Re: [asterisk-users] Music On Hold
On Mon, Dec 17, 2007 at 05:28:12PM -0500, itgasterisk wrote: Hello everyone, I am having a bit of problem getting MusicOnhold to play. I am running Asterisk 1.4 with MPG123 0.59 installed. Any specific reason you want to use mp3 format? If you downsample this to a 8kHz 16 bits per sample mono wav file you'll get a file which may be even smaller and will not take any special transcoding to play by Asterisk on each time. -- Tzafrir Cohen icq#16849755 jabber:[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] Upgrade to Asterisk 1.4 - it's one year's old!
Maybe some of the developers could work on stability and reliability while others work on a smooth upgrade process and yet others work on usability. Still others might look at enhancements, rather than considering a PBX as an appliance like a toaster: works fine for bread, but when bagels come along, scrap it and plug in the new model. In today's environment, I think any technology needs to be considered inadequate to begin with. We can't always anticipate all of next-year's requirements, and don't want every enhancement to require what was known in the PBX world as a forklift upgrade. --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, December 17, 2007 1:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! I would rather the Developers spend their precious time improving the stablilty and reliability than creating a smooth upgrade process. Not that I don't think it is at least as reliable and stable as 1.2 right now. It seems to be for me in a low call volume environment. A PBX should be looked at as more of an appliance than an application server IMHO. You shouldn't have to upgrade it unless it was inadequate to begin with. If that is the case you should be doing an install of 1.4 from scratch anyways. Just my opinion. -Original Message- From: Phil Knighton [mailto:[EMAIL PROTECTED] Sent: Monday, December 17, 2007 4:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Hello As a person who is somewhat a newbie to Asterisk, I have been given the task of preparing our 1.2 installation for upgrade. The thing that has slowed me down is some of the gaps in information on the upgrade process. What's on the Wiki might make complete sense to both experienced Linux users, and Asterisk users but as someone who is feeling there way through - it's a bit daunting! Considering how important a phone system is to a business, I'm loathed to rush the upgrade through and have instead opted to install 1.4 on a different box, and port our existing setup over to it. This is a time consuming process and has taken quite a low priority. As Olle says - 1.2 works just fine. Personally speaking, the upgrade process has to be even easier if people are going to jump for it. Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E Sent: 15 December 2007 10:57 To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! Friends in the Asterisk community, I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 and 1.4 there's been a lot of important development. New code cleanups, optimization, new functions. I realize that 1.4 at release time wasn't ready for release, but we've spent one year polishing it, working hard with bug fixes. The 1.4 that is in distribution now is very different from the young and immature product that was release before Christmas in 2006. Testing, testing, testing and hard work from developers has changed this and the 1.4 personality is now much more grown-up and mature :-) I wonder if there are any major obstacles for upgrading. - Bugs that are still open? - Bugs that are not reported? - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 When responding, remember that we don't add new features to 1.4 after release, so I'm not looking for a wishlist - that's for the coming release. We need to make a released product stable, not add new features and potential scary bugs. Success stories with 1.4 are also welcome. Upgrading to 1.4 doubled our revenues in a month and gave us 200% more quality in the voice channels or Asterisk 1.4 gave us more reliable pizza deliveries and also fixed the bad taste of the coffee in our vending machine. Anything. Also, I would like input on what you consider the most important new feature in 1.4. I will try to make a list based on the feedback. Feel free to send feedback to the list or in a private e-mail to me directly. Let's make 1.4 the choice for everyone's PBX - from small home systems to large scale carrier platforms! /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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
[asterisk-users] Using MysqlPool Application 1.4
Hi, Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore for select queries :-( I'm using dbquery from MysqlPool Application 1.4 and selecting something from a table returns nothing even if I try to do a query like SELECT 1; Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? For information, I'm using MySQL 5.1 (for xml) on a ubuntu gutsy server. Regards. -- Cyril SCETBON ___ --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] All trunk are busy please try your call again later
Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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] All trunk are busy please try your call again later
Post: Asterisk CLI : sip show peers Asterisk CLI : zap show channels Asterisk CLI: zap show status As well as your extensions.conf Are you able to ping you GSM gateway? is connected via SIP or Telephony interface card? Best regards, Mouta On Dec 18, 2007 10:47 AM, Lolu Gbenga [EMAIL PROTECTED] wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] Dial() Macro option error in 1.4.15
On Thursday 06 December 2007 03:03:35 pm Anthony Messina wrote: What was I trying to do???... Using the M option is probably not the best way to set the CDR(userfield) anyway. What I was trying to accomplish was to have inbound DUNDi calls define something like dundi-in in the userfield, and ENUM would say enum, or something like that. The trouble is, I use a grabber script which grabs all the ATT numbers local to my pstn and store those area codes/prefixes in realtime for use with DUNDi or local calls. I know I could use a Set(CDR(userfield)=dundi-in) as priority 1, and have the Dial as priority 2 in MySQL, but I have to figure out how to fix my grabber script for that. Is there a way that an inbound DUNDi call could have an accountcode based on the peer which is connecting to place the call? after i wrote that question, i realized (duh!) that i could simply put the accountcode in the [dundi] user section in iax.conf! some things end up being so simple... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --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] All trunk are busy please try your call again later
Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success You need to at least post some verbose from the console and explain how you are connecting to the PSTN. It would greatly help if you included the relevant portions of your extensions.conf. Thanks, Steve Totaro ___ --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] Trixbox Phones Home
In http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home is said Kerry Garrison that: Both trixbox and FreePBX have phone-home mechanisms in them. So does FreePBX phones home too? On Dec 17, 2007 4:27 AM, Than Taro [EMAIL PROTECTED] wrote: As I pointed out here last night, there is also a very serious security vulnerability associated with this. Example: An attacker could compromise the script that is used on the remote host, and set it to force clients that connect to run a command such as rm -rf /. There are about half a dozen ways I could see this being abused - in either a one off or an every installation scenario. Fonality has yet to acknowledge this aspect of the issue - and I fear that they never will. See: http://voipsa.org/pipermail/voipsec_voipsa.org/2007-December/002522.html P.S.: On behalf of Rob (of FreePBX fame), I'd like to also point out this this is something that was added to trixbox, and not FreePBX. Quoting Rob: when someone mistakenly says 'trixbox does...' they usually mean 'freepbx does...' as FreePBX is the GUI Trixbox uses to configure Asterisk. In this instance, that is not the case - it is only a trixbox issue. From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com; [EMAIL PROTECTED] Date: Sun, 16 Dec 2007 20:53:53 -0500 Subject: [asterisk-users] Trixbox Phones Home I just read on Slashdot (at http://yro.slashdot.org/article.pl?sid=07/12/16/43 ) that Trixbox has been phoning home with statistics about their installations, as a Trixbox user exposed in Trixbox Phones Home at http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home. -- (C) Matthew Rubenstein ___ --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 -- The best games are on Xbox 360. Click here for a special offer on an Xbox 360 Console. Get it now! http://www.xbox.com/en-US/hardware/wheretobuy/ ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] Trixbox Phones Home
