Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..
Diego Iastrubni wrote: I am seeing on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 (repeated much more then what I will show here). I see that it comes from static void* do_monitor(void *data) in chan_zap.c, but I do not understand what does it mean, and now why is it spamming my logs. Because you are logging debug messages :-P ---cut--- ; Debug mode turns on a LOT of extra messages, ; most of which you are unlikely to understand without an understanding of ; the underlying code. Do NOT report debug messages as code issues, unless ; you have a specific issue that you are attempting to debug. They are ; messages for just that -- debugging -- and do not rise to the level of ; something that merit your attention as an Asterisk administrator. Debug ; messages are also very verbose and can and do fill up logfiles quickly; ; this is another reason not to have debug mode on a production system unless ; you are in the process of debugging a specific issue. ---cut--- So if there is no problem with your system just don't enable debug mode. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method for scripting options specified in make menuconfig
On Wed, Aug 08, 2007 at 09:52:51AM -0400, James FitzGibbon wrote: On 8/8/07, arkda [EMAIL PROTECTED] wrote: I've been digging around and I haven't found a way to do this, but I have a feeling I'll feel like an idiot because it's something I'm over looking. Normally if I need to specify an additional option (such as different language sound files) or I'm building an Asterisk server with a lean configuration and need to remove some modules I do so with 'make menuconfig'. I've ran into a need however to install Asterisk entirely from the command line, so I'm looking for the method of accomplishing what I've normally done through 'make menuconfig' solely from the command line. Anyone know how this is accomplished? After you run make menuselect, you'll have a file 'menuselect.makeopts' in your asterisk source dir. Copy that to /etc/asterisk.makeopts (or ~/.asterisk.makeopts) and it will be used for future builds. Once you've copied the file over, do a 'make distclean ; ./configure ; make' to check that it worked. Hmmm why distclean ? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on the Monitor command on AMI
Try MixMonitor() l. In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu [EMAIL PROTECTED] ha scritto: Hi all, Is there a way to have this command to record a mixed file instead of one for in and one for out? I have set the Mix parameter to 1, but it is still generating two files. I would prefer it to have the in and out files mixed. Thnx. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..
On Wed, Aug 08, 2007 at 12:56:33PM +0300, Diego Iastrubni wrote: Hi all, I am seeing on my logs this message: Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 Jun 13 09:14:51 DEBUG[4944] chan_zap.c: Monitor doohicky got event Event 160 on channel 3 It means that either: * Asterisk is drunk * Asterisk is getting strange events from the zaptel channels. The trigger here is an even from Zaptel. But what are those events? But where does Zaptel send those events? I don't see any similar ZT_EVENT_* value in zaptel.h and I don't see any value sent through zt_qevent_lock that is not a ZT_EVENT_* . Hmmm... leftovers in the events buffer? Time for a bug report, I guess. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
Cant help you with storm issue but second problem you have is coming from bad FXO module. Replacing that module should fix it. On 8/8/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. Second problem, the card with only 1 fxo port has gone crazy, its permenantly busy, no matter if i reboot the system, even if the system is off, the line is still busy until i unplug it from the digium card. i have no idea whats making the line always busy, this just happened out of no where. again reloading modules, rebooting or even shutting down the system does not make the line un-busy until its unplugged from the card, big problem since its the only line at the location. I appreciate your help everyone. thank you. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..
On Thursday 09 August 2007 09:40, Philipp Kempgen wrote: Because you are logging debug messages :-P ---cut--- ; Debug mode turns on a LOT of extra messages, ; most of which you are unlikely to understand without an understanding of ; the underlying code. Do NOT report debug messages as code issues, unless ; you have a specific issue that you are attempting to debug. They are ; messages for just that -- debugging -- and do not rise to the level of ; something that merit your attention as an Asterisk administrator. Debug ; messages are also very verbose and can and do fill up logfiles quickly; ; this is another reason not to have debug mode on a production system unless ; you are in the process of debugging a specific issue. ---cut--- And that is my question, what is that message. My first assumption is that debug messages are for debugging - which means something can be wrong , and those messages will help me. As a develop (he wrong list...?) I would like to know what help can I gain from those messages. So if there is no problem with your system just don't enable debug mode. See Tzafrir's reply, that would explain much more. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] a couple of new tutorials
Hello list, I posted a couple of tutorials lately, maybe someone can benefit from them: The first tutorial explains how to transform your Asterisk call recordings (in WAV or GSM) to lo-fi MP3 to save a lot of space. It's actually pretty easy to implement using a makefile. http://astrecipes.net/index.php?n=294 The other tutorial lets you implement a way to monitor all outgoing traffic for a set of extensions - what is nice is that it's pretty easy to implement and you can decide which extensions are monitored through the asterisck CLI without touching the extensions.conf files. http://astrecipes.net/index.php?n=293 Any comment is welcome. And feel free to vote them up on AstPligg http://oinko.net/astpligg :-) l. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a couple of new tutorials
On Thu, Aug 09, 2007 at 09:41:44AM +0200, lenz wrote: Hello list, I posted a couple of tutorials lately, maybe someone can benefit from them: The first tutorial explains how to transform your Asterisk call recordings (in WAV or GSM) to lo-fi MP3 to save a lot of space. It's actually pretty easy to implement using a makefile. http://astrecipes.net/index.php?n=294 Is mp3 better than gsm (with regards to compression ratio)? Converting gsm to mp3 doesn't sound like a good idea to me (sorry for the pun). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..
Diego Iastrubni wrote: On Thursday 09 August 2007 09:40, Philipp Kempgen wrote: So if there is no problem with your system just don't enable debug mode. See Tzafrir's reply, that would explain much more. That message did not yet make it to me. The list still seems to have delivery problems. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] usage of each field
Hi all,
From the web, I can find a table scheme of sipusers for ARA using.
However, I can't find any meaning of each field, especially for the
field regserver which is new in the table. Can any tell me more
detail about the usage of each field?
CREATE TABLE `sip_buddies` (
`id` int(11) NOT NULL auto_increment,
`name` varchar(80) NOT NULL default '',
`host` varchar(31) NOT NULL default '',
`nat` varchar(5) NOT NULL default 'no',
`type` enum('user','peer','friend') NOT NULL default 'friend',
`accountcode` varchar(20) default NULL,
`amaflags` varchar(13) default NULL,
`callgroup` varchar(10) default NULL,
`callerid` varchar(80) default NULL,
`cancallforward` char(3) default 'yes',
`canreinvite` char(3) default 'yes',
`context` varchar(80) default NULL,
`defaultip` varchar(15) default NULL,
`dtmfmode` varchar(7) default NULL,
`fromuser` varchar(80) default NULL,
`fromdomain` varchar(80) default NULL,
`insecure` varchar(4) default NULL,
`language` char(2) default NULL,
`mailbox` varchar(50) default NULL,
`md5secret` varchar(80) default NULL,
`deny` varchar(95) default NULL,
`permit` varchar(95) default NULL,
`mask` varchar(95) default NULL,
`musiconhold` varchar(100) default NULL,
`pickupgroup` varchar(10) default NULL,
`qualify` char(3) default NULL,
`regexten` varchar(80) default NULL,
`restrictcid` char(3) default NULL,
`rtptimeout` char(3) default NULL,
`rtpholdtimeout` char(3) default NULL,
`secret` varchar(80) default NULL,
`setvar` varchar(100) default NULL,
`disallow` varchar(100) default 'all',
`allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
`fullcontact` varchar(80) NOT NULL default '',
`ipaddr` varchar(15) NOT NULL default '',
`port` smallint(5) unsigned NOT NULL default '0',
`regserver` varchar(100) default NULL,
`regseconds` int(11) NOT NULL default '0',
`username` varchar(80) NOT NULL default '',
PRIMARY KEY (`id`),
UNIQUE KEY `name` (`name`),
KEY `name_2` (`name`)
) TYPE=MyISAM ROW_FORMAT=DYNAMIC;
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Re: [asterisk-users] Macro Overlap
Mojo with Horan Company, LLC wrote: set your own mutex using astdb? It may just be atomic enough for you to get by. atomic enough - that's a nice term :-) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Overlapping Playback() with Dial()?
