Re: [asterisk-users] dial, answered and then hangup
On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- 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!
On Dec 15, 2007 11:57 AM, Johansson Olle E [EMAIL PROTECTED] wrote: I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 SNIP - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 Hi Olle, It's very simple in my case. I did the install very recently on our company pbx. When asterisk runs, I get a segfault I was not able to debug. It doesnt seem to happen in the same place in the startup each time. If it was after a line that loaded some module, sure, but this didn't seem to be the case. I have no time or knowledge to chase issues like this, so I immediately went back to the latest 1.2. ___ --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!
Hi. The only problem i have is with sending and recieving Faxes. Right now i'm using spandsp an app_rtxfax. This works fine. But there seem to be no spandsp and app_rtx packages in my gentoo. ciao 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] [Zaptel] Why no port to Windos?
On Dec 15, 2007 6:06 PM, Michael Graves [EMAIL PROTECTED] wrote: When an Asterisk appliance and associated phones can compete with a Panasonic KXTG-4000 (or similar) on terms including price, ease of use reliabilitythat's when Asterisk for every grandma, aunt, uncle counsins (who never finished high school) will be viable for the broader home/residential market. The other aspect of this question is that more and more, like computing in the cloud and storage in the cloud, VOIP in the cloud is taking over. How many people now have unlimited dialing on VOIP routers that have replaced phone lines for consumers, giving them similar flexibility? How many on this list have played with services like Grand Central or TringMe? What about these cell phone providers that give unlimited Wifi calling at anyone's home when they have the right router and hotspot? These and many other services are around the corner. Having hardware at home may become a thing of the past for the basic consumer. I love having asterisk in the office and playing with the dialplan, but for those who have no desire to play with technology, I see no future at all in hardware, other than better phones. ___ --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!
I wonder if there are any major obstacles for upgrading. My reasons for not moving to 1.4 : - fear of possible instability problems, my 1.2 servers are rock solid - fear of goofing up with the new way you have to configure asterisk at install time (tell it which modules to build or not build) - no real new functionality i really, REALLY need One feature that would immediately draw attention and would greatly enhance upgrade enthusiasm for a new release would be better fax support. chan_mobile looks nice, would be nifty to be able to use gsm phones, i will probably look into that Just my $0.02 :) ___ --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!
My reasons for not yet upgrading to 1.4: - I have a lot of customisations to app_meetme, which I will need to port to 1.4. I have procrastinated about doing so because of all the SLA stuff that got grafted into app_meetme during the early 1.4 versions. If I had developed the SLA code, I would have made a separate app_sla or res_sla with copies of only those parts of app_meetme that were actually needed (e.g. leaving out DTMF menus, participant announcements, etc.), and left app_meetme to do only real conferencing. - Scare stories about IAX-related lockups in 1.4, due to the new multi-threaded implementation. It looks like the latest versions should have got this sorted, especially with the use of astobj2, but I haven't had time to try it out yet. - So far, 1.2 is doing everything we need, and has been rock solid. Another problem is, when I do move to 1.4, any customisations or new features I do create would still need to be ported to trunk before they would have any chance of making it into Asterisk. This takes time, which is always in short supply, and means that some cool features remain mine only :-( Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Cannot allocate memory
Hello, I have a problem with our asterisk server. (Version 1.4.15 RPM Build running on Fedora Core 7) After 3 day's running, i'v got this message in my logfile: Dec 14 13:14:09 amsec-tk1 asterisk[4321]: VERBOSE[11363]: -- Executing [EMAIL PROTECTED]:1] System(SIP/SN2400-67ec6aa0, echo -e Eingehender Support Anruf am 14.12.2007 um 13:14 durch die Nummer 06 \n /drbd/ams-support-hotline.txt) in new stack Dec 14 13:14:09 amsec-tk1 asterisk[4321]: WARNING[11363]: asterisk.c:820 in ast_safe_system: Fork failed: Cannot allocate memory Reserved memory was about 1,5GB of RAM! After restarting Asterisk, the system is normally running until yet. Is there a memory leak in this version? thanks for answer Thomas ___ --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] Problem in Recording file on Hangup ?????
Hi guys I am facing a problem in recording voice on Hangup. I am using php agi class for this purpose. Currently its voice message is being recording when i used to press 1. For this purpose i am using record_file() function with its respective parameters. Is there any way that i can be able to record my voice message on hangup (don't want to send 1 as DTMF to asterisk.). Can you guys guide me towards its solution. Thanx in Advance -- 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] 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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mail Test
Sorry, I'm doing a mail test since I was not able to send any mails to the mailing list for about a week... 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] Upgrade to Asterisk 1.4 - it's one year's old!
On Monday 17 December 2007 04:17:32 Thomas Stein wrote: The only problem i have is with sending and recieving Faxes. Right now i'm using spandsp an app_rtxfax. This works fine. But there seem to be no spandsp and app_rtx packages in my gentoo. That sounds more like an issue in Gentoo than in Asterisk. -- 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!