On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote: In http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home is said Kerry Garrison that: Both trixbox and FreePBX have phone-home mechanisms in them. So does FreePBX phones home too? And if you read further down that thread you would have seen the reply by philippel of FreePBX: ... | The only time this happens is when an online update is initiated by you, | or if you have chosen to receive update notifications since those are | nothing more then a cron Job that does exactly what Check for Online | Updates does in the GUI. ... -- Tzafrir Cohen icq#16849755 jabber:[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] Trixbox Phones Home
Thanks Tzafrir! I really appreciate Free PBX. Keep on going your good job. Best regards, Mouta On Dec 18, 2007 11:59 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Dec 18, 2007 at 11:38:03AM +, Marco Mouta wrote: In http://www.trixbox.org/forums/trixbox-forums/open-discussion/trixbox-phones-home is said Kerry Garrison that: Both trixbox and FreePBX have phone-home mechanisms in them. So does FreePBX phones home too? And if you read further down that thread you would have seen the reply by philippel of FreePBX: ... | The only time this happens is when an online update is initiated by you, | or if you have chosen to receive update notifications since those are | nothing more then a cron Job that does exactly what Check for Online | Updates does in the GUI. ... -- Tzafrir Cohen icq#16849755 jabber:[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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] Asterisk GUI - Call Waiting
Has anyone tested disabling call waiting for a SIP extension via the GUI ? I have deselected call waiting for a user with a SNOM 360 and applied my changes but they still get calls waiting and are reporting that 80% of the time when they get the bleeping in their ear when the new call comes in and that it kills the current call before they get chance to respond in any way ___ --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] BLF trouble
On Tue, Dec 18, 2007 at 03:53:14AM +0100, Philipp Kempgen wrote: Bristuff should have a Devstate() application. show application Devstate http://www.das-asterisk-buch.de/stable/snom-leds.html#snom-leds-custom Mmmh. The Xorcom version does not include the Devstate application. I will try to add it by hand. But thanks a lot. This seems to be exactly what I need. Lars -- Indifference: It takes 43 muscles to frown and 17 to smile, but it doesn't take any to just sit there with a dumb look on your face. -- Despair INC (http://www.despair.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] BLF trouble
On Tue, Dec 18, 2007 at 07:45:12AM +, Thomas Kenyon wrote: I have some trouble with the BLF indicator. If you are using Grandstream Phones with firmware 1.1.5.15, you will find that the BLF implementation no longer works. Yes, I'm using 1.1.5.15. But this would explain one of the problems (missing the fast hint updates). But asterisk displays the wrong state information for outgoing calls as well. 'core show hints' lists phones as idle that have dialed out. It's just working for incoming calls. I will make some more tests and gather some CLI output. Lars -- Everyone hates me because I'm paranoid. ___ --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] BLF trouble
On Tue, Dec 18, 2007 at 02:02:20PM +0100, Lars Bensmann wrote: On Tue, Dec 18, 2007 at 03:53:14AM +0100, Philipp Kempgen wrote: Bristuff should have a Devstate() application. show application Devstate http://www.das-asterisk-buch.de/stable/snom-leds.html#snom-leds-custom Mmmh. The Xorcom version does not include the Devstate application. I will try to add it by hand. But thanks a lot. This seems to be exactly what I need. devstate is available as a separate application from http://sourceforge.net/projects/agx-ast-addons Or the specific patch that adds it to bristuff: misc-app-devstate from: http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/bristuff/ svn://svn.debian.org/svn/pkg-voip/asterisk/trunk/debian/patches/bristuff/ -- Tzafrir Cohen icq#16849755 jabber:[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
[asterisk-users] How to automaticaly close calls when Asterisk didn't receive the bye request ?
Hi, I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? Thanks, -- Anthony Chapellier - MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : [EMAIL PROTECTED] Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.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] Music On Hold
On Tue, 2007-12-18 at 13:28 +0530, Godson Gera wrote: On Dec 18, 2007 3:58 AM, itgasterisk [EMAIL PROTECTED] wrote: Hello everyone, I am having a bit of problem getting MusicOnhold to play. I am running Asterisk 1.4 with MPG123 0.59 installed. And here's what i see in the debugging window of asterisk: -- Started music on hold, class 'default', on channel 'SIP/x123-082043d0' -- Stopped music on hold on SIP/x123-082043d0 Any idea why it is not playing the file at all? Hi Eric, Try to install asterisk-addons which can play mp3 (using format_mp3.so) files directly, instead of depending on mpg123. Once you install addons don't forget to set mode=files in musiconhold.conf Even better, don't use mp3 at all. Iirc variable bit rate mp3s could cause Asterisk to blow up. Don't know if that has been fixed but do you want to run that risk with a PBX? Just convert your music on hold files to the native formats you use like ulaw, alaw, gsm, g729 etc. and configure musiconhold.conf to use those files. 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] How to automaticaly close calls when Asterisk didn't receive the bye request ?
On Tue, 2007-12-18 at 15:20 +0100, Anthony Chapellier wrote: I'd like to know if it's possible to configure Asterisk to automaticaly close calls when the BYE request hasn't been sent by any clients and the call still exists for Asterisk ? There is a SIP timers patch in the bug tracker (see http://bugs.digium.com/view.php?id=10665) that currently implements this, and it's being tested in the team/group/sip_session_timers/ branch in SVN. Please test this out and help provide feedback, so that we can get this put into Asterisk in time for the next major release. I'd also like to take this opportunity to thank John Todd and Raj Jain for their hard work on this feature -- this is a great example of patches being submitted to Asterisk with great documentation, a detailed explanation of the current limitations, an explanation of the standard, implementation details, and a test plan. Good job guys! --- Jared Smith Community Relations Manager Digium, Inc. ___ --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] All trunk are busy please try your call again later
lolu, sounds more like a telco/itsp problem then *. I would tcpdump -i eth0 port 5060 to make sure it is actually going out... change 5060 if you have changed your port to your itsp, of course. to see what is going on as well as the other debugging notes mentioned in this thread. daveC Lolu Gbenga wrote: Good Day all Please I am having some issues on my voip asterisk server I make internal calls on extensions configured ie extension 192 can call extension 195 etc But each time i try to make calls outside the extension ie calling a GSM or an external line ,i always hear this response all trunk calls are busy please try your call again later Please how can i resolve this problem . I will appreciate your response. Best Regards Success ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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] AsteriskNOW release date???
Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul ___ --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] Using MysqlPool Application 1.4
On Tuesday 18 December 2007 03:59:04 Cyril SCETBON wrote: Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? See func_odbc.conf. -- Tilghman ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Monday 17 December 2007 19:30:46 Don Kelly wrote: Maybe some of the developers could work on stability and reliability while others work on a smooth upgrade process and yet others work on usability. Still others might look at enhancements, rather than considering a PBX as an appliance like a toaster: works fine for bread, but when bagels come along, scrap it and plug in the new model. Actually, all of the developers have their own pet projects and enhancements. We'd go stark raving loony if we all had to only fix bugs all day. Instead, we share the load. -- Tilghman ___ --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] Dropped Calls
Hi all, I have a problem with some asterisk boxes. I have a standard installation with 1.4.14 (I also test with 1.4.4) in core duo Mac Mini on Debian Etch. I use SJphone softphone, Linksys SPA921 or Thomson 2030 for phones. All my phones are in a LAN with good status of 2ms max. Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? Thanks in advance, Jeremy ___ --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] AsteriskNOW release date???
I asked the Adtran/Digium guys a similar question at the end of What's New at Digium at this year's Astricon in Arizona, which was just the announecment of the aquisition of SwitchVox. The reply was less than encouraging for future dev on the free GUI which is what I expected. While very vague and obviously a question they were not ready for, the answer was something along the lines of Mumble, mumble, it will be up to the community to continue development. I ask, what is the incentive to put out a full featured GUI for free at this point? Take into consideration the SwitchVox aquisition and aslo the recent news from 3Com. I see no reason to continue dev on AsteriskNOW except the fact that it is rigged to move the end user to order phones through a link that will obviously take business away to anyone who has not sworn allegiance to Digium (and their first born) to become part of their food chain. FreePBX seems to be the most logical choice to me. Thanks, Steve Totaro - Original Message - From: Raúl Gómez C. To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, December 18, 2007 10:16 AM Subject: [asterisk-users] AsteriskNOW release date??? Hi list, Anyone knows about the date of the official (stable) release (v1.0) of AsteriskNOW??? It's supposed to be at the end of this year, which is very close now with no signs of it. Thanks... Raul ___ --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] Dropped Calls
On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org wrote: Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? The first place I'd look would be the Asterisk CLI. Make sure you turn up the CLI verbosity first by typing core set verbose 5 before the call. If that doesn't offer any clues, I'd next look at the SIP signaling. You can see that by typing sip set debug at the Asterisk CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. --- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Dropped Calls
Jared Smith wrote: On Tue, 2007-12-18 at 17:43 +0100, Administrateur www.jeremy-salmon.org wrote: Randomly I have dropped calls during communication. No absolutetimeout or other calling limitation options. Any ideas on how to solve this problem? The first place I'd look would be the Asterisk CLI. Make sure you turn up the CLI verbosity first by typing core set verbose 5 before the call. If that doesn't offer any clues, I'd next look at the SIP signaling. You can see that by typing sip set debug at the Asterisk CLI, or by using a network tool such as Wireshark, tcpdump, or ngrep. --- Jared Smith Community Relations Manager Digium, Inc. I would also ask that all user's keep a log or send an email to you with their extension, if the call was internal or external, time, and how long into the call that it dropped. Collecting this data might help you figure out a trend. I would open an SSH session with txt logging and ask everyone to submit a dropped call report and see if you can link up some common events or errors. You may find it is only happening on external calls which may look like a normal hangup and could indicate a problem with your PSTN connectivity. Thanks, Steve Totaro ___ --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] AsteriskNOW release date???
Would it be possible to install FreePBX on an AsteriskNOW system? The one thing I really like about AsteriskNOW is the reduced attack surface b/c it is running on an rpath appliance. Are there any appliances out there that combine asterisk with freepbx? Regards, Zaheer _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro FreePBX seems to be the most logical choice to me. Thanks, Steve Totaro ___ --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] About Pickup Grandstream
On Friday 23 March 2007, Lukas wrote: Hahaha don't worry my friend. We're not the only ones. I'm on Asterisk 1.2.16 For this job i think it's the best one. Gonna Sleep (here in Spain it's 5 am) See you later. If i find the solution i'll post again. Any news on this topic? Run into the same problem. cheers t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --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] SIP call interrupted after 64 seconds
Jaswinder Singh schrieb:
Can you post the part of your dialplan which causes this behaviour
Hi,
I've found, what's causing the problem:
My dialcommands are always of the type:
Dial(IAX2/user:[EMAIL
PROTECTED]/12345678,120,gS(${maxduration})M(connect^${some_params}))
or
Dial(SIP/[EMAIL PROTECTED],120,gS(${maxduration})M(connect^${some_params}))
${maxduration} is set to 86400 in most cases, sometimes to 3600
or 7200 (but never to 64). I checked this from within
the console.
When I leave the S() parameter away, there is no call, stopping after
64 secs. When I have the S() parameter, about every 10th call stops
after exactly 64 secs.
Thus, I assume a bug with the S() parameter in asterisk-1.4.x.
Can maybe someone check this on his machine, before I open a bug
report!
Roger.
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Re: [asterisk-users] AsteriskNOW release date???
On Tuesday 18 December 2007 10:13:59 Steve Totaro wrote: I asked the Adtran/Digium guys a similar question at the end of What's New at Digium at this year's Astricon in Arizona, which was just the announecment of the aquisition of SwitchVox. The reply was less than encouraging for future dev on the free GUI which is what I expected. While very vague and obviously a question they were not ready for, the answer was something along the lines of Mumble, mumble, it will be up to the community to continue development. That's incorrect. At this time, there are no plans to discontinue development of the free GUI. I ask, what is the incentive to put out a full featured GUI for free at this point? Well, for one, the appliance and any other hardware that is not beefy enough to run Switchvox. Take into consideration the SwitchVox aquisition and aslo the recent news from 3Com. I see no reason to continue dev on AsteriskNOW except the fact that it is rigged to move the end user to order phones through a link that will obviously take business away to anyone who has not sworn allegiance to Digium (and their first born) to become part of their food chain. Not exactly a fair attack. The GUI was designed to respond to a direct end user complaint that they didn't know from whom to purchase extra phones. The appliance is meant to help those specific people who do not know where to go buy phones or how to provision them (and don't care to learn). It's interesting, is it not, that Digium is simultaneously attacked in one thread for not responding to user demands and another thread for giving end users exactly what they demanded? FreePBX seems to be the most logical choice to me. Which is being leveraged to take away business to anyone who has not sworn allegiance to Fonality. Sorry, couldn't resist. ;-) -- Tilghman ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
At 10:33 AM 12/17/2007, you wrote: At 02:55 AM 12/17/2007, you wrote: I wonder if there are any major obstacles for upgrading. Because of your message I tried upgrading to 1.4 again Saturday. That was the third or fourth time I've tried and the first time it's lasted more than a few hours before segfaulting and causing me to step back to 1.2. It seems like I might be staying with 1.4 this time as 2 days later it's still working. I did find one last deprecated function in the startup logs and fixed that so I should now be good for the 1.6 upgrade. Well, I spoke too soon. This morning I'll be going back to 1.2 as 1.4.15 just segfaulted. It always happens when something is going on with a Zap call, this time it was hanging up a call. I've no idea what might be the problem or how to even begin to troubleshoot. And it's my business so I can only play on Saturdays, gives me 2 days to fix it if there's a problem, sadly this time, it took 2 days to break Ira ___ --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 Recording on Hanup
Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --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 Recording on Hanup
What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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] Doorbell Siedle DCA 612 and Asterisk?
