Re: [Asterisk-Users] Asterisk MGCP register
On Sat, 2003-12-20 at 03:22, Senad Jordanovic wrote: Hi, I am trying to figure out if * can register as a client on a remote MGCP service. Just like SIP and other protocols Do. Anyone tried this? No I don't believe it can. The MGCP implementation in Asterisk is a CallAgent not a UserAgent. --Karl Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: eagi-sphinx-test.c has hardcoded values for the sphinx server. Change the values to suit your network and installation and try again. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sphinx
On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: eagi-sphinx-test.c has hardcoded values for the sphinx server. Change the values to suit your network and installation and try again. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] issue recording files in wav49 from AGI
Following is a log from an attempt to record and playback a file in
wav49 format from an AGI script.
COMMAND: stream file aa/after_the_tone 0
RESULT_LINE: 200 result=0 endpos=41920
RESULT_DICT: {'result': ('0', ''), 'endpos': ('41920', '')}
COMMAND: record file /activity_alerts/wavs/123456_1_1_0.745781945801 wav49 # 2 0
beep
-- Playing 'beep' (language 'en')
RESULT_LINE: 200 result=35 (dtmf) endpos=0
RESULT_DICT: {'result': ('35', 'dtmf'), 'endpos': ('0', '')}
COMMAND: stream file /activity_alerts/wavs/123456_1_1_0.745781945801 0
WARNING[2160657]: File format_wav_gsm.c, Line 194 (check_header): Does not say data
WARNING[2160657]: File file.c, Line 379 (ast_filehelper): Unable to open fd on
/activity_alerts/wavs/123456_1_1_0.745781945801
WARNING[2160657]: File app_agi.c, Line 322 (handle_streamfile): Unable to open
/activity_alerts/wavs/123456_1_1_0.745781945801
RESULT_LINE: 200 result=0 endpos=0
== Spawn extension (activity-alerts, s, 2) exited non-zero on '[EMAIL PROTECTED]/2'
At the point where it tries to playback the file the warnings are spit
out and the call is disconnected.
Any ideas?
--Karl
--
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[Asterisk-Users] pyst the Python for Asterisk project
A new minor release of pyst is available at http://sourceforge.net/projects/pyst/ Completed is the manager interface allowing access to all manager commands. The manager interface is event driven and designed to be useful for driving gui applications. Also included is a small bug fix to agilib for channel names that had a : as part of the name... most notably IAX channels. Docs for agilib are complete and available using pydoc. The manager module still needs to be documented and provided with samples. Enjoy, --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Predictive Dialer
On Thu, 2003-10-02 at 17:09, C. Maj wrote: 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. That list was set up by me back in April. There wasn't much traction at that time, but for those interested in the subject of predictive dialing and Asterisk the subscription page for the list is http://www.putland.linux-site.net/mailman/listinfo/astdialer-dev The best thing I can say for contribution is either to post it to the dialer list or open a sourceforge project for it. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SoundPoint 500 with Asterisk
On Wed, 2003-08-27 at 09:59, Timothy Soos wrote: Hello All, Does anyone use a Plolycom SIP-based phone with Asterisk? Does anyone use a Polycom SoundPoint 500 with Asterisk using SIP? If so, please share your experiences, both good and bad. I tried to contact Polycom regarding their VoIP products and they were less than helpful. They will not offer any documentation about the phones. That being said... Feel free to give them a shot, but realize that supply and support can ba hard to come by for those phones. --Karl Thanks, Tim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] mgcp problems
On Fri, 2003-07-11 at 08:42, Pavel Zheltouhov wrote: When I connected over two mgcp channels and sending numerical indication to cisco ata it seems hangup one channel (receving ) and generate 'fast busy' tone. I hack chan_mgcp and my threewaycalling works ok! But why indications are sent after I press hookflash on answering end? indications are sent to provide a dialtone after flashhook. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hook Flash INFO messages
On Fri, 2003-07-11 at 22:12, Sean P. Robertson wrote: Here is a question that needs a few opinions... Recently we installed a couple of FXS gateways into a site with a SIP Proxy/Softswitch other than Asterisk. One of the things that the users on this site need to do is receive calls on single line phones on the FXS gateways and then hookflash and transfer them to other SIP users. We found that the FXS units, true to their nature as VoIP gateways, saw the hookflash and passed a SIP INFO (event hookflash) back to the Proxy. The Proxy sent this message on to the calling SIP phone which replied that this feature is not implemented. The gateway manufacturer says that the Proxy should process the INFO packet, place the calling endpoint on hold (as a PBX would), stream dialtone to the gateway prompting the user to dial the digits indicating the destination to whom the calling party should be transferred, and then do a transfer. The Proxy manufacturer says that the gateway should see the hookflash, Hold the caller locally (as a SIP phone would), and give new dialtone to the single line phone prompting the user to dial the digits digits indicating the destination to whom the calling party should be transferred, and then send a complete transfer sequence to the Proxy. My question is, how would Asterisk handle a situation like this? Are there any opinions as to how this scenario should be handled? Asterisk currently only handles dtmf INFO messages. --Karl Sean -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] switch = priority in the dialplan..(probably an issue for Mark)
On Fri, 2003-07-04 at 02:01, WipeOut . wrote:
Hi,
It seems that the switch parameter has a priority in the dialplan
that is higher than the wildcard extensions.. This I am finding to be
a problem..
switches are actually searched after the local dialplan.