17 dec 2007 kl. 10.45 skrev randulo: On Dec 15, 2007 11:57 AM, Johansson Olle E [EMAIL PROTECTED] wrote: I'm kind of interested in the slow uptake of Asterisk 1.4. Between 1.2 SNIP - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 Hi Olle, It's very simple in my case. I did the install very recently on our company pbx. When asterisk runs, I get a segfault I was not able to debug. It doesnt seem to happen in the same place in the startup each time. If it was after a line that loaded some module, sure, but this didn't seem to be the case. I have no time or knowledge to chase issues like this, so I immediately went back to the latest 1.2. I do understand you :-) I hope that after some time, you will try 1.4 again. When you have time, please report the bugs and crashes in the bug tracker. We do read all bug reports and the sum of all reports help us. Sometimes we can't solve the issue based on one bug report, but after a while we see a pattern and can solve the issue based on many - so keep reporting bugs like crashes, please! Thanks for the 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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Tony, Thanks for the feedback! 17 dec 2007 kl. 12.40 skrev Tony Mountifield: - I have a lot of customisations to app_meetme, which I will need to port How about sharing them so we can maintain them in the open source base? :-) - Scare stories about IAX-related lockups in 1.4, due to the new multi-threaded implementation. It looks like the latest versions should have got this sorted, especially with the use of astobj2, but I haven't had time to try it out yet. I hope you get time soon. As always, your input is appreciated. - So far, 1.2 is doing everything we need, and has been rock solid. Great! Another problem is, when I do move to 1.4, any customisations or new features I do create would still need to be ported to trunk before they would have any chance of making it into Asterisk. This takes time, which is always in short supply, and means that some cool features remain mine only :-( Even if you haven't got the time, if you contribute them in 1.4 versions and they're interesting enough we can publish them and ask if there are contributors willing to port them to trunk. Cheers, /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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
17 dec 2007 kl. 14.11 skrev Tilghman Lesher: On Monday 17 December 2007 04:17:32 Thomas Stein wrote: The only problem i have is with sending and recieving Faxes. Right now i'm using spandsp an app_rtxfax. This works fine. But there seem to be no spandsp and app_rtx packages in my gentoo. That sounds more like an issue in Gentoo than in Asterisk. But on the other hand, if people rely on third-party distributions we might want to set up some kind of peer pressure on the maintainers - and possibly identify them so we can support them and speed up their process. /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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Monday 17 December 2007, Tilghman Lesher wrote: On Monday 17 December 2007 04:17:32 Thomas Stein wrote: The only problem i have is with sending and recieving Faxes. Right now i'm using spandsp an app_rtxfax. This works fine. But there seem to be no spandsp and app_rtx packages for 1.4 in my gentoo. That sounds more like an issue in Gentoo than in Asterisk. You're right. But you know 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] Upgrade to Asterisk 1.4 - it's one year's old!
Johansson Olle E wrote: 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. For Asterisk users in countries that use the MFC/R2 protocol on E1 channels, it took a couple of months so we could start testing/using 1.4 because the lack of official support for MFC/R2. But thanks to some users/developers the UniCall channel driver was ported to 1.4 and there are many using it now with Digium cards. Today, there are E1 cards available that have native MFC/R2 support from some companies like DigiVoice (http://www.digivoice.com.br/english.php) , so many users are simply avoiding Digium cards/UniCall and buying this cards. We hope that when 1.6 comes up the company that make the card and maintains the channel driver update it quickly so their users can upgrade faster. So, it's not a problem with 1.4 at all, sometimes the reason is that users just depend on external code. Leonardo ___ --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!
Remco Barendse wrote: - fear of goofing up with the new way you have to configure asterisk at install time (tell it which modules to build or not build) This step is completely optional. If you don't do anything, it will build the same way that Asterisk 1.2 did (i.e. it will build every module that is capable of being built on your system). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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, answered and then hangup
On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) Yes [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- 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 ___ --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!
- Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 Funny, but my results have been different. I was running on 1.2.17 (and on to 22) for a year and had all sorts of lockups. For me, when I switched to 1.4.5 these things went away. I did find some bugs and switched to branch/1.4 in SVN. While some have considered this bleeding edge, I figure since there is no NEW code development, these are all bug fixes. So far, I have yet to find something that broke since I polled SVN every week. I do review the changes and implement as needed. I have yet to have a major problem with the server that was not caused because of my config. Our site might be small in number of connections in real time, but it is much more stable since 1.4 I think if you polled the 1.2 community, not everyone is running 1.2.25. I think that the bugs you know are the bugs you love. I don't know if it is just 1.4 but I think that anything past version 1.2.X is considered dangerous. I also wonder if because 1.2 had such success, that there are many who use this code who are not programmers and have trouble diagnosing Open Source bugs. Not that they need to be, but if that were me, it would change my opinion of the code and the support given if I didn't know how to debug a C program. Just some thoughts. Thanks to the team for their hard work on 1.4. My experience with 1.4 makes me hunger for 1.6 a short time after release. ___ --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!
