirley O'Neill" <100> >context=internal >[EMAIL PROTECTED] >qualify=yes > > -- > Steve Foy| http://www.unite.net > UNITE Solutions | Tel: 028 9077 7338 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by The CCIS.net MailScanner, and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by the Bugs.Hamel.Net MailScanner, > and appears to be clean. > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you turn on the 7960 msg waiting light?
I can only speak for the SIP IOS load on the 7960's (We're running 6.1 ) but if you add: [EMAIL PROTECTED] It "should" work Note: 7188 being the mail box number and "ContextInVoicemailConf " being the context in the "voicemail.conf" file where the mail box 7188 exists. Example: [7188] type=friend username=7188 secret=7188 host=dynamic nat=no dtmfmode=inband context=mycontext callerid="Bubba" < (555)-555-1212 > [EMAIL PROTECTED] canreinvite=no amaflags=default disallow=all allow=ulaw allow=alaw ;End HTH -b Quoting Paul Mahler <[EMAIL PROTECTED]>: > Does anyone in Asterisk land know how to turn on the message light on the > back of the earpiece of a cisco 7960 when a message is waiting? -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls dropping off
Hi, Have you checked for IRQ conflicts ? -b Quoting Steve Foy <[EMAIL PROTECTED]>: > Hi, > > On Mon, Feb 02, 2004 at 06:04:40PM +0100, Philipp von Klitzing wrote: > > Steve, > > > > this really is a FAQ. You need add to EACH (!) sip user something like > > > > disallow=all > > allow=ulaw > > allow=alaw > > allow=gsm > > I do have that in my sip.conf. I am using ulaw. > > Calls from the SIP phones through Asterisk and out one of my X100P cards are > working 95% of the time and also, incoming calls through the X100P cards to > the SIP phones are the same. > > The only problem is that every once in a while, without any odd circustances > that I can see, the call just drops and the remote user is gone. > > The box running asterisk isn't under heavy load, so I can't see why this is > happening. > > I am not using g.729 or 723, just plain old ulaw, which I have got enabled > in > sip.conf > > Cheers, > Steve > > -- > Steve Foy| http://www.unite.net > UNITE Solutions | Tel: 028 9077 7338 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by The CCIS.net MailScanner, and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by the Bugs.Hamel.Net MailScanner, > and appears to be clean. > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-grandstream call
I am trying to muddle my way tthrough getting something - actually anything to work - with Asterisk. I've acquired a Grandstream phone and I've got * on a Red Hat 9 box. I've gotten to a point where I can see (via ethereal) that the phone REGISTER's successfully with asterisk, and then I try to dial into voicemail. This is what I observe in the packet trace... GS: INVITE -> * *: Status 100 (Trying) -> GS *: Status 200 (OK with session description) -> GS So far, seems reasonable - but I'm a complete novice with this protocol. Then I see a huge stream of UDP packets sent by * to the GS on port 5004, but the GS only replies with an ICMP destination unreachable to each packet. I'm guessing that this is an RTP audio stream, but I don't know why the GS is not ready or otherwise unwilling to receive the packets. Examining the GS config, I've confirmed that the "local RTP port" is set to 5004. I have many questions about how this should work, but I'll save some bandwidth and leave it to someone here to suggest what should be checked next. Thanks. -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-grandstream call
Right - OK - sans comments for brevity: sip.conf: [general] port = 5060
; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context =
default ; Default for incoming calls [248379] username=billdesk
type=friend host=dynamic canreinvite=no mailbox=1234 context=demo
extensions.conf: [general] static=yes writeprotect=no [globals]
CONSOLE=Console/dsp ; Console interface for demo IAXINFO=guest ; IAXtel
username/password TRUNK=Zap/g2 ; Trunk interface TRUNKMSD=1 ; MSD digits
to strip (usually 1 or 0) [iaxtel700] exten =>
_91700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider] [trunkint] exten =>
_9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_9011.,2,Congestion [trunkld] exten =>
_91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91NXXNXX,2,Congestion [trunklocal] exten =>
_9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_9NXX,2,Congestion [trunktollfree] exten =>
_91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91800NXX,2,Congestion exten =>
_91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91888NXX,2,Congestion exten =>
_91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91877NXX,2,Congestion exten =>
_91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) exten =>
_91866NXX,2,Congestion [international] ignorepat => 9 include =>
longdistance include => trunkint [longdistance] ignorepat => 9 include
=> local include => trunkld [local] ignorepat => 9 include => default
include => parkedcalls include => trunklocal include => iaxtel700
include => trunktollfree include => iaxprovider [macro-stdexten]; exten
=> s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten
=> s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ u\
navail announce exten => s,3,Goto(default,s,1) ; If they press #, return
to start exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail
w/ busy \ announce exten => s,103,Goto(default,s,1) ; If they press #,
return to start [demo] exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line exten => s,3,DigitTimeout,5 ; Set
Digit Timeout to 5 seconds exten => s,4,ResponseTimeout,10 ; Set
Response Timeout to 10 seconds exten => s,5,BackGround(demo-congrats) ;
Play a congratulatory message exten => s,6,BackGround(demo-instruct) ;
Play some instructions exten => 2,1,BackGround(demo-moreinfo) ; Give
some more information. exten => 2,2,Goto(s,6) exten =>
3,1,SetLanguage(fr) ; Set language to french exten => 3,2,Goto(s,5) ;
Start with the congratulations exten => 1000,1,Goto(default,s,1) exten
=> 1234,1,Playback(transfer,skip) ; "Please hold while..." ; (but skip
if channel is not up) exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail exten =>
1236,1,Dial(Console/dsp) ; Ring forever exten => 1236,2,Voicemail(u1234)
; Unless busy exten => #,1,Playback(demo-thanks) ; "Thanks for trying
the demo" exten => #,2,Hangup ; Hang them up. exten => t,1,Goto(#,1) ;
If they take too long, give up exten => i,1,Playback(invalid) ; "That's
not valid, try again" exten => 500,1,Playback(demo-abouttotry); Let them