Hi All, Can I overlap Playback() with Dial() in a dialplan? For example, I have this scenario: A call comes in, Asterisk picks it up, does Background(enter_number), then does Playback(bulletin_message), and while the Playback() is still going, I want to execute Dial() to the target extension so it overlaps with the Playback() and the call will be bridged instantly upon completion of Playback(). Is this possible in Asterisk? I am trying to save callers long distance charges by eliminating wait time as much as possible. Thank you. Jeng - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] generating a GUID
I have a need to have a GUID (for example, bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the -]) generated in the dialplan. Is there any asterisk function that would do this ? I would prefer not to have to shell out every time a call comes in. Julian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
Please stop advertising your forums/services on every single chance u get on users list . On 08/08/07, Al Bochter [EMAIL PROTECTED] wrote: That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange warning
Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED]' I dont know what is the problem. Can somebody explain me this? Below is my client configuration. [general] bindport=9060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm context=incoming allowexternalinvites=yes register= diet:[EMAIL PROTECTED]:9060 registertimeout=10 ;(default 20 secs) registerattempts=10 ;set it to zero for infinit attempts Following is the server sip account im using for my client asterisk to register: [diet] username=diet type=friend secret=pepsi qualify=no nat=yes mailbox=12129339033 insecure=invite,port call-limit=2 host=dynamic dtmfmode=rfc2833 context=local canreinvite=no callerid=formula one 13232044055 accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
EVERYONE should be using surge protection on their incoming circuits. Doing so may fix your problems once you replace your bad FXO module. Thanks, Steve Al lists wrote: Cant help you with storm issue but second problem you have is coming from bad FXO module. Replacing that module should fix it. On 8/8/07, *Michael J. Liberatore * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. Second problem, the card with only 1 fxo port has gone crazy, its permenantly busy, no matter if i reboot the system, even if the system is off, the line is still busy until i unplug it from the digium card. i have no idea whats making the line always busy, this just happened out of no where. again reloading modules, rebooting or even shutting down the system does not make the line un-busy until its unplugged from the card, big problem since its the only line at the location. I appreciate your help everyone. thank you. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
Issue #1 may be a grounding issue. When a storm front comes through the front of the storm carrier high charges of electricity. This is normally bled off with a proper ground. Without a proper ground, it may linger on equipment causing all kinds of noise. On 8/8/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: Hi, I am having some major problems with 2 digium cards in two seperate servers they are both TDM400P cards one has 4 fxo ports and the other has 1 fxo port. First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. Second problem, the card with only 1 fxo port has gone crazy, its permenantly busy, no matter if i reboot the system, even if the system is off, the line is still busy until i unplug it from the digium card. i have no idea whats making the line always busy, this just happened out of no where. again reloading modules, rebooting or even shutting down the system does not make the line un-busy until its unplugged from the card, big problem since its the only line at the location. I appreciate your help everyone. thank you. Mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 705 DIDs for Collingwood Ontario?
Hi, Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Thanks -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FSK callerid
Am Mittwoch, den 08.08.2007, 23:55 +0900 schrieb Balgansuren Batsukh: Hello, I installed Asterisk on Dell Precision workstation and configured with sample configuration. I have two TDM400 board and one with 4xFXO and second one 4xFXS module installed. I made call to telephone line connected to FXO port and never seen callerid on those lines. I tested cidsignalling and cidstart types and all doesn't work. Just a guess. Try a Wait(2) in the dialplan before Answer()ing the line (or doing anything else). The CID might be sent in or after the first ring... so if you immediately answer the line is already up and no CID can be read from it. BR Anselm ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Allison Smith?
Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pick sip channel whn two party talking
google for ASTERISK CMD CHANSPY and follow voip-info link in search results . On 08/08/07, satish patel [EMAIL PROTECTED] wrote: Dear all i need this feature in asterisk whn 2 party calling that time i pickup call and listen conversation of that party spoofing like is it possible in asterisk Rgds satish patel -- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool.http://us.rd.yahoo.com/evt=48518/*http://autos.yahoo.com/carfinder/;_ylc=X3oDMTE3NWsyMDd2BF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsDY2FyLWZpbmRlcg--+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 330 Speakerphone
Anyone who has experience with the Polycom 330 know if the speakerphone is loud enough to be heard in a 20 foot x 20 foot room? The context is a classroom where announcements will need to be made. The phone will be wall mounted at the front of classroom. Thanks, Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Level3 WIreless
Does anyone have any idea why Level3 refuses to port in wireless numbers? I know on several occasions we've had wireless port-ins fail. Does anyone know the reasoning on Level3's end? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
Ok, Maybe I read it wrong. When you go to the digium website, it no longer goes to thevoice.digium.com. In fact it says you can get credit for anything still outstanding... I did see on Allison's website, that she is still a Digium partner. Confusion, I guess... I didn't see any easy link from Digium's website to Allison, or a way to purchase IVR from her... It looked like the partnership had been severed. On 8/9/07, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Order of matching SIP packet to sections in sip.conf
Hi, When Asterisk receives SIP INVITE packets, it tries to match the packet to a section on sip.conf, so that it can know what context of the dialplan should be used, what codec's are allowed, etc. (what else does it do here?) I would like to know what is exactly the order for this matching considering Asterisk 1.4. I guess it's something like this: 1. It tries to find type=peer sections where the host=... setting is the same as the Host: header on the SIP packet. 2. It tries to find type=user sections where the username or the thing in [...] is the same as the authenticated username on the SIP packet. 3. It tries to find domain=... on the [default] section, where the configured domain is the same as the @... part on the To: header on the SIP packet. I guess it's more or less like this, but I'm not certain of the details... could someone tell me exactly how it's done? If you can point me to the code that does it, it would be fine. Also, regarding authentication, as far as I know, usually SIP INVITE packets are sent unauthenticated, then Asterisk will reply with an 403 Proxy Auth Required, and then the original UAC will retry, now sending a SIP INVITE packet with authentication information. Where, in the algorythm above, will Asterisk know that the user should authenticate and issue the 403 response? Thanks, Filipe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 705 DIDs for Collingwood Ontario?
On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote: Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I have the same issue with the ringing currently, so I force a ring. Stephen Bosch wrote: Wes Baehr wrote: I had a lot of problems with garbled IAX calls (even when calling into just the IVR). The problem was compacted when I would bridge an incoming IAX call to an outgoing SIP call, though that may be a fault of Asterisk. Since using SIP everything has been working perfectly. I never had any real problems with dropping calls (that weren’t on my end). However, I don’t use IAX anymore, so I am not aware of any current issues. This is interesting information -- I've had similar problems with IAX trunks on totally different carriers. Example: Callers do not hear the remote ringing, or only some of the rings, or don't hear the beep tone for voice mail. IAX is easier if you're behind a firewall :( -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call forward at telco
Hello, I want to enable call forwarding at my telco. In Germany you can press *21*destination# and all calls will be redirected to the destination without interaction with any equipment on my side. How to dial this with Asterisk and Zap-Channels? It can not be send as called number, it has to be send as keypad facility. Anyone here with some hints? The application ZapSendKeypadFacility in Asterisk 1.4 only supports answered channels if I read it correctly. But my channel is not answered before sending *21*destination# (I get a voice telling me the call forwarding is activated). Thanks, Gunnar ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring Jitter in Asterisk
I have used this freeware tool in the past: http://sineapps.com/sinestatiax.php maybe you can have a look at it as well l. In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED] ha scritto: At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote: At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. You could look at the txjitter and rxjitter values (and other values) stored in the CHANNEL() function, and those values you're looking for were previously known as RTPAUDIOQOS. Or is this not sufficient? Are txjitter and rxjitter working reliably? These calls are going to be placed from AMI and bridged together. Do you think the variables would be correctly set for each leg of the call? Doug. I think the best way to determine this would be to compare the numbers provided by CHANNEL() versus the numbers provided by something with a little more reliability, such as wireshark, in a controlled set of circumstances. Please post your results here - it would be an interesting test. JT ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a couple of new tutorials
It depends - I believe mp3 8k has about the (poor) quality of gsm, but takes about half the disk space yes, I would not routinarily save in gsm and then turn it to mp3, but I developed a script for some guys who had a lot of existing gsm files they wanted transcoded :-) l. In data Thu, 09 Aug 2007 09:56:42 +0200, Tzafrir Cohen [EMAIL PROTECTED] ha scritto: Is mp3 better than gsm (with regards to compression ratio)? Converting gsm to mp3 doesn't sound like a good idea to me (sorry for the pun). -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
On 8/8/07, Michael J. Liberatore [EMAIL PROTECTED] wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. I have experienced that same state randomly on one phone connected to a TDM400 with three FXS. It happens one one particular phone and changing the phone to a different module doesn't help. I can only assume the phone has some characteristic that the FXS doesn't like. Only an unload-reload of zaptel cures the problem. Note that unplugging or plugging in a phone will often result in this condition as well. /r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
I have been a VP connect customer for a few years, mow traffic, outgoing only. I have had very good experiences and they are usually the lowest cost for a USA route, often less than .01/min retail. /r On 8/8/07, John Meksavan [EMAIL PROTECTED] wrote: Has anybody use Voicepulse Connect for Asterisk? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] generating a GUID
On 8/9/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I have a need to have a GUID (for example,
bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the
-]) generated in the dialplan. Is there any asterisk function that
would do this ? I would prefer not to have to shell out every time a
call comes in.
There's nothing built in that I know of. I had mused with the idea of
wrapping the available UUID generator code out there into a function and
offering it as a patch, but it's a low priority thing for me.
In the meantime, you could achieve what you want without the cost of
spinning up a shell process by writing a FastAGI app in Perl. Using the
modules Asterisk::FastAGI and Data::UUID, you could get a UUID back for the
cost of the socket connection.