All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. Just a thought for 1.6 (and maybe a backport to 1.4 and should have been in 1.2) What if the warning messages about deprecated functions were able to be tracked in a separate file. I can see on some busy machines that these warning messages get lost. I thought I had all of them handled on our dial plan, but learn 4 new locations I was using old functions just last month. This way, when the users look to upgrade, they change the reporting location for these commands, and then check them in a month to see what they need to fix in the dial plan. Just a thought. ___ --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
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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Tony Plack wrote: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. Just a thought for 1.6 (and maybe a backport to 1.4 and should have been in 1.2) What if the warning messages about deprecated functions were able to be tracked in a separate file. I can see on some busy machines that these warning messages get lost. I thought I had all of them handled on our dial plan, but learn 4 new locations I was using old functions just last month. This way, when the users look to upgrade, they change the reporting location for these commands, and then check them in a month to see what they need to fix in the dial plan. In logger.conf: warning = warning,error Maybe it's worth to add it in logger.conf.sample and enable by default? Regards, Atis ___ --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] VoIP service providers/PSTN termination points
I use Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com On 12/16/07, Benjamin Jacob [EMAIL PROTECTED] wrote: Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
phil, I think you are on to it... the best path is to load a new system up with 1.4.x and port your existing dialplan over, test it out, lock it down and then roll it out... I've worked as a UNIX system integrator for 20+ years, worked with open source and custom developed C/C++ code, Ada, and a slew of developers (150 developers at two gov't projects).on new projects, developers don't usually provide enough info in the first releases to even install the product. you always have a gotcha. upgrades are better by far but still lack those things that developers take for granted. developers conceived the idea and they have been talking about them for months... to the integrators, the release is new and that is when the difficulty arises. as an integrator, we are charged with making the product stable in the target environment whereas developers are charged with making the product stable in their development environment... it is two different scenarios. when an integrator sets up a test/QA environment, things that the developers never invisioned come to light. then it is a find, fix, retest cycle until all is well. it is time consuming but well worth the effort as your support/help desk calls are greatly reduced... so now that I am talking about this, perhaps I should offer a migration/integration/test lab service :)... since I've been through it a hundred times... daveC Phil Knighton wrote: 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 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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On 12/15/07, Johansson Olle E [EMAIL PROTECTED] wrote: 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! We have switched to 1.4 some half year ago, and main motivation was some stability issues with 1.2 (and few new features), so having 1.4 for us means - we're actually having support - we can post bugs to Mantis, and got them solved. Our migration is not yet completely over, last step is getting rid of AgentCallbackLogin, that we plan to do in beginning of next year. However 1.4 since release have had some serious changes that blocked our planned upgrades - for example some memory corruption that raised between 1.4.10 and 1.4.12 that was very hard to track down. This shows that having 1.4 in bugfix-only state is not actually working that good - we have to test each new release very carefully. In total 1.4 have helped us to get rid of twice-per-week crashes we experienced on 1.2, so i would call it more stable than 1.2. 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] dial, answered and then hangup
Rilawich, We use a TDM400P here in the UK, and if you set hanguponpolarityswitch=yes in zapata.conf, we get the same result. I think it is country specific, but try switching this to no and see what happens. This cured our problems. I have a note in zapata.conf (not sure if its from the release confs or from someone internal here) that says Fatal on TDM400P if set to yes. Good Luck Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: 17 December 2007 14:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial, answered and then hangup On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) Yes [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- 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 ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Phil Knighton wrote: 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 Agreed. Given that our group has many 1.2 versions working well on CentOS 3.x boxes, and that 1.4 requires either 4 or 5, your option of starting all over is about all that will work. Also, given that the few in the group that HAVE migrated, have now uncovered a new issue that I am sure isn't unique, where changes made from 1.4.13 to 1.4.15 cause a macro related to ENUM to fail. Smarter heads than I have so far been unable to uncover the cause. Even for those who don't place our business in the hands of the whims of Asterisk, there are few reasons to make the change simply to have the latest and greatest? John Novack -- 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] VoIP service providers/PSTN termination points
Same here - Gafachi has been great. Decent rates, very stable and great voice quality. I use Gafachi.com http://Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com http://www.gafachi.com On 12/16/07, *Benjamin Jacob* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere http://www.iconnecthere.com http://www.iconnecthere.com Vonage http://www.vonage.com Teliax http://www.teliax.com I found something known as Inphonex http://www.inphonex.com. These had the cheapest rates and quite a good coverage too. Anyone with experience on this one? I am looking at a combination of decent prices and good quality. Any other suggestions or ideas welcome too. TiA - Ben. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday @12 PM EST VOIP Users Conference + Aus/NZ/India/Japan conference event