The problem is that _90027. is a can_match not exact_match so *
continues to search the switch for an exact match until the digit
timeout or until you're done dialing at which point it uses the local
match of _90027..
After dialing take a look at iax2 show cache to see the results of the
lookups against the switch.
--Karl
My setup..
UA1--[AST1]--{IAX}--[AST2]--UA2
| |
PSTN1 PSTN2
I use switch on AST1 to connect to AST2... As you can see I have PSTN
connections on both and also the IAX connection is not permanent..
I have wildcard extensions that define which PSTN line to use when
dialing out..
For example I have the following on AST1 in extensions.conf..
[extensions]
switch = IAX2/user:[EMAIL PROTECTED]/extensions
and my local extensions are in this context..
[dialout-uk]
exten =
_90044[1-9].,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED])
[dialout-int]
exten = _90027.,1,Dial(Zap/1/${EXTEN})
and some other country definitions..
What is happeneing is that when I break the IAX connection(this way I
can see the errors) and I dial 9002731555 from UA1, AST1 tries to
look for extension 9 on AST2 via switch and times out then tries
extention 90 on AST2 then 900 then 9002 then 90027 etc... you get the
idea.. Only once it has tried every number will it move to the _90027.
definition and use the PSTN line attached to AST1 as it is supposed
to.. This whole sequence takes a while to run through and by then you
would have hung up the phone and tried again..
I have tried moving things around in the extensions.conf file but it
seems that switch has a higher priority than local wildcard
extension definitions..
Surely ALL local definitions (static or wildcard) should take priority
over switching to the remote Asterisk box?? so switch should be the
last thing to try when all else has failed... or am I missing
somthing??
Later..
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RE: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wed, 2003-07-02 at 13:12, Joe Antkowiak wrote:
How do you tell asterisk to detect for fax tones?
create and exten for fax
exten = fax,1,Dial(${MYFAXDEVICE})
--Karl
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, July 02, 2003 2:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] BIG problem with multiple rings before pickup
On Wed, 2003-07-02 at 13:34, Jim Archer wrote:
Hi All...
I have a maddening problem...
I have Asterisk configured to pick up a line after 4 rings. I do this to
allow my fax machine to pick up a particular distinctive ring pattern, so
I
don't have to pay for a dedicated fax line.
If someone calls the line, lets it ring 3 times and then hangs up,
Asterisk
answers the line, and holds it off hook forever, constantly playing the
prompts.
My hardware is 2 X100P cards.
Any ideas?
Get a TDM10B, cancel your distinctive ring, and let asterisk answer
immediately and detect fax tones and forward it to your fax machine.
--
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[Asterisk-Users] mec3 experiment
I was fooling around with echo cans today. I found that I could reproduce this problem regularly t400p adit600 for fxs mec3 enabled Two speaker phones side by side within about 8 of each other. Put phone 1 offhook with speakerphone Dial phone 2 Phone 2 rings and is loud enough to be picked up by the mic in phone 1 By the second ring the echocan thinks it's figured things out, but the ringtone on phone 1 suddenly changes to a different sound. Asterisk box is now completely locked up and the phones continue to ring regardless of whether they are answered of hungup. Same scenario with mec2 and all is well with the world. Looks like I won't be using mec3 anytime soon. Is anyone else able to confirm this issue with mec3? --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy message with call waiting?
