[Asterisk-Users] (no subject)
Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe(H323:996, ) in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno' -- Playing 'conf-getconfno' Thanks -- Jordan In a world without windows, who needs gates? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
I don't know what that is, so probably not. Is that a conference type board? Is there a way to make conferencing work or to assign an extension to a h323 connection? Thanks On Mon, 2003-06-23 at 23:42, Jeremy McNamara wrote: Do you have a Zaptel device in this machine? Jeremy McNamara Jordan Peterson wrote: Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe(H323:996, ) in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno' -- Playing 'conf-getconfno' Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordan In a world without windows, who needs gates? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and passwords
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi. Perhaps I just didn't found it, but there are two things I'm currently missing in asterisk: - - The user passwords for SIP are written *plain ascii* in the sip.conf... is there any possibility to store them encryptet? - - How can a user change his(her) SIP-Password? Is there any key in the voicemail/agent-system which allows changing the SIP-Password? The only thing I've found there is the possibility of changing the voicemail password. Or exists somewhrer an web-based form (like the SIPExpressRouter has) to do so? Thanks for your help Rainer - -- http://graphics.cs.uni-sb.de/~rainer/ -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.6 (GNU/Linux) Comment: For info see http://www.gnupg.org iD8DBQE+9/fKH/5+6U9F1qYRAnlLAJwN2sjtyXLwR1LzCqBu7Xg+EltuzQCfaDyU y39p7YeM5X8yhNtt7D+7veo= =Um1g -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] databases for billing
hostname=localhost dbname=asteriskcdrdb password= user=asteriskcdruser carlos del mayor wrote: I'm only asking for some examples of cdr_mysql.conf, nobody has done anything with cdr and mysql? If you think is better another DB,,, tell me, please! thanks in advance carlos carlos del mayor [EMAIL PROTECTED] wrote: can you be more explicit, please? or give me some examples? please, i'm little lost! thanks a lot carlos Martin Pycko [EMAIL PROTECTED] wrote: cdr_mysql.conf On Fri, 20 Jun 2003, carlos del mayor wrote: hi I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands? Any help would be so apreciated! Thanks a lot carlos - Yahoo! Sorteos Juega a la Lotera Primitiva sin salir de casa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Ya! hoo! Sorteos Juega a la Lotera Primitiva sin salir de casa Yahoo! Sorteos Juega a la Lotera Primitiva sin salir de casa
Re: [Asterisk-Users] (no subject)
You must have a Zaptel device installed in your computer or load ztdummy module to get conferencing to work... Jordan Peterson wrote: I don't know what that is, so probably not. Is that a conference type board? Is there a way to make conferencing work or to assign an extension to a h323 connection? Thanks On Mon, 2003-06-23 at 23:42, Jeremy McNamara wrote: Do you have a Zaptel device in this machine? Jeremy McNamara Jordan Peterson wrote: Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe(H323:996, ) in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno' -- Playing 'conf-getconfno' Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Active ISDN PCMCIA card
One more thing, just to be sure. Are these (AVM B1, Fritz) Cardbus cards? Nothing is mentioned on the web site. Thanks, Michael. Michael Manousos wrote: Thanks for the replies. It seems that AVM B1 is the only active PCMCIA card that can be used with Asterisk. The kernel supports this card, so I guess that the driver can be built on non-x86 systems. Regards, Michael. Olaf Menzel wrote: On Friday 20 June 2003 13:28, Michael Manousos wrote: Are there any suggestions for active ISDN CAPI PCMCIA cards that are known to work with Asterisk? You can try AVM B1 PCMCIA. This card is fully I4L compliant but AVM has developed a LINUX capi 2.0 stack. http://www.avm.de/en/products/hardware/active/B1_PCMCIA/index.html The Linux Capi driver you find here: ftp://ftp.avm.de/cardware/b1_pcm/linux/ Be aware that the Capi4Linux driver is distributed only as binary and especially prepared for Suse distributions. WIth some adaptations it should work with other distributions as well. Otherwise you should use I4L for this card. BTW. The Capi4Linux driver works also for the AVM Fritz which is much cheaper than the B1 device and supports full CAPI functionality such as G3 Fax. regards Olaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ringing tones oh323
Jorge Cisneros wrote: When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress This is generated by the IP phone. You don't have to do anything special. In the case that the call has already been answered, and you want to give back some kind of indication, you could playback a prerecorded tone. thanks Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up the E100P
Martin Pycko wrote: THat's not it. in zapata.conf you *also* need to have signalling=pri_cpe or pri_net that did the work, I already had ( I guessed about this :) the pri_cpe option in zapata.conf. so having in zaptel.conf: bchan=1-15,17-31 dchan=16 and in zapata.conf: signaling=pri_cpe made E1 work with PRI signaling Also my guess that Cisco`s switchtype primary-net5 ~= Asterisk EuroISDN am I right? This at least works with Seimens EDSW I think we should have this added to documentation in /etc/asterisk/zapata.conf thank a lot to Michael Blelicki and Martin Pycko!! Still does anybody know what do all this stuff in zttool mean? Thnks!! -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App queue only + waiting call pickup
Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives - put it in a queue - remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone encountered a similar request, or is working on it. I could start writing a new app, perhaps based on app_queue , or is better an agi script ? Or I'm missing a clever method to do it? Matteo. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App queue only + waiting call pickup