Hi, Kerry Garrison from Fonality will be with us live to address the Trixbox so-called phone home script issue. I'm going to try to have something about the year 2007 in review for any and all VOIP and Asterisk-related events, so if anyone wants to report on what they've been doing in 2007, you're welcome to chime in. We're also talking about doing a second conference call beginning next year to be held weekly some time between 09:30-11:30 UTC for callers in Australia, China, Japan, New Zealand, India, Pakistan, etc. If you are interested in this, please let me know you day and start time preference. This can be done as long as there are enough callers. As usual, the pre and post conference times are also open for any and all discussion. r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk: Call Flow
I'm slowly working on AstSee, found at http://www.astsee.com - Sorry for the .commercial domain name, AstSee is currently free! but it has not been ported to Windows yet. Moj bilal ghayyad wrote: Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
17 dec 2007 kl. 15.26 skrev Tony Plack: - Not enough reasons to upgrade, since 1.2 really works well - Just a bad karma for 1.4 Funny, but my results have been different. I was running on 1.2.17 (and on to 22) for a year and had all sorts of lockups. For me, when I switched to 1.4.5 these things went away. I did find some bugs and switched to branch/1.4 in SVN. While some have considered this bleeding edge, I figure since there is no NEW code development, these are all bug fixes. So far, I have yet to find something that broke since I polled SVN every week. I do review the changes and implement as needed. I have yet to have a major problem with the server that was not caused because of my config. Our site might be small in number of connections in real time, but it is much more stable since 1.4 I think if you polled the 1.2 community, not everyone is running 1.2.25. I think that the bugs you know are the bugs you love. I don't know if it is just 1.4 but I think that anything past version 1.2.X is considered dangerous. I also wonder if because 1.2 had such success, that there are many who use this code who are not programmers and have trouble diagnosing Open Source bugs. Not that they need to be, but if that were me, it would change my opinion of the code and the support given if I didn't know how to debug a C program. Just some thoughts. Thanks to the team for their hard work on 1.4. My experience with 1.4 makes me hunger for 1.6 a short time after release. Please observer that this man was not bribed by the Asterisk Developer Team :-) Thanks a lot! More postive and negative feedback is appreciated. /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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
17 dec 2007 kl. 15.42 skrev Tony Plack: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. Just a thought for 1.6 (and maybe a backport to 1.4 and should have been in 1.2) What if the warning messages about deprecated functions were able to be tracked in a separate file. I can see on some busy machines that these warning messages get lost. I thought I had all of them handled on our dial plan, but learn 4 new locations I was using old functions just last month. This way, when the users look to upgrade, they change the reporting location for these commands, and then check them in a month to see what they need to fix in the dial plan. Just a thought. We could certainly consider adding a new logger channel for this. Thanks for the feedback and the suggestion! /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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
However 1.4 since release have had some serious changes that blocked our planned upgrades - for example some memory corruption that raised between 1.4.10 and 1.4.12 that was very hard to track down. This shows that having 1.4 in bugfix-only state is not actually working that good - we have to test each new release very carefully. Hmm. That's important feedback. Release testing has been a topic for discussion for a long time and it's very hard to get done in an open source community. We have to come back to that later on and see what to do. In total 1.4 have helped us to get rid of twice-per-week crashes we experienced on 1.2, so i would call it more stable than 1.2. That's important news. Thanks for the 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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
Agreed. Given that our group has many 1.2 versions working well on CentOS 3.x boxes, and that 1.4 requires either 4 or 5, your option of starting all over is about all that will work. I would like to know a bit more on why Asteirsk 1.4 means that you have to upgrade Centos? (obviously not a Centos user here :-) ) Also, given that the few in the group that HAVE migrated, have now uncovered a new issue that I am sure isn't unique, where changes made from 1.4.13 to 1.4.15 cause a macro related to ENUM to fail. Smarter heads than I have so far been unable to uncover the cause. ...but has of course reported it to the bug tracker, right? :-) Even for those who don't place our business in the hands of the whims of Asterisk, there are few reasons to make the change simply to have the latest and greatest? Agreed. There's no need for us revenue-wise to push the user base forward, since the revenue for Open Source licensing is zero. But there's always a need to understand the user base and see what we can do to help them. On the wishlist are maintainers of old versions and a test team, but that's not very sexy roles for Open Source contributors, it requires special souls who love maintaining a code base and working with the community, or just get their kicks out of testing. People that get no thrills out of being coding wizards, famous specialists focusing on adding new buggy code... :-) 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
Re: [asterisk-users] Trixbox Phones Home
On Sun, Dec 16, 2007 at 10:27:36PM -0600, Than Taro 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. Ok, then I *didn't* misread the advisory. Yes: who ever thought that *retrieving commands to execute in a privileged fashion from an non-authenticated remote source* was a pretty neat idea? *This* is the thing for which Fonality should be hoist, not the phone home aspect, per se. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Witty slogan redacted until AMPTP stop screwing WGA ___ --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 call interrupted after 64 seconds
Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger. ___ --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!