This is a quick example that I coded up to do that - it was actually more
painful to install the modules from CPAN than code up the server itself:
--START--
#!/usr/bin/perl
#
use strict;
use warnings;
MyAGI-run( port = 4574 );
package MyAGI;
use base 'Asterisk::FastAGI';
use strict;
use Data::UUID;
my $uuid;
sub child_init_hook
{
$uuid = Data::UUID-new;
}
sub fastagi_handler
{
my $self = shift;
$self-agi-set_variable( UUID = $uuid-create_str() );
}
---END---
When run, this creates a pre-forking server with 5 children, which makes the
individual UUID generation about as cheap as you're going to get going
outside of the Asterisk process. When I execute that with agi debugging
turned on from this diaplan snippet:
exten = 7993,1,Answer
exten = 7993,n,AGI(agi://127.0.0.1:4574/fastagi_handler)
exten = 7993,n,SayAlpha(${UUID})
exten = 7993,n,Hangup
I get this:
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/427-9df490e0, )
in new stack
-- Executing [EMAIL PROTECTED]:2] AGI(SIP/427-9df490e0,
agi://127.0.0.1:4574/fastagi_handler) in new stack
AGI Tx agi_network: yes
AGI Tx agi_network_script: fastagi_handler
AGI Tx agi_request: agi://127.0.0.1:4574/fastagi_handler
AGI Tx agi_channel: SIP/427-9df490e0
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1186667018.723
AGI Tx agi_callerid: 427
AGI Tx agi_calleridname: James FitzGibbon
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 7993
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from-internal-admin
AGI Tx agi_extension: 7993
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx CLI
AGI Rx SET VARIABLE UUID 88AEDB9A-467E-11DC-9F13-8E31D47CEF85
AGI Tx 200 result=1
-- AGI Script agi://127.0.0.1:4574/fastagi_handler completed, returning
0
-- Executing [EMAIL PROTECTED]:3] SayAlpha(SIP/427-9df490e0,
88AEDB9A-467E-11DC-9F13-8E31D47CEF85) in new stack
And then Allison starts chattering out the digits of the UUID.
Hope that gives you something to work with.
--
j.
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[asterisk-users] Friday Aug 10 @ 12:30 PM EDT - Asterisk Users Conference
This week, the second part of connecting to the outside world using TDM, ATA and even... IAX hardphones with compilable software. More on topics and guests: http://groups.google.com/group/asterisk-users-conference Instructions: http://www.AsteriskUsersConference.org IRC on freenode.net: #asterisk-users-conference /r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfer/conference
Hi All- I have an asterisk server and GXP2000. If I want to send a call to someone else (external), I can transfer the call where I can announce it, and then send it over. But what I'd like is to start a 3-way conference, and then drop out. But if I do a conference button on the phone, and then drop out, the other two are not left to finish their conversation (the call is ended). Should I be using the MeetMe application instead or is there a different way to fix this problem? thanks Todd ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which spandsp unicall version to use with 1.2?
On Tue, 2007-08-07 at 17:15 +0300, Tzafrir Cohen wrote: On Tue, Aug 07, 2007 at 04:01:54PM +0200, Patrick wrote: Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/ Nither. Use spandsp 0.0.3 for asterisk 1.2 . Thanks for the feedback Tzafrir. Steve answered also and mentioned that for * 1.2 I should use spandsp 0.0.2 so I will go with that one. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which spandsp unicall version to use with 1.2?
On Wed, 2007-08-08 at 22:30 +0800, Steve Underwood wrote: Patrick wrote: Hi all, Anyone have an idea which version of spandsp, libunicall, libmfcr2, libsupertone, app_rxfax/app_txfax and chan_unicall I should use for the latest asterisk 1.2? Would that be the ones listed below? http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre4.tgz http://www.soft-switch.org/downloads/snapshots/spandsp/test-apps-asterisk-1.2/ http://www.soft-switch.org/downloads/unicall/unicall-0.0.5pre1/ http://www.soft-switch.org/downloads/snapshots/unicall/asterisk-1.2.x-20060205/ For * 1.2 use: spandsp-0.0.2 and the apps that accompany it. unicall-0.0.3pre11 and the chan_unicall that accompanies it. Thanks Steve. Most helpful. Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging Application - Polycom 601
Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. I would be curious to see how you set up the phones to accept paging, just to make sure there isn't something iffy with your phone configuration. I'm trying to get more info on how the phones are set up. This is a commercial product and I have a web GUI to enable/disable paging. What that actually does? I don't know yet. I'll find out. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. Is the 601 calling the page, or receiving a page from another phone? The 601 is Calling the page. (601 is Receptionist) This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. You feel like it? Do you know for sure? OK, now I know for sure... Had the 601 crash again this morning and I used your help in see the meetme info. This is roughly 20 minutes after the 601 crashed... Conf Num PartiesMarked Activity Creation 1913938683d0006 0001 00:19:07 Dynamic * Total number of MeetMe users: 6 User #: 01 9403225392 Reception Channel: SIP/7110-b2e11758 (unmonitored) User #: 03 7137 no name Channel: SIP/7137-b2f63e80 (Listen only) (unmonitored) User #: 05 7129 no name Channel: SIP/7129-b2ca1c78 (Listen only) (unmonitored) User #: 09 7121 no name Channel: SIP/7121-b2c6a0e0 (Listen only) (unmonitored) User #: 15 7117 no name Channel: SIP/7117-0855e960 (Listen only) (unmonitored) User #: 20 7136 no name Channel: SIP/7136-b2f09b58 (Listen only) (unmonitored) 6 users in that conference. If the phone does not show an active call, it's not connected to anything. I don't see how it would be in a conference after a reboot. Your problems below are probably caused by something else. The spontaneous reboot is telling. I appears it is still in the conference, even after reboot. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. The 601s with sidecars have been problematic. I'm finding that out the hard way! What Polycom firmware are you using? 1.6.7.0098 1) Has anyone ever seen this problem? Other users have reported problems with 601s crashing. Check your firmware. AFAIK, the current firmware is 2.1.3. My vendor tried to move to a 2.x firmware, but it had a real bad delay when reading keys. It would miss about ever 3rd or 4th key you pressed. Sometimes, the keys would stick and you'd hear the touchtone for 10 seconds or so. They had me move back to 1.6.7 and it all went away... 2) Is there a way from the CLI to show and kill a page? 'show channels' will show you active calls (in 1.2; in 1.4, use 'core show channels') 'meetme kick' lets you kick channels/users from a conference. Thanks. Helped alot. Still, I don't think that's what's happening here. I'm no so sure. The one way audio seems to show it's face within an hour or so after a page that crashes the 601. I kicked everyone off the meetme this time within 20 minutes and it's been 2 hours now. No one way audio yet... Thanks for the help. Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending reset all channels signals to my system, to which he's getting an establish remote response from my asterisk box. I've been running a packet dump (wanpipemon -i w1g1 -c trd) of my d-channel this whole time and have yet to see a single incoming packet. I believe I *should* be seeing an incoming packet when he sends the reset, correct? Is there any way to do a completely raw dump of the d-channel? Thanks to everyone who offered suggestions on how to troubleshoot this issue. After working with the telco for over a week on this, I finally got them to admit today that they have a configuration problem. I had been telling this since day 1, but they didn't listen to me. Their change in perspective came when they had a tech come on-site with a PRI emulator device. He connected that directly to my asterisk server and was able to make calls with no issues whatsoever. Fortunately after this final test, they admitted that the problem must be on their end. Hopefully they'll get it sorted today. As an aside, I had a quick question regarding smartjacks. Is there a jumper or something on the smartjack itself to change from an old-style EM T1 to a PRI? I'd think that change would happen in the telco's switch, but I just thought it might be a possibility. In my case, as I stated in my original email, the bchannels come up fine, but not the dchannel. This makes me think it could be something simple... -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
So I see: /*Note:* The site, TheVoice.digium.com and its credit system for purchasing voice prompts, has been discontinued. For customers who have outstanding credits through the site, please contact Customer Service http://www.digium.com/en/company/contact.php to receive a refund. / To me, that indicates less a cessation of contract with Digium/Allison and more a modification of the way things are handled. But who knows. N. Matt wrote: She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call forward at telco
On Thu, 9 Aug 2007, Gunnar Schaller wrote:
Hello,
I want to enable call forwarding at my telco. In Germany you can press
*21*destination# and all calls will be redirected to the destination
without interaction with any equipment on my side.
How to dial this with Asterisk and Zap-Channels? It can not be send as
called number, it has to be send as keypad facility.
Anyone here with some hints? The application ZapSendKeypadFacility in
Asterisk 1.4 only supports answered channels if I read it correctly.
But my channel is not answered before sending *21*destination# (I get
a voice telling me the call forwarding is activated).
This doesn't work?
exten = _*21*X.,1,Dial(Zap/1/*21*${EXTEN:4})
Then you can dial
*21*destination#
then just push 'send' on your SIP phone and the system will dial it out
for you... ??
or am I missing something...