Personally, I'd like a feature similar to the app_queue but which doesn't require the agent's phone to be off-hook all the time. This way, the agent's phone can ring like normal and they can answer it. This better suits an 'office' environment where the 'agents' are not on the phone all day, but might get 2 or 3 calls per hour (or even less)... It's there. eg (from queues.conf) [sales] music = moh-queue-announce member = SIP/7011 member = SIP/7012 member = SIP/7013 member = Agent/17 member = Agent/18 member = Agent/19 context = callqueues-breakout You can also put Zaptel (eg: Zap/1/whatever) in there. Regards, Shaun Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] App queue only + waiting call pickup
Hi Il mar, 2003-06-24 alle 14:32, Adam Goryachev ha scritto: Today I was asked about a function of asterisk. That's what it should to: a call arrives - put it in a queue - remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone encountered a similar request, or is working on it. I could start writing a new app, perhaps based on app_queue , or is better an agi script ? Or I'm missing a clever method to do it? Just setup a extra parking range of extensions, say 750 -799 (depends on how many calls you might have queud at a time). Then, just have another extension (say 800 to pull a call from the 'queue') which just runs through from 750 to 799, the first call it finds is connected through to the user. mmh, and if I free the first parking exten, like 750, then another call arrives, is parked into 750... so I don't have a real FIFO... and ..how can I pull a call from the parking lot, without dialling the parked extension ? Personally, I'd like a feature similar to the app_queue but which doesn't require the agent's phone to be off-hook all the time. This way, the agent's phone can ring like normal and they can answer it. This better suits an 'office' environment where the 'agents' are not on the phone all day, but might get 2 or 3 calls per hour (or even less)... That should be accomplished with the standard app_queue, just define members as channels (not agents) like Zap/xx or SIP/blah or anything you would use in a standard Dial argument, and put incoming calls in the queue... then the member tecnologies will start to ring ;) That's what I use in standard office enviroments ;) Regards, Adam Matteo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip NAT
Hervé Thibaud wrote: Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit : ... I try to connect directly the both to fwd.pulver.com and now i have a perfect sound but the question is perhaps links after opening session is only on the local networks with 10Mb/s. Once i can (when i'll have an external user to call) i'll try. It's like i thought, the sound is nasty with many blanks when i try a connection on internet and ISDN bandwith on one channel 64kb/s is not enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and use a bandwith of 64kb/s so. Is anybody that has a free or a very cheap solution (it's to try asterisk) to have an IP phone hardware or software with G.723.1 codec Other thing, I would like to try X-Lite but i have pb with registration, i don't know how to write settings, i have an error (for example) : File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau sip:[EMAIL PROTECTED],0,1 failed for '192,168,0,1' I try many form of settings but didn't succeed. X-Lite support speex (SPX on the X-Lite screen). This is a very impressive codec that will even allow you to talk with someone over a _modem_ with a little bandwidth to spare for other stuff. Obviously the modem adds a degree of latency but it's still impressive. So with the latest technology, etc we've managed to get a _voice_ conversation to travel over a standard phone line. ;-) But seriously, it is impressive. Regards, Andrew Radke ,-_|\ [EMAIL PROTECTED] mobile: +61 412 798593 / \ Member, System Administrators Guild of Australia \_,-._* o I didn't know it was impossible when I did it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] databases for billing
Don't forget you mst have MySQL-dev package installed for the cdr_mysql app to be compiled when you build Asterisk. The Makefile looks for the MySQL includes and libraries. Good luck. Jim Friedeck Ing. Angel Gomez Garcia wrote: hostname=localhost dbname=asteriskcdrdb password= user=asteriskcdruser carlos del mayor wrote: I'm only asking for some examples of cdr_mysql.conf, nobody has done anything with cdr and mysql? If you think is better another DB,,, tell me, please! thanks in advance carlos */carlos del mayor [EMAIL PROTECTED]/* wrote: can you be more explicit, please? or give me some examples? please, i'm little lost! thanks a lot carlos */Martin Pycko [EMAIL PROTECTED]/* wrote: cdr_mysql.conf On Fri, 20 Jun 2003, carlos del mayor wrote: hi I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands? Any help would be so apreciated! Thanks a lot carlos - Yahoo! Sorteos Juega a la Lotería Primitiva sin salir de casa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users *Ya! hoo! Sorteos* *Juega a la Lotería Primitiva sin salir de casa* http://es.rd.yahoo.com/mail_es/tagline/lottery/*http://es.lottery.yahoo.com/v24/primisistema.html *Yahoo! Sorteos* *Juega a la Lotería Primitiva sin salir de casa* http://es.rd.yahoo.com/mail_es/tagline/lottery/*http://es.lottery.yahoo.com/v24/primisistema.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk hogging CPU resources
Below is the portion of the ps information about Asterisk. Note that Asterisk started out at less than 5% CPU, and after 1075:35, it is now at 74.6%. The interfaces I am using are 2 X100P, and a TDM40B. I know there had been a bug some time back that caused every asterisk thread to open handles on /dev/zap/timer repeatedly and at some point my system had run out of file handles to give out and performance started sucking. A CVS upgrade fixed that. I upgraded asterisk yesterday, and this problem still occurs. Also, I am not sure what is meant by opening handles on /dev/zap/timer; I don't have anything located at /dev/zap/timer. USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 9911 74.6 6.3 113296 7976 ? RJun23 1075:35 asterisk -vvvcg root 9912 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9913 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9914 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9915 0.0 6.3 113296 7976 ? SJun23 0:17 asterisk -vvvcg root 9916 0.0 4.0 7148 5176 ?SJun23 0:01 /usr/bin/mpg123 - root 9917 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9918 0.0 0.8 3952 1076 ?SJun23 0:00 /usr/bin/mpg123 - root 9919 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9920 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9921 0.7 6.3 113296 7976 ? SJun23 10:07 asterisk -vvvcg root 9922 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9923 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9924 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9925 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9926 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg root 9928 0.1 6.3 113296 7976 ? SJun23 1:34 asterisk -vvvcg root 9929 0.0 6.3 113296 7976 ? SJun23 0:00 asterisk -vvvcg Any help is always appreciated. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer Sent: Monday, June 23, 2003 2:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk hogging CPU resources What appears to be hogging CPU? What interfaces are you running? Mark On Fri, 20 Jun 2003, Derek Beaumont wrote: Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A question about cdr table
Hi, I have a little question: the calls from an 'oh323' terminal to another 'oh323' are they inserted into the cdr table in MySQL? CDR table has only the calls, to or from a zap channel, but it doesn't have any 'oh323' call. Any idea about this? Regards. Rafael Gonzalez Lomeña ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
You risk hanging up on your other 911 callers... but everything is always a tradeoff. In my experience, the 911 dispatcher can (does) pin the call, so that even though the remote side hangs up, the line is not available for use again until the dispatcher releases it. I'd expect this to mean that the proposed hangup would end up with the 911 operator transferred from caller-caller-caller if asterisk were configured to re-use a line for a new outbound 911 call... chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dynamic queue channels
Title: Message
There
may be some trickiness that can be done with "chan_local" asagents of the
call queue. However, a much more elegant way to do this would be to create
an app_addagent and app_removeagent that allows the dynamic addition and removal
of extensions from the agent pool for a given queue. addagent(${CHANNEL},
techsupport) or something like that.
Ben
-Original Message-From: Paulo Mannheimer
[mailto:[EMAIL PROTECTED] Sent: Monday, June 23, 2003 6:36
PMTo: [EMAIL PROTECTED]Subject:
[Asterisk-Users] dynamic queue channels
Hi, Im trying to build a call
center application that allows attendants to come in the morning and dial a
certain extension to make their extension available.
I wouldnt like to use the AgentLogin app because their line would need to stay
off-hook (is this correct?)
Is there any SET channel status
command that would allow me to do something like
this?
PauloHM
[Asterisk-Users] Chan Local Examples
There's anyone that's using chan_local that would provide to the community some example on how to use it or what's needed for ? Just for knowledge. Matteo. -- Matteo Brancaleoni Espia System Administrator http://www.espia.it - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NoOp gives an ringing indication ?
Hi all,
i want lock Zap channels via global var FREE1
if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel
But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication !?
The immediate property in the zapat.conf is yes
[tel1]
exten = s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten = s,2,Playback,gesperrt
exten = s,3,Hangup
exten = s,4,NoOp
exten = _X.,1,Dial,Modem/g1:${EXTEN}
exten = _X.,102,Hangup
What can i do ?
Thanks for help,
Thomas.
***
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[Asterisk-Users] PHP MySQL cdr interface?
Title: Message Before I "reinvent-the-wheel", does any one know of PHPbasedinterface to the CDRtable? If not, I will get started writing one. Thanks, Marcus
[Asterisk-Users] Conference calls on Pingtel Phones
Has anyone been able to get conference calls to work on the Pingtel Phones? I assume this feature works with their implementation, but connected to my asterisk box it doesn't work. The Pingtel phone thinks it is making a second call, but asterisk never sees anything about a second call. Any help would be appreciated. Sincerely, Andy Hester Consero ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
That would be the case if calls are dropped at random to clear the way for 911 calls. With some form of access control (NCOS, Calling Search Space/Partitions, priority levels) you would be able to drop the least important calls. BTW, how are trunk restrictions managed right now? How can I specify which phones/extensions can make local, long distance or international calls? Can this be controlled by a time-of-day schedule, to change restrictions after regular business hours (cleaning crew calling LD)? Dylan. Chris Witte wrote: You risk hanging up on your other 911 callers... but everything is always a tradeoff. In my experience, the 911 dispatcher can (does) pin the call, so that even though the remote side hangs up, the line is not available for use again until the dispatcher releases it. I'd expect this to mean that the proposed hangup would end up with the 911 operator transferred from caller-caller-caller if asterisk were configured to re-use a line for a new outbound 911 call... chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP MySQL cdr interface?
Roy Sigurd Karlsbakk posted a php utility to calculate call costs to this list a while back. I hacked it for my own use and you can have that if you'd like to improve it/make it general purpose. Iain --On Tuesday, June 24, 2003 11:18 am -0400 Marcus Adolfsson [EMAIL PROTECTED] wrote: Before I reinvent-the-wheel, does any one know of PHP based interface to the CDR table? If not, I will get started writing one. Thanks, Marcus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Polycom
I am trying to configure Asterisk to use a Polycom SoundPoint IP 500 phone. Does anyone know where I can get the software and configuration file for this phone. I spoke to Polycom support and they say it's up to the SIP vendor to provide this.
Re: [Asterisk-Users] NoOp gives an ringing indication ?