Hi, To summurize, it seems that one thing preventing people from upgrading is the lack of an upgrading tool : somehow, it should be possible and easy to : - install 2 different versions of Asterisk on the same hardware, - interactively translate config files from one version to another - load balance between them. The lack of incentive to move is another problem that should be kept apart from ease of upgrading. And a third type of issue is that some features are missing in 1.4 version. Regards ___ --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] Calling Party Category Field
For the past month I've been having trouble dialing certain numbers. We have Asterisk 1.4.14, Zaptel 1.4.7, Libpri 1.4.2 on a CentOS 5 server. We are using PRI on a TE110P card with a provider called Alestra in Monterrey, Mexico. There are some numbers that whenever we dial them we always get a busy tone. These numbers do not all belong to the same provider, but they all do not belong to Alestra. In the CLI Asterisk says that the number is not available, like the number does not exist. After some testing the provider has told us that the problem is on our side and gave this explanation: Whenever we dial one of those numbers Asterisk is sending the following: 52 Calling Party Category Field 0x00 When it should be sending: 52 Calling Party Category Field 0x0a As they explain it our server is sending the wrong signal and that is causing the other side to drop the call. Where can I check on this? Is it possible to change this behavior? Here is the relevant part of the config: /etc/zaptel.conf # Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 span=1,0,0,ccs,hdb3 #,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = mx defaultzone=mx /etc/asterisk/zapata.conf language=es usecallerid=yes callwaiting=no callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=no echotraining=yes rxgain=0.0 txgain=0.0 immediate=no context=e1-incoming accountcode=Alestra group=1 switchtype=euroisdn callerid=asreceived signalling=pri_cpe pridialplan=unknown faxdetect=both channel=1-15,17-31 Any ideas on how to solve this problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 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] SIP call interrupted after 64 seconds
Can you post the part of your dialplan which causes this behaviour ? On Dec 17, 2007 11:19 PM, Roger Schreiter [EMAIL PROTECTED] wrote: Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? My os is Suse-Linux 10.2. Thanks for any hints! Roger. ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
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. 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] 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? ___ --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
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 -- 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? ___ --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!
17 dec 2007 kl. 18.57 skrev Olivier: Hi, To summurize, it seems that one thing preventing people from upgrading is the lack of an upgrading tool : somehow, it should be possible and easy to : - install 2 different versions of Asterisk on the same hardware, - interactively translate config files from one version to another - load balance between them. Ok, I see what you mean. The lack of incentive to move is another problem that should be kept apart from ease of upgrading. Absolutely. And a third type of issue is that some features are missing in 1.4 version. That's something I would like to know a bit more about. Thanks! /Olle ___ --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!
17 dec 2007 kl. 19.33 skrev Ira: 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. That makes me very happy to hear! And you're proving that one can run 1.2 and 1.4 with the same configuraiton files. But not the config files written with 1.0 syntax. Thanks! /Olle ___ --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
17 dec 2007 kl. 18.49 skrev Roger Schreiter: Hi, some months ago, I had the problem with an asterisk-1.4.x- Version, that some calls (but not all) were interrupted 64 seconds after connect (a call limit of 86400 seconds was installed using the S()-parameter). It was just a test machine, and later, I switched to callweaver, and the problem had gone. Thus, I never investigated this problem. Now, I upgraded a machine for production use to asterisk-1.4.8, and do encounter the same problem. I have other asterisk machines running, using the same dialplan, without this problem. Did anyone else observe this strange behaviour of calls ending after 64 secondes of uptime? There is a hidden reason somewhere and you need to add verbose logging to your Asterisk, maybe also debug logging so that you can find out what's going on - where the call fails. With the log files, it's often very simple for a trained eye to spot what goes on. It seems like some kind of signalling problem as it is kind of close to the SIP timeouts. If you think it is a bug, don't hesitate to file a bug report and add your log output with verbose set to 4 and debug set to 4, sip debug also turned on! /Olle ___ --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!
Olle E Johansson wrote: *snipped But on the other hand, if people rely on third-party distributions we might want to set up some kind of peer pressure on the maintainers - and possibly identify them so we can support them and speed up their process. /O that is a very important, 'so we can support them' part. all i have experienced so far is the 'peer pressure' part, and frankly it tends to leave a bad taste in my mouth. just my -0.02, as i am less than broke. ___ --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
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 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] Upgrade to Asterisk 1.4 - it's one year's old!