know what's going on exten =>
500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the Asterisk
de\ mo exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo
site exten => 500,4,Goto(s,6) ; Return to the start over message. exten
=> 600,1,Playback(demo-echotest) ; Let them know what's going on exten
=> 600,2,Echo ; Do the echo test exten => 600,3,Playback(demo-echodone)
; Let them know it's over exten => 600,4,Goto(s,6) ; Start over exten =>
8500,1,VoicemailMain exten => 8500,2,Goto(s,6) [default] include => demo
From: "Glenn Dalgliesh" <[EMAIL PROTECTED]> To:
<[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users]
asterisk-grandstream call Date: Mon, 9 Feb 2004 15:27:55 -0500 Reply-To:
[EMAIL PROTECTED] Please include your sip.conf and
extension.conf files. Hard to say what is wrong without seeing the
configuration - Original Message - From: "Bill Michaelson"
<[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday,
February 09, 2004 3:15 PM Subject: [Asterisk-Users] asterisk-grandstream
call
I am trying to muddle my way tthrough getting something - actually
anything to work - with Asterisk. I've acquired a Grandstream phone and
I've got * on a Red Hat 9 box. I've gotten to a point where I can see
(via ethereal) that the phone REGISTER's successfully with asterisk, and
then I try to dial into voicemail. This is what I observe in the packet
trace...
GS: INVITE -> *
*: Status 100 (Trying)
Re: [Asterisk-Users] asterisk-grandstream call
Arg.. my posting was mangled by text-wrapping. Sorry.
Here again...
sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
[248379]
username=billdesk
type=friend
host=dynamic
canreinvite=no
mailbox=1234
context=demo
extensions.conf:
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[iaxtel700]
exten => _91700NXX,1,Dial(IAX2/${[EMAIL PROTECTED]/${EXTEN:1[EMAIL PROTECTED])
[iaxprovider]
[trunkint]
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
exten => _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXX,2,Congestion
[trunklocal]
exten => _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXX,2,Congestion
[trunktollfree]
exten => _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXX,2,Congestion
exten => _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXX,2,Congestion
exten => _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXX,2,Congestion
exten => _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXX,2,Congestion
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maxi\
mum
exten => s,2,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w\
/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten => s,102,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ bu\
sy announce
exten => s,103,Goto(default,s,1) ; If they press #, return to start
[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats) ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct) ; Play some instructions
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,2,Goto(s,6)
exten => 3,1,SetLanguage(fr) ; Set language to french
exten => 3,2,Goto(s,5) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,2,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,2,Voicemail(u1234) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,2,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call the Asterisk\
demo
exten => 500,3,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,4,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,2,Echo ; Do the echo test
exten => 600,3,Playback(demo-echodone) ; Let them know it's over
exten => 600,4,Goto(s,6) ; Start over
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
[default]
include => demo
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2785 - 6 msgs
From: "Glenn Dalgliesh" <[EMAIL PROTECTED]> I am assuming the problem you are trying to solve is no audio. Are both = the phone and asterisk on public ip address? - The problem is the ICMP messages in response to what presumably is an audio stream, as originally described. Both devices are on the same LAN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk-grandstream call
>I am trying to muddle my way tthrough getting something - actually >anything to work - with Asterisk. I've acquired a Grandstream phone and >I've got * on a Red Hat 9 box. I've gotten to a point where I can see >(via ethereal) that the phone REGISTER's successfully with asterisk, and >then I try to dial into voicemail. This is what I observe in the packet >trace... > >GS: INVITE -> * >*: Status 100 (Trying) -> GS >*: Status 200 (OK with session description) -> GS Does the GS then send an ACK? It should. If it doesn't then this probably means that it hasn't received the 200 response. (firewall issue?) If it is sending the ACK, then it is probably a codec issue, as has been already mentioned. GS doesn't always seem to do very well in codec selection. Doug - Thanks for that hint. I see what you mean. When configured for FWD, the GS does indeed send an ACK at an equivalent point in the protocol. But no, the GS does not send an ACK when configured for my * box. I suppose the * box is expecting it, because about one second later, the * box resends the 200 message - this in spite of the fact that has started spewing RTP furiously. Both devices are on the same LAN, with no intervening firewall, and the OK ought to be visible to the GS (the packet even contains the expected destination MAC ID, derived earlier via ARP). That makes two mysteries: 1) why doesn't the GS seem to see the OK? and 2) why does * send the RTP stream in spite of the fact that it has not received the ACK from the GS? Shouldn't it wait? Regarding codec selection, I see a minor difference between the FWD and the local * box test cases, but I know nothing about the negotiation protocol... With FWD, the OK message lists 3 Media Formats: Media Description, name and address (m): audio 10496 RTP/AVP 0 8 101 Media Type: audio Media Port: 10496 Media Proto: RTP/AVP Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 But with the local box, it lists one other - note the addition of GSM... Media Description, name and address (m): audio 16708 RTP/AVP 3 0 8 101 Media Type: audio Media Port: 16708 Media Proto: RTP/AVP Media Format: 3 Media Format: 0 Media Format: 8 Media Format: 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Don't see much else different in the packets. It might also be relevant that the FWD connection, which works successfully, is through a firewall with NAT. Still fishing... thanks for your attention - much appreciate not being alone here!