Hi, has anyone already set up a configuration between the doorbell Siedle DCA 612 and an Asterisk Server? I have used a Grandstream HT 286 to connect the doorbell and the asterisk. When I press the button, the phone ring and when I pick up the call I hear a beeping. At the door I hear nothing. According to the wiki, this doorbell should work with Asterisk, but I haven't found a dialplan or a howto on how the configuration should look like. Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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] Softphone
Hi List; I was knowing when asterisk started, there was a softphone that has an text messages feature, voice calls, knowing who are online with u, look like messanger. Where that softphone? I do not see it any more in Asterisk. Regards Bilal Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ --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 change sendmail return path
Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --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] SIP Anonymous auth
Hi all, i am new with VOIP and use asterisk with Trixbox ;) When i have a look in my logs i see the following problem: chan_sip.c: Failed to authenticate user Anonymous sip:[EMAIL PROTECTED] I never setup a user Anonymous which is communicating with dus.net ;) Can anyone explain why it will authentificate with the User Anonymous? My SIP/Trunk conf looks as the following: register=USER:[EMAIL PROTECTED]/USER fromdomain=voip.dus.net fromuser=user host=voip.dus.net nat=yes qualify=true secret=pass type=friend username=user context=induscalls fromdomain=voip.dus.net fromuser=user host=voip.dus.net insecure=very nat=yes secret=pass type=friend username=user -- Mit freundlichen Grüßen Daniel mailto:[EMAIL PROTECTED] ___ --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] Asterisk 1.4.16 and 1.2.26 released
The Asterisk.org development team has released Asterisk versions 1.4.16 and 1.2.26. Both releases contain a fix for a security vulnerability. The 1.4.16 release also contains a number of other bug fixes made over the past few weeks. The details of the security issue have been published in a security advisory: http://downloads.digium.com/pub/security/AST-2007-027.pdf The issue affects users of the dynamic realtime configuration method for IAX2 or SIP that use host based authentication. Systems that do not use host based authentication with realtime are not affected. A full list of changes is available in the ChangeLog, which is distributed with the release and is also available on the downloads page. http://downloads.digium.com/pub/telephony/asterisk/ChangeLog-1.4.16 The releases are available for immediate download from http://downloads.digium.com/. Thank you for your support! ___ --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 change sendmail return path
shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 The option you are looking for is called serveremail. By default, if this is not set, it will be set to asterisk. Set this in the [general] section of voicemail.conf. Mark Michelson ___ --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] AST-2007-027 - Database matching order permits host-based authentication to be ignored
Asterisk Project Security Advisory - AST-2007-027 ++ | Product | Asterisk | |+---| | Summary | Database matching order permits host-based| || authentication to be ignored | |+---| | Nature of Advisory | Logic error | |+---| | Susceptibility | Remote Unauthenticated Sessions | |+---| | Severity | Moderate | |+---| | Exploits Known | No| |+---| |Reported On | October 30, 2007 | |+---| |Reported By | Tilghman Lesher tlesher AT digium DOT com | |+---| | Posted On | December 18, 2007 | |+---| | Last Updated On | December 18, 2007 | |+---| | Advisory Contact | Tilghman Lesher tlesher AT digium DOT com | |+---| | CVE Name | CVE-2007-6430 | ++ ++ | Description | Due to the way database-based registrations (realtime) | | | are processed, IP addresses are not checked when the | | | username is correct and there is no password. An | | | attacker may impersonate any user using host-based | | | authentication without a secret, simply by guessing the | | | username of that user. This is limited in scope to | | | administrators who have set up the registration database | | | (realtime) for authentication and are using only | | | host-based authentication, not passwords. However, both | | | the SIP and IAX protocols are affected. | ++ ++ | Resolution | As a workaround, administrators may set a password for| || all users and peers in their registration realtime | || database. A fix is included in the newest release of | || Asterisk, as provided below. | ++ ++ | Affected Versions| || | Product | Release | | || Series| | |+-+-| |Asterisk Open Source|1.0.x| Not affected| |+-+-| |Asterisk Open Source|1.2.x| All versions prior to | || | 1.2.26 | |+-+-| |Asterisk Open Source|1.4.x| All versions prior to | || | 1.4.16 | |+-+-| | Asterisk Business Edition |A.x.x| Not affected| |+-+-| | Asterisk Business Edition |B.x.x| All versions prior to | || | B.2.3.6 | |+-+-| | Asterisk Business
Re: [asterisk-users] Call Recording on Hanup
yes, the senario is this when user gets a call IVR starts playing and after hearing beep user starts recording message for 30 seconds(call duration is for 30 seconds). What i want is During 30 seconds if user does hangup his/her call then message should be recorded otherwise(after timeout) message is discarded. Is there any thing that will help me...??? currently I am doing the same thing on pressing 1 with php agi script and its working fine. On 12/19/07, Marco Mouta [EMAIL PROTECTED] wrote: What do you mean with record a call on hangup? If the calling party ends the call you want to keep recorded file? On Dec 18, 2007 6:27 PM, Jamshed Zaidi [EMAIL PROTECTED] wrote: Hello everyone out there, I am having a problem in call recording with php agi library. I have already recorded voice after playing an IVR, to accept the recording user need to press one. but I need to record a call on hangup, Is there any way to do it. Currently i am using record_file() function in php. Is there any way to record voice by using record_file() function with hangup. can anyone helps me in resolving this problem ??? -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --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 -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --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] Cisco 7970 BLF/Presence