On Saturday 15 December 2007 08:42, Steve Totaro wrote: Johansson Olle E wrote: 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. When Digium starts using 1.4 in ABE then I would consider using it in a production environment. All I ever hear is soon, and I have heard that for months if not the whole year. Until Digium itself is comfortable selling and supporting this version, then neither am I. There is exactly one feature left that is still in testing, relating to the automatic detection of hardware in the GUI. Other than that one issue, ABE version C.1.0 is ready to go. Also, note that for existing users of Business Edition, builds of C have been available in the software portal since August. Version C has also been shipped in the Asterisk Appliance. So yes, while Digium isn't selling ABE C as a standalone product yet, it is supporting it in a commercial environment. I hope to hear of your successful conversion to 1.4 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] 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 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!
I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. -Original Message- From: Atis Lezdins [mailto:[EMAIL PROTECTED] Sent: Monday, December 17, 2007 6:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old! On 12/15/07, Johansson Olle E [EMAIL PROTECTED] wrote: 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! We have switched to 1.4 some half year ago, and main motivation was some stability issues with 1.2 (and few new features), so having 1.4 for us means - we're actually having support - we can post bugs to Mantis, and got them solved. Our migration is not yet completely over, last step is getting rid of AgentCallbackLogin, that we plan to do in beginning of next year. However 1.4 since release have had some serious changes that blocked our planned upgrades - for example some memory corruption that raised between 1.4.10 and 1.4.12 that was very hard to track down. This shows that having 1.4 in bugfix-only state is not actually working that good - we have to test each new release very carefully. In total 1.4 have helped us to get rid of twice-per-week crashes we experienced on 1.2, so i would call it more stable than 1.2. 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] 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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
shadowym wrote: I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. Digium uses an Empirix Hammer (which is an actual product, not just a codename) to test Asterisk Business Edition and verify that it will handle the call loads and scenarios we sell it for. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ --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] VoIP service providers/PSTN termination points
At 10:02 12/17/2007, mail-lists wrote: Same here - Gafachi has been great. Decent rates, very stable and great voice quality. I use Gafachi.com http://Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com http://www.gafachi.com Triple Ditto for Gafachi. ___ --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] [Zaptel] Why no port to Windos?
On Mon, 17 Dec 2007 11:18:08 +0100, randulo wrote: On Dec 15, 2007 6:06 PM, Michael Graves [EMAIL PROTECTED] wrote: The other aspect of this question is that more and more, like computing in the cloud and storage in the cloud, VOIP in the cloud is taking over. How many people now have unlimited dialing on VOIP routers that have replaced phone lines for consumers, giving them similar flexibility? How many on this list have played with services like Grand Central or TringMe? What about these cell phone providers that give unlimited Wifi calling at anyone's home when they have the right router and hotspot? These and many other services are around the corner. Having hardware at home may become a thing of the past for the basic consumer. I love having asterisk in the office and playing with the dialplan, but for those who have no desire to play with technology, I see no future at all in hardware, other than better phones. That is a significant insight. Better phones...this is a very nice idea. I wonder if anyone has seriously considered the possibilities. The hardware world, at least with respect to IP phones, seems to be in a rut. Expect for G.722 I can't think of any significant improvement in recent times. Michael -- Michael Graves mgravesatmstvp.com o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ --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 - testing
17 dec 2007 kl. 21.00 skrev shadowym: I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. Then we must have different opinions on how an Open Source project works. Digium contributes a lot every day with a large team of developers and give the community this work - new code and bug fixes - for free. As a community developer I contribute my own time and time paid for by customers who contract me. The community is the power of Asterisk and what makes Asterisk what it is. I don't know who whoever is. If you where talking about the Business Edition, you would be right. And in fact, Digium has got a test team for that. And, as an additional plus, all the things they find are fixed in the Open Source edition. I have not seen any bug reports coming from Fonality - at least not any bug reports or patches that I can trace from that source. I do hope that they want to join the community so that the Open Source version of Asterisk can benefit from their extensive tests! Our problem is that very few in the community test beta releases or development code. I want to send a big thank you to all that do, you are very important in this process. And for those of you who want to join, go to www.asterisk.org and find instructions on how to download development code for testing. Join the whoever tests this stuff group today :-) Thank you for your thoughts on this matter! /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
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Mon, 2007-12-17 at 12:00 -0800, shadowym wrote: I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. This hammer of which you speak is a commercial program from Empririx, part of their Hammer line of VoIP testing products.[1] Just to be fair and honest, Digium has a copy of the Empirix Hammer software and uses it to test Asterisk. They also spend countless hours testing Asterisk in other ways as well. Part of the problem of testing comes from high number of combinations of different components that must be tested. Just testing calls between the three most common channel drivers (SIP, IAX2, and Zap) involves nine tests at a minimum: SIP-SIP SIP-Zap SIP-IAX2 IAX2-IAX2 IAX2-SIP IAX2-Zap Zap-Zap Zap-SIP Zap-IAX2 Obviously, within each of those tests, there are lots of different options that could be tested as well (such as methods for sending DTMF). I've offered to start pulling together a community-driven set of tests that we can automate and run against Asterisk on a regular basis, but so far nobody has offered up any help in this regard, and I've been busy with other things (like teaching Asterisk training classes) that I haven't had any time to devote to it myself. I'm hoping to be able to start working on a testing framework sometime in January, as long as I don't get too many other things put on my plate between now and then. [1] http://www.empirix.com/products-services/voip_and_ims.asp --- 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] Upgrade to Asterisk 1.4 - it's one year's old!