[Asterisk-Users] Re: Asterisk<->GS and codec selection
> Regarding codec selection, I see a minor difference between the FWD > and the local * box test cases, but I know nothing about the > negotiation protocol... > > With FWD, the OK message lists 3 Media Formats: > > Bingo...GS chokes with GSM...just disallow it in your sip.conf: disallow=all allow=alaw allow=ulaw Thank you, very much. That got it working. Actually, I used disallow=gsm as suggested by someone else. Please forgive my ignorance, but this leaves open questions which are nagging me... I expected that the SIP dialog would be a negotiation such that the devices agree on a mutually acceptable encoding. And I think it's obvious (correct me if I'm missing any key points) that such a negotiation would involve selecting one of the encoding formats which appears in both lists presented by each side. It doesn't seem reasonable that the GS should just "flake out" as it seems to do, simply because it is offered an option it can't accept amongst ones that it can. Is this indeed what I am seeing, or am I mischaracterizing it? Also, as I noted earlier, shouldn't * wait for the ACK before spewing the audio stream? It appears to be missing the ACK because it retransmits the OK shortly after it begins sending the RTP data. These loose ends make me very uncomfortable.
Re: [Asterisk-Users] Digium connectivity issue?
I observed a packet routing endless loop at: 16 host-63-108-128-153.apid.com (63.108.128.153) This happened with traceroute from two distinct origination points. Seems to have been resolved. Message: 3 Date: Fri, 13 Feb 2004 20:11:44 -0500 From: John Fraizer <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Digium connectivity issue? Reply-To: [EMAIL PROTECTED] Rich Adamson wrote: > Are others seeing hugh delays and/or lack of connectivity to Digium? > > Rich > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > I saw an over 12hr outage to Digium and IAXtel and I know it wasn't OUR network so, they must have had some sort of problem. John [EMAIL PROTECTED]
[Asterisk-Users] Is there a MaxQueueTime for Queues ?
Hi, Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the "n" option with "queue" (if I don't the 2nd caller will stay in the queue infefinately) eg: exten => 401,1,Queue(support1|n) The problem with using "n" is that with one agent logged into the queue and he is "busy" on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten => 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold => default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail Password Digit Timeout
FromJim Burwell, Dec 21,2003 __ I had the same problem with Grandsteam phones and *. No other hard or soft phones have the 'double digit' problem with *. I don't think Asterisk can do both RFC2833 and in-band DTMF at the same time. It does, however, do RFC2833 and SIP Info at the same time (SIP Info method seems to be on all the time, even when RFC2833 is selected in the sip.conf file). Switching the Grandsteam to SIP Info allowed it to talk to Asterisk and fixed the double digits problem. - Jim __ Date: Sat, 14 Feb 2004 10:56:39 -0600 From: Rob Fugina <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail Password Digit Timeout Reply-To: [EMAIL PROTECTED] On Thu, Feb 12, 2004 at 04:30:19PM -0600, Ryan R. Fligg wrote: I was wondering if there was any way to change the digit timeout or some setting of that sort on the voicemail password entry. Currently when our users enter their passwords they have to enter them very rapidly, otherwise asterisk will log the number twice. So if someone entered a voicemail password of 1234 slowly and deliberately on our system the asterisk receives it as the following number, 11223344 and thus returns the passcode invalid message. System: Asterisk CVS-02/10/04-13:27:57 built by [EMAIL PROTECTED] on a i686 running Linux 3 X100P cards 5 Snom200 phones I can't help you, but I can "me too". I have a TDM400, and accessing voice-mail from these extensions is always fine. I also have a Grandstream SIP phone, and it behaves exactly as you describe. It has to do with how long the number buttons are pressed. To make it work, you have to key your PIN like the buttons are too hot to touch... I'm running the latest (.46) Grandstream firmware. I'm using "dtmfmode=rfc2833" in sip.conf, and the matched setting on the phone. Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent / Queue help
Hi, First let me apologize if I sent this to the list twice. Is it possible to kick a caller out of a queue after 5 minutes and goto the next priority in the context where they were assigned to the queue ? My desired result is that even though one agent is dynamically logged into the queue and is on a call, I would like the 2nd caller to stay in the queue for 5 minutes and then timeout to the next priority if the agent is still busy and can't get to the call. Some observations: I have tried the "n" option with "queue" (if I don't the 2nd caller will stay in the queue infefinately) eg: exten => 401,1,Queue(support1|n) The problem with using "n" is that with one agent logged into the queue and he is "busy" on a call, when the 2nd call is placed in the queue it immediately timesout and goes to the next priority in the context even if timeout=300 is set in queue.conf. Any help appreciated. -bh Here are the configs: extensions.conf [supportq] exten => 401,1, Queue(support1|t) agents.conf [agents] autologoff=15 ackcall=no ;wrapuptime=5000 musiconhold => default queues.conf general] [support1] music = default strategy = leastrecent ;context = leavemessage timeout = 300 retry = 2 maxlen = 0 -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inbound IAX to SIP
I've a SIP phone (GS 100) which dials out fine through a Voicepulse
Connect account via *.
