Re: [asterisk-users] retry loop in ansible ?
On 12/6/23 02:08, Axel Rau wrote:
I have a simple config with some phones ringing simultaneously.
Some of them are softphones (zoiper apps on iPhone w/o push notification).
If such an app did bot register in time, it has no chance to pick up the call.
If I could configure a retry loop checking for registered candidates,
say once a second until one phone takes the call, this would allow me
to pick up the call with zoiper app registered late.
How could this be done in ansible?
Did you mean asterisk ?
If so, then you might look into the While()/EndWhile() applications,
combined with timeouts to Dial() application, starting with something
very basic such as the following:
same = n,Set(tries=0)
same = n,While($[${INC(tries)}<99])
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(1234)})
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(5678)})
same = n,Dial(${team},10)
same = n,Wait(1)
same = n,EndWhile()
...at most that would be 11 seconds in between registration of x5678 and
the next time it gets called when x1234 is not answering.
Other approaches might involve Queue()'s with some ChannelRedirect()'s
or even Bridge()'s, maybe AGI/ARI, etc.
BTW the Asterisk Forums are a great place to post these kinds of
questions in the future: https://community.asterisk.org
Regards,
--
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Beyond 1-833-PNGN-PBX
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Re: [asterisk-users] 60+ devices in confbridge and dropping audio
On 8/21/23 08:23, Jerry Geis wrote: I am using asterisk 18.14.0 and chan_sip. confbridge has dsp_drop_silence=yes The conf joins all the endpoints in a one-way conf. 60+ devices and packets choppy or dropping audio. The CPU is decent at Intel(R) Xeon(R) CPU E3-1240 v5 @ 3.50GHz What else might I tweak to get this working without audio dropping ? not much else is running on the server besides - asterisk. What else have you tried ? jitterbuffer ? mixing_interval ? codecs ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Alternative to Local channel
On 8/17/23 05:04, Federico wrote: Yes that are, but how do I use them to execute a part of the dialplan, once, when Asterisk starts up. Systemd provides "ExecStartPost=" option to run more commands eg. cp /usr/local/src/asterisk/sample.call /var/spool/asterisk/outgoing/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault
On 8/17/23 12:44, John Harragin wrote: You should be able to define multiple data sources. However I'm having my own issues. I have my dialplan accessing one maria database which is hosted locally on the asterisk server then logging cdr with odbc adaptive which connects to maria on a remote machine. This works fine except when the remote server is out of reach the calls zombie and eventually lead to a fault. Setting "batch=yes" in your /etc/asterisk/cdr.conf file may fix this by moving the database interaction to separate threads away from the call. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Noob Service Provider T1/T400p physical interfacing question
On Wed, 14 Jul 2004, Kris Boutilier waxed: Is it enough to simply plug an incoming T1 line in to a Digium T100p card or should I pass the connection through some form of local CSU to provide isolation, buffering, local diags and so on? Plug it right in there. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Music on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed: FC1 What I don't understand is why it works using the -vgcd but not when just running asterisk ? Are there any log messages about the mp3 player not being spawned ? Like Fork failed or unable to spawn mp3player ? I am unfamiliar with how FC1 starts a service. Is this something you added yourself ? --Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Maj Sent: Wednesday, July 14, 2004 5:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Music on hold On Wed, 14 Jul 2004, Hall, Eric M. waxed: I have been working on the music on hold part for a few hours today and I found something that just doesn't sound right. If I just run asterisk via service service asterisk start' everything work but MOH If I run it via asterisk -vgcd MOH works... Any idea what the difference is ? MOH is done via external mpg123 processes, maybe the service stuff doesn't like spawning external processes ? What distro are you running ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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 -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Music on hold
On Thu, 15 Jul 2004, Hall, Eric M. waxed:
Nothing in the logs about mp3. the startup script is
/etc/rc.d/init.d/asterisk
How about fork failed ?
Here is the file
Hmm, this is similar, but not the same, to the file found in
the asterisk source contrib/init.d/ directory for RH. I
guess services are still the same as in Fedora, tho.
[EMAIL PROTECTED] root]# cat /etc/rc.d/init.d/asterisk
#!/bin/bash
#
# chkconfig: 2345 99 15
# description: Open source PBX
# processname: asterisk
# source function library
. /etc/rc.d/init.d/functions
RETVAL=0
case $1 in
start)
echo -n Starting Asterisk PBX:
/sbin/modprobe ixj
daemon /usr/sbin/asterisk
Try changing this line to daemon safe_asterisk and make
sure safe_asterisk is installed. You'll find that in the
asterisk source contrib/scripts/ directory. Put it
somewhere in the path, like /usr/sbin would be good.
I don't actually use this script on RH, I only use the
safe_asterisk script from S99local. But that's pretty
non-standard.
RETVAL=$?
echo
[ $RETVAL -eq 0 ] touch /var/lock/subsys/asterisk
;;
stop)
echo -n Shutting Asterisk PBX:
killproc asterisk
/sbin/rmmod -r ixj
RETVAL=$?
echo
[ $RETVAL -eq 0 ] rm -f /var/lock/subsys/asterisk
;;
restart|reload)
$0 stop
$0 start
RETVAL=$?
;;
status)
status asterisk
RETVAL=$?
;;
*)
echo Usage: asterisk {start|stop|status|restart|reload}
exit 1
esac
exit $RETVAL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Thursday, July 15, 2004 3:37 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Music on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
FC1
What I don't understand is why it works using the -vgcd but not
when just running asterisk ?
Are there any log messages about the mp3 player not being spawned ?
Like Fork failed or unable to spawn mp3player
?
I am unfamiliar with how FC1 starts a service. Is this something you
added yourself ?
--Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of C. Maj
Sent: Wednesday, July 14, 2004 5:26 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Music on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed:
I have been working on the music on hold part for a few hours today
and I found something that just doesn't sound right.
If I just run asterisk via service service asterisk start'
everything
work but MOH If I run it via asterisk -vgcd MOH works...
Any idea what the difference is ?
MOH is done via external mpg123 processes, maybe the service stuff
doesn't like spawning external processes ? What distro are you
running ?
--Chris
--
Chris Maj, Rochester
cmaj_at_freedomcorpse_dot_com
Pronunciation Guide: Maj == May
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Re: [Asterisk-Users] DID AND EXTENSION DIALED NUMBER FORWARD
On Thu, 15 Jul 2004, Celedonio Albarran waxed: We have look at wiki and in the list archive and have not found any related to it. Have you looked at the consultants page on the wiki ? The question you pose is fairly basic setup. It doesn't look like you've even *tried* to do any configuration. Might as well pay someone to get you up and running, at least at first. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] can you trust CDR for billing information?
On Wed, 14 Jul 2004, Johannes van Hulst waxed: 8's I called out on my X100T card. It's not as reliable on an analog interface versus a digital one, ie. a PRI. You will need to look at things like callprogress in your config files and google for more. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Music on hold
On Wed, 14 Jul 2004, Hall, Eric M. waxed: I have been working on the music on hold part for a few hours today and I found something that just doesn't sound right. If I just run asterisk via service service asterisk start' everything work but MOH If I run it via asterisk -vgcd MOH works... Any idea what the difference is ? MOH is done via external mpg123 processes, maybe the service stuff doesn't like spawning external processes ? What distro are you running ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Re: Audio filters (was: feature - VM gain adjust?)