Where in here are you waiting for digits? I don't see anything in
here that allows input from the user. You are getting the ring
because the s extension is completed, and ${EXTEN} is set to
something, and then you're jumping right to _X. which is finding a
match and executing the Dial application.
You need a Background application in the s extension range
somewhere that waits for digits.
JT
Hi all,
i want lock Zap channels via global var FREE1
if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel
But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication !?
The immediate property in the zapat.conf is yes
[tel1]
exten = s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten = s,2,Playback,gesperrt
exten = s,3,Hangup
exten = s,4,NoOp
exten = _X.,1,Dial,Modem/g1:${EXTEN}
exten = _X.,102,Hangup
What can i do ?
Thanks for help,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] asterisk and passwords
Currently there is no way to store SIP passwords in an encrypted form, and there is no user interface to change passwords that is built into the Asterisk system. You will need to build your own user interface. Note that you could, if you really, REALLY wanted to, build your own password encryption method using some of the tools in the ser toolkit. An easier way to do it would be to allow users to change their passwords via an AGI script (perl?) that had an interface through a voice prompt menu within Asterisk. JT Hi. Perhaps I just didn't found it, but there are two things I'm currently missing in asterisk: - - The user passwords for SIP are written *plain ascii* in the sip.conf... is there any possibility to store them encryptet? - - How can a user change his(her) SIP-Password? Is there any key in the voicemail/agent-system which allows changing the SIP-Password? The only thing I've found there is the possibility of changing the voicemail password. Or exists somewhrer an web-based form (like the SIPExpressRouter has) to do so? Thanks for your help Rainer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
make a context for l/d dialing and include it for the phones / times of day, when it is actually supposed to be used, not otherwise. At 09:52 AM 6/24/2003 -0600, you wrote: That would be the case if calls are dropped at random to clear the way for 911 calls. With some form of access control (NCOS, Calling Search Space/Partitions, priority levels) you would be able to drop the least important calls. BTW, how are trunk restrictions managed right now? How can I specify which phones/extensions can make local, long distance or international calls? Can this be controlled by a time-of-day schedule, to change restrictions after regular business hours (cleaning crew calling LD)? Dylan. Chris Witte wrote: You risk hanging up on your other 911 callers... but everything is always a tradeoff. In my experience, the 911 dispatcher can (does) pin the call, so that even though the remote side hangs up, the line is not available for use again until the dispatcher releases it. I'd expect this to mean that the proposed hangup would end up with the 911 operator transferred from caller-caller-caller if asterisk were configured to re-use a line for a new outbound 911 call... chris. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PHP MySQL cdr interface?
I don't know of one but would be very interested in getting hold of one.. If you are going to realese it please let me know.. Thanks.. Before I reinvent-the-wheel, does any one know of PHP based interface to the CDR table? If not, I will get started writing one. Thanks, Marcus -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update
TC, Thank you very much for the pointer. I wanted to acknowledge how valuable it was. I was able to monitor the channel. I did not know that tool exsited or what it did. Now I do. I actually do not know what an FSK tone is. I gather, from google, that it is some audible tone between the first and send ring. I was not able to to hear anything between the rings. However, my caller ID stopped working in loopstart mode as well. So... I have to dig deeper. Thanks again, Jason On Friday, June 20, 2003, at 10:47 AM, TC wrote: Callerid issue 1) if you run ztmonitor on the fxo line call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail else do you get check sumfails or sum fin else else zaptel is not detecting the ring/fsk correct might need to tunimng on ZT_RINGTRAILER else CAC is hooped -Original Message- From: Surfer Dude [EMAIL PROTECTED]> To: [EMAIL PROTECTED] [EMAIL PROTECTED]> Date: June 20, 2003 10:24 AM Subject: Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update Thanks for the responses. Here is an update on my zap groundstart, loopstart, disconnect supervision woes. Apparently, the groundstart mode in the CAC FXO module works on loopstart lines. I still don't understand how and why. I have loopstart lines. The only way disconnect supervision works on my system is if I set the FXO module to groundstart. However! There is a big caveat. In groundstart mode, incoming caller ID stops working! This is something I am going to have to live with for now as disconnecting the line on hang-up is far more important than caller ID. I have no idea if the caller ID problem is a problem with CAC FXO modules or the zap channel software or the fact that we are using groundstart signaling on a loopstart line. I can confirm that this is the behavior with both CAC I and CAC II units as I have a CAC II unit and I spoke with Michael on this list who has a CAC I unit. Both of us are in the same predicament. Anyway, I hope others find this info useful. If anyone has any other ideas on how I can get my caller ID to work I would be really excited about that. Thanks, Jason PS: I have seen people mention that, when you have many lines, PRI is cheaper. This is not our case at all. San Francisco. We have eight lines at the cost of $0.01 per line. All we pay for really is the call time. - Original Message - From: Surfer Dude To: [EMAIL PROTECTED] Sent: Tuesday, June 17, 2003 4:05 PM Subject: [Asterisk-Users] Question :: groundstart and loopstart Hello Astrites, I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines. I had tried many configurations. I had attached a toner to the line to see if I was getting this disconnect supervision signal after hang-up. (toner has a line powered light in the off position) It seemed that I was getting the signal, as: The light was steady on onhook When the line rang the light dimmed. When I picked up the line stayed dim (If I remember correctly) When the other end hung up the light stayed on Then turned off momentarily a few seconds later. (I suspected this was disconnect supervision) Anyway asterisk never detected hang-up. Everything else worked. I was frustrated as hell. After a few days of trying everything, I tried to set my CAC II FXO module to Groundstart and configured zaptel.conf and zapata.conf to groundstart signaling. Voila! Everything work! The question is how does groundstart and loopstart work? How was I able to have a somewhat working system using loopstart when it seems that the phone company signaling was groundstart? Can I assume that my lines are indeed groundstart? I am really happy that it works. I can now begin the process of deployment with a much greater feeling of security. Jason
AW: [Asterisk-Users] parsing bug? (using PGSQL)
OK. I see.