I have been trying to get the 7970 (running SIP firmware) to display presence information about other extensions. Thus far, I have been unsuccessful. Does anyone have BLF working on the SIP-loaded 7941/7961/7970/7971? I have been using the following as a guide for my work: http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones I have successfully implemented every step (although there are a few minor errors in the code on that page that I have corrected) except for the part where the 7970 has to connect to Asterisk via SIP/TCP. I have chan_sip.c patched with the SIP/TCP patch to allow for connectivity, but I cannot seem to force the 7970 to connect that way. I have read several posts that indicated that the SIP firmware does not support presence information as of yet, but supposedly, whoever wrote that article above has it working somehow. Does anyone know how to make the phone connect over TCP? Or, better yet, does anyone have a working method that they would be willing to share? These are great phones but in the environment that we're in they are almost useless if we don't know who's on a call when. I'd rather not go the SCCP route unless I absolutely have to. 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] How to change sendmail return path
Have a look at serveremail = [EMAIL PROTECTED] and fromstring = The Asterisk PBX in voicemail.conf. On Dec 18, 2007, at 2:28 PM, shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --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] SIP Anonymous auth
Guten Tag Daniel, am Dienstag, 18. Dezember 2007 um 20:35 schrieben Sie: Hi all, i am new with VOIP and use asterisk with Trixbox ;) When i have a look in my logs i see the following problem: chan_sip.c: Failed to authenticate user Anonymous sip:[EMAIL PROTECTED] Anonymous is the phone number from the person who is calling. When i try to call with my mobile phone i see sip:(my-number)@dus.net I have the same config with sipgate and that works fine for me. -- Mit freundlichen Grüßen Daniel mailto:[EMAIL PROTECTED] ___ --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 1.4.16 and 1.2.26 released
The releases are available for immediate download from http://downloads.digium.com/. Could someone make sure the files are actually available BEFORE sending these out? John ___ --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 1.4.16 and 1.2.26 released
John Faubion wrote: The releases are available for immediate download from http://downloads.digium.com/. Could someone make sure the files are actually available BEFORE sending these out? That's just a way to find out how much traffic the PHP scripts on the download server can handle. ;) 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 http://www.kempgen.net/asterisk/current/ 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] Leading 0 in PRI outbound
All We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. National Numbering Plan for Landline in India is typically 0+Area Code + phone number. If it's a local number, you just dial the number without the area code. So for instance, if you want to call a number 42121234 in Delhi (Area Code 11), from any place outside of Delhi, you'd dial 01142121234 but only 42121234 within Delhi. Because of the prefix, when dialed locally, the number appears as 042121234 (which is not a valid number as there's a 0 without an area code!) There's nothing in the dial plan that is doing it. In fact, set verbose and pri intense debug indicate that the channel that's originating the cal is Zap/g0/42121234 but somehow there's a zero that gets prefixed :( Tried changing zapata.conf to include prilocaldialplan and so on but to no avail! Any help appreciated! thanks rajeev ___ --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 1.4.16 and 1.2.26 released
On Tuesday 18 December 2007 14:36:22 John Faubion wrote: The releases are available for immediate download from http://downloads.digium.com/. Could someone make sure the files are actually available BEFORE sending these out? Sorry, the process that normally syncs the files out to the public repository had mysteriously died. It's fixed now. -- Tilghman ___ --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 Center Setup on asterisk
http://astguiclient.sourceforge.net/vicidial.html - supports both inbound and outbound http://queuemetrics.com/ - excellent set of metrics to measure your agents' performance! good luck -r On Dec 17, 2007 8:14 PM, Jared Smith [EMAIL PROTECTED] wrote: On Sat, 2007-12-15 at 19:06 +0200, Dovid B wrote: http://www.h6315.com/ast_docs/Asterisk%20TFOT%20v2.pdf I'm not sure who is running this website, but I'd kindly ask them to please point people to the official download at http://www.asteriskdocs.org/ instead of being an unofficial mirror. One of the important reasons for this is so that O'Reilly can better measure how many people are downloading the free version of the book versus how many people are buying the paper copy. Thanks! -Jared Smith ___ --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] resync linksys SPA9XX config file from Asterisk
Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Mail Test
Anthony Chapellier wrote: Sorry, I'm doing a mail test since I was not able to send any mails to the mailing list for about a week... Tell me about it. I've just given up on numerous posts because they'd vanish into cyberspace. Doubt this will show up since now even replies are being eaten, but what the heck I'll give it a try. -- Does your Canadian VoIP service need CRTC-compliant 9-1-1 services? Please visit http://www.digitalcon.ca/voip9-1-1/ to find out 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
Re: [asterisk-users] resync linksys SPA9XX config file from Asterisk
I would like to know as well, it has never worked for me. On Dec 18, 2007 4:27 PM, JR Richardson [EMAIL PROTECTED] wrote: Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses ___ --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] Asterisk/iaxclient IAX2 source port
All,
I have a simple question and a complicated reason for asking:
Is it possible to change asterisk's source port for outbound IAX2
connections?
I've tried using sourceaddress to no avail. I can set it to:
proper.ip.of.box:4569
or
0.0.0.0:4569
and it works as expected. But if I try to set it to:
proper.ip.of.box:5000
or
0.0.0.0:5000
it fails around line 8536 in channels/chan_iax2.c, function
peer_set_srcaddr, specifically:
if (ast_netsock_find(netsock, sin)) {
always returns false unless the port is set to 4569. Thus tripping the
error message:
chan_iax2.c:8940 peer_set_srcaddr: Non-local or unbound address specified
(0.0.0.0:5000) in sourceaddress for 'test-trunk', reverting to default
Is there any way to get asterisk to listen for inbound connections on
4569, but to use a non-4569 source port? (Ephemeral ports would be great)
Below is the reason for my asking, for the curious:
Currently, asterisk uses port 4569 as both the source and
destination port for all its outbound connections. This is generally
fine, but I find myself in a very frustrating NAT issue as a result of
iaxclient also defaulting to using 4569 for both source and destination
ports. We run several sites around the world, all using ENUM to place
calls between sites. Thus, none of the sites register with each other.
Thus, until a call is made, there is no connection between site A and site
B, and thus no NAT entries in the router at site B for site A.