On Wed, 14 Jul 2004, [EMAIL PROTECTED] waxed: On Wed, 14 Jul 2004, John Todd wrote: This second method also assumes that Dial is capable of intercepting DTMF, doing some dialplan logic, and then re-connecting the two legs together without hanging up either leg (see my many, many previous discussions on this topic.) Yeah! I'd love to be able to do this - basically to be able to have a context eavesdropping on each side of the call - ordinary pbx.c logic could match digits heard on the call. What fun could be had! Wouldn't using # to transfer (then doing some extension magic) help accomplish this ? Or have I missed something ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] ZapBarge and SIP Channels
On Mon, 12 Jul 2004, Mamadou Lamine KA waxed: Is there any alternative to Asterisk ZapBarge command for SIP and IAX channels? Set up a silent meet me conference for the channels you want to listen in on. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] zaptel debugging tools
On Mon, 12 Jul 2004, Glen Hinkle waxed: Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? * zttool is in the zaptel source directory * you could issue set verbose 10 on the asterisk CLI * you could issue pri debug span x on the asterisk CLI Also, try getting a PRI trace from your telco. But what exactly are you trying to debug ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Manager help
On Mon, 12 Jul 2004, Kyle Hagan waxed: Can someone give me an example of what comes from the manager when a call comes in on a PRI./T1 when a call comes in? On the x100p i ger the following: Event: Newchannel Channel: Zap/1-1 - I need what would be populated in here Would it be same? ZAP/(card)-(1 to 100, for 100 block DID)? No. Take Zap/X-Y: X is the channel number (1-23 for a PRI T1, see zapata.conf) Y is either 1 (usual) or 2 (for three way calling) The DID would most likely be the Exten: value. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Two outbound calls at once
On Thu, 8 Jul 2004, David Goldfein waxed: Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI 8's It doesn't look like you have a channel collision problem, other than the same far end number being dialed. Are you able to place at least one call with success ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Predictive Dialers
On Fri, 9 Jul 2004, mattf waxed: - Shady-dial (http://shadydial.sourceforge.net/) Lead by some nice Europeans, they have a beta of it up and running supposedly handling up to 10 agents per server, although I'm not sure of exactly what level of 'Predictive' the dialer is(whether it detects voicemail/answering machines and such). This dialer alters the code of Asterisk and is dependant upon PostgreSQL as a database backend. There is some documentation on installation and usage and it is released mostly under the GPL. This dialer does not restrict what kind of phones you can use with it(I'd love to hear more from them or people using this system on the specifics of their project) Damn, I wish I was European some days ! Stupid Bush. But alas, I am an imperialist American, whose lowly shady dial is really the bare bones of predictive dialers. I have been customizing the system for a client, however, and can't decide yet how much should go back into the project CVS. Of course, any changes to the * code base will always be there; specifically these are changes to allow phone key results to PostgreSQL. I think the rest have been incorporated thru the bug tracker by now as they were generic enough to be useful by others. The changes outside CVS are many, including reporting, book management, and utilizing 2 line LCD displays for name and address information. The code is all still written in Tcl. But while the core dialing routines will be updated in CVS, it's very much customized for this client's particular needs in other areas, so I'm not sure all of that code will make it back in. However, I am planning to fork off my *-Tcl manager interface in the very near future, since it's functional enough to be useful in other projects of mine. That will probably be released under a BSDish license soon. There is no capacity yet for answering machine detection, and frankly, none is planned. The predictive nature is simply counting the agents, counting the calls (busy, no answer, timeout, etc.), timing the calls, and making sure the abandon rate doesn't get too high. Currently, the only known system that I'm aware of has been running 12 hours a day for the past few months with 6 agents. I am rolling out another one next month of the same size for the same client. Connections between the dialer(s) and the main switch is over IAX, but the agents use analog phones so those are Zaptel channels. You could probably use any connection you want, tho. It's pretty much hands off at this point, except when someone gets upset by my music on hold selection. (I personally think that Herb Alpert and the Tijuana Brass rule.) For something similar to shady dial, take a look at this: http://www.predictivedialers.com/products/index.html - VICIDIAL (http://astguiclient.sf.net/) Lead by my company, we are currently developing our 6th production release of a one-call-at-a-time dialer(this is NOT predictive). We have placed over one million calls through this system in the last 12 months mostly to the UK and Australia. It has complete installation instructions, full web-based administration as well as a cross-platform GUI client. It is mostly written in perl, runs on top of an unaltered Asterisk codebase, is dependant on a MySQL as it's backend database and is released under the GPL. If you use this you are limited to SIP clients and Zaptel channels. There are currently 6 companies that are using this system in production envoronments that I know about.(I'd love to hear from other companies using Matt-- Is VICIDIAL a turn key system ? Shady dial is getting closer, but still not there yet. I mean, I can't ship somebody a big brown box with instructions and have that be it. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Shady dial anyone??
On Thu, 8 Jul 2004, Nauman Farooq waxed: wondering if anybody knows this..does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. It is interface ambivalent. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Call queing
On Mon, 5 Jul 2004, Jeremy Kenney waxed: Hello all I have a issue I am wondering if someone can help me Here is my problem, I have several queues setup for different numbers I want each queue to play a custom message to the caller when calling in and then to the called extention when the person answers how is this done and can I specify a customer directory for each one This is possible, just Playback different messages before Queue'ing them. Change the music on hold class if you want to play different stuff while they are waiting. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] recording an on-going call
On Wed, 7 Jul 2004, lenz waxed: Hello list, I wonder if this is possible with Asterisk: - While talking through Asterisk, I would like a client to start recording a call by typing, say, #99# I know it is possible to do it using an external monitoring application, but I want to know if it's possible to have Asterisk silently monitoring an on-going call and responding to DTMF tones within it. How do things like call forwarding work with Asterisk? (I could not find much on the docs) Thank you l. Unless you hack the Dial application, it is not possible to listen for arbitrary DTMF. I think the only DTMF you can do something with in the stream is '#'; provided you supply the 't' or 'T' options to Dial. That would initiate a transfer, and then you could rig up some extensions magic to begin the monitoring. Or you could try taking a look at the manager interface, if by 'typing' you meant computer keyboard and not telephone keypad. That would have the advantage of not playing music on hold to the caller while transfering to a recording mode. There would be no interruption in the call at all. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] iax or sip
On Mon, 5 Jul 2004, Randy Bush waxed: i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. Are we all supposed to guess what your needs are ? iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. I think you could argue that IAX loathes NATs, too. That's why it traverses them. That's a loathing way about it, eh ? trunking will save some bytes in flight iff one has four or more streams moving between two pbxes. but who would want to have the pbxes in the data stream anyway? reinvite rules, especially in a geographically distributed use scenario. You *can* set up IAX to by-pass intermediate PBXes for direct, end-to-end communication. I think the default conf files actually ship that way. now, i could see a network of iaxen if there was some way to negotiate call routing with costs etc. but trip looks a bit ugly and kinda far away. and it certainly is not part of current play. So future expansion with the protocol is not your concern ? what am i missing here? The answer to the question: what kind of VoIP do you want ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] IRQ Misses and Dropped Calls?
On Fri, 2 Jul 2004, Brian D'Arcy waxed: I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. 8's Can you try changing motherboards ? Just a guess, but it seems like you've already made it quite a few steps. Might be time to blame some hardware. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] execute a context from cron
On Thu, 1 Jul 2004, Michael George waxed: I want to have call forwarding (from the POTS) turned on at the close of work and turned off automatically by *. I can create a context that should do just that, but I need a way to have that context spontaneously executed at a specific time. I figured that one way to do it would be to have cron run asterisk -rxsome command if there were some command that would tell asterisk to go to a specific context,extension,priority, but I cannot find that command. Does such a command exist? Or is there a better way to do this that I have overlooked? Try looking at sample.call in the top asterisk source directory. Set up cron to create this file to connect to the specific extention and dump it into: /var/spool/asterisk/outgoing --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] PostgreSQL
On Wed, 26 May 2004, Fabio Donaggio waxed: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! You need to make sure that PostgreSQL is running with the '-i' option for net connections OR that postgresql.conf contains the line 'tcpip_socket = 1' Are you sure that PostgreSQL is running on the same machine that Asterisk is running on ? How did you connect to the database to create the CDR table initially ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] CTI (Computer-Telephony Integration) with Asterisk ?