This works.
Thank you,
Thomas.
-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Martin
Pycko
Gesendet: Dienstag, 24. Juni 2003 18:03
An: Asterisk User
Betreff: Re: [Asterisk-Users] parsing bug? (using PGSQL)
If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)
That should work
regards
Martin
On Tue, 24 Jun 2003, Thomas Haeger wrote:
Hi all again,
if i make a query with
...
exten = _X.,2,PGSQL,Query resultid ${connid} SELECT
getdest('${EXTEN}');
...
an error like
WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No
application
'PGSQL,Query resultid ${connid} 'SELECT getdest'
for extension (tel1, 00905888, 2)
occurs.
This looks like an parsing bug. As if the brackets be cut.
Can somebody help?
Thanks,
Thomas.
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RE: [Asterisk-Users] Asterisk and Polycom
Title: Message Yeah, we tested this. Polycom was a real butt about it too. They have sealed some agreements with vendors that sell their phones with their proprietary systems. While Polycom admits the phone does "SIP" they will not disclose any info about configuring it and the vendors phone systems that configure it have system specific firmware and xml config files. So, my take on the Polycom is that they say "SIP" but it is not really usable by anything but their partners phone systems. So my advice is don't waste your time. Ben -Original Message-From: Chris [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 24, 2003 10:30 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] Asterisk and Polycom I am trying to configure Asterisk to use a Polycom SoundPoint IP 500 phone. Does anyone know where I can get the software and configuration file for this phone. I spoke to Polycom support and they say it's up to the SIP vendor to provide this.
[Asterisk-Users] Asterisk vs. system user accounts
I've been scouring the archives for discussions on this: Why doesn't Asterisk use system user accounts for each extension/mailbox? That would add the benefit of encrypted passwords, logical grouping, unified mail/voice mail accounts (using /var/spool/mail instead of /var/spool/asterisk). I can already imagine Festival reading my emails to me, HylaFAX faxing documents to me while I'm on the road :). Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
Because you haven't written and contributed that functionality yet. (smiley face goes here) That sounds pretty sweet. I'm wondering if LDAP might be the more correct thing to use though. -reed At 10:50 AM 6/24/2003 -0600, you wrote: I've been scouring the archives for discussions on this: Why doesn't Asterisk use system user accounts for each extension/mailbox? That would add the benefit of encrypted passwords, logical grouping, unified mail/voice mail accounts (using /var/spool/mail instead of /var/spool/asterisk). I can already imagine Festival reading my emails to me, HylaFAX faxing documents to me while I'm on the road :). Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
I'm wondering if LDAP might be the more correct thing to use though. Absolutely! Reed Wade wrote: Because you haven't written and contributed that functionality yet. (smiley face goes here) That sounds pretty sweet. I'm wondering if LDAP might be the more correct thing to use though. -reed At 10:50 AM 6/24/2003 -0600, you wrote: I've been scouring the archives for discussions on this: Why doesn't Asterisk use system user accounts for each extension/mailbox? That would add the benefit of encrypted passwords, logical grouping, unified mail/voice mail accounts (using /var/spool/mail instead of /var/spool/asterisk). I can already imagine Festival reading my emails to me, HylaFAX faxing documents to me while I'm on the road :). Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working Clients for Linux?
All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is trying to connect. LINPHONE -- Linphone does not work with ALSA, nor with ALSA's OSS emulation. I've used pre-packaged version of Linphone and compiled my own, using --enable-alsa to get it up and running. NOTE -- not all versions of ALSA let you do this! There's a silent failure to compile -- check config.log to make certain that the ASOUND_H is defined. GNOPHONE -- Gnophone also does not work with ALSA, although I haven't yet tried to compile it from a tarball/CVS. With ALSA version 0.9.4, GnoPhone won't even give me a chance to configure the sound card. IIRC GnoPhone did work with 0.9.2 and possibly 0.9.3. 1. I invite comments from people who can get these clients to work with ALSA 0.9.4. I am working with the latest CVS of ALSA. 2. Are there other open-source clients that I've missed? Regards, Moshe P.S. I've fiddled with the CVS versions only enough to get things to compile. E.g., in linphone, configure.in must be changed to find alsa/asoundlib.h instead of sys/asoundlib.h. Well, ok, I did add -mfmath=sse to some of the Makefiles but that should be harmless enough... -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ALSA module not working
Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or chan_alsa.so; as of this morning, only chan_oss.so will load. The symptoms are very straightforward: If chan_alsa.so loads, no subsequent module loads. An outward sign is that show dialplan will have just a few items and not the entire dialplan because pbx_config.so doesn't load, and important functions like stop now do not exist at the console. There are no error messages, and just the ordinary debug messages. I haven't had any success deciphering the problem -- I'm still figuring out its extend and ramping up on new sections of Asterisk code. I am using the latest (2003-06-24) CVS of asterisk. Question: Is anyone using Asterisk's chan_alsa.so and ALSA version 0.9.4 from a later CVS? I like to know if it's some problem with my local configuration or a bug I haven't found. Related question: Are OSS and ALSA mutually incompatible? I would think so, but there's nothing in the documentation. If they should *not* be loaded simultaneously, I will submit a few lines to patch modules.conf. -- Moshe Yudkowsky * http://www.Disaggregate.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patching Festival
I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. tar xvzf festival-1.4.2-release.tar.gz patch -p0 /usr/src/asterisk-ng/festival-1.4.2-diff (or wherever the patch is located) When I run the patch command, I get the following: [EMAIL PROTECTED] src]# patch -p0 /usr/src/asterisk/festival-1.4.2.diff can't find file to patch at input line 4 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff -u -r festival-1.4.2/lib/tts.scm festival-1.4.2-asterisk/lib/tts.scm |--- festival-1.4.2/lib/tts.scm Wed Jan 8 09:54:14 2003 |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan 7 08:51:44 2003 -- File to patch: Which file am I supposed to patch? Thanks for your help. -Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP REGISTER script
Some of you have unusual SIP configurations, and this SIP perl script
may be useful to get remote devices registering with your Asterisk or
other SIP server. Most Cisco routers, as an example, are too stupid
to REGISTER, so this script would be required to dynamically register
them with a remote server. This may not be 100% applicable to
Asterisk, since static registrations are possible, but who knows?