On Fri, 2003-06-13 at 16:07, John Todd wrote: Hmm... this gets quickly back to my long-standing desire to have more comprehensive call completion codes being handed back by the channels to the dialplan. Just a couple of comments. I agree with jtodd about the call completion codes, but I'd like to put this out for some thought as well. Why is it that Dial is the thing that waits for the call to be completed? Why not have dial just dial, then have applications like WaitForAnswer, WaitForDisconnect etc...? This would give more granularity to the call flow control and allow someone to get brave and write a WaitForHuman or whatever. Or for that matter... since extensions.conf is starting to look like a scripting language, Why not embed a perl or Python interpreter into Asterisk to allow for programming extension logic in whatever your favorite language is. Just some food for though. --Karl The current method of throwing certain replies into a big bucket called Busy and others into a big bucket called Error and auto-jumping to certain priorities based on those two results is probably getting towards the end of it's useful life as people get more sophisticated with their dialing plans and error control. Perhaps a method of selectably moving to a new method of error handling is in order. Instead of building a million little exception cases, why not hand some values back to the dialplan logic and let the person building the IVR create their own GotoIf tree? See my post on the topic: http://lists.digium.com/pipermail/asterisk-users/2003-April/009797.html JT There's not really a way to do that that right now, although we could add something like AST_CONTROL_INUSE which could represent that the channel is in use actually. Wouldn't be extremely difficult to do, but would INUSE and BUSY be the same? If not, where do we jump to? Mark On Wed, 11 Jun 2003, Derek Beaumont wrote: Is it possible to have both a busy and an away message when the call waiting feature is enabled? extensions.conf ... exten=403,1,Dial,Zap/3|10 exten=403,2,Voicemail2,u403 exten=403,103,Voicemail2,b403 ... Because I have enabled call waiting, I can't see how it will be possible to get the busy message to play (because there will always be a dial tone). Am I right, or do I have incorrect configurations? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Colorado Asterisk Users
Just a quick query to find out if there are any other Asterisk users in Colorado. If you're out there, drop me a line off-list. I'd like to start a user group if there is anyone else out there. --Karl -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over IAX
On Thu, 2003-04-03 at 08:56, Jon Pounder wrote: I think the real solution is some piggy backed protocol that can be told this is fax information at one end, digitize the fax as if it were a faxmodem, stream it to the other end using a non-realtime protocol, and then initiate a fax call at the other end and restream out the data, all while possibly holding open the original call to indicate reception confirmation at the end. Well there's T.30 which is a store and forward mechanism, or T.38 which is a realtime fax relay but they cost $ for licensing of the protocols. --Karl At 10:20 AM 4/3/2003 -0500, you wrote: From what I have heard packetizing fax does not work well, does not matter if it is IAX or SIP. I think that was straight from digium tech support. On Wednesday, April 2, 2003, at 09:53 AM, John Harragin wrote: Hi, We are looking at consolidating our lines with PRI. This will allow the elimination of many fax lines. Some of them will be replaced with this type of config ... PRI * IAX * Channel-Bank FAX We will have daggressor suppressor enabled. Is anyone doing this and should I expect smooth operation? John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-798-9106 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017 ___ 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 -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dlink DG-104S
On Fri, 2003-03-28 at 02:44, [EMAIL PROTECTED] wrote: Sort of. There are couple of things broken if you want to make calls from MGCP to SIP, and some things need to be implemented (hook flash in MGCP), retransmission (or at least timing out) of MGCP messages. I have preliminary patch which is not correct by any means but at least makes it usable. I've got two dg-104s on the way. Can you send me your patch I can help get it working once they arrive. --Karl On Fri, 28 Mar 2003, Brian Capouch wrote: Does anyone know if this unit works with Asterisk? Thx. B. ___ 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 -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP Gui for Asterisk (AGI questions)
On Tue, 2003-03-18 at 00:10, Chris Albertson wrote:
The best thing might be to seporate the two types of data.
The simple thing to do is use a preprocessor like M4.
Defin a macro for a type-D phone and then have lines like
extn-type_D(6578)
extn-type_D(6579)
Then to change the flw of control for all 100 extension you
only have to change the macro definition. Still you are keeping
two diferent kinds of data in the same file but at least you've
factored out the redundent information
extensions.conf now has powerful variables and macros for the last couple of weeks.
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the interface,
20 seconds maximum
exten = s,2,Voicemail(u${ARG1}); If unavailable, send
to voicemail w/ unavail announce
exten = s,3,Goto(default,s,1) ; If they press#,
return to start
exten = s,102,Voicemail(b${ARG1}) ; If busy, sendto
voicemail w/ busy announce
exten = s,103,Goto(default,s,1); If they press#,
return to start
exten = 1234,2,Macro(stdexten,1234,${CONSOLE})
;exten = 6275,Macro(stdexten,6275,${MARK})
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Re: [Asterisk-Users] Proposed IAX2 Name
What about ITP Internet/IP Telephony Protocol On Thu, 2003-03-13 at 09:40, Mark Spencer wrote: LIghtweight Voice over IP Exchange Or: Lightweight Internet Voice Exchange Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: PRI costs in US
On Wed, 2003-03-05 at 08:33, Steven Critchfield wrote: What I learned from home, you do not want X running at all on your phone system. The VoIP quality on my home system took a severe drop when my screensaver would activate. It would also clear as soon as the screen saver would go away. This was on a 1ghz AMD T'bird with 256 megs of ram. The X screen savers aren't exactly cpu friendly. Set the screensaver to just blank screen then X doesn't eat cpu while your machine is idle. -- Karl Putland [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