On Wed, 26 May 2004, Florent Guiliani waxed: Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent, Yes, buy a computer and install Asterisk on it. C = computer T = Asterisk I = install -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] bug or feature?
On Wed, 26 May 2004, Maveric waxed: I've noticed that when i pass a wait in an exten = that it doesn't allow Are you talking about the Wait() application ? 'show application wait' for dtmf tone input. Also on another note i've noticed that when using Background() is what you want if you want to *wait* for DTMF. gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem? GotoIf should execute a lot faster than your fingers can push buttons to send DTMF. Can you post the relevant section(s) of your extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Zapata.conf setup for TE410P
On Sun, 23 May 2004, William Zhang waxed: Hi, I have a TE410P with 3 E1 being enabled, some how it crashes for 2 times lately, I suspect it might be the channel setup issue, can Does it crash immediately or after a fixed amount of time ? anyone tell me if following part in zapata.conf is correct? switchtype = euroisdn signalling = pri_cpe pridialplan=local group = 1 context = incoming channel = 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 These channels look fine for E1. You said 3, but there are 4 configured right there. Shouldn't matter much. What does your zaptel.conf look like ? Can you post that ? Also, is there way to log the reason why Asterisk is crashed? Thank you. asterisk -vvgc That will give you a core file and lots of output, plus connect you to asterisk so you can watch it running on a console. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Fw: setting the number of rings befor asterisk picks up?
Hank-- I was waiting for the 4th, 5th, or 6th email to reply... BUT Have you looked at the Wait(seconds) application ? show application wait from the Asterisk CLI ? Maybe try that before you issue an Answer() on the line ? --Chris On Mon, 24 May 2004, hank waxed: - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 6:19 PM Subject: setting the number of rings befor asterisk picks up? hello how do I set the number of rings picks up on? I am using a single port fxo card and currently asterisk is answering after 1 or 2 rings and I want it answering after 4 5 or 6 rings thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ 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 -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] persistant call variables
On Wed, 19 May 2004, Mike Sturdee waxed: Are there any variables or structure elements unique to a call that stay till the end of a call -- including when caller enters a queue and then bridged with agent. I am trying to get some variables about the caller in I think the account code sticks pretty well. CDR user field, too. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] recommended hardware for quad E1 system
On Mon, 17 May 2004, Robert Almeida waxed: Could anyone tell me which is the recommended hardware to a system running voicemail and conference, attending four E1 trunks and, another, attending only one E1? Can I use a PIII 850Mhz? Maybe for a single port E1 card, maybe. You'll definitely have problems with 4 trunks, voicemail, conference, etc., on a P3. Need at least a P4 for all of that. Maybe dual if you are doing lots of codec translation. If you are really going to put asterisk to work with 120 lines, buy the fastest box your budget will allow, 1 gig ram, SCSI, the works. If you've already got the P3, then by all means, try that out first before spending your money. You have to buy the card either way. Might as well test your configs on that. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] TDMoE hangs the machine
On Sun, 16 May 2004, Bruno Fontana waxed: I was trying to use TDMoE and I lasted with two problems. First of all I can't configure the dynamic span to use CAS signalling but documentation (by Mark) says that you can use any type of signalling (and this includes CAS I guess). Well just pick one that works I guess. You aren't limited to T1/E1 d-channel / b-channel ratios, either. Meaning you can create 200 b-channels and still have the signalling over only 1 d-channel. Not that you'll ever be able to push that much over the wire... My second problem is related that my Linux system crashes frequently due to ztdynamic and friends. I'm using a 2.4.26 version kernel and zaptel drivers 0.9.1 (Gentoo distro). I've looked for a solution or a clue in list archives but there was nothing. Any ideas?, someone who had a similar problem and have found a solution? I've found it unstable on high loads, in excess of 50 channels between smoking P4s, with or without any zaptel card drivers loaded. I would like to try blaming it on the switch, and possibly the NICs both facing the public network, but it was still 0 hops between boxes. Might have better luck on separate, better ethernet cards over a private network. TDMoE is very noisy: http://www.marko.net/asterisk/archives/0301/0566.html There's an ancient TDMoE howto here: http://www.convergence.com.pk/TDMoE-HOWTO It's old, but so is the code for ztdynamic, so I assume it just works for a lot of people. That or people have dropped it in favor of something like trunking on IAX2. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] dialing a remote phone system and then entering an extension
On Mon, 3 May 2004, Joel Duffield waxed: I am trying to get a way to have * forward calls that are dialed to an extension, to end up at an extension on my old analog phone system. I will have 7 lines coming into * using the new Digium cards via PSTN, and then lines coming from * into the PSTN lines on the analog system. So that if for example someone dials extension 110: The system will call the analog system, the system will assume that a call is coming from the telco as always, pick up right away, and then listen for an extension to be entered. This should then connect the incoming call to the extension on the analog system. My question is, does my logic work, and also if I use the dial command, and I set the analog system to pick up immediately, will wait long enough before it dials? If that wouldn't work is there a way that I can tell * to dial then wait and then send digits? So, the legacy PBX already provides an analogue to *'s Background app, ie., you dial in and it sits to wait for the entered extension ? Then you might as well just bridge it right through * exten = s,1,Dial(Zap/g1/legacy_background_extension) Since the Dial app isn't eating the DTMF, it should just pass thru to the legacy PBX. If, however, you want to use * to do the Background, then dial that extension on the legacy PBX: exten = 110,1,Dial(Zap/g1/110) Probably a better option because it gets you migrating to * quicker. And I assume you are upgrading. I don't think you need to worry about the wait, * handles analog interfaces and this is a requirement of such an interface. Meaning it won't send audio while on hook, but wait for an answer. There's even support for pulse dialing in *, if it is that much of a legacy PBX. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Start recording during call by pressing button sequence
On Fri, 30 Apr 2004, Dean Collins waxed: Ian, I'd love to see an example of this. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Iain Stevenson Sent: Friday, 30 April 2004 1:47 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Start recording during call by pressing button sequence --On Thursday, April 29, 2004 3:21 pm +0300 Vladyslav [EMAIL PROTECTED] wrote: On Thu, 2004-04-29 at 15:06, Andrew Kohlsmith wrote: Thank U for your reply, however I was asking about recording during call (for example I don't need record all calls, but only some of them and I want start recording during actual call process). You can activate call recording with a php script from a web page too. You can turn recording on and off without the called party knowing and at any time in the call. Iain Those in the US might want to check on what sort of laws affect recording of telephone conversations: http://archive.aclu.org/issues/cyber/phonelaw.html I recall it being mentioned on this list that people wished to spy on their kids with *, and that's specifically forbidden in most states, as it would be zero-party consent. You can log their IM all you want, tho. Then wonder why they hate you. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Playing with time ranges...