Maybe someone will find this useful. There is no password support on
this version. I haven't fooled around extensively with the script -
feel free to modify and re-submit a better working copy.
If anyone wants to build the password-capable version of this, some
tools can be found in the SER package (http://www.iptel.org/ser/) in
the utils/gen_ha1 directory - there is a password generator in
there, along with C routines for WWW-Authorize responses.
Notes:
Replace 123.123.123.123 with the IP address of the remote SIP
server that you're trying to REGISTER with.
Replace sipdomain.company.com with the domain you're using. This
may not matter much for some SIP servers, but others are fussy about
it.
Change the Expires: value to whatever you think is useful. It's
measured in seconds.
Change John Doe sip:[EMAIL PROTECTED] to your contact info
(probably something like sip:[EMAIL PROTECTED])
JT
Original script by Marian Durkovic
cut here ---
#!/usr/bin/perl
use Socket;
# USAGE register { local IP address, Extension number, SIP contact address }
register(1.2.3.4, 9, John Doe sip:[EMAIL PROTECTED]);
sub register {
$local_ip = shift;
$ext_number = shift;
$contact = shift;
$proxy_ip = 123.123.123.123;
$tm = time();
$seq = $tm % 65536;
$MESG=REGISTER sip:sipdomain.company.com SIP/2.0
Via: SIP/2.0/UDP $local_ip:5060
From: sip:[EMAIL PROTECTED]
To: sip:[EMAIL PROTECTED]
Contact: $contact
Call-ID: [EMAIL PROTECTED]
CSeq: $seq REGISTER
Expires: 3700
Content-Length: 0
;
$proto = getprotobyname('udp');
socket(SOCKET, PF_INET, SOCK_DGRAM, $proto) ;
$iaddr = inet_aton($local_ip);
$paddr = sockaddr_in(5060, $iaddr);
bind(SOCKET, $paddr) ;
$port=5060;
$hisiaddr = inet_aton($proxy_ip) ;
$hispaddr = sockaddr_in($port, $hisiaddr);
send(SOCKET, $MESG, 0,$hispaddr ) || warn send $host $!\n;
}
cut here ---
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Re: [Asterisk-Users] Patching Festival
What directory are you in while running this? If you are in the festival directory try: patch -p1 /usr/src/asterisk-ng/festival-1.4.2-diff The -p options strips directory names from the patch. Hope that helps. -Steve On Tue, 24 Jun 2003, Derek Beaumont wrote: I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. tar xvzf festival-1.4.2-release.tar.gz patch -p0 /usr/src/asterisk-ng/festival-1.4.2-diff (or wherever the patch is located) When I run the patch command, I get the following: [EMAIL PROTECTED] src]# patch -p0 /usr/src/asterisk/festival-1.4.2.diff can't find file to patch at input line 4 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff -u -r festival-1.4.2/lib/tts.scm festival-1.4.2-asterisk/lib/tts.scm |--- festival-1.4.2/lib/tts.scm Wed Jan 8 09:54:14 2003 |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan 7 08:51:44 2003 -- File to patch: Which file am I supposed to patch? Thanks for your help. -Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
LDAP is a _MUCH_ better solution to that problem than user accounts on a machine. Even if you have a /bin/false shell, users could cause trouble with your system. For security reasons you want to keep the number of accounts low and the number of accounts with password protected access even lower. So using the built-in passwords for a system would only make you more likely to be rooted on a machine that your business will depend upon. After saying LDAP is a better choice than system users, I still wonder why it is important to have users be able to change passwords here. In small deployments, it isn't going to happen often enough to bother the administrator. In office environments, you shouldn't have to modify passwords. Only when you have people roving around should you be worried about password changes to keep them secure. OF course you should also first try to see how hard LDAP is to get setup and secured before trying to link it into asterisk. On Tue, 2003-06-24 at 12:07, Reed Wade wrote: Because you haven't written and contributed that functionality yet. (smiley face goes here) That sounds pretty sweet. I'm wondering if LDAP might be the more correct thing to use though. -reed At 10:50 AM 6/24/2003 -0600, you wrote: I've been scouring the archives for discussions on this: Why doesn't Asterisk use system user accounts for each extension/mailbox? That would add the benefit of encrypted passwords, logical grouping, unified mail/voice mail accounts (using /var/spool/mail instead of /var/spool/asterisk). I can already imagine Festival reading my emails to me, HylaFAX faxing documents to me while I'm on the road :). Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patching Festival
Make sure the directory that festival is in is named festival-1.4.2. One other user had the same problem. The directory was called festival. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Derek Beaumont Sent: Tuesday, June 24, 2003 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Patching Festival I just wanted to try out Festival, but I can't get it patched. I'm thinking that there is something missing from the steps listed at http://www.marko.net/asterisk/archives/0209/0389.html. tar xvzf festival-1.4.2-release.tar.gz patch -p0 /usr/src/asterisk-ng/festival-1.4.2-diff (or wherever the patch is located) When I run the patch command, I get the following: [EMAIL PROTECTED] src]# patch -p0 /usr/src/asterisk/festival-1.4.2.diff can't find file to patch at input line 4 Perhaps you used the wrong -p or --strip option? The text leading up to this was: -- |diff -u -r festival-1.4.2/lib/tts.scm festival-1.4.2-asterisk/lib/tts.scm |--- festival-1.4.2/lib/tts.scm Wed Jan 8 09:54:14 2003 |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan 7 08:51:44 2003 -- File to patch: Which file am I supposed to patch? Thanks for your help. -Derek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog 2x8