And I've got a phone number which does DID in via IAX from Voicepulse.
I want it to ring the GS phone for now.
I have this in extensions.conf:
[voicepulse-incoming]
; This context tells Asterisk what to do with
; incoming calls from VoicePulse (if you have signed
; up for DIDs
;
; We should now hear a "congratulations" recording
; on incoming calls to our VoicePulse phone number.
; Once we know that's working, we'll change this to a
; "Dial" statement (or something else depending on our
; needs).
;exten => _NXXNXX,1,Playback(demo-congrats)
exten => _NXXNXX,1,Dial(SIP/248379)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
; busy condition N+101...
exten => _NXXNXX,102,Playback(demo-congrats)
And sip.conf:
[248379]
type=friend
host=dynamic
canreinvite=no
mailbox=1234
context=demo
disallow=gsm
dtmfmode=inband
But the phone won't ring... it acts busy and I don't understand why.
Here is some console info...
-- Accepting AUTHENTICATED call from 66.234.228.132, requested
format = 4, actual format = 4
-- Executing Dial("[EMAIL PROTECTED]/2", "Sip/248379") in
new stack
Feb 17 18:17:56 NOTICE[1209214528]: app_dial.c:506 dial_exec: Unable to
create channel of type 'Sip'
== Everyone is busy at this time
-- Executing Playback("[EMAIL PROTECTED]/2",
"demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
== Spawn extension (voicepulse-incoming, 6094556707, 102) exited
non-zero on '[EMAIL PROTECTED]/2'
-- Executing Hangup("[EMAIL PROTECTED]/2", "") in new stack
== Spawn extension (voicepulse-incoming, h, 1) exited non-zero on
'[EMAIL PROTECTED]/2'
-- Hungup '[EMAIL PROTECTED]/2'
There is also:
*CLI> sip show peers
Name/usernameHost Mask Port Status
248379 (Unspecified) (D) 255.255.255.255 0Unmonitored
Clues gratefully accepted.
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[Asterisk-Users] System cmd usage
Using John Todd's example for recording, from his cleanup/conversion macro...
; Turn the two in/out
.wav files into a single .wav file with both channels
exten => s,3,System(/usr/local/bin/wmix ${MONITORDIR}/${CALLFILENAME}-in.wav
${\
MONITORDIR}/${CALLFILENAME}-out.wav > ${MONITORDIR}/${CALLFILENAME})
;
; Remove the old .wav files - we don't need them anymore.
exten => s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav ${MONITORDIR}/\
${CALLFILENAME}-out.wav)
;
; This part of the routine compresses the .wav files into a .gsm file for
; better storage (about 1/5 the size of a .wav file). Use "untoast" to
restor\
e
; to normal wav file format. (toast and untoast are fairly standard on Linux
s\
ystems)
;
exten => s,5,System(/usr/bin/toast -F ${MONITORDIR}/${CALLFILENAME})
The wmix runs successfully (it produces the mixed file), and running "by
hand" from the shell indicates that it returns 0 to the shell. But the *
console log seems to think it failed...
-- Executing System("SIP/248379-fe6e",
"/usr/local/bin/wmix /var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav >
/var/spool/asterisk/monitor/20040220-121235-111-916095326873") in new stack
Feb 20 12:12:56 WARNING[1209214528]: app_system.c:57 system_exec: Unable
to execute '/usr/local/bin/wmix /var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav >
/var/spool/asterisk/monitor/20040220-121235-111-916095326873'
== Spawn extension (macro-record-cleanup, s, 3) exited non-zero on 'SIP/248379-fe6e'
in macro 'record-cleanup'
== Spawn extension (intern-post, s, 1) exited non-zero on 'SIP/248379-fe6e'
Any ideas why?
[Asterisk-Users] Problem playing the first voice mail prompt
I dial an extension that starts up VoiceMailMain. When the call comes in the following lines are written to /var/log/messages: Feb 20 11:01:37 redhat2 kernel: Zapata Telephony Interface Registered on major 196 Feb 20 11:01:37 redhat2 kernel: No ISA tormenta card found at d Feb 20 11:01:37 redhat2 kernel: Zapata Telephony Interface Unloaded Feb 20 11:01:37 redhat2 insmod: /lib/modules/2.4.20-8/misc/torisa.o: init_module: Input/output error Feb 20 11:01:37 redhat2 insmod: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg Feb 20 11:01:37 redhat2 insmod: /lib/modules/2.4.20-8/misc/torisa.o: insmod char-major-196 failed With "sip debug" on the CLI interface I get the warning at line 521 in file.c, something about the "file can't be written". I have a system that has no Zapata cards at all. Do I need to have one? Any ideas as to what may be wrong? Thanks, Bill Hamlin Globalnet ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: System call succeed, asterisk sees failure
Then I infer that the asterisk process is improperly retrieving or interpreting the System process completion code. That would be a serious bug that could break a lot of applications. I wonder if it is specific to some installations or more widespread.