Normally, this is fine. A call is placed from A to B and the
packets come into the router at B and get NATed properly:
A.ext:4569 - B.ext:4569 - A.ext:4569 - B.int.asterisk:4569
The trouble though, comes when someone who normally works at site
A vists site B, but has their IAX softphone (zoiper) set to register back
to site A. By default, this softphone, like asterisk, uses 4569 for both
the source and destination port. Thus, if there is no call between site A
and site B and a softphone registers back to site A, a NAT mapping gets
created that looks like:
A.ext:4569 - B.ext:4569 - A.ext:4569 - B.int.softphone:4569
Now, for the life of this NAT entry, if someone at site A dials
site B, their call will be routed to the lucky softphone that got this
entry, and not to the asterisk server at site B. Of course, calls out
from site B to site A still work properly, since the NAT device just
changes the port number on the fly since 4569 already has a mapping:
B.int.asterisk:4569 - A.ext:4569 - B.ext:65535 - A.ext:4569
There are three options I see that would fix this:
1. Prevent the linux router at site B from giving the 4569/4569 conntrack
entry to a softphone. Would be great, but as far as I can tell, there's
no way to do this using a standard distribution kernel. (Hopefully I'm
wrong, but my research hasn't turned up anything at all useful in this
regard)
2. Reconfigure all softphones to use a port other than 4569 as their
source port. In theory this is possible, but a huge pain to find/change
every existing softphone, as well as to ensure that people don't
accidentally end up with the default config in the future causing the
same problem.
3. Reconfigure asterisk to use a port other than 4569 for its source port
on outbound connections. The number of asterisk servers relative to
softphones is small, and the asterisk servers are configured/controlled by
admins, not end users. Thus we could have some guarantee that this
solution couldn't be circumvented.
Am I overlooking something? Is there an obvious solution here
that's escaped me?
(Ugh, why couldn't iaxclient/zoiper/asterisk all just follow the
RFCs and use ephemeral source ports to begin with?)
Thanks,
Chris
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Re: [asterisk-users] Asterisk/iaxclient IAX2 source port
On 13:52, Tue 18 Dec 07, Chris Tracy wrote: All, Below is the reason for my asking, for the curious: Currently, asterisk uses port 4569 as both the source and destination port for all its outbound connections. This is generally fine, but I find myself in a very frustrating NAT issue as a result of iaxclient also defaulting to using 4569 for both source and destination ports. We run several sites around the world, all using ENUM to place calls between sites. Thus, none of the sites register with each other. Thus, until a call is made, there is no connection between site A and site B, and thus no NAT entries in the router at site B for site A. Normally, this is fine. A call is placed from A to B and the packets come into the router at B and get NATed properly: A.ext:4569 - B.ext:4569 - A.ext:4569 - B.int.asterisk:4569 The trouble though, comes when someone who normally works at site A vists site B, but has their IAX softphone (zoiper) set to register back to site A. By default, this softphone, like asterisk, uses 4569 for both the source and destination port. Thus, if there is no call between site A and site B and a softphone registers back to site A, a NAT mapping gets created that looks like: A.ext:4569 - B.ext:4569 - A.ext:4569 - B.int.softphone:4569 Now, for the life of this NAT entry, if someone at site A dials site B, their call will be routed to the lucky softphone that got this entry, and not to the asterisk server at site B. Of course, calls out from site B to site A still work properly, since the NAT device just changes the port number on the fly since 4569 already has a mapping: B.int.asterisk:4569 - A.ext:4569 - B.ext:65535 - A.ext:4569 There are three options I see that would fix this: 1. Prevent the linux router at site B from giving the 4569/4569 conntrack entry to a softphone. Would be great, but as far as I can tell, there's no way to do this using a standard distribution kernel. (Hopefully I'm wrong, but my research hasn't turned up anything at all useful in this regard) 2. Reconfigure all softphones to use a port other than 4569 as their source port. In theory this is possible, but a huge pain to find/change every existing softphone, as well as to ensure that people don't accidentally end up with the default config in the future causing the same problem. 3. Reconfigure asterisk to use a port other than 4569 for its source port on outbound connections. The number of asterisk servers relative to softphones is small, and the asterisk servers are configured/controlled by admins, not end users. Thus we could have some guarantee that this solution couldn't be circumvented. Why not let the softphones register to the closest asterisk box and use dundi to route the calls to the box where the softphone is registered ? We use this in a couple of setups with great success. Not with softphones, but with philips dect phones. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Leading 0 in PRI outbound
On Tuesday 18 December 2007 15:22:18 Rajeev Natarajan wrote: We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. What is pridialplan set to in zapata.conf? This value sets an extra 4 bits in the PRI dialog between you and the telco. And typically, if you have it set to something like 'national', the telco will tell you you have numbers prefixed, even when you don't, because their switch software is written to make the translation. So what most people do (and what works most often) is to set pridialplan to 'unknown', which sets the bit field to all zeros and the number isn't prefixed at all at the telco switch, but simply routed based upon the number sent. -- Tilghman ___ --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 1.4.16 and 1.2.26 released
Could someone make sure the files are actually available BEFORE sending these out? I apologize for the way that sounds. It certainly sounded a lot more tongue-in-cheek in my head. John ___ --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] resync linksys SPA9XX config file from Asterisk
JR Richardson wrote: Hi All, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR The Header is: Event: resync You will have to set the parameter Auth Resync-Reboot: to NO on the phone so it will not ask for credentials. Andres. ___ --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 change sendmail return path
servermail= changes what shows up in the from section of the email. It doesn't change what shows up in the email header which is what the mail system looks at as the REAL return path. -Original Message- From: Mark Michelson [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 18, 2007 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 The option you are looking for is called serveremail. By default, if this is not set, it will be set to asterisk. Set this in the [general] section of voicemail.conf. Mark Michelson ___ --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 change sendmail return path
Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. -Original Message- From: Forrest Beck [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 18, 2007 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to change sendmail return path Have a look at serveremail = [EMAIL PROTECTED] and fromstring = The Asterisk PBX in voicemail.conf. On Dec 18, 2007, at 2:28 PM, shadowym wrote: Is there a way to change the return path sendmail uses when sending out voicemail to email? Currently the voicemails my asterisk system emails out have a return path of [EMAIL PROTECTED] I would like the return path to be [EMAIL PROTECTED] I cannot find any place where I can change that. I tried adding a sendmail alias to send asterisk to noreply and even tried root There are no config options anywhere in any asterisk *.conf or *.inc file which affect this There is nothing in my etc/hosts file which would cause the asterisk. I'm running CentOS 5.1, Asterisk 1.4.15, FreePBX 2.3.1 ___ --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] G.278 RTP conversation capture, please.