Is anyone on the list running an Asterisk system with 2 x X100P and 2 x TDM40 (4 port) cards? I am interested in your hardware setup. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk vs. system user accounts
On Tue, 2003-06-24 at 15:47, Dylan VanHerpen wrote: After saying LDAP is a better choice than system users, I still wonder why it is important to have users be able to change passwords here. It would greatly simplify unified messaging: one account, all your messages (email, voice, fax) in one mailbox. This doesn't mean there needs to be a real account on the system. Exim can deliver to non login accounts, Cyrus IMAP doesn't need real user accounts, and I forget the pop3 daemon that works the same way. Hylafax doesn't need a user to send you the faxes. So again, you only need some simple way of keeping user/password mappings straight. LDAP would be okay, if it was easy enough to setup and use. Not to mention for your SIP users, LDAP can be used as a phone directory. I hope this helps you move forward with something usable and secure. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Macro Example
Cool trick!
You could simplify this:
[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten = s/_,1,Dial(${ARG1}r2,${ARG2})
; Caller*ID is not 4 digits (external call)
exten = s,1,Dial(${ARG1},${ARG2})
; Both of the above lines go here next
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
For those of you running Cisco 7960's and using the ALERT_INFO stuff,
You can use this version of the same thing. (I am now using this in my
config thanks to Eric's example):
[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten = s/_,1,SetVar(ALERT_INFO=1)
; Caller*ID is not 4 digits (external call)
exten = s,1,NoOp
; Both of the above lines go here next
exten = s,2,Dial(${ARG1},${ARG2})
exten = s,3,Voicemail(u${MACRO_EXTEN})
exten = s,4,Hangup
exten = s,103,Voicemail(b${MACRO_EXTEN})
exten = s,104,Hangup
John
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Tuesday, June 24, 2003 4:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Distinctive Ring Macro Example
I use the following macro for my extensions. It only works with Zap
channels and assumes that any Caller*ID number that is 4 digits is an
internal call and all other calls are external calls.
Use like this: exten = 1234,1,Macro(std-exten,Zap/4,20)
[macro-std-exten]
;
; Caller*ID is 4 digits (internal call)
;
exten = s/_,1,Dial(${ARG1}r2,${ARG2})
exten = s/_,2,Voicemail(u${MACRO_EXTEN})
exten = s/_,3,Hangup
exten = s/_,102,Voicemail(b${MACRO_EXTEN})
exten = s/_,103,Hangup
;
; Caller*ID is not 4 digits (external call)
;
exten = s,1,Dial(${ARG1},${ARG2})
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup
--Eric
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)
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Re: [Asterisk-Users] Asterisk vs. system user accounts
No-one has mentioned PAM yet. (pluggable authentication modules). If you implement PAM in Asterisk, then you have LDAP/passwd shadow windows etc. in one step. Maybe the phone numbers in /etc/passwd will get used! cheers, Woody On Tue, Jun 24, 2003 at 04:17:15PM -0500, Steven Critchfield wrote: On Tue, 2003-06-24 at 15:47, Dylan VanHerpen wrote: After saying LDAP is a better choice than system users, I still wonder why it is important to have users be able to change passwords here. It would greatly simplify unified messaging: one account, all your messages (email, voice, fax) in one mailbox. This doesn't mean there needs to be a real account on the system. Exim can deliver to non login accounts, Cyrus IMAP doesn't need real user accounts, and I forget the pop3 daemon that works the same way. Hylafax doesn't need a user to send you the faxes. So again, you only need some simple way of keeping user/password mappings straight. LDAP would be okay, if it was easy enough to setup and use. Not to mention for your SIP users, LDAP can be used as a phone directory. I hope this helps you move forward with something usable and secure. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling Asterisk under Yellow Dog
On Tue, 2003-06-24 at 17:59, Serge Mankovski wrote: Hi, I am trying to compile Asterisk under Yellow Dog 3.0 distributionn. I am getting an error gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm /usr/bin/ld: cannot find -lgsm May be I need packages that my distribution does not include? What do I need to download to get it compiled? Just a bit of education for you so you can help yourself later, -l flags to gcc are to link against libraries and the name of the library is lib + what ever is appended to the -l flag. It appears you have the headers for libgsm since it didn't complain and die on lack of headers, but it appears that gcc and ld are unable to find libgsm. It might be that it is installed to a directory that ld needs to be made aware of in /etc/ld.so.config. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP-to-SIP proxy