Olle E Johansson [EMAIL PROTECTED] writes: But on the other hand, if people rely on third-party distributions we might want to set up some kind of peer pressure on the maintainers - and possibly identify them so we can support them and speed up their process. Third-party distributions are very important, and Asterisk has for various reasons done relatively badly there. Fedora still doesn't have Asterisk, but does have CallWeaver. Asterisk isn't even available in the most popular extra repositories, but only in ATrpms, my least favourite of the larger repositories. Hey, I just discovered https://admin.fedoraproject.org/pkgdb/packages/name/asterisk Brilliant! I hope it gets in soon! It has a proper init script too, and it's split in subpackages, and, and... Thanks a lot! /Benny ___ --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] Off topic...AOCN wanted
For those CLECs out there, if you know of a contract AOCN that you have personal experience with and would recommend, please reply. For those who don't know what an AOCN is, please delete this message. Bruce Komito WPTI Telecom (775) 236-5815 ___ --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] Music On Hold
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? thanks Eric ___ --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] Automatic tests (was Re: Upgrade to Asterisk 1.4 - it's one year's old!)
On 12/17/07, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-12-17 at 12:00 -0800, shadowym wrote: I do wish Digium or whoever tests this stuff had a more reliable way of testing software releases rather than relying on feedback from the community. Fonality, for example use what they call a hammer which sounds to me like a bunch of servers running various stress tests on the software to try break it. This hammer of which you speak is a commercial program from Empririx, part of their Hammer line of VoIP testing products.[1] Just to be fair and honest, Digium has a copy of the Empirix Hammer software and uses it to test Asterisk. They also spend countless hours testing Asterisk in other ways as well. Part of the problem of testing comes from high number of combinations of different components that must be tested. Just testing calls between the three most common channel drivers (SIP, IAX2, and Zap) involves nine tests at a minimum: SIP-SIP SIP-Zap SIP-IAX2 IAX2-IAX2 IAX2-SIP IAX2-Zap Zap-Zap Zap-SIP Zap-IAX2 Obviously, within each of those tests, there are lots of different options that could be tested as well (such as methods for sending DTMF). I've offered to start pulling together a community-driven set of tests that we can automate and run against Asterisk on a regular basis, but so far nobody has offered up any help in this regard, and I've been busy with other things (like teaching Asterisk training classes) that I haven't had any time to devote to it myself. I'm hoping to be able to start working on a testing framework sometime in January, as long as I don't get too many other things put on my plate between now and then. Have you seen testing framework we created? I posted it some time ago, but unfortuneately i didn't get much feedback. It might be not so intuitive to configure, but it really helped us to find memory corruption problems under high load. Please see: http://lists.digium.com/pipermail/asterisk-users/2007-November/200429.html 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] Off topic...AOCN wanted
On Monday 17 December 2007 16:19, Bruce Komito wrote: For those CLECs out there, if you know of a contract AOCN that you have personal experience with and would recommend, please reply. For those who don't know what an AOCN is, please delete this message. I know what an AOCN is, but please use the -biz list in the future for these types of queries. That is what it is there for. -- 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] Echo - when pressing digits
I'm units Digium ATA adapter and Sipura adapter. After I dial a number and get connected when I any digit, I can here the echo. Is there a way to cancel DTMF echo? -- #Joseph ___ --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] BLF trouble
Hello, I have some trouble with the BLF indicator. I have two phones that use the same hint: 13 = hint,1,SIP/phone13SIP/phone13-wlan This works great from the asterisk side, but it seems the status change is too quick for the attached Grandstream-phones. When I ring the extension the hint changes to Ringing. The Grandstream blinks. Great. Now, when someone picks up one of the phones the hint changes to InUse+Ringing and half a second or so later changes just to InUse. It seems the Grandstream-phones drop the later, so they signal ringing for the whole phone call. Is there some way to delay the updates between status changes for a hint? Or does somebody know, how I can make this work? Another thing that doesn't work (this time from the asterisk side): The hints aren't updated for outgoing phone calls. A calls B. Hint B gets updated to Ringing and then InUse, but A stays in state Idle. How do I convince asterisk to update the outgoing phone hint as well? And then a last thing. This is more like a wishlist item: I would like to program a call-forward feature in the dialplan. To give the user some kind of visual feedback that they don't forget the call-forwarding when they return, I would like to light up some BLF light on the Grandstream phones (GXP-2000 and GXP-2010). Is this somehow possible? The phones have four options for each button: Speed dial (doesn't light up the LED at all), Asterisk BLF (which shows the status of the hints in the dialplan), Presence Watcher (I have no idea what this does) and Eventlist BLF (I don't really know what this is, but it seems to be related RFC 4662, which does not seem to be implemented in Asterisk). Does anybody have an idea how I can toggle a light on the phone? I'm using Asterisk 1.4.19 with bristuff-0.4.0-test4-xr3 (Xorcom version). Thanks, Lars -- We're not lost. We're locationally challenged. -- John M. Ford ___ --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