The validity of the code in app_system.c is unclear to me at first glance...
res = system((char *)data);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
} else {
if (res && ast_exists_extension(chan, chan->context, chan->exte\
n, chan->priority + 101, chan->callerid))
chan->priority+=100;
res = 0;
}
My reading of man pages indicates that the status return by system(2) (refer to wait()) is more than just the value set by an exit() call or returned by a main() function, which seems to be restricted to the low-order byte. I haven't studied it through, but I'm wondering if the hi-order bit can be set, thus causing (res < 0) in spite of successful process completion (returning 0).
Could this be the problem?
From: Eric Stanley <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System cmd usage
Date: Fri, 20 Feb 2004 12:22:12 -0600
Reply-To: [EMAIL PROTECTED]
I saw the same thing. I think I determined that it always failed at the same
point in the macro, no matter what command was being executed. I just put
the whole cleanup process in a shell script and I execute the shell script
from the macro.
Eric
Message: 2
Date: Fri, 20 Feb 2004 12:48:36 -0500
From: Bill Michaelson <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] System cmd usage
Reply-To: [EMAIL PROTECTED]
--03060507040102040002
Content-Type: text/plain; charset=us-ascii; format=flowed
Content-Transfer-Encoding: 7bit
Using John Todd's example for recording, from his cleanup/conversion
macro...
; Turn the two in/out .wav files into a single .wav file with both channels
exten => s,3,System(/usr/local/bin/wmix
${MONITORDIR}/${CALLFILENAME}-in.wav ${\
MONITORDIR}/${CALLFILENAME}-out.wav > ${MONITORDIR}/${CALLFILENAME})
;
; Remove the old .wav files - we don't need them anymore.
exten => s,4,System(/bin/rm ${MONITORDIR}/${CALLFILENAME}-in.wav
${MONITORDIR}/\
${CALLFILENAME}-out.wav)
;
; This part of the routine compresses the .wav files into a .gsm file for
; better storage (about 1/5 the size of a .wav file). Use "untoast" to
restor\
e
; to normal wav file format. (toast and untoast are fairly standard on
Linux s\
ystems)
;
exten => s,5,System(/usr/bin/toast -F ${MONITORDIR}/${CALLFILENAME})
The wmix runs successfully (it produces the mixed file), and running "by
hand" from the shell indicates that it returns 0 to the shell. But the
* console log seems to think it failed...
-- Executing System("SIP/248379-fe6e", "/usr/local/bin/wmix
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav >
/var/spool/asterisk/monitor/20040220-121235-111-916095326873") in new stack
Feb 20 12:12:56 WARNING[1209214528]: app_system.c:57 system_exec: Unable
to execute '/usr/local/bin/wmix
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-in.wav
/var/spool/asterisk/monitor/20040220-121235-111-916095326873-out.wav >
/var/spool/asterisk/monitor/20040220-121235-111-916095326873'
== Spawn extension (macro-record-cleanup, s, 3) exited non-zero on
'SIP/248379-fe6e' in macro 'record-cleanup'
== Spawn extension (intern-post, s, 1) exited non-zero on
'SIP/248379-fe6e'
Any ideas why?
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[Asterisk-Users] System called seems forked up
Attempting to correct the problem about which I earlier posted - wherein
a system() call which apparently succeeds is perceived to have failed by
the * process, I changed code in app_system.c so that it would be more discerning...
res = system((char
*)data);
/*
if (res < 0) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
} else if (res == 127) {
ast_log(LOG_WARNING, "Unable to execute '%s'\n", (char *)data);
res = -1;
*/
if (res == -1) {
ast_log(LOG_WARNING, "Fork failed for '%s'\n", (char *)data);
res = -1;
} else if (WEXITSTATUS(res) != 0) {
ast_log(LOG_WARNING, "Error completion for '%s'\n", (char
*)data);
res = -1;
It is now indeed more discerning, but it has reported Fork failed. But the
fork most certainly has not failed! The shell command invoked has run, and
what's more, completed successfully, producing the expected files.
Referring to the system(2) man page (Red Hat 9, stock)...
RETURN VALUE
The value returned is -1 on error (e.g. fork failed), and the return
status of the command otherwise. This latter return status is in
the
format specified in wait(2). Thus, the exit code of the command
will
be WEXITSTATUS(status). In case /bin/sh could not be executed,
the
exit status will be that of a command that does exit(127).
...and noting that "fork failed" is only an example of an error, I'm wondering
what *other* condition might cause the -1 return value.
Does anyone have any ideas?