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a proxy, but when connecting SIP to SIP it works only as 'SIP registrar' forwarding SIP requests to client. Is it possible to make Asterisk work as a 'proxy' so that any incoming calls would be ade to Asterisk and then internally forwarded to receiver? (for eg. when Asterisk is on host that is doing masquerade for internal network and receiver/caller is in this masqueraded network) - -- Witold Piotr Krecicki (adasi) adasi [at] culm.net GPG key: 7AE20871 http://www.culm.net -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.1 (GNU/Linux) iD8DBQE++N659le3g3riCHERAom9AKCF460/VHVaVRBfCgRANvoJz7BjxACghbkf S1LWBrHtnR661pQDQBls/MU= =dSDZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SIP-to-SIP proxy
On Tue, 2003-06-24 at 18:28, Witold P. Krecicki wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a proxy, but when connecting SIP to SIP it works only as 'SIP registrar' forwarding SIP requests to client. Is it possible to make Asterisk work as a 'proxy' so that any incoming calls would be ade to Asterisk and then internally forwarded to receiver? (for eg. when Asterisk is on host that is doing masquerade for internal network and receiver/caller is in this masqueraded network) Use the reinvite=no or canreinvite=no options. Search list for more detail on these options. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with # and extensions.
I get the following message when I dial 74#. Does anyone have any ideas
on what might be going on? If I don't require numbers to be terminated
with # everything works as expected, but you have to wait for the digit
timeout, of course.
MESSEGE:
DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got
something to jump out with ('#')!
-- Invalid extension '#' in context 'speeddial' on SIP/2111-076f
CONFIG:
[default]
exten = 74#,1,Goto(speeddial,s,1)
[speeddial]
exten = s,1,Background(${MYSOUNDS}/enterspeedcode)
exten = 1#,1,Background(${MYSOUNDS}/speedhelp)
exten = 1#,2,Goto(s,1)
exten = _[2-9]#,1,SetVar(SPEEDCODE=${EXTEN})
exten = _[2-9]#,2,Background(${MYSOUNDS}/enterspeednumber)
exten = _X.#,1,SetVar(SPEEDNUMBER=${EXTEN})
exten = _X.#,2,Background(${MYSOUNDS}/speedcode)
exten = _X.#,3,SayDigits(${SPEEDCODE})
exten = _X.#,4,Background(${MYSOUNDS}/assignedto)
exten = _X.#,5,SayDigits(${SPEEDNUMBER})
--
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)
___
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[Asterisk-Users] Tor2 modprobe problem
I just received my dev kit and installed the hardware. I followed the directions Digium sends out to the letter and when I ran modprobe tor2 I got the following error: /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod /lib/modules/2.4.18-19.7.x/misc/tor2.o failed /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed What have I missed? -- Ryan V. Howerton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_oh323.c Segmentation fault during Openphone/Gnomemeetingconnect during module loading...
My apologies if this question has been answered previously. However, I found that it was nearly impossible to search and find since anything can cause a segmentation fault. Problem. When Asterisk is booting up the h323 modules and a client tries to connect before Asterisk/h323 is finished booting, the program seg faults out and doesn't load. I thought about putting this into the inittab, but wouldn't that be more of a pain to manage from the initttab (respawn, etc)? Other than that, it boots up fine and works with h323 clients just fine..Can you imagine the hell a person would go through trying to boot up asterisk once it becomes more popular to use by clients? For now, it works fine because i am the only person who is connecting to it.. haha.. Is there a simple fix to this, or is everyone having this problem? Thanks -- Jordan In a world without windows, who needs gates? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tor2 modprobe problem
OK...I think I have found that answer to one problem, but now more have cropped up. Apparently I was using the wrong module. What modules do I need to load? Are there any parameters that need to be passed? Where can I find more information about my cards. I assumed that there would be more information with the cards than I received. Ryan I just received my dev kit and installed the hardware. I followed the directions Digium sends out to the letter and when I ran modprobe tor2 I got the following error: /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod /lib/modules/2.4.18-19.7.x/misc/tor2.o failed /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed What have I missed? -- Ryan V. Howerton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tor2 modprobe problem
On Tue, 2003-06-24 at 22:19, [EMAIL PROTECTED] wrote: I just received my dev kit and installed the hardware. I followed the directions Digium sends out to the letter and when I ran modprobe tor2 I got the following error: /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod /lib/modules/2.4.18-19.7.x/misc/tor2.o failed /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed What have I missed? This would be because your card was not detected. Since I haven't kept up with what was in the dev kit, do you have the 4 port T1 card? Tor2 is for the 4 port T1/E1 cards. If it is a single port T1 card, you need to use the wct1xxp module. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