Hello all, I have a bit of a request. I need a wireshark capture of a SIP conversation using g.728. I don't need anything fancy, just a call and have both ends say hi to each other. hopefully someone out there can help me. Thank you all. This list has been of use many times in the past, even though I tend to stay quiet. ___ --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] Queue calls drop to voicemail intermittantly
That’s what I thought, but they all deny that they are pressing any buttons on the phone. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Norton - SophMedia LLC Sent: Monday, December 17, 2007 12:45 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Queue calls drop to voicemail intermittantly Are the agents “ignoring” the calls while their ringing? -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 HYPERLINK http://www.XStreamHost.comhttp://www.XStreamHost.com - Web Hosting HYPERLINK http://www.SophMedia.comhttp://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Malsack Sent: Monday, December 17, 2007 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Queue calls drop to voicemail intermittantly Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 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
Re: [asterisk-users] Queue calls drop to voicemail intermittantly
That is the same thing I thought as well. However the queue is set to an 18 second timeout and the voicemail is set to a 20 second timeout. I'll increase the voicemail timeout so there is a little more play there just to see if that helps. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Monday, December 17, 2007 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue calls drop to voicemail intermittantly On Mon, 2007-12-17 at 11:45 -0700, Robert Norton - SophMedia LLC wrote: Are the agents “ignoring” the calls while their ringing? -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development I had this problem. What was happening was that the timeout on the dial command for the extension where the agent is was lower than the time the queue waits for the agent to answer before returning the call to the queue. The voicemail timeout should be higher than the time the queue waits until the agent answers. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 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
Re: [asterisk-users] Queue calls drop to voicemail intermittantly
I would use extensions, but they want the agents logging in and out of the queue so they can pull reports on when the agents are waiting for calls. The channels that are assigned to the queues are Agent/. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Monday, December 17, 2007 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue calls drop to voicemail intermittantly On Monday 17 December 2007 12:35, Gregory Malsack wrote: Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? Probably you're putting Local channels into the queue. Any answer event at all generated by the Local channel, including one generated by Voicemail, is considered a pickup by the Queue app. Note that if you use the raw channel (SIP/IAX/Zap/whatever), then this will not happen when a queue member fails to answer their phone. Or create extensions that do not end in Voicemail for the use of Queue. -- Tilghman ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 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
Re: [asterisk-users] Asterisk Qeueu with static agent
The grammar makes it hard to understand the question, but if I’m understanding this right, this will probably to the trick. In the queue config file add: member = Agent/(agent’s id number) to the end of the queue directives. Otherwise if you are trying to say that you want the agent always in the queue, change the line listed above to: member = SIP/(Extension Number) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of satish patel Sent: Friday, December 14, 2007 7:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk Qeueu with static agent Dear all I have asterisk every time my Agent login in queue and useing queue but i want to staticly map that agent in queue so how do it possible and what configuration required for it ??? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org _ Looking for last minute shopping deals? HYPERLINK http://us.rd.yahoo.com/evt=51734/*http:/tools.search.yahoo.com/newsearch/category.php?category=shoppingFind them fast with Yahoo! Search. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.4/1188 - Release Date: 12/17/2007 2:13 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
Re: [asterisk-users] resync linksys SPA9XX config file from Asterisk
Andres wrote: Anyone know the sip header to send to a Linksys to resync it's config file? You will have to set the parameter Auth Resync-Reboot: to NO on the phone so it will not ask for credentials. Or you can use patch for asterisk that enable authorization of outgoing sip notify: http://bugs.digium.com/view.php?id=9896 This is more secure way to notify devices. There are more event in linksys devices for cold and war reboot. -- Best regards, Igor A. Goncharovsky ICQ: 648337 mailto: [EMAIL PROTECTED] ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Steve Edwards wrote: On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? I mean, Seriously? (in a Grey's Anatomy kind of way...) The old syntax was inconsistent -- show manager command vs sip show channels and just plain bad -- for example sip reload should have been reload sip. The new syntax continues down the noun-verb path instead of correcting itself and using verb-noun like most other applications (MySQL, GDB, Oracle, etc.) Then, just to make it worse, now I have to learn which commands somebody (arbitrarily) decided are core and which are not -- for what benefit? Certainly doesn't make MY job easier! Approach the command line like a noob. I want Asterisk to show me something so I'll start the command line with show. I'm not quite sure what I'm doing, so I'll press TAB to see what I can show. Oh, channel looks like what I want. Hmm, too much. Maybe I should have qualified what kind of channel I'm looking for BEFORE the word channel. Here's a suggestion -- stop thinking like a parser and start thinking like a person :) Which makes more sense (at least in English)? 1) show black dogs -- show sip channels 2) black show dogs -- sip show channels 3) dogs black show -- channels sip show 4) show dogs black -- show channels sip 5) black dogs show -- sip channels show 6) dogs show black -- channels show sip Is it too late to fix this for 1.6? Are there going to be Black Dogs in 1.6?? WOOF -- Dog is my co-pilot ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
ok, here is my $0.02... I created a script since I had to
install/update so often and for various reasons...
you can choose to compile automatically or manually...
modify the current release numbers, your repository, and source root...
all else is automated..
which is unloading zap driver, stopping a running asterisk, getting the
current release, untar'ng it and compiling it...
enjoy,
daveC
#!/bin/sh
#
#get_latest_rel.sh
#
# Dave Cantera: [EMAIL PROTECTED]
#
#get the current asterisk release components, put them in our REPOSITORY
#and unpack them in SRC_ROOT
--- Change to suite between these lines --
VER_AST=1.4.16
VER_ZAPTEL=1.4.7.1
VER_LIBPRI=1.4.3
VER_ADDONS=1.4.5
REPOSITORY=/root/tarballs
SRC_ROOT=/usr/local/src
--- Change to suite between these lines --
HTTP_SITE=http://downloads.digium.com;
PUB_DIR=/pub
TARBALL_AST=/asterisk/releases/asterisk-${VER_AST}.tar.gz
TARBALL_LIBPRI=/libpri/releases/libpri-${VER_LIBPRI}.tar.gz
TARBALL_ZAPTEL=/zaptel/releases/zaptel-${VER_ZAPTEL}.tar.gz
TARBALL_ADDONS=/asterisk/releases/asterisk-addons-${VER_ADDONS}.tar.gz
COMPONENTS=${HTTP_SITE}${PUB_DIR}${TARBALL_AST}
${HTTP_SITE}${PUB_DIR}${TARBALL_ZAPTEL}
${HTTP_SITE}${PUB_DIR}${TARBALL_LIBPRI}
${HTTP_SITE}${PUB_DIR}${TARBALL_ADDONS}
echo
echo
echo we are prepared to get the complete current release
echo of asterisk, libpri, zaptel, and addons
echo the tarballs will be placed in our REPOSITORY and
echo then extracted to our SRC_ROOT
echo
echo --- Activity Recap
echo
echo TARBALL REPOSITORY: ${REPOSITORY}
echoSRC_ROOT: ${SRC_ROOT}
echoasterisk tarball: ${TARBALL_AST}
echo libpri tarball: ${TARBALL_LIBPRI}
echo zaptel tarball: ${TARBALL_ZAPTEL}
echo addons tarball: ${TARBALL_ADDONS}
echo
echo -n Are You Ready? Y to procced:
read ANSWER
if [ null${ANSWER} == nullY ]
then
echo
echo -
echo stopping asterisk
echo
echo choose your poison:
echo a) /usr/bin/asterisk -xr stop now
echo b) /etc/init.d/asterisk stop
echo
echo -n which one?