Gordon
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Re: [asterisk-users] Overlapping Playback() with Dial()?
Jeng Yu wrote: Hi All, Can I overlap Playback() with Dial() in a dialplan? For example, I have this scenario: A call comes in, Asterisk picks it up, does Background(enter_number), then does Playback(bulletin_message), and while the Playback() is still going, I want to execute Dial() to the target extension so it overlaps with the Playback() and the call will be bridged instantly upon completion of Playback(). Is this possible in Asterisk? I am trying to save callers long distance charges by eliminating wait time as much as possible. Thank you. Jeng Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just execute the dial after the playback. Otherwise use a queue and put the audio as the announcement. Anthony ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] usage of each field
Rilawich Ango wrote:
Hi all,
From the web, I can find a table scheme of sipusers for ARA using.
However, I can't find any meaning of each field, especially for the
field regserver which is new in the table. Can any tell me more
detail about the usage of each field?
CREATE TABLE `sip_buddies` (
`id` int(11) NOT NULL auto_increment,
`name` varchar(80) NOT NULL default '',
`host` varchar(31) NOT NULL default '',
`nat` varchar(5) NOT NULL default 'no',
`type` enum('user','peer','friend') NOT NULL default 'friend',
`accountcode` varchar(20) default NULL,
`amaflags` varchar(13) default NULL,
`callgroup` varchar(10) default NULL,
`callerid` varchar(80) default NULL,
`cancallforward` char(3) default 'yes',
`canreinvite` char(3) default 'yes',
`context` varchar(80) default NULL,
`defaultip` varchar(15) default NULL,
`dtmfmode` varchar(7) default NULL,
`fromuser` varchar(80) default NULL,
`fromdomain` varchar(80) default NULL,
`insecure` varchar(4) default NULL,
`language` char(2) default NULL,
`mailbox` varchar(50) default NULL,
`md5secret` varchar(80) default NULL,
`deny` varchar(95) default NULL,
`permit` varchar(95) default NULL,
`mask` varchar(95) default NULL,
`musiconhold` varchar(100) default NULL,
`pickupgroup` varchar(10) default NULL,
`qualify` char(3) default NULL,
`regexten` varchar(80) default NULL,
`restrictcid` char(3) default NULL,
`rtptimeout` char(3) default NULL,
`rtpholdtimeout` char(3) default NULL,
`secret` varchar(80) default NULL,
`setvar` varchar(100) default NULL,
`disallow` varchar(100) default 'all',
`allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw',
`fullcontact` varchar(80) NOT NULL default '',
`ipaddr` varchar(15) NOT NULL default '',
`port` smallint(5) unsigned NOT NULL default '0',
`regserver` varchar(100) default NULL,
`regseconds` int(11) NOT NULL default '0',
`username` varchar(80) NOT NULL default '',
PRIMARY KEY (`id`),
UNIQUE KEY `name` (`name`),
KEY `name_2` (`name`)
) TYPE=MyISAM ROW_FORMAT=DYNAMIC;
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Here you go!
http://www.voip-info.org/wiki-Asterisk+config+sip.conf
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Re: [asterisk-users] Allison Smith?
linky http://www.digium.com/en/products/voice/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, August 09, 2007 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Allison Smith? She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging Application - Polycom 601
About your keys sticking...I had the same issue when I moved my 501s up to 2.x, and after a lot of fiddling around I realized that the problem was my register timeout (my phones would register every 30 seconds) which overloaded the phones CPU, resulting in what appeared to be sticky keys. I simply changed the register attempts to a longer delay (I actually think I removed them completely to be honest). I used reregister for NAT traversal, but in 2.x there is a NAT keepalive functionality, which has been working fine for me. It might be worth trying that out, it would allow you to move to firmware 2.x and get whatever benefits you can get from that. Regards, Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Andersen Sent: Thursday, August 09, 2007 10:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging Application - Polycom 601 Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a Page. From what I understand, the Page is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. I would be curious to see how you set up the phones to accept paging, just to make sure there isn't something iffy with your phone configuration. I'm trying to get more info on how the phones are set up. This is a commercial product and I have a web GUI to enable/disable paging. What that actually does? I don't know yet. I'll find out. The problem I am having is about 1 out of 25 pages will crash the Polycom 601 (receptionist) and the phone will reboot. Is the 601 calling the page, or receiving a page from another phone? The 601 is Calling the page. (601 is Receptionist) This leaves all the extensions in the conference room and each party must hit end call on their phone to get out of the conference. However, the receptionist can't do that because that phone restarts. Once it has rebooted, it does not show to be connected to the conference room. However, I feel like it is still in the conference - with no way out. You feel like it? Do you know for sure? OK, now I know for sure... Had the 601 crash again this morning and I used your help in see the meetme info. This is roughly 20 minutes after the 601 crashed... Conf Num PartiesMarked Activity Creation 1913938683d0006 0001 00:19:07 Dynamic * Total number of MeetMe users: 6 User #: 01 9403225392 Reception Channel: SIP/7110-b2e11758 (unmonitored) User #: 03 7137 no name Channel: SIP/7137-b2f63e80 (Listen only) (unmonitored) User #: 05 7129 no name Channel: SIP/7129-b2ca1c78 (Listen only) (unmonitored) User #: 09 7121 no name Channel: SIP/7121-b2c6a0e0 (Listen only) (unmonitored) User #: 15 7117 no name Channel: SIP/7117-0855e960 (Listen only) (unmonitored) User #: 20 7136 no name Channel: SIP/7136-b2f09b58 (Listen only) (unmonitored) 6 users in that conference. If the phone does not show an active call, it's not connected to anything. I don't see how it would be in a conference after a reboot. Your problems below are probably caused by something else. The spontaneous reboot is telling. I appears it is still in the conference, even after reboot. After one of these crashes, the 601 phone will start having one way audio (can't hear caller), various other weirdness (side car status wrong) and the only way to completely correct the problems are to restart asterisk - which I assume kills the rogue page application. The 601s with sidecars have been problematic. I'm finding that out the hard way! What Polycom firmware are you using? 1.6.7.0098 1) Has anyone ever seen this problem? Other users have reported problems with 601s crashing. Check your firmware. AFAIK, the current firmware is 2.1.3. My vendor tried to move to a 2.x firmware, but it had a real bad delay when reading keys. It would miss about ever 3rd or 4th key you pressed. Sometimes, the keys would stick and you'd hear the touchtone for 10 seconds or so. They had me move back to 1.6.7 and it all went away... 2) Is there a way from the CLI to show and kill a page? 'show channels' will show you active calls (in 1.2; in 1.4, use 'core show channels') 'meetme kick' lets you kick channels/users from a conference. Thanks. Helped alot. Still, I don't think that's what's happening here. I'm no so sure. The one way audio seems to show it's face within an hour or so after a page that crashes the 601. I kicked everyone off the meetme this time within 20 minutes and it's been 2 hours now. No one way audio yet... Thanks for the help. Bill
[asterisk-users] Asterisk Help
Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Method for scripting options specified in make menuconfig
On 8/9/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: After you run make menuselect, you'll have a file 'menuselect.makeopts' in your asterisk source dir. Copy that to /etc/asterisk.makeopts (or ~/.asterisk.makeopts) and it will be used for future builds. Once you've copied the file over, do a 'make distclean ; ./configure ; make' to check that it worked. Hmmm why distclean ? 'clean' doesn't remove the generated menuselect.makeopts: clean: $(SUBDIRS_CLEAN) rm -f defaults.h rm -f include/asterisk/build.h rm -f include/asterisk/version.h @$(MAKE) -C menuselect clean cp -f .cleancount .lastclean distclean: clean @$(MAKE) -C menuselect dist-clean @$(MAKE) -C sounds dist-clean rm -f menuselect.makeopts makeopts menuselect-tree menuselect.makedeps rm -f makeopts.embed_rules rm -f config.log config.status rm -rf autom4te.cache rm -f include/asterisk/autoconfig.h rm -f include/asterisk/buildopts.h rm -rf doc/api rm -f build_tools/menuselect-deps So if you go through this cycle: untar ./configure make menuselect ...make module choices... cp menuselect.makeopts /etc/asterisk.makeopts make clean ./configure Then the automated run of menuselect is going to have two makeopts files that it might pull from: the generated one left over from the first run of configure, and the one in /etc. But since the files should be identical, you won't be absolutely sure that your file in /etc is the one driving the module choices. If you changed cp menuselect.makeopts... to mv menuselect.makeopts... in the above snippet, then I suppose 'make clean' would suffice. But 'make distclean' doesn't do any harm - it should return the directory to it's post-untar state, right? -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
On Thu, 9 Aug 2007, John Meksavan wrote: Is my network to slow? Or is there something else going on? It sounds like there may have been a temporary period of high utilisation that created this distortion. However, in general, the Asterisk 'qualify' mechanism (which consists of a SIP OPTIONS ping, nothing more) is not a terribly reliable metric of network throughput or performance, especially when constrained to extremely low latency. Nor does Asterisk always accurately determine that a peer is UNREACHABLE only when it is, in fact, UNREACHABLE. I encourage you to set the qualify= value for each peer in sip.conf very high, or otherwise omit the setting entirely. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
On 8/9/07, Matt [EMAIL PROTECTED] wrote: She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. But if you click Record Prompt it lets you enter the text to have Allison record. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging Application - Polycom 601
Bill, I've had that problem, too. It was caused by too frequent of a registration and something goofy in the Polycom software. 2.1.2 (the latest) does not have this problem and I would definitely suggest moving to it. It is doubtful that you need that high of a registration period, anyway. Is 3600 seconds too high for you? Do the phones move? I have mine set to 90 seconds to allow for external failover of their internet connection. Kevin Bill Andersen wrote: What Polycom firmware are you using? 1.6.7.0098 1) Has anyone ever seen this problem? Other users have reported problems with 601s crashing. Check your firmware. AFAIK, the current firmware is 2.1.3. My vendor tried to move to a 2.x firmware, but it had a real bad delay when reading keys. It would miss about ever 3rd or 4th key you pressed. Sometimes, the keys would stick and you'd hear the touchtone for 10 seconds or so. They had me move back to 1.6.7 and it all went away... ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The quest for making hint more flexible continues - using Realtime now
Ok, now that I've learned I cannot use any variables when using the `hint`
priority (for BLF), I figured I'd try to use the next best thing: hardcoded
values using realtime. This way I avoid variables such as ${ACCOUNTCODE}
but I can at least change the DB more easily than text files. This is the
appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata |
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 | |
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from xx.xxx.xx.xx, but there is
no hint for that extension
Wellthere is! Is there any way I can do this?