Lars Bensmann wrote: I would like to program a call-forward feature in the dialplan. To give the user some kind of visual feedback that they don't forget the call-forwarding when they return, I would like to light up some BLF light on the Grandstream phones (GXP-2000 and GXP-2010). Is this somehow possible? The phones have four options for each button: Speed dial (doesn't light up the LED at all), Asterisk BLF (which shows the status of the hints in the dialplan), Presence Watcher (I have no idea what this does) and Eventlist BLF (I don't really know what this is, but it seems to be related RFC 4662, which does not seem to be implemented in Asterisk). Does anybody have an idea how I can toggle a light on the phone? I'm using Asterisk 1.4.19 with bristuff-0.4.0-test4-xr3 (Xorcom version). Bristuff should have a Devstate() application. show application Devstate http://www.das-asterisk-buch.de/stable/snom-leds.html#snom-leds-custom Grüße, 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 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dial, answered and then hangup
Phil, Thanks. The problem gone if it set to no. I want to know what is Polarity Reversal. I can't find it in the web. On Dec 18, 2007 12:02 AM, Phil Knighton [EMAIL PROTECTED] wrote: Rilawich, We use a TDM400P here in the UK, and if you set hanguponpolarityswitch=yes in zapata.conf, we get the same result. I think it is country specific, but try switching this to no and see what happens. This cured our problems. I have a note in zapata.conf (not sure if its from the release confs or from someone internal here) that says Fatal on TDM400P if set to yes. Good Luck Phil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rilawich Ango Sent: 17 December 2007 14:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial, answered and then hangup On Dec 17, 2007 4:32 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Dec 17, 2007 at 11:34:42AM +0800, Rilawich Ango wrote: Below is the log I got. It seems related to Polarity Reversal. --zapata.conf-- ;answeronpolarityswitch=yes hanguponpolarityswitch=yes --full log-- [Dec 15 19:35:35] DEBUG[2195] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM oi_systemalias WHERE alias = '2272' [Dec 15 19:35:35] VERBOSE[2195] logger.c: No Realtime Matches Found. [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:2 ] GotoIf(SIP/114-b7d06198, 1?:hu_trunkhk) in new stack [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:3 ] Dial(SIP/114-b7d06198, Zap/g0/2272|25) in new stack [Dec 15 19:35:35] DEBUG[2195] dsp.c: dsp busy pattern set to 0,0 [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Dialing '2272' [Dec 15 19:35:35] DEBUG[2195] chan_zap.c: Deferring dialing... [Dec 15 19:35:35] VERBOSE[2195] logger.c: -- Called g0/2272 [Dec 15 19:35:38] VERBOSE[2195] logger.c: -- Zap/1-1 answered SIP/114-b7d061 98 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Ignore switch to REVERSED Polarity on channel 1, state 6 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 1: channel 1, state 6, pol= 1, aonp= 0, honp= 1, pdelay= 600, tv= 864361 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal detected and now Han ging up on channel 1 [Dec 15 19:35:41] DEBUG[2195] chan_zap.c: Polarity Reversal event occured - DEBU G 2: channel 1, state 6, pol= 0, aonp= 0, honp= 1, pdelay= 600, tv= 864361 Answer and then an immediate hangup? (as signalled by the provider) Yes [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Hungup 'Zap/1-1' [Dec 15 19:35:41] VERBOSE[2195] logger.c: == Spawn extension (internal, 922720 000, 3) exited non-zero on 'SIP/114-b7d06198' [Dec 15 19:35:41] VERBOSE[2195] logger.c: -- Executing [EMAIL PROTECTED]:1] Hangup (SIP/114-b7d06198, ) in new stack On Dec 17, 2007 10:29 AM, dave cantera [EMAIL PROTECTED] wrote: rilawich, can you post the CLI output so we can see what is going on? from the exten, it is doing exactly what you tell it to do... dial then hangup daveC ___ --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 -- 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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP service providers/PSTN termination points
Thanks good ppl! Doug wrote: At 10:02 12/17/2007, mail-lists wrote: Same here - Gafachi has been great. Decent rates, very stable and great voice quality. I use Gafachi.com http://Gafachi.com and have good quality with no minimum requirements. Try them at www.gafachi.com http://www.gafachi.com Triple Ditto for Gafachi. ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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
Lars Bensmann wrote: Hello, 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. Grandstream are aware of this and working on a solution. ___ --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 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 -- Godson Gera, http://godson.in ___ --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