[Asterisk-Users] Re: System called seems forked up
>It is now indeed more discerning, but it has reported Fork failed. But >the fork most certainly has not failed! The shell command invoked has >run, and what's more, completed successfully, producing the expected files. > Does anyone have any ideas? [EMAIL PROTECTED] suggested: Can you check the errno? strerror(errno); should give you a string of why it failed. (Just be careful not to use other stuff which touches errno after the fork() Of course - very good suggestion (embarrassed I didn't think of it)... anyway... it returns 10, which perror tells me is "No child processes". Sooo, I suppose the spawned process is somehow disassociated from the process group prior to execution of the wait() embedded within the system()? Duuh... I'm still stumped, but I guess we are on to something? On the other hand, if a fork does really fail, one might expect errno to be 10 in that case too. I've half a mind to break it out into a fork/exec/wait for myself, but, uh, ugh. I guess I'm lazy. Please, briliant insights, anybody?
[Asterisk-Users] Re: System call forked - more stuff
It gets better (worse)...
I had been testing with console (-c) mode. When I allow * to run background,
it crashes after the system() call (which succeeds, by the way). The -vvv
option yields these final messages before *poof*...
== Spawn extension
(intern-post, 112, 1) exited non-zero on 'SIP/248379-bcdc'
-- Executing Macro("SIP/248379-bcdc", "record-cleanup") in new stack
-- Executing SetVar("SIP/248379-bcdc", "MONITORDIR=/var/spool/asterisk/monitor")
in new stack
Feb 21 09:56:57 WARNING[1209214528]: ast_expr.y:346 ast_yyerror: ast_yyerror():
syntax error: parse error
-- Executing GotoIf("SIP/248379-bcdc", "0?6:3") in new stack
-- Goto (macro-record-cleanup,s,3)
-- Executing System("SIP/248379-bcdc", "/usr/local/bin/wmix /var/spool/asterisk/monitor/20040221-095652-111-112-in.wav
/var/spool/asterisk/monitor/20040221-095652-111-112-out.wav > /var/spool/asterisk/monitor/20040221-095652-111-112")
in new stack
I don't know what the yyerror is about either.
[Asterisk-Users] outdial broadcast
Can someone refer me to an example of an automated broadcasting operation that sends a canned voice message to a list of phone #'s? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hotel wake-up
Anybody know how to implement a hotel wake-up call feature with *? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] outgoing spool parallelism
Thanks for the suggestions on the hotel wake-up! Actually, I don't have a hotel, but my earlier request was unanswered because I suppose it was uninspiring. So I used a hard example that was readily identifiable. Your helpful responses led me to the facility I had not managed to find by myself in the docs. Now that I've tried it, and it works, I've got some more specific questions about it's operation... How does * manage concurrency when processing files in the outgoing directory? Does it have some kind of intelligence or controlling mechanism which serializes requests based on the capacity of resource combinations required to satisfy the requests? Or is it just a single thread/processing queue for all requests found in the spool dir? Also, is there any way to control the sequencing (priority) of the "enqueued" requests? Or is it a random free-for-all? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Outgoing parallelism
Thanks, Scott. I'm in a general exploration mode, but I do have a small
broadcast application in mind. My limited experimentation leads me to
suspect that there is no queue management at all. I was testing with
only a single call file just minutes ago, and the system tried to redial
the destination as a retry (60 second interval had been spec'ed), even
though the first call was still in progress!
I suppose I will have to manage throttling with some kind of completely
external process, which is likely to be cumbersome. For the immediate
application, and given my current facilities, single threading will be
adequate (and necessary), but from what I've seen, even this could be
challenging. If I put together anything generally useful, I'll share it.
From: "Scott Stingel" <[EMAIL PROTECTED]>
Hi Bill-
I've built some load testers for asterisk, using the outgoing call facility.
It's been a little while, so you may want to test this yourself, but I
recall finding a couple of problems:
(a) I don't think it manages queuing very well if there are a limited amount
of outbound resources. For example (again, from memory), if you define a
group ("g9" for example) of two lines for use in outbound calling, it works
fine if the number of outbound calls to be made at any moment never exceeds
2. A third call file in this example, will be grabbed by asterisk, but will
fail immediately. So I had to create a mechanism in my Perl script to
ensure that the outbound calls actually completed - no easy feat since I
couldn't tell when that occurs from the perl script too easily.
(b) There was a problem dumping more than about 12-15 outbound calls at once
in the outgoing directory, even if there were plenty of channels available
to make the calls. Asterisk would grab them but would not process some of
them. This is a load-testing scenario, and not too common I realise, but
something to be aware of. It didn't seem to matter if I switched to a more
powerful processor.
These problems occurred using a December release of asterisk - maybe they
are fixed now??
Please let me know if you are doing any load testing, and I'll send you some
simple scripts if you like.
The outgoing facility works fine at lower call volumes, if you stagger the
creation of the files in the outgoing directory.
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[Asterisk-Users] IAX Native bridge
I coded a dialplan that conditionally forwards a call to my cell phone if no answer on site. During a test, I received a call (via Voicepulse IAX) which correctly Dial'ed out to my cell phone (also via Voicepulse) as expected. Fine - it worked - except that the voice delay was so extreme (> 1sec). But the interesting part came next... -- Call accepted by 66.234.228.132 (format ULAW) -- Format for call is ULAW -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 is ringing -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 stopped sounds -- IAX2[voicepulse]/6 answered [EMAIL PROTECTED]/1 -- Attempting native bridge of [EMAIL PROTECTED]/1 and IAX2[voicepulse]/6 -- Channel '[EMAIL PROTECTED]/1' ready to transfer -- Channel 'IAX2[voicepulse]/6' ready to transfer -- Releasing IAX2[voicepulse]/6 and [EMAIL PROTECTED]/1 -- Hungup 'IAX2[voicepulse]/6' Native bridge? Cool! I says to myself. I figure the call will be released from * and handled entirely by Voicepulse, who I assume will bill me appropriately for the remainder of the "outgoing" call. And I'll get better quality for the remaining duration. But the call instead is dropped at this point instead - both sides disconnected from the cloud. Anybody know why and how this is controlled and what my options are? -- Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED] Thanks for putting up with my spam filter!