On Fri, 30 Apr 2004, Mark Elkins waxed: Playing with time ranges - using the examples found in one of the asterisk cook books... (pdf - page 17) ; After Hours include = night_menu|00:00-08:00|Tue-Fri|*|* include = night_menu|17:00-24:00|Mon-Thu|*|* this gives... ... pbx.c:2962 get_timerange: 24:00 isn't a valid end time -- Including context 'night_menu|17:00-24:00|Mon-Thu|*|*' in context 'default' Looking at pbx.c - I'm not sure if I should change the end time (ie midnight) to either 23:59 -or- 00:00. 23:59 will work - but what happens to calls then between 23:59 and midnight? (Be nice if the people who wrote cookbook examples tested them too) Well, considering you only have 60 seconds to test this every night, good luck ! But seriously, it sounds like a bug. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Very basic questions
On Wed, 21 Apr 2004, Laurent BURGY waxed: Hi, I am new in asterisk and i've bought a X100p and a TDM400... First, you are probably eligible for support from digium directly if you bought the hardware from them. First of all, how can i verify my config files ? You could try attaching them to the email so people can take a look. zapata.conf, zaptel.conf, extensions.conf, indications.conf are probably the minimum that you need to alter. Secondly, when i'm trying to pass a call to the outside, i ve a Notice about appdial.c (l 554) telling me: unable to create channel of type Zap ...and i don't understand... That's a problem, but possibly for lots of reasons. No zaptel kernel module, no asterisk driver, bad extensions.conf format, etc. Finally, when i plug my analog phones in RJ45 of my TDM400, there is no tonality ( i'm not sure that it is the right word in english , but i can't hear any tut-tut or any noise...) ... Did you compile the kernel modules in the zaptel source directory for that card ? What does lsmod show you ? Maybe, it's obvious but i can't succeed... No, it's still quite difficult, especially if english is not your first language. I am completely unaware of any non-english documentation. Good luck, --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] A few questions
On Wed, 21 Apr 2004, Ben Merrills waxed: Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). That's great news. MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems like a common feature in most hardware PBX systems we've used that support Call Conferences. Has anyone found a way of doing this? Is there an alternative to MeetMe that would support this feature (that's as good if not better?). I don't think this is currently supported, could be wrong, tho. Would take some modification to app_meetme.c, or else just have people say there name when it beeps them in. :) Sort of the flip side, but maybe it would be more helpful to have the person entering the conference hear the name of everyone already in it. That could be done via Record and Playback apps, before executing MeetMe. Every time someone enters, have Record take their name. Then, run Playback for each of the Recorded files. Queues: I'm running the 1.0 stable from the cvs server, and I've added the queue status announcement directives to the queues.conf - yet asterisk gives me the following errors: Apr 21 11:22:58 WARNING[950286]: Unknown keyword in queue 'Sales': monitor-format at line 9 of queue.conf I think this only works in development, not stable, CVS. :( --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] PBX - AST - AST - PBX
On Wed, 14 Apr 2004, Ezequiel Golub waxed: Im trying to come up with a cost effective way to unite two PBX using VOIP. My idea is that since most companys here (Argentina) are not ready cough up the money to go to full-fledged VOIP, they might be willing to pay for a hybrid-solution: a kind of point-to-point line using VOIP, which let's them dial an extension on the other PBX. This is *exactly* what Asterisk is for -- and a billion other things. Could someone please tell me that im not crazy or trying to reinvent the wheel ? You are not crazy, nor are you really re-inventing the wheel. You would be crazy to not read a lot more on the wiki and elsewhere before spending your money. Your diagram could use a little work, since Asterisk==PBX. Try this: analog phones - */PBX - INTERNET - */PBX - analog phones That's being done right now by lots of people. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] call queue list members using sql query
On Mon, 12 Apr 2004, Dragan Mickovic waxed: Is it possible for asterisk to do an sql query in order to get the member list of a call queue? No, you will have to write code besides SQL in order to do it. To go the C route, try modifiying app_queue. To use a different language, you could code something over the manager interface that interacts with AddQueueMember and RemoveQueueMember in extensions.conf. That would even give you some more dynamic control of the members. There's lots more ways to do it, tho. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] controlling call duration
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed: In other words can I receive information which we are usually getting in CDRs during the time when the call is still active? Yes, via the manager interface. Check manager.conf, it lets * talk on port 5038. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] CallerID
On Mon, 5 Apr 2004, AstGrp waxed: I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. callerid=asreceived That's a good line to have. context=default-nga signalling=featd group=2 channel = 5-8 context=default-tne signalling=featd group=1 channel = 1-4 context=default-pb signalling=featd group=3 channel = 9-12 context=default=ctm signalling=featd group=3 channel = 13-14 What context is the company in that gets the cid right ? Maybe you are only receiving the cid on certain channels ? Why do you have 3 groups, but 4 contexts ? Is everything hooked up to a channel bank ? What kind of hardware is installed on the box ? Are you explicitly setting the cid in extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Change IP info.
On Mon, 5 Apr 2004, William C. Ray waxed: Hello i was wondering how i can change the IP address information for my Asterisk box, IP addy, Gateway, DNS. I have a smoothwall router that i am using and i am tring to put the Asterisk box on the orange interface so if anyone can help me please i can use it. I was wondering what a smoothwall was, so I checked it out. smoothwall.org first, then when I saw the full color 86 page administrator guide, I thought, wow, this can't be all of it, so I found smoothwall.net, where they will sell you all manner of 'smoothie' flavors AKA linux. Neat. I saw one of these at a tradeshow the other day for $2500, so I took a second look, at least that's my excuse to the list for taking a look and helping out. :P I'm assuming this is the commercial version in question. And from question 1 on the FAQ at smoothwall.net, it says that the orange interface is the DMZ and green is the local protected interface. There are also red IP's, not sure what they do. But this appears to be a traffic light running some form of embedded Linux. Also in the FAQ, Mel Gibson gets props for his appearance as Mad Max, in Mad Max 2: The 'Road Warrior', which is what you use 'on the road' to get at your 'smoothie' -- one must hope it will still work after the apocalypse. Myself, I would rather use industrial hemp for fuel than fight over dinosaur bones, but that would make a pretty boring action movie. Maybe fodder for a Mel Brooks comedy ? Anyhow, back to the question at hand, if you can change the IP address for a windows box via the smoothie windows GUI, chances are you can do the same for the Asterisk box. That's if you are running Asterisk on Linux, which I assume you didn't get it to compile on FreeBSD, so it's probably Linux. You might need to know the MAC address of the network card on the Asterisk box to change it's IP addy. Gateway and DNS should be the same as what you set on the windows boxes. Apart from not knowing or being able to understand how exactly you got Asterisk running (configs, etc.), if you can do the IP stuff with smoothie for windows, you can probably do it with smoothie for linux. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Quick Caller ID and Voicemail ?s
On Tue, 6 Apr 2004, Ryan Thrash waxed: The number shows up, but I can't get the words to show on a local bell line. The text always comes up as unavailable. In sip.conf for each extension, I've tried: callerid=VERTEX 2142618000 callerid=VERTEX 2142618000 Neither one works. Suggestions? Your telco is probably setting the callerid name to whatever their own database maps that number to. Also, I'm not sure if that numbering scheme will work, I would try adding ()'s. Something like this in zapata.conf: callerid=VERTEX (NNN) NNN- Can't say about anything about sip, tho. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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] Help setting up asterisk
On Wed, 18 Feb 2004, Jason Miller waxed: Hello all I am new to this and was wondering if anyone could assist me or point me in the right direction as to how to setup asterisk for a T-1 to be able to call a PSTN number? I see plenty of walk throughs if you have analog cards installed but none with just the T-1 card or am I missing something and dont know it or just uninformed on what this takes? All the information I have read says all I need is a Digium T100P card terminating my T1 and a ethernet card for my local network for my VOIP phones, is this correct? The T1 is your link to the PSTN. And yes, you can use a T100P card for the T1, and ethernet to your hub out to VoIP phones. If you are completely new to telephony, *, etc., then you might want to check the Asterisk consultants page on the wiki and pay someone to set it up right. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] PRI error or what?