Mike
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Re: [asterisk-users] Allison Smith?
More specifically: https://www.digium.com/en/wheretobuy/digiumdirect/voice_prompt.php On 8/9/07, Cory Andrews [EMAIL PROTECTED] wrote: linky http://www.digium.com/en/products/voice/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, August 09, 2007 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Allison Smith? She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Terrible clicking on T1
Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. We've been through tech support with Sangoma and we are confident it isn't the Sangoma card. We've had the TMDI card replaced in the Nortel and we still have the problem. Pure IP calling on the Asterisk box works fine so it isn't between the phones and Asterisk. I'm now completely out of ideas and I'm looking for some direction to go here. Does anybody have any ideas? I desperately need some help. TIA, Jason Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-05-08 22:33:23 UTC # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:14 bus:0 span: 1] span=1,0,0,esf,b8zs bchan=1-23 dchan=24 ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] generating a GUID
James FitzGibbon wrote: On 8/9/07, *Julian Lyndon-Smith* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a need to have a GUID (for example, bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the -]) generated in the dialplan. Is there any asterisk function that would do this ? I would prefer not to have to shell out every time a call comes in. There's nothing built in that I know of. I had mused with the idea of wrapping the available UUID generator code out there into a function and offering it as a patch, but it's a low priority thing for me. In the meantime, you could achieve what you want without the cost of spinning up a shell process by writing a FastAGI app in Perl. Using the modules Asterisk::FastAGI and Data::UUID, you could get a UUID back for the cost of the socket connection. This is a quick example that I coded up to do that - it was actually more painful to install the modules from CPAN than code up the server itself: [snip] Wow! That is way, way beyond what I asked for - many thanks indeed. I'll start playing with this. Thanks again Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
I have the same debian and asterisk version combo running in more than one location. Some are T1 and some are in data centers. There have been times when I got such messages and some simple ping/traceroute testing showed obvious problems at my end or the provider end. Problems at the provider end were confirmed by testing from multiple locations. John Meksavan wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
That happened last year. I remember getting emails about the site being discontinued and the prompts being added to digiums store. On 8/9/07, SIP [EMAIL PROTECTED] wrote: So I see: /*Note:* The site, TheVoice.digium.com and its credit system for purchasing voice prompts, has been discontinued. For customers who have outstanding credits through the site, please contact Customer Service http://www.digium.com/en/company/contact.php to receive a refund. / To me, that indicates less a cessation of contract with Digium/Allison and more a modification of the way things are handled. But who knows. N. Matt wrote: She's sort of on the website... click 'Purchase and Price', then 'Buy Online', You will see there is no place to purchase it. On 8/9/07, SIP [EMAIL PROTECTED] wrote: Steve Totaro wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve http://www.digium.com/en/products/voice/ She's still on the website. N. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on the Monitor command on AMI
Is MixMonitor a Manager API command? Last I checked, it is just script application. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: Thursday, August 09, 2007 3:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question on the Monitor command on AMI Try MixMonitor() l. In data Thu, 09 Aug 2007 00:24:47 +0200, Wai Wu [EMAIL PROTECTED] ha scritto: Hi all, Is there a way to have this command to record a mixed file instead of one for in and one for out? I have set the Mix parameter to 1, but it is still generating two files. I would prefer it to have the in and out files mixed. Thnx. -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
On Wed, Aug 08, 2007 at 11:44:51PM -0400, Michael J. Liberatore wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. First, find the knob in your mailer that says send messages as HTML and turn it off, please? HTML is bad for mailing lists. Secondly, remember: this is a *phone* system now; you're hooking it up to several kilofeet of antenna. If you don't have telco-quality lightning protection and grounding on the box, you can expect this sort of thing. You can't find practices handbooks anymore (damnitall), but if you've ever looked at a professionally installed key system backboard, and seen those Porta-Systems gas-tubes, and the size of the grounding wire, then you may get an inkling of a) why you're having problems, and b) why traditional PBX's cost so much to buy and install. It's not *all* extra markup, folks. Cheers, -- jr 'hobby horse' a -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible clicking on T1
On Thu, Aug 09, 2007 at 11:39:38AM -0400, Gleim, Jason wrote: I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. My bet is clock-slip due to a fight over who's clocking the line. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 705 DIDs for Collingwood Ontario?
Its a small company with an office in Collingwood, and they were looking into getting a local VoIP number if possible. On 8/9/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote: Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) -A. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
On Thu, 9 Aug 2007, Paul wrote: I have the same debian and asterisk version combo running in more than one location. Some are T1 and some are in data centers. There have been times when I got such messages and some simple ping/traceroute testing showed obvious problems at my end or the provider end. Problems at the provider end were confirmed by testing from multiple locations. The 'mtr' command is handy here, although it can generate measurable traffic if you care to count every byte ;-) Just run mtr to the hostname of your upstream VoIP provider (netlogic?) and leave it running for a day or 2... If you don't have mtr, then: apt-get install mtr-tiny and you soon will have :) (mtr-tiny which is the text/curses version - the 'full' mtr is a GTK application, and running X applications on your asterisk box probably isn't what you want to do! Mtr *really* doesn't warrant a GUI IMO!!!) Gordon John Meksavan wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allison Smith?
Maybe the Adtran brains decided that they should bring someone on staff and stop having to split the fees? Makes sense to me. Hire someone who has multiple job descriptions including doing recordings. If there are not enough recordings to be done, he/she can do the other tasks in the job description. Thanks, Steve Matt wrote: Ok, Maybe I read it wrong. When you go to the digium website, it no longer goes to thevoice.digium.com. In fact it says you can get credit for anything still outstanding... I did see on Allison's website, that she is still a Digium partner. Confusion, I guess... I didn't see any easy link from Digium's website to Allison, or a way to purchase IVR from her... It looked like the partnership had been severed. On 8/9/07, Steve Totaro [EMAIL PROTECTED] wrote: Matt wrote: Did I miss something? I see Digium no longer contracts with Allison to record IVR prompts, was there a falling out? Where do you see that? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] low-level dump for PRI dchan debugging
On Aug 9, 2007, at 9:37 AM, Erik Anderson wrote: On 8/6/07, Erik Anderson [EMAIL PROTECTED] wrote: I've been going back and forth with my telco for several days, trying different configurations to get a new PRI to come up. The bchannels are all up and the T1 is not in alarm status. The dchannel refuses to come up however. We've tried ni2, qsig, and now dms100 for the switchtype. The telco tech I've been working with says that he's been sending reset all channels signals to my system, to which he's getting an establish remote response from my asterisk box. I've been running a packet dump (wanpipemon -i w1g1 -c trd) of my d- channel this whole time and have yet to see a single incoming packet. I believe I *should* be seeing an incoming packet when he sends the reset, correct? Is there any way to do a completely raw dump of the d-channel? Thanks to everyone who offered suggestions on how to troubleshoot this issue. After working with the telco for over a week on this, I finally got them to admit today that they have a configuration problem. I had been telling this since day 1, but they didn't listen to me. Their change in perspective came when they had a tech come on-site with a PRI emulator device. He connected that directly to my asterisk server and was able to make calls with no issues whatsoever. Fortunately after this final test, they admitted that the problem must be on their end. Hopefully they'll get it sorted today. As an aside, I had a quick question regarding smartjacks. Is there a jumper or something on the smartjack itself to change from an old-style EM T1 to a PRI? I'd think that change would happen in the telco's switch, but I just thought it might be a possibility. In my case, as I stated in my original email, the bchannels come up fine, but not the dchannel. This makes me think it could be something simple... It will be something simple, like getting a clueful tech on their end. No the smart jack has no bearing on d channel. Old style or new style the T1 is used however the gear on either end says it should be. The smart jack just passes info through itself. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The quest for making hint more flexible continues - using Realtime now
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than text files. This is the appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata |
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 | |
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
mailto:Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for
that extension
Wellthere is! Is there any way I can do this?