[Asterisk-Users] Dial via X100P
Just connected my X100P to Verizon. I stumbled across a config that
works, for the moment, with this Dial command:
;this works, because it prefixes a 1 on the dialing. But why does it?...
exten => _NXX,1,Dial(Zap/1/609${EXTEN}|55)
The comment says it all. The card/SW seems to dial a 1 before it dials
the 609${EXTEN}
Unless I'm misinterpreting what is happening?
This obviously limits my possibilities. Can somebody explain to me why
it dials 1, or appears to?
--
Bill Michaelson - COS, Incorporated - Software Development - [EMAIL PROTECTED]
Thanks for putting up with my spam filter!
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[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3092 - 11 msgs
I know that the 1 denotes the Zap channel number. That's why I would
not expect it to dial a 1. But it apparently does dial a one. Hence my
original question.
If it did not dial a 1, it would not work because a 1 is required for
the called number, as coded, to work properly with the local phone service.
Furthermore, I discovered this because I originally coded it this way:
exten => _NXX,1,Dial(Zap/1/${EXTEN}|55)
...which simply timed out on the line and failed. Experimentally, I
determined that the telco was expecting 3 more digits, in spite of the
fact that 7 digit dialing is normal for the line.
From: Asterisk Learner <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Dial via X100P
It does not dial a 1. The '1' denotes the Zap channel number which in
this case is probably your X100P. Zap channels are assigned to Zap ports
depending on the order in which you do a modprobe on them.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill
Michaelson
Sent: Saturday, March 13, 2004 2:18 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dial via X100P
Just connected my X100P to Verizon. I stumbled across a config that
works, for the moment, with this Dial command:
;this works, because it prefixes a 1 on the dialing. But why does
it?...
exten => _NXX,1,Dial(Zap/1/609${EXTEN}|55)
The comment says it all. The card/SW seems to dial a 1 before it dials
the 609${EXTEN}
Unless I'm misinterpreting what is happening?
This obviously limits my possibilities. Can somebody explain to me why
it dials 1, or appears to?
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[Asterisk-Users] Re: SIPURA 2000 Problems (Senad Jordanovic)
I have had a similar problem upgrading to .24 . Sipura support suggested using tftp which worked successfully. On the tftp server you use the URL http://aaa.bbb.ccc.ddd/upgrade?/path_name/spa.bin where aaa.bbb.ccc.ddd is the IP address of the Sipura. Do not know why these instructions are not in the manual. -- Bill From: "Senad Jordanovic" <[EMAIL PROTECTED]> To: "'Stefan Meier'" <[EMAIL PROTECTED]> Cc: <[EMAIL PROTECTED]> Date: Tue, 16 Mar 2004 16:50:08 - Subject: [Asterisk-Users] SIPURA 2000 Problems Reply-To: [EMAIL PROTECTED] * I can not update device to latest .31 firmware. It just sits there waiting for SPA 2000 to "not to be in use". I waited and waited for many many minutes... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk in the news
http://www.tmcnet.com/tmcnet/articles/2004/031704rt.htm Previous article by same author: http://www.tmcnet.com/it/0104/0104PO.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] question about CPU usage
I've had my asterisk running for a couple of weeks and just noticed that it takes about 98% of the CPU time (Linux RH9). Is this what you would expect? Is it just that the program is polling for things to do, calling "sleep(0)" or something simlar so as to relinquish the machine but otherwise polling like crazy? Thanks, Bill Hamlin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question about CPU usage
What is it about asterisk that makes this happen? My other apps that wait on a select take hardly any CPU time at all. I didn't find anything like "ldassume" using google. Can you tell me more about that? Thanks, Bill. > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Steven > Critchfield > Sent: Monday, March 22, 2004 4:07 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] question about CPU usage > > > On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: > > I've had my asterisk running for a couple of weeks and just > noticed that it > > takes about 98% of the CPU time (Linux RH9). Is this what you > would expect? > > Is it just that the program is polling for things to do, > calling "sleep(0)" > > or something simlar so as to relinquish the machine but > otherwise polling > > like crazy? > > Do a google search. I believe there is a export line you need for RH to > behave more sanely. Something like ldassume_2_4_1. Or you could switch > to a more free distro and it will fix itself. > -- > Steven Critchfield <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200
You must have port mapping in the Linux NAT that allows the SIP-level packets to get to the * Server, so you need to add a port mapping for the RTP packets. I may be wrong but I think * sends RTP on the same port that it receives RTP on, so once the phone sends some RTP to * then the RTP coming back should work. Turn on "sip debug" to see the packets and cut and paste here if you're still having a problem. Bill > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Geert Nijpels > Sent: Monday, March 22, 2004 4:25 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Snom 200 > > > Barry Fawthrop wrote: > > >Progress > > > >It seems I can't hear the Say Time, due to RTP Double NAT > >I'm guess this is both problem 1 and 2 really issue. > > > >My config: > >IP Phone <-> Router (Nat) <-> Internet <-> Linux (NAT) <-> * Server > > > >ANyone know of work arounds the double NAT? or other methods > >to route RTP with snom 200s, to work around this? > > > > > I think you can make progress with the following link: > http://www.voip-info.org/tiki-index.php?page=NAT%20and%20VOIP > > Have fun, > > Geert > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] question about CPU usage