On Tue, 17 Feb 2004, Tomica Crnek waxed: I have TE410P with two E1 links connected. It is working ok, but suddenly, from time to time I got this and it goes on and on for a few minutes during which period I can't establish new calls == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up == D-Channel on span 1 up == D-Channel on span 2 up This could be a problem with your telco. I had the same thing for a couple of months before they finally identified it as their bad cable pair in outside wiring. Run a trace on the PRI from *, and find someone at the telco to do the same on their end. Good luck, --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Asterisk for a call center?
On Mon, 16 Feb 2004, Jim Archer waxed: First, can Asterisk be configured accept calls on a bunch of incoming lines, answering with a greeting and telling the person that they will be transferred to the next available operator. Then, can it watch all the extensions, and route the calls to these extensions on a first in, first out basis? Can operators somehow tell Asterisk they are ready for another call or are on break? Yes to the above, using Agents and Queues. See apps/app_queue.c and channels/chan_agent.c --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] More external call control
On Thu, 12 Feb 2004, [EMAIL PROTECTED] waxed: My questions are as follows, (but before I begin; I know there is queueing and some ACD functionality in *, but I need to do this externally. I want the queueing decisions to be external because my central queue engine handles things like email, chat, etc as well as calls): You might then want to consider just putting people in MusicOnHold extensions, Parking extensions, etc. Putting them in a * queue -- when they are already in your own external queue -- would be a flawed redundancy, owing to differences in queue logic. In other words, can I send some message to * that will tell it to route a call in queue to a specific extension by a unique ID (because there may be los of calls queued). This is possible. But not really through AGI. You would need to use the manager interface, which is more for external control. Not to say that you couldn't create manager commands that would in turn put you in the dialplan to run a specific AGI... While the call is in queue, can I send commands to have different announcements played? There's no moh stuff for the manager interface, but you could look at the 'Redirect' manager command. Maybe consider moving someone from one queue to another, where there happens to be different moh. If a call hangs up while in queue, is that a step in extensions.conf so I can call my script with that info? Extension 'h' is reserved for hangups, but it isn't limited to just queues. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Automated Dialing / Recording ?
On Mon, 2 Feb 2004, Rick Smith waxed: We have 1000's of Remote Call Forward #'s across the USA / Canada, which forward into 1000's of 800 #'s in our call center. Is it possible to automate a solution where Asterisk could dial a given number, record the first 3 seconds of the call, save it to disk, and then go on to the next number, and just do this all day long ? We need to regularly check that the numbers work, for billing and payment purposes as well as operational purposes, and I thought this would be the perfect situation ! Sounds like a great project for *. Only thing I would question is listening to those first 3 seconds. If your intent is to listen for some sort of voice/data in that 3 seconds, that's a lot of sitting around and just listening to 3 seconds. Probably would want a solution that would do some sort of automatic sampling of the recordings, running another app over them to detect noise. Besides that, you just need to create a bunch of 'sample.call' files, and place them in /var/spool/asterisk/outgoing/ to dial the numbers. Make sure they connect to an extension tho, and not directly to an application. Heck, you could just check CDR logs to see if the calls went through, for a first trial run of the system. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] TE410P E1 PRI problem
On Sat, 31 Jan 2004, Tomica Crnek waxed: Hi everyone! Here is my configuration and messages taken from Asterisk startup. The E1 PRI trunk is connected to our national telecom company here in Croatia. When I call from outside over this trunk to my company I get 'error in connection' respnse. In the same moment I can't see anything in Asterisk, nothing that will tell me that the call reached Asterisk. I think there is a problem with PRI synchronization or PRI to zap communication. The card I am using is TE410P, the first port is the one that I use. /etc/zaptel.conf - # port 1: trunk to telecom span=1,0,0,ccs,hdb3 Try this instead: span=1,1,0,ccs,hdb3 (Note the second 1, that sets your timing to the telecom.) --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Migrating home POTS VM to Asterisk VM
On Thu, 29 Jan 2004, Chris Hirsch waxed: Currently a caller can press *3 to leave me a message in my own mailbox on the FXS machine. Is there any way that I can make Asterisk monitor the line after the machine has picked up and if it detects a *3 dump the user into my Asterisk voice mail box but for everything else just let the user leave a message on the normal machine? Maybe try it the other way round, for a two-tiered menu approach. Try to Dial first, and if that times out after n seconds, play a message press 1 to leave chris a message, 2 for everybody else. Then connect yours to * VM, and everybody else's connects to the external answering machine. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Zapateller
On Mon, 26 Jan 2004, Steve Foy waxed: I'm just wondering about 'Zapateller'. How exactly does it work!? I might be interested in employing it at work here, but wondering if anyone's using it? I think you can just put it in your dial plan: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Zapateller exten = s,4,Dial(whatever) It fakes a disconnected number by playing the same tones you would get when dialing, for example, 555-1234. If a computer was calling you, sometimes they are programmed to listen for those tones. No sense keeping the connection with a disconnected number. I mean, if the National Weather Service is trying to call everyone in the path of a hurricane as fast as possible with information about nearby storm shelters and emergency plans, they want their computer to connect to as many valid phone numbers as it can and play their warning message. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?
On Wed, 21 Jan 2004, Bill Hamel waxed: 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 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 cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] need help configuring IAX to make outbound calls through a remote server
On Wed, 21 Jan 2004, Paul Mahler waxed: I am trying to make outbound calls from my Asterisk client through a remote Asterisk server with IAX. In iax.conf on both sides [dar] context=trusted secret=xx type=friend host=192.168.1.1 I'm not going to try and fix all of this, but if you've got the same hostname on both hosts, one of them doesn't know about the other. You need to set the host differently on each of the hosts. Ie, on 192.168.1.1, you need to set host=192.168.1.2 and on 192.168.1.2, you need to set host=192.168.1.1. in extensions.conf at the client making the call Exten=_1NXXNXX,1,Dial(IAX2/dar:[EMAIL PROTECTED]/) What goes in extensions.conf at the remote server? What is needed for the remote server to accept the call from my client, figure out the dialed number and then dial it outbound on some line? You'll need to have a trusted context in each, for starters. But there's a lot more dialplan work you'll need to do, depending on where your outbound lines are, what numbers they can dial without incurring toll charges, etc. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Is there a way to # of agents logged into a queue ?