Mike
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I personally opened a bug in the bugtracker about this and it was closed
as wont fix. You simply cannot use the hint priority in realtime with
out a major change to the API. So until the code is changed, you are
going to have to have a separate hint context with nothing but hint
priority extensions and set the subscribe context in sip.conf for all
concerned devices to that context.
This is how I am running in production now.
Anthony
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Re: [asterisk-users] Major Digium Card Problems
Jay R. Ashworth wrote: On Wed, Aug 08, 2007 at 11:44:51PM -0400, Michael J. Liberatore wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. First, find the knob in your mailer that says send messages as HTML and turn it off, please? HTML is bad for mailing lists. Secondly, remember: this is a *phone* system now; you're hooking it up to several kilofeet of antenna. If you don't have telco-quality lightning protection and grounding on the box, you can expect this sort of thing. You can't find practices handbooks anymore (damnitall), but if you've ever looked at a professionally installed key system backboard, and seen those Porta-Systems gas-tubes, and the size of the grounding wire, then you may get an inkling of a) why you're having problems, and b) why traditional PBX's cost so much to buy and install. It's not *all* extra markup, folks. Cheers, -- jr 'hobby horse' a I was not aware that ground wire was very expensive or difficult to ground correctly. I do not see how that adds very much to the dealer's cost. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
Gordon, Thanks for tip. Using this tool mtr makes it a whole lot easier to figure what is really going on. Thanks again. Best Regards, john From: Gordon Henderson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Help Date: Thu, 9 Aug 2007 17:09:05 +0100 (BST) On Thu, 9 Aug 2007, Paul wrote: I have the same debian and asterisk version combo running in more than one location. Some are T1 and some are in data centers. There have been times when I got such messages and some simple ping/traceroute testing showed obvious problems at my end or the provider end. Problems at the provider end were confirmed by testing from multiple locations. The 'mtr' command is handy here, although it can generate measurable traffic if you care to count every byte ;-) Just run mtr to the hostname of your upstream VoIP provider (netlogic?) and leave it running for a day or 2... If you don't have mtr, then: apt-get install mtr-tiny and you soon will have :) (mtr-tiny which is the text/curses version - the 'full' mtr is a GTK application, and running X applications on your asterisk box probably isn't what you want to do! Mtr *really* doesn't warrant a GUI IMO!!!) Gordon John Meksavan wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible clicking on T1
Gleim, Jason wrote: Hey All, I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. We've been through tech support with Sangoma and we are confident it isn't the Sangoma card. We've had the TMDI card replaced in the Nortel and we still have the problem. Pure IP calling on the Asterisk box works fine so it isn't between the phones and Asterisk. I'm now completely out of ideas and I'm looking for some direction to go here. Does anybody have any ideas? I desperately need some help. TIA, Jason Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-05-08 22:33:23 UTC # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:14 bus:0 span: 1] span=1,0,0,esf,b8zs bchan=1-23 dchan=24 ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf Have you done a PRI intense debug on that span? Maybe you will find something there. Have you looked at the logs? Obviously, Asterisk/Sangoma is not the problem. What options do you have on the Nortel for the card? Is there a way to remotely connect to the Nortel at a given time and issue the disable/enable for the tdmi card (like after it does it's maintenance)? Can you define the maintenance routines and see if there is something that can be disabled that would affect the card? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible clicking on T1
Jay R. Ashworth wrote: On Thu, Aug 09, 2007 at 11:39:38AM -0400, Gleim, Jason wrote: I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. My bet is clock-slip due to a fight over who's clocking the line. Cheers, -- jra He means to set the Nortel to be the master for timing and Asterisk to be the slave. That means change: span=1,0,0,esf,b8zs to: span=1,1,0,esf,b8zs .. and whatever you have to do on the Nortel. Kevin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Terrible clicking on T1
On Thu, Aug 09, 2007 at 11:39:38AM -0400, Gleim, Jason wrote: I have an Asterisk box connected to a Nortel Option 11C via a T1. In the Asterisk box we have a Sangoma A101C and in the Nortel we have a TMDI card. The Nortel is also hooked to the PSTN via a T1 on a different NTAK09 PRI card. I've included the Zapata.conf and zaptel.conf files below. Our issue is that when a call is sent over the tie line between the two systems, the audio on the Asterisk side is terrible. There are rapid 'clicks' on it similar to when you have a cell phone close to an analog phone or a set of computer speakers. The clicks start as soon as the audio channel is opened (when I start to get rings) and it only affects the Asterisk side of the call. But, it affects both inbound and outbound audio on that side. On the Nortel side, the audio they hear is soft and distorted. On the Asterisk side, the audio they hear is full of the clicking but broken thru when the caller speaks. It's almost like the 'silence packets' are being interpreted wrong by Asterisk. If I put the Asterisk box on the T1 for the PSTN, it works perfect. The best part of all this... if we disable the TMDI card in the Nortel and then re-enable it, the audio is pristine... until the Nortel runs it's nightly maintenance routines. Then the noise is back the next day. We can always clear the problem with the disable/re-enable trick but it always come back after maintenance. My bet is clock-slip due to a fight over who's clocking the line. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 I thought that might be an issue too... and it was originally. When we started out, I had the Sangoma card generating the timing for the span but we could never get the d-channel to come up. Turns out that since we were connected to the PSTN, we had to let the Nortel set the timing on the span because it was receiving the timing from the CO. (Essentially the timing needed to 'flow' away from the CO) But, since we got that fixed and the span started working, I felt that timing wasn't the source of the problem. Plus, if we dump the error counters on both ends, they are not incrementing... even if the span is up for several days and we clearly have the audio problems. The slip counters, framing error, etc all stay at 0 and you would figure that if it was timing slip, those would be incrementing on at least one of the sides. J. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need Help in changing Voice message
Hi, Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan beep beep beep The person you are calling is unavaiable, please try again. I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no log appears in the asterisk cli command just get this message so i am not getting any idea which macro or application generating this message. Anybody have any idea about how to change this? Thanks Regards Farooq -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange warning
OK...does anybody know what this message means: -- Got SIP response 489 Bad event back from 68.209.117.205 This is displayed whenever a notify sip msg is sent to the client server. in response the client sends back the bad event message.i dont why why it does not understand the request. Can anybody explain? On 8/9/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce received from 'sip:[EMAIL PROTECTED] ' I dont know what is the problem. Can somebody explain me this? Below is my client configuration. [general] bindport=9060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm context=incoming allowexternalinvites=yes register= diet:[EMAIL PROTECTED]:9060 registertimeout=10 ;(default 20 secs) registerattempts=10 ;set it to zero for infinit attempts Following is the server sip account im using for my client asterisk to register: [diet] username=diet type=friend secret=pepsi qualify=no nat=yes mailbox=12129339033 insecure=invite,port call-limit=2 host=dynamic dtmfmode=rfc2833 context=local canreinvite=no callerid=formula one 13232044055 accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm allow=g729 -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to push callerid for each user from sip phone on one side through asterisk (Digium) to E1 card running application on other side
On Thu, Aug 09, 2007 at 11:07:50AM +0530, [EMAIL PROTECTED] wrote:
Hi,
I am running asterisk PBX ( digium TE120P card configured) on one
system. It is connected to E1 card running application on the other system.
After establishing sync between two card, I am able to place call from sip
phone to E1 card running application. I want to pass the callerid, when
calling from sip phone to E1 card running application. Which all
configuration files is to be changed in the asterisk.
I am doing the following changes in extensions.conf
exten=115,1,SET(set(CALLERID(num)=2)
exten=115,2,Dial(ZAP/g1/115,20)
So, when dialling from sip phone to extension 115
it pushes the callerid hardcoded for that extension to E1 card running
application, not for each user in sip.conf.
Can anybody tell me how to insert the callerid to each users? Which all
are the configuration files, where changes are to be made? So that, when I
call from sip phone through asterisk PBX to E1 card running application,
callerid for each user from sip phone called should be forwarded to E1
card running application side. thanks and regards sanchal
There are two ways, depending on your setup.
Easiest method if you have a straightforward SIP config, set, for each
extension in sip.conf, for example:
[1234]
callerid=Fred User 2
...
Then this callerID string should be used.