read STOPCMD
if [ null${STOPCMD} == nulla ]
then
/usr/bin/asterisk -r -x 'stop now'
fi
if [ null${STOPCMD} == nullb ]
then
/etc/init.d/asterisk stop
fi
echo
echo -
echo get the current asterisk component releases and put them in
our repository ${REPOSITORY}
# lets go to the repository directory
cd ${REPOSITORY}
for TARBALL in `echo ${COMPONENTS}`
do
echo getting component: ${TARBALL}
#wget ${TARBALL}
done
TARFILES=
asterisk-${VER_AST}.tar.gz
libpri-${VER_LIBPRI}.tar.gz
zaptel-${VER_ZAPTEL}.tar.gz
asterisk-addons-${VER_ADDONS}.tar.gz
echo
echo -
echo unpack the current asterisk component tarballs into our
source root ${SRC_ROOT}
# lets go to the source root directory
cd ${SRC_ROOT}
for TARBALL in `echo ${TARFILES}`
do
echo untar'ng component: ${TARBALL}
#tar xzf ${TARBALL}
done
echo
echo -
echo unloading Zap drivers
# unload the zaptel drivers
ZAP_MODULES=`lsmod | grep zap | awk '{printf(%s,,$4)}' | sed 's/,/
/g'`
for MODULE in `echo ${ZAP_MODULES}`
do
echo unloading zap module: ${MODULE}
#modprobe -r ${MODULE}
done
echo
echo now you are ready to compile at ${SRC_ROOT}
echo
echo -n Shall I continue with the compile? Y?
read COMPILE
if [ null${COMPILE} == nullY ]
then
echo Compiling Zaptel version ${VER_ZAPTEL}
cd ${SRC_ROOT}/zaptel-${VER_ZAPTEL}
make;make; make install
echo Compiling libpri version ${VER_LIBPRI}
cd ${SRC_ROOT}/libpri-${VER_LIBPRI}
make; make install
echo Compiling Asterisk version ${VER_AST}
cd ${SRC_ROOT}/asterisk-${VER_AST}
make; ./configure; make; make install
echo Compiling Asterisk Addons version ${VER_ADDONS}
cd ${SRC_ROOT}/asterisk-addons-${VER_AST}
echo make disabled...
#make; make install
else
echo Ok, compile it yourself!
fi
else
echo Aborted by user
fi
exit
--
Ira wrote:
At 10:33 AM 12/17/2007, you wrote:
At 02:55 AM 12/17/2007, you wrote:
I wonder if there are any major obstacles for upgrading.
Because of your message I tried upgrading to 1.4 again Saturday. That
was the third or fourth time I've tried and the first time it's
lasted more than a few hours before segfaulting and causing me to
step back to 1.2. It seems like I might be staying with 1.4 this time
as 2 days later it's still working. I did find one last deprecated
function in the startup logs and fixed that so I should now be good
for
Re: [asterisk-users] Leading 0 in PRI outbound
Yeah: we are using pridialplan=local - am using AsteriskNOW by the way. Does it require some kind of a patch? for it to understand 'pridialplan' ? My pri intense debug shows: Calling Number (len= 4) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Number not available (67) '' ] [70 0b a1 39 37 38 39 30 39 31 30 31 31] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '9789091011' ] Thanks Rajeev On Dec 19, 2007 3:47 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 18 December 2007 15:22:18 Rajeev Natarajan wrote: We have a PRI line setup on an asterisk box using TE110P. Both outbound and inbound are working fine BUT the provider claims that all our numbers come prefixed with a '0' (in India a 0 prefix indicates long distance) and that could become an issue with local calls. What is pridialplan set to in zapata.conf? This value sets an extra 4 bits in the PRI dialog between you and the telco. And typically, if you have it set to something like 'national', the telco will tell you you have numbers prefixed, even when you don't, because their switch software is written to make the translation. So what most people do (and what works most often) is to set pridialplan to 'unknown', which sets the bit field to all zeros and the number isn't prefixed at all at the telco switch, but simply routed based upon the number sent. -- Tilghman ___ --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] Using MysqlPool Application 1.4
On 12/18/07, Cyril SCETBON [EMAIL PROTECTED] wrote: Hi, Since I've upgraded to Asterisk 1.4 I can't use a MySQL database anymore for select queries :-( I'm using dbquery from MysqlPool Application 1.4 and selecting something from a table returns nothing even if I try to do a query like SELECT 1; Is anyone in the same troubles ? Do you advice me another solution to connect to my database ? app_addon_sql_mysql from asterisk-addons - it works fine for me. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --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 change sendmail return path
19 dec 2007 kl. 01.07 skrev shadowym: Unfortunately that only changes the from field. So if you were to reply to the email that is the one Outlook would use. The receiving mail system looks at the return path in the header of the email to determine if it is valid. serveremail and fromstring do not change that. Again, the return path in the email is set to [EMAIL PROTECTED]. I can easily change mydomain.com in sendmail but cannot figure out how to change asterisk. Sendmail has a notion of trusted users that are allowed to change the envelope sender's address. Your Asterisk process userid propably does not belong to that group. Add it to the group in the wonderfully elegant and simple sendmail configuration and change the mailcommand in voicemail.conf so that you specify another sender. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
19 dec 2007 kl. 04.43 skrev Steve Edwards: On Sat, 15 Dec 2007, Johansson Olle E wrote: I wonder if there are any major obstacles for upgrading. How about the change from a bad command line interface to a really bad command line interface? Steve, While I don't believe the CLI syntax stops you from upgrading, you are joining a very old discussion. Please discuss this on asterisk-dev if you want to re-open it. There's also an open bug in the bug tracker that you can help resolving. The old way was a mess. We had two different systems, one like your old show and one syntax starting with the module name. We had to move forward with only one syntax and decided to go for modulename verb which is not human language-like, but we haven't really clamed that the CLI is a human language parser. Maybe we should go for an avatar approach... -Hello, I'm your Asterisk assistant. What do you want to do today? -Why do you want to reload SIP? Having a bad day, are you? - Are you really sure you want to load the IAX2 module? Don't you prefer meeting your shrink instead? I can schedule a meeting? - Please don't hurt my calls that way, don't stop Asterisk now! I can hear the Allison voices coming out of my system... I do understand the pain with changing the CLI though, I hate to switch from Asterisk 1.0 to 1.2 to 1.4 and trunk and have different commands. Old men have a problem learning they say to me... :-) Thanks for your feedback! /O ___ --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