Nope same problem. I just started it and did a couple of ps aux's and got this output: [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.6 1.3 115880 6676 ? R15:43 1:10 asterisk -vgcd root 20221 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.3 1.3 115880 6676 ? R15:43 1:13 asterisk -vgcd root 20223 0.0 0.1 3568 624 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 91.7 1.3 115880 6676 ? R15:43 1:16 asterisk -vgcd root 20225 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.4 1.3 115880 6676 ? R15:43 1:18 asterisk -vgcd root 20227 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# ps aux|grep ast root 20140 92.6 1.3 115880 6676 ? R15:43 1:20 asterisk -vgcd root 20229 0.0 0.1 3572 628 pts/2S15:44 0:00 grep ast [EMAIL PROTECTED] root]# > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel > Sent: Monday, March 22, 2004 4:36 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] question about CPU usage > > > I think Steve is referring to the following line: > > export LD_ASSUME_KERNEL=2.4.1 > > If you put this in your command line before starting asterisk, > you will get > around the RH9 problem of leaving zombies when AGI processes quit. Other > than that, I don't think it influences CPU load. > > Note that the line is not necessary for Fedora Core 1 > > regards > Scott > > Scott M. Stingel > Emerging Voice Technology Inc. > Palo Alto, California and London, England > > Email: scott "at" evtmedia.com > URL:www.evtmedia.com > > > > >-Original Message- > >From: [EMAIL PROTECTED] > >[mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamlin > >Sent: Monday, March 22, 2004 9:22 PM > >To: [EMAIL PROTECTED] > >Subject: RE: [Asterisk-Users] question about CPU usage > > > >What is it about asterisk that makes this happen? My other > >apps that wait > >on a select take hardly any CPU time at all. > > > >I didn't find anything like "ldassume" using google. Can you > >tell me more > >about that? > > > >Thanks, > >Bill. > > > >> -Original Message- > >> From: [EMAIL PROTECTED] > >> [mailto:[EMAIL PROTECTED] Behalf Of Steven > >> Critchfield > >> Sent: Monday, March 22, 2004 4:07 PM > >> To: [EMAIL PROTECTED] > >> Subject: Re: [Asterisk-Users] question about CPU usage > >> > >> > >> On Mon, 2004-03-22 at 14:49, Bill Hamlin wrote: > >> > I've had my asterisk running for a couple of weeks and just > >> noticed that it > >> > takes about 98% of the CPU time (Linux RH9). Is this what you > >> would expect? > >> > Is it just that the program is polling for things to do, > >> calling "sleep(0)" > >> > or something simlar so as to relinquish the machine but > >> otherwise polling > >> > like crazy? > >> > >> Do a google search. I believe there is a export line you > >need for RH to > >> behave more sanely. Something like ldassume_2_4_1. Or you > >could switch > >> to a more free distro and it will fix itself. > >> -- > >> Steven Critchfield <[EMAIL PROTECTED]> > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> To UNSUBSCRIBE or update options visit: > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > >___ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Watchguard Firebox 1000 and Asterisk
The firebox has the UDP timeout set pretty low by default, this is a good thing to help prevent DOS attacks, but isn't a really good thing for a SIP device. There is no option in the GUI to set this. However you can go into the config file itself and modify the following: options.masquerade.udp.timeout: 30 options.services.dynamic.timeout.udp: 25 Set them higher than your "register timeout" on your 7960. Then save the config file and upload to the firebox. HTH -bh Quoting Glenn Dalgliesh <[EMAIL PROTECTED]>: > Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I > have asterisk on public side and phones on the private side. I am able to > get the phones to register and make outbound calls but the inbound calls are > intermittent. I have NAT enable in asterisk and on the Cisco 7960. > > Any insight would be appreciated. > > Thanks > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G726 not working ?
Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == Registered file format g726-40, extension(s) g726-40 == Registered file format g726-32, extension(s) g726-32 == Registered file format g726-24, extension(s) g726-24 == Registered file format g726-16, extension(s) g726-16 I 'Ass'ume this indicates that g726 is installed.. So in my sip.conf I put many variations of what I thought should go in there, finally includeing (to no avail): disallow=all allow=g726-40 allow=g726-32 allow=g726-24 allow=g726-16 allow=g726 allow=ima-adpcm (Also tried G.726-xx etc... ) And none seem to work because when I dial out I get Mar 30 15:49:29 WARNING[-1137677392]: chan_sip.c:2128 process_sdp: No compatible codecs! I must not be putting the correct "allow=" value in sip.conf or possibly missing something. Can anyone point me in the right direction ? Thanks -bh -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