(This sounds like what you have at the moment)
If you need to change it to something different depending on the trunk being
used etc there are a variety of ways to do it, depending on how many users
etc. You could use an asteris db lookup instead before you place the call in
extensions.conf to overwrite what is set in sip.conf, to translate SIP extn
caller id to something else (in the right place in extensions.conf):
exten = 115,1,Set(CALLERID(number)=${DB(${CALLERID(num)}/callerid)}}
exten = 115,2,Dial(ZAP/g1/115,20)
Then write a db entry for each client from the CLI:
asterisk -r
asterisk*CLI database put 1234 callerid 2
Updated database successfully
asterisk*CLI database show 1234
/1234/callerid: 2
In this example a call with the callerid of 1234 would get changed to
2 for that call.
This would be a good approach if there was no apparent relationship between
1234 and 2.
There are ways to modify the callerID on the way out based on the extension,
say for example you have a callerID of 43703 and you just want to translate
that to 703 on the way out, you could extract the digits and replace.
In this example if the callerid is in the range I want, translate the
callerID to something else (in fact we just take the last three digits)
exten = 115,n,ExecIf($[$[${CALLERID(num)} = 43000] $[${CALLERID(num)}
= 43999]],Set,CALLERID(number)=${CALLERID(num):-3})
Hope one of these answers gives you some inspiration...
Rob
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[asterisk-users] LIBPRI - video calls over ISDN
Hello! I have following scenario: PBX - Asterisk - ISDN E1 line The asterisk box relays calls from the E1 to the PBX and vice versa. Additionally some outgoing calls of the PBX are being sent over VoIP providers instead of using the E1 line. I have one problem: Video calls starting from the PBX and sent over the ISDN E1 line are not working. With zaptel version 1.2.7 and libpri 1.2.3 they were working fine. I upgraded to zaptel version 1.2.19 and libpri 1.2.5 and these calls stopped working. This is the error that client displays: * H.221 negotiation timeout: Turn off and restart the system and try again.* Is this a known problem? Is there any configuration that must be done to allow video over ISDN? Is any driver alteration/recompilation needed for these calls to work? Is this merely related to libpri or do the zaptel drivers have influence in this functionality? Thanks in advance for your info, Best regards, Óscar Patrício ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable DND feature key in Polycom Phone
Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable DND feature key in Polycom Phone
Farooq Ahmed wrote: Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq Not sure there is an option other than opening the phone and removing the contacts. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible clicking on T1
On 8/9/07, Gleim, Jason [EMAIL PROTECTED] wrote: [snip] I thought that might be an issue too... and it was originally. When we started out, I had the Sangoma card generating the timing for the span but we could never get the d-channel to come up. Turns out that since we were connected to the PSTN, we had to let the Nortel set the timing on the span because it was receiving the timing from the CO. (Essentially the timing needed to 'flow' away from the CO) But, since we got that fixed and the span started working, I felt that timing wasn't the source of the problem. Plus, if we dump the error counters on both ends, they are not incrementing... even if the span is up for several days and we clearly have the audio problems. The slip counters, framing error, etc all stay at 0 and you would figure that if it was timing slip, those would be incrementing on at least one of the sides. Yes, but the audio artifact might be caused if Asterisk is using a different internal clock to your hardware card, particularly if transcoding is occuring. (This would not cause errors on the card, but would cause distortion) A standard E1/T1 install will have the Telco providing clock to your E1/T1 card, which then acts as the master clock to Asterisk. See the email from Kevin Bockman previously in this thread - I believe he has the answer. Cheers, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible clicking on T1
Gleim, Jason wrote: I thought that might be an issue too... and it was originally. When we started out, I had the Sangoma card generating the timing for the span but we could never get the d-channel to come up. Turns out that since we were connected to the PSTN, we had to let the Nortel set the timing on the span because it was receiving the timing from the CO. (Essentially the timing needed to 'flow' away from the CO) But, since we got that fixed and the span started working, I felt that timing wasn't the source of the problem. Plus, if we dump the error counters on both ends, they are not incrementing... even if the span is up for several days and we clearly have the audio problems. The slip counters, framing error, etc all stay at 0 and you would figure that if it was timing slip, those would be incrementing on at least one of the sides. Okay -- if it's not clock slip (also my first inclination): Your observation that the problem goes away after the card is disabled and re-enabled, then returns after the maintenance routine runs, is a major clue. Here's what you need to find out: -Where does the card get its configuration at start time? -What is in that configuration? -What procedures does the maintenance routine perform? In this case, you want as much detail as you can get. Talk to Nortel if necessary. As an experiment -- can you disable the maintenance routine entirely? If so, do it -- see whether the problem remains gone the following day. You want to confirm that it's that routine that is causing the change, and not something else. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Phones Call Hold Reminder function problem
I want to enable on hold reminder function on polycom 430 phones. I have enabled it in sip.cfg using this setting hold localReminder call.hold.localReminder.enabled=1 call.hold.localReminder.period=60 call.hold.localReminder.startDelay=90/ /hold But still if the call is on hold the phones does not remind about the on hold call. Any idea? Regards Farooq -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 705 DIDs for Collingwood Ontario?
Andrew Kohlsmith wrote: On Thursday 09 August 2007 8:15:09 am Zeeshan Zakaria wrote: Does anyone provide 705441XXX, 705444XXX or 705446XXX DIDs? This is for Collingwood area in Ontario. Why would anyone want a Collingwood DID? I don't answer calls from Collingwood simply because I am plain old not interested in the free vacation weekend I keep winning. :-) Are there lots of boiler rooms in Collingwood? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use OpenVPN with Asterisk
Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The quest for making hint more flexible continues - using Realtime now
I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Thursday, August 09, 2007 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more flexible
continues - using Realtime now
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than text files. This is the appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata |
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 | |
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
mailto:Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for
that extension
Wellthere is! Is there any way I can do this?
Mike
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I personally opened a bug in the bugtracker about this and it was closed as
wont fix. You simply cannot use the hint priority in realtime with out a
major change to the API. So until the code is changed, you are going to have
to have a separate hint context with nothing but hint priority extensions
and set the subscribe context in sip.conf for all concerned devices to that
context.
This is how I am running in production now.
Anthony
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Re: [asterisk-users] LIBPRI - video calls over ISDN
On 8/9/07, Oscar Patricio [EMAIL PROTECTED] wrote: Hello! I have following scenario: PBX - Asterisk - ISDN E1 line The asterisk box relays calls from the E1 to the PBX and vice versa. Additionally some outgoing calls of the PBX are being sent over VoIP providers instead of using the E1 line. I have one problem: Video calls starting from the PBX and sent over the ISDN E1 line are not working. With zaptel version 1.2.7 and libpri 1.2.3 they were working fine. I upgraded to zaptel version 1.2.19 and libpri 1.2.5 and these calls stopped working. This is the error that client displays: * H.221 negotiation timeout: Turn off and restart the system and try again.* Is this a known problem? Is there any configuration that must be done to allow video over ISDN? Is any driver alteration/recompilation needed for these calls to work? Is this merely related to libpri or do the zaptel drivers have influence in this functionality? Thanks in advance for your info, Hi, Sorry, this is not an answer to your question, but I would be REALLY interested how you ever made that work at-all. It there a FAQ you could point me at? How do you initiate an H.221 ISDN call via Zaptel for example? I have never heard of this feature, so perhaps this is some external Zaptel/Asterisk patch that was in your 1.2.7 build that is not in your new build? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
We can port most of these numbers, give us a call to see how fast we can switch this over, Meanwhile we know Les, so we can ask them to push temporariliy to our switches while it's being transfered. On 8/9/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Please stop advertising your forums/services on every single chance u get on users list . On 08/08/07, Al Bochter [EMAIL PROTECTED] wrote: That is why you need to start posting info about the providers at http://www.bochterservices.com/phpbb/ so everyone knows This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. There will be more added by your request Best regards, Al Bochter http://www.BochterServices.com --- See what we are selling at auction http://www.epier.com/auctions.asp?bochterservices --- Take a look at our online store http://www.bochterservices.com/onlinestore/ --- Join our forum. This is where you can talk about VOIP You can overview some providers others have used. http://bochterservices.com/phpbb/ --- Stephen Bosch wrote: Mail list wrote: Just got mail from them saying my NY DID will be deactivated in few days . Funny thing is their site is still showing orderable DID's of same area code . Anybody else got this ? Wow. That is totally unacceptable. Are they going to give you the option of porting the DID? -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 000764-2, 08/08/2007 - 8/8/2007 5:31:56 PM ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The quest for making hint more flexible continues - using Realtime now
is subscribe context an addiotional switch/field ?
or its the peer context ?
On 8/9/07, Mike [EMAIL PROTECTED] wrote:
I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Thursday, August 09, 2007 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more flexible
continues - using Realtime now
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than text files. This is the appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata
|
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 |
|
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
mailto:Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for
that extension
Wellthere is! Is there any way I can do this?