Nope, I'm an idiot. Here's the patch :P
On Thu, 22 Jan 2004, Bill Hamel waxed:
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 cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide: Maj == May
Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146
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Asterisk-Users mailing list
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Chris Maj cmaj_hat_freedomcorpse_hot_info
Pronunciation Guide: Maj == May
Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146
--- ../../../../asterisk/asterisk/apps/app_queue.c 2004-01-06 16:45:20.0
-0500
+++ app_queue.c 2004-01-22 11:05:49.0 -0500
@@ -1403,6 +1403,57 @@
return RESULT_SUCCESS;
}
+static int agent_queues_show(int fd, int argc, char **argv)
+{
+ struct ast_call_queue *q;
+ struct queue_ent *qe;
+ struct member *mem;
+ time_t now;
+ char max[80];
+int agents=0;
+int callers=0;
+
+ time(now);
+ if (argc != 2)
+ return RESULT_SHOWUSAGE;
+ ast_mutex_lock(qlock);
+ q = queues;
+ if (!q) {
+ ast_mutex_unlock(qlock);
+ ast_cli(fd, No queues.\n);
+ return RESULT_SUCCESS;
+ }
+ while(q) {
+ ast_mutex_lock(q-lock);
+ if (q-maxlen)
+ snprintf(max, sizeof(max), %d, q-maxlen);
+ else
+ strcpy(max, unlimited);
+ ast_cli(fd, Queue: %s\r\n, q-name);
+ if (q-members) {
+mem = q-members;
+while (mem) {
+mem = mem-next;
+agents++;
+}
+}
+ ast_cli(fd, Agents: %d\r\n, agents);
+ if (q-head) {
+qe = q-head;
+while (qe) {
+qe = q-next;
+callers++;
+}
+}
+ ast_cli(fd, Callers: %d\r\n, callers);
+ ast_mutex_unlock(q-lock);
+ q = q-next;
+ }
+ ast_mutex_unlock(qlock);
+ ast_cli(fd, \n);
+ return RESULT_SUCCESS;
+}
+
/* JDG: callback to display queues status in manager */
static int manager_queues_show( struct mansession *s, struct message *m )
{
@@ -1410,6 +1461,11 @@
return queues_show( s-fd, 2, a );
} /* /JDG */
+static int manager_agent_queues_show( struct mansession *s, struct message *m )
+{
+ char *a[] = { show, agentqueues };
+ return agent_queues_show( s-fd, 2, a );
+}
/* Dump queue status */
static int manager_queues_status( struct mansession *s, struct message *m )
@@ -1473,6 +1529,7 @@
ast_cli_unregister(cli_show_queues);
ast_manager_unregister( Queues );
ast_manager_unregister( QueueStatus );
+ ast_manager_unregister( AgentQueues );
return ast_unregister_application(app);
}
@@ -1484,6 +1541,7 @@
ast_cli_register(cli_show_queues);
ast_manager_register( Queues, 0, manager_queues_show, Queues );
ast_manager_register( QueueStatus, 0, manager_queues_status, Queue
Status );
+ ast_manager_register( AgentQueues, 0, manager_agent_queues_show,
AgentQueues );
// [PHM 06/26/03]
ast_register_application(app_aqm, aqm_exec, app_aqm_synopsis,
app_aqm_descrip) ;
Re: [Asterisk-Users] VOIP on linux
On Tue, 20 Jan 2004, Ralph Blach waxed: I would like to start using VOIP on Linux but I would like to start out just using my sound card in my Linux box. Is there anyway to do this. Yes, try using Asterisk: http://www.asterisk.org/ Also, what service provider would I use to get my Voice over IP service Try searching on the wiki: http://www.voip-info.org/tiki-searchresults.php?words=VoIP+providerwhere=pagessearch=go --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Can asteric be used with just a voice card
On Tue, 20 Jan 2004, Ralph Blach waxed: Can asteric be used with just a voice card. If so, how would I get this going? Also, what carrier would I use connect to? You'll want to do it on your Linux box. Ie would would be my carrier. Maybe these links would be your carrier: http://www.voip-info.org/tiki-searchresults.php?words=VoIP+providerwhere=pagessearch=go --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Zap show channel
On Wed, 21 Jan 2004, dkwok waxed: What are the meaning of these Zap show channel output? Caller ID string: Owner: None y Real: None ? Callwait: None y Threeway: None y Confno: -1 n Propagated Conference: -1 n Real in conference: 0 n DSP: no n Relax DTMF: no ? Dialing/CallwaitCAS: 0/0 ? Default law: ulaw n Fax handled: no n Pulse phone: no y Echo Cancellation: 0 taps unless TDM bridged, currently OFF y Actual Confinfo: Num/0, Mode/0x n Actual Confmute: No n Are these settings configurable in /etc/asterisk/zapata.conf? See marked above, but I might be wrong on a couple of them, and some I didn't know. Just off the top of my head. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Parked Calls Settings
On Wed, 21 Jan 2004, [EMAIL PROTECTED] waxed: Is there settings that can be adjusted in the Parked calls timeout before it hangs up i want to try and hold the call for atlest 5-10 mins .. but holds the call for about less then 1minute? any suggestions or ideas? Do you have an AbsoluteTimeout set somewhere in your extensions.conf ? --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Making a call with sample.call
On Wed, 21 Jan 2004, Steve Foy waxed: Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/TEL NUMBER HERE MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into /var/spool/asterisk/outgoing, it does call the number, but when I pick up the phone, I just get a dial tone as if Asterisk has maybe hung up before playing the sound. Can anyone see what I'm doing wrong here? The extension does work correctly when called normally. Regards, Steve Try putting a Wait(2) in at extension 502, priority 1; and move the Playback(sndfile) down to priority 2. Maybe it's a real short sound ? Take a look at what's happening on the * console with sudo asterisk -r --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] ADSI phone vs. IP phone
On Mon, 19 Jan 2004, Ted Cabeen waxed: Andrew Kohlsmith [EMAIL PROTECTED] writes: Why wouldn't you just use your existing Ethernet infrastructure putting the IP phones inline between the wall jack and the PC? There are a number of IP phones that have builtin switch/hub that allows the PC to daisy chain off the IP phone. To quote myself: True, but I don't have to retool my office and install POE switches to use ADSI phones, either. No, I will not put a hub/switch at every desk and then use wall-warts for every phone to get around retooling the office. :-) I'm not going to bastardize my network by placing the equivalent of a 3-port switch or hub at every desk to have the phone system compete with our heavy network users (CAD mostly), and I will fight tooth and nail against having to put a goddamned wall-wart at every station just to power the damned IP phones. :-) Do ADSI phones need wall-warts, or can they drive themselves from the line power? You can get dial tone on ADSI w/o a wall-wart, just like a regular analog phone. But you need a wall-wart to give you power for the screen and ADSI functionality, at least on the Nortel Vista 350. Since there's no Ethernet, I don't think it would be practical to do POE. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] ADSI phone vs. IP phone
On Sun, 18 Jan 2004, Ulexus waxed: On Sunday, 18 January, 2004 02:04, Ken Alker wrote: Assuming the price of an ADSI screen phone (say, Aastra 390) was the same as an IP screen phone (say, Cisco 7960) and someone was setting up an * server for their 20 employees (each of whom would have either an ADSI or IP phone on their desk), would there be advantages to using the ADSI phones over the IP phones, or vice-versa? For discussion, let's assume that the hardware needed to patch the ADSI phones back into * was not a cost concern. I'm looking for differences between the technologies independent of cost. Pretty much no. The ADSI specification was crippled from the start to specificly not compete with PBX offerings. It has one advantage of (very limited) programmability, but a phone like the SNOM has an open-source core. It also has the dubious value of being interchangeable with a regular analog phone, but that is about it. You will not get anything near the functionality and feature set of a SIP phone, and it has the further irritation that much of its signalling is both in-band and audible. I completely second that. My original thought was to use ADSI for enhanced caller id information -- like an address or whatever from my own database. But the data/voice context switch takes 5-7 seconds, and it's loud and annoying. Best use would be a one-way setup, where you are simply dialing into your voicemail, checking your bank account, etc. I have yet to get my hands on an IP phone, so I can't compare. It is too bad. If it were properly implemented, the concept behind ADSI is great. Unfortunately, Telcordia strikes again. Yeah, it seems like ADSI was way ahead of its time, but they played it like a monopoly before it was big enough to do so. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] * For Call Center
On Thu, 15 Jan 2004, Steve waxed: On Thursday 15 January 2004 12:03 am, [EMAIL PROTECTED] wrote: sounds like one of those pesky auto dialers the simpsons make fun of. It sure does... The AT-5000 was Prof. Frink's first patent, and it was designed to alert children of snow days and such. I think Homer bought it at one of those pesky police auctions, you know, the ones where the liberty and freedom loving US government says your property is guilty of a crime and theirs to sell... But don't forget that Prof. Frink went on to invent such wonders as the Flying Motorcycle, a Matter Transporter, and the Frinkahedron: http://www.internerd.com/frink.retired/frinkv.3/inventions/ --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
RE: [Asterisk-Users] * For Call Center
On Thu, 15 Jan 2004, mattf waxed: 8's There is a group of Asterisk users that decided to modify the code of Asterisk to try to make it a predictive dialer, called shady_dial I believe, but I haven't heard anything about it lately. http://shadydial.sourceforge.net/ Lots of recent updates made in CVS, and it works with the latest and greatest * CVS, too. No screen pops yet, but that is the next step. Call results are simply logged in the phone, which is pretty sloppy since it resides in the agent hangup function. Francois Lambert posted some time ago on -dev that his company had worked on a predictive dialer with answering machine detection. Said they hacked * code a little, too, and since it's GPL I would be interested in seeing it. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Basic Asterisk capabilities question
On Wed, 14 Jan 2004, Gary Franczyk waxed: I'd like to configure a voice recording system using Asterisk and a Tormenta2 Quad T1 card. A co-worker was able to create this system a while back with Bayonne and a Dialogic card, but I would like to do the same thing with much cheaper hardware. I do not believe Bayonne supports the Tormenta2. (I think it is also known as the Wildcard TE410P) Yes, but the TE410P is not required. Only if you want T1 connections. You could just use a regular linux box and VoIP, for example. When the user dials in: - He enters his user id - he enters his password - he begins his recording. - while recording, he has the option to pause, continue, rewind by 4 seconds and review his recording, - press a button to start another recording. - press a button to recieve a recording ID number In addition to this, I would like the user to be able to call in again later and listen to his previous recordings by entering his recording id numbers. So, Is this possible and resonable to accomplish with the Asterisk system? This system is possible with Asterisk, using the Asterisk Gateway Interface (AGI). You could code it in your favorite language, too. Or is Asterisk more of an out-of-the-box PBX rather than a voice application system? It's not a turnkey system as downloaded from CVS. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
RE: [Asterisk-Users] linux journal article on asterisk
On Wed, 14 Jan 2004, calvis waxed: Thanks for the link. This is an interesting article on Asterisk. I was hoping to send him some kudos, but his website isn't working at http://www.bschwarz.com/. And I just noticed the guy lives near me! Does anyone know if he hangs out on the list? Yeah, he's posted, from time to time. Awesome Tcl coder, too ! Hi Brett :) --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Asterisk on FreeBSD 4.9?