Mike
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I personally opened a bug in the bugtracker about this and it was closed
as
wont fix. You simply cannot use the hint priority in realtime with out a
major change to the API. So until the code is changed, you are going to
have
to have a separate hint context with nothing but hint priority extensions
and set the subscribe context in sip.conf for all concerned devices to
that
context.
This is how I am running in production now.
Anthony
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--
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] How to use OpenVPN with Asterisk
MOSBAH ABDELKADER wrote: Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. 1. get openvpn up and running. That will give you a secure tunnel between server#1 and server#2. 2. whatever it is you need asterisk to do, make sure it uses the tunnel endpoints for networking. /Per Jessen, Zürich -- http://www.spamchek.com/ - your spam is our business. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The quest for making hint more flexiblecontinues - using Realtime now
subscribecontext (one word) is another attribute of a peer (sip.conf). I am
using it as part of a MYSQL table that holds all my sip registrations, and
that works fine. I did have to add the column, since it wasn't part of the
table construct that can be found on the wiki.
Mike
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Lynchfield
Sent: Thursday, August 09, 2007 13:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more
flexiblecontinues - using Realtime now
is subscribe context an addiotional switch/field ?
or its the peer context ?
On 8/9/07, Mike [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I feared so, but I have already started working on this. Thanks for the
confirmation.
Too bad, the rest of my design was relatively elegant (IMO) and easily to
modify.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Francis
Sent: Thursday, August 09, 2007 12:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] The quest for making hint more flexible
continues - using Realtime now
Mike wrote:
Ok, now that I've learned I cannot use any variables when using the
`hint` priority (for BLF), I figured I'd try to use the next best
thing: hardcoded values using realtime. This way I avoid variables
such as ${ACCOUNTCODE} but I can at least change the DB more easily
than text files. This is the appropriate line in the DB:
+--+--+---+--++-+
| id | context | exten | priority | app| appdata |
+--+--+---+--++-+
| 2000 | hint-context | 705 | hint | SIP/test-1 | |
+--+--+---+--++-+
This is what I put in mt hint-context in extensions.conf:
[hint-context]
switch = Realtime/[EMAIL PROTECTED]
mailto: mailto:Realtime/[EMAIL PROTECTED]
Realtime/[EMAIL PROTECTED]
And this is what I get from the CLI:
Aug 9 11:34:14 NOTICE[19894]: chan_sip.c:11187
handle_request_subscribe: Got SUBSCRIBE for extension [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] from xx.xxx.xx.xx, but there is no hint for
that extension
Wellthere is! Is there any way I can do this?
Mike
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I personally opened a bug in the bugtracker about this and it was closed as
wont fix. You simply cannot use the hint priority in realtime with out a
major change to the API. So until the code is changed, you are going to have
to have a separate hint context with nothing but hint priority extensions
and set the subscribe context in sip.conf for all concerned devices to that
context.
This is how I am running in production now.
Anthony
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] Measuring Jitter in Asterisk
Oh jeez. Another GUI... -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of lenz Sent: Thursday, August 09, 2007 6:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Measuring Jitter in Asterisk I have used this freeware tool in the past: http://sineapps.com/sinestatiax.php maybe you can have a look at it as well l. In data Thu, 09 Aug 2007 02:07:49 +0200, John Todd [EMAIL PROTECTED] ha scritto: At 3:33 PM -0700 2007/8/3, Douglas Garstang wrote: At 12:31 PM -0700 2007/8/3, Douglas Garstang wrote: How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. You could look at the txjitter and rxjitter values (and other values) stored in the CHANNEL() function, and those values you're looking for were previously known as RTPAUDIOQOS. Or is this not sufficient? Are txjitter and rxjitter working reliably? These calls are going to be placed from AMI and bridged together. Do you think the variables would be correctly set for each leg of the call? Doug. I think the best way to determine this would be to compare the numbers provided by CHANNEL() versus the numbers provided by something with a little more reliability, such as wireshark, in a controlled set of circumstances. Please post your results here - it would be an interesting test. JT ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Help in changing Voice message
On 8/9/07, Farooq Ahmed [EMAIL PROTECTED] wrote: Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan beep beep beep The person you are calling is unavaiable, please try again. I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no log appears in the asterisk cli command just get this message so i am not getting any idea which macro or application generating this message. Anybody have any idea about how to change this? This is probably not coming from Asterisk. It's probably generated by your phone when Asterisk responds with a 5xx or 4xx response code to your INVITE. Depending on your phone you may or may not be able to change it. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
MOSBAH ABDELKADER wrote: Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. As far as asterisk is concerned OpenVPN is just another interface. There isn't really anything you need to do to asterisk to make it work. As for OpenVPN, their site has some excellent HOWTO's on getting started with OpenVPN. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. Except now, right, pal? Your site is down, you see. A shame, that. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major Digium Card Problems
Does this mean that the server itself may not be grounded? (as in the outlet isnt properly grounded) That would obviously be the easiest thing to fix. Assuming it is grounded, I guess the first place I should check is the outside telco box? Make sure its grounded? Its strange this just started out of no where though, either it was always grounded or it always wasn't. Thanks for your help. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, August 09, 2007 12:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Major Digium Card Problems Jay R. Ashworth wrote: On Wed, Aug 08, 2007 at 11:44:51PM -0400, Michael J. Liberatore wrote: First problem, the card with 4 FXO ports is fine until there is a storm in the area, then all 4 lines are massively static filled making phone calls barely understandable until the system is rebooted or the zaptel modules are unloaded and reloaded. There is no problem with other phones or the previous phone system on these landlines, so i dont think there is a problem with the lines. First, find the knob in your mailer that says send messages as HTML and turn it off, please? HTML is bad for mailing lists. Secondly, remember: this is a *phone* system now; you're hooking it up to several kilofeet of antenna. If you don't have telco-quality lightning protection and grounding on the box, you can expect this sort of thing. You can't find practices handbooks anymore (damnitall), but if you've ever looked at a professionally installed key system backboard, and seen those Porta-Systems gas-tubes, and the size of the grounding wire, then you may get an inkling of a) why you're having problems, and b) why traditional PBX's cost so much to buy and install. It's not *all* extra markup, folks. Cheers, -- jr 'hobby horse' a I was not aware that ground wire was very expensive or difficult to ground correctly. I do not see how that adds very much to the dealer's cost. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
Per Jessen escreveu: MOSBAH ABDELKADER wrote: Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. 1. get openvpn up and running. That will give you a secure tunnel between server#1 and server#2. 2. whatever it is you need asterisk to do, make sure it uses the tunnel endpoints for networking. what is the overhead added (proc and bandwidth)? Tom Lobato ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] les.net losing DID's
Jay Moore wrote: This is a FREE SERVICE provided by Bochter Services and it is not going away any time soon. Except now, right, pal? Your site is down, you see. A shame, that. Site came right up for me Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Help in changing Voice message
James FitzGibbon wrote: On 8/9/07, *Farooq Ahmed* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan beep beep beep The person you are calling is unavaiable, please try again. I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no log appears in the asterisk cli command just get this message so i am not getting any idea which macro or application generating this message. Anybody have any idea about how to change this? This is probably not coming from Asterisk. It's probably generated by your phone when Asterisk responds with a 5xx or 4xx response code to your INVITE. Depending on your phone you may or may not be able to change it. -- j. Is it Allison's voice saying to try again? If so, then there is a good chance that it is Asterisk but and a small chance that it might not be. Grandstream phones use Allison's voice for inbound callerID announcement if you enable it. What kind of phone are you using? Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use OpenVPN with Asterisk
On Thu, 9 Aug 2007, MOSBAH ABDELKADER wrote: Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. If it's purely between 2 Linux boxes, then you might want to look into using the TUN/TAP interfaces and running vtund which is far easier to setup. See http://vtun.sourceforge.net/ Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable DND feature key in Polycom Phone
I'm not sure of the correct wording in ipmid.cfg or sip.cfg, but I think you'd be most successful using the keys/ block. A probably wrong example might be: key.IP_500.9.function.prim=Null for a soundpoint 50x and 60x. or key.IP_300.7.function.prim=Null for a soundpoint 30x But it at least might get you pointed in the right direction. If Null isn't what you want you could map it to an arrow key or something else... Mojo Farooq Ahmed wrote: Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable DND feature key in Polycom Phone
to clarify what I'm talking about: I'm referring to the soundpoint ip admin guide for version 1.5 for example. The key/ wording is in section 4.6.1.15, or page 113. The key *numbers* referred to, however, are found in section 3.1.7, beginning on page 21. Moj Mojo with Horan Company, LLC wrote: I'm not sure of the correct wording in ipmid.cfg or sip.cfg, but I think you'd be most successful using the keys/ block. A probably wrong example might be: key.IP_500.9.function.prim=Null for a soundpoint 50x and 60x. or key.IP_300.7.function.prim=Null for a soundpoint 30x But it at least might get you pointed in the right direction. If Null isn't what you want you could map it to an arrow key or something else... Mojo Farooq Ahmed wrote: Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable. Any idea regarding this. Regards Farooq ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