On Wed, 14 Jan 2004, [EMAIL PROTECTED] waxed: On Wed, Jan 14, 2004 at 07:34:20AM -0500, Troy Settle wrote: I'm not completely conversant on how GPL software can be committed to the kernel, but I believe it can be done under the contrib/ directory. I think the *BSD kernel device drivers should be BSD licensed. BSD and Linux device drivers are different enough that this can probably be done without re-using code, so shouldn't be a big problem. There is also precedent with EXT2FS support for GPL drivers being included (in /usr/src/sys/gnu on FreeBSD), but it will be much easier if the drivers are BSD licensed. If the drivers are that different, perhaps the old Zapata Telephony BSD drivers might help as a starting point ? They look very well-documented in the source, but some of it is in Mexican: http://www.zapatatelephony.org/ I think the GPL'd Asterisk Zaptel stuff has its roots in that code. I confess to knowing nothing about BSD drivers, tho. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] inbound call routing problem
On Tue, 13 Jan 2004, Lane Hoskins waxed: We have 8 lines coming into an ADTRAN channelbank that then goes to the * server via a T100P card. I need to route lines 1 and 2 to everyone when a call comes in on either of them. I also need lines 3 - 8 to ring first at specific sip extensions (direct dials for staff here) and then to go to voicemail or fwd to a cellphone after that if the extension is not answered. Has anyone done this that could provide an example for me or point me to better documentation? We have searched extensively and not found anything yet. Here's Rich Adamson's A WORKING EXAMPLE from September: http://lists.digium.com/pipermail/asterisk-users/2003-September/020944.html I see SIP and Voicemail in there, but I haven't tried it myself. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] max queue time; newbie question (fwd)
On Tue, 13 Jan 2004, Ted Cabeen waxed: Martin Pycko [EMAIL PROTECTED] writes: sure, use the 'n' option of the queue and put voicemail app as the next priority Will that work? From my read of the code, the timeout parameter is only checked while the call is being sent to an agent's phone (inside the try_calling function). The timeout doesn't seem to be checked while the user is waiting to get to the head of the queue (inside the wait_our_turn function). Unless the ast_waitfordigit function checks the timeout and I missed it, this solution won't work. Am I reading the code right? Set up a queue, with the n option, and dial into it, without having any agents logged in. It should drop out after the timeout expires. Least that's what my slightly hacked version does. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] T1 Sync clarification
On Sun, 11 Jan 2004, John Brown (CV) waxed: THank you. Thats what I thought it should be. Off to call the telco and tell them they are mucked up. I'm wondering if I should do the same for my T400, as I seem to be getting similar errors. Might not be just the telco. I set one span to 1, to sync off the telco, and the second to 0, since the clock is already set by the first span. The other 2 spans are channel banks with sync set to 0. Does anyone else have 2 t1's plugged into their T400 ? If so, how are they synced ? This was just happening at night, but I lost the second span a dozen times already today, all within less than an hour earlier this afternoon. Thanks, --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] ADSI. used beyond own phone network?
On Mon, 12 Jan 2004, Andrew Thompson waxed: I am curious. I understand that features can be pushed to an ADSI phone that make navigating your own voicemail easier, and for other internal things. But, does anyone push this data outside of their own phone network? Example: I am at home with my spiffy new ADSI compatible handset and dial up my bank, OneWorldBankingConglomerate. Would they be able to push me their menu? Button 1 for TeleBanking. Button 2 for New Accounts. Button 3 for Customer Service. Button 4 Check Verification, etc... Is there a protocol that supports this? Yeah, ADSI :) Could it be passed in sideband somehow? ADSI works on standard POTS lines. You can hear it pop and whistle. Would normal people actually buy phones to have access to this? I think this is probably one of the original marketing goals with ADSI. Unfortunately, it didn't really take off. Several numbers to call and download were posted about a year ago to the list. I think I got a few to work. Might check the archives. What kind of security implications would this have? Probably the same as using DTMF when you call the bank to check on your credit cards. As long as everybody can be trusted at the telco, you're probably fine. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] New Installation problem
On Mon, 12 Jan 2004, marin blu waxed: I'm trying to install * on Mandrake 9.2/P4, but under asterisk - make clean;make install there is the following error: How about: make then: make install -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] ADSI Configs
On Sat, 10 Jan 2004, Lee Redmayne waxed: Hi All If I want to get my ADSI Phones (successfully connected off a Rhino Channel Bank and TE410P) to connect to Asterisk to get their config downloaded, is there something specific needed in extensions.conf for them to dial to get this? Thanks :) You'll need to set them up with adsi=yes in zapata.conf, then try making an extension for VoiceMailMain and dial into it from your ADSI phone. I think that's a good start. But if your phone is locked, you might run into snags. Check the list archives for locked ADSI if that's the case. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] drop calls with T100P / PRI
On Sat, 10 Jan 2004, Steven Critchfield waxed: On Sat, 2004-01-10 at 15:19, John Brown (CV) wrote: busydetect=yes callprogress=yes musiconhold=default signalling=pri_cpe group=1 channel= 1-4 Well seems you haven't been on the list, or maybe you haven't been paying attention since we have been covering that problem for a while lately. PRI has busydetect and callprogress built into the D channel and is absolutely known. Those 2 options are for analog links where the signaling is not always accurate. Easy, now. I just added another T1, and I have noticed the D-Channel dropping for a few seconds then coming right back, fortunately in the wee hours when no one's on the lines. But I have both busydetect and callprogress off, with a T400 tho not a T100. Here's some zaptel.conf: span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs span=3,1,0,esf,b8zs span=4,0,0,esf,b8zs First 2 spans are channel banks, span 3 is local T1, and span 4 is long distance T1. The D-Channel only appears to drop on span 4, although I can't yet get the time to play wire/card-swap much due to the machine being in production. Both T1's are from the same switch, so I'm told, so it should be the same clock. Here's some log: Jan 9 04:22:05 WARNING[5126]: File chan_zap.c, Line 5759 (zt_pri_error): PRI: Read on 106 failed: Unknown error 500 Jan 9 04:22:05 WARNING[6151]: File chan_zap.c, Line 4708 (handle_init_event): Detected alarm on channel 73: Red Alarm Jan 9 04:22:05 WARNING[6151]: File chan_zap.c, Line 1101 (zt_disable_ec): Unable to disable echo cancellation on channel 73 Jan 9 04:22:05 WARNING[6151]: File chan_zap.c, Line 4708 (handle_init_event): Detected alarm on channel 74: Red Alarm Jan 9 04:22:05 WARNING[6151]: File chan_zap.c, Line 1101 (zt_disable_ec): Unable to disable echo cancellation on channel 74 ...last 2 lines repeated for each channel on span 4 (up to channel 95) all at the same time Jan 9 04:22:11 NOTICE[6151]: File chan_zap.c, Line 4703 (handle_init_event): Alarm cleared on channel 73 Jan 9 04:22:11 NOTICE[6151]: File chan_zap.c, Line 4703 (handle_init_event): Alarm cleared on channel 74 ...again repeated at the same time for every channel on span 4 Jan 9 04:22:11 WARNING[5126]: File chan_zap.c, Line 5759 (zt_pri_error): PRI: Read on 106 failed: Unknown error 500 Jan 9 04:22:14 VERBOSE[5126]: == D-Channel on span 4 down Jan 9 04:22:20 VERBOSE[5126]: == D-Channel on span 4 up ...and then the B-Channels start coming back up I'm trying to blame it on the mobo and the telco, but now that someone else is seeing it, maybe it's * ? It wasn't a problem when I had one T1, and span 3 doesn't seem to ever drop, so maybe it's something with the span 4 port on the T400 card ? Or is this some side-effect of the known buffering problem ? --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info 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
Re: [Asterisk-Users] Re: Asterisk , Video Switching
On Tue, 23 Dec 2003, Hubert Kiyimba waxed: Dear members, I am writing to inquire whether Asterisk can serve as video switching software for the purposes of video conferencing over IP on a campus network. Hubert http://www.gnophone.com/ -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key 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
Re: [Asterisk-Users] Seting callerID on outgoing calls
On Wed, 17 Dec 2003, Anton Yurchenko waxed: Hello, I have a E1 with PRI signaling, is there a way to set that some of the channels when the call is made from them to outside PSTN, will show one callerID, and other group of channels will show other callerID. Right now I have it like this group=1 callerid=777 channel=1-10 group=2 callerid=888 channel=11-20 When I call to outside the caller id shows only the same one You might not be able to set the callerid to anything you like. That would be a limitation of your E1 provider. Probably they only let you set the callerid to numbers in your DID block. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key 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
RE: [Asterisk-Users] ALL incoming Zap channel calls are getting picked up as FAX calls!
On Wed, 17 Dec 2003, Patrick Cantwell waxed: FYI: Doing an rm -rf zaptel asterisk in /usr/src, then doing a checkout from 12/08/2003, I no longer have this problem.. so it's something with the newer code? My nose is bleeding from CVS. Same thing with a T400, had to comment out all fax extensions. Updated to CVS of 12/16. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key 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
Re: [Asterisk-Users] Queue only ringing one agent at a time
On Fri, 12 Dec 2003, Derek Barber waxed: but, if this is case then how can you run a call center with asterisk? What if you have 40 simultaneous calls coming into the call center, most calls would be missed, even if you have 40 available agents. Of course one call should go to one agent, but if a second call, or a third call joins the queue, shouldn't they also go to agents as well? Don't use the callback option. If the call center is that large, and that busy, those agents need the phone to their ear all the time. Let agents log in and wear headsets, so they can hear the 'beep' and immediately be connected to a customer in the queue. Beats 40 phones ringing in a room. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key 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
Re: FW: [Asterisk-Users] recording calls
On Thu, 6 Nov 2003, mattf waxed: Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: 8's I record all outgoing calls using Kostya V. Ivanov's 'R' patch to the Dial application, which was posted to the list about a year ago. Then I wrote a Tcl script which uses soxmix to merge the conversations and burn them to CD every night. Even with 5000 calls per day, using gsm the CD isn't even half full. And there's plenty of room to store the past month or two on the hard drive. So I slapped a 'Play' button on the customer information form in another application. The only down side is that employees' personal calls are recorded, too. But they know it's happening, and anyhow, they should be waiting until a scheduled break when they can use their cell phone. --Chris -- Chris Maj cmaj_hat_freedomcorpse_hot_info Pronunciation Guide: Maj == May Key ID: 0xF0DEC146 Key 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
Re: [Asterisk-Users] Predictive Dialer
Hi James-- I got a dialer working without too many hiccups about two months ago. It relies on changes to chan_agent, app_queue, a PostgreSQL backend, a Tcl-* manager interface, a bunch of Tcl glue, and some cron jobs. The results for each call are logged in right through the phone key pad, and the algorithm for prediction looks at number of agents logged in, average length of calls, and a magic number the boss man can set to speed it up or slow it down, plus a couple others I forget. Although it relies on some bastardization of the Caller-ID (who doesn't), it is in compliance with all the latest FCC rules. A key to making it stable was the recent placement of extra locks in the queue and agent code. It still gets some frozen lines, but I blame it on the Zhone, and they seem to thaw out when you power cycle the POS channel bank. I know there was a separate list setup for discussions about a predictive dialer, and I would like to contribute my code there but don't remember who made the list or if it has ever seen any traffic. Not to make a meta-comment on this thread, but whenever the discussion of a predictive dialer does arise, it seems to get spit on by those who aren't fans of the technology. I think that's a real shame as it represents a huge market for *. I had some moral qualms about it, too, but they pale in comparison to those I would have if, say, I was hacking on voicemail for the Pentagon or rolling out a PBX at Fox News. --Chris On Thu, 2 Oct 2003, James Coberly waxed: Hi, Some time ago there were posts about Predictive dialing. Has anyone seen or made any forward progress on this ability? I would be very interested in any further info regarding the ability. Thanks, James- -- Chris Maj cmaj_hat_freedomcorpse_hot_info 0xC0051F6A 5EBB 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 4-port T1 cards
On 28 May 2003, Steven Critchfield waxed: 8's While I'm on the postgres bandwagon for now, I wouldn't want it in the middle of a phone system doing heavy call loads either. Postgres also has some downsides too. As I understand it, postgres doesn't understand prepared statements, or at least it doesn't via the perl DBI. Regardless I've seen our postgres database eat +2600 updates in under 2 seconds from a remote host on the same exact hardware that mysql choked on and not cause any degredation of access times for any other user. I've been using my own PostgreSQL CDR backend -- patches submitted -- without a hitch for months now, peak load is ~1000 calls/hr. The server, an Athlon XP1800+ w/256MB and a 40gig IDE HD, also serves up data for a calling card service with PGSQL, and hosts 3-6 clients for data entry. Up for 100+ days. I was going to put * on it, too ;) I'm running PostgreSQL 7.2, but 7.3 does provide for PREPARE'd queries on a per-connection basis, although they are dropped after the connection is closed and I'm not sure what fancy footwork Perl-DBI does. --Chris ps. While on this thread, I'm attaching my PG CDR patches again. -- Chris Maj cmaj_hat_freedomcorpse_hot_info 0xC0051F6A 5EB8 2035 F07B 3B09 5A31 7C09 196F 4126 C005 1F6A cdr_postgres.patches.tar.gz Description: Binary data
