Re: [asterisk-users] hint priority with 50 channels
Here is the scenario. My customer has 3 groups each with 10 users. Every user has the users of his group programmed on the snom phone for pickup etc... This works perfectly. Now they want one global button for each group. so if any phone in another group rings then they can pick it up. Why do we put 80 character limits, computers have GB or memory? Loic On Fri, 2008-11-21 at 13:19 -0700, Anthony Francis wrote: Just curious but why would you want to have a lot of devices all have the exact same state information? Philipp Kempgen wrote: Loic Didelot schrieb: I noticed that my hint priority stops working when I add to many extensions/channels. It looks like everything exceeding 80 characters is discarded. By stop working I mean the status is and stays Unavailable. This works exten = *1,hint,SIP/loicvoip1_1IAX2/loicvoip1_1SIP/loicvoip1_1_a1 This does not work: exten = *1,hint,SIP/bla1SIP/bla2SIP/bla3SIP/bla4SIP/bla9SIP/bla5SIP/bla6SIP/bla7SIP/loicvoip1_1IAX2/loicvoip1_1SIP/loicvoip1_1_a1 I tested on several asterisk 1.4 versions like 1.4.21*. Is this a bug or something like working as designed? It's by design. 80 characters is likely to be the limit. Is there another possibility to monitor a bigger number of channels? In Asterisk 1.6 you could build something with Custom hints and DEVICE_STATE(). Philipp Kempgen ___ -- Bandwidth and Colocation Provided 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
On 22 Nov 2008, at 00:06, Michael Collins wrote: Date: Fri, 21 Nov 2008 16:20:28 -0600 From: Terry Wilson [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes Yehavi Bourvine wrote: OK, but I still did not get a reply to my original question: Why using SIP registrar in front of Asterisk and not simply use bare Astersik? can't it handle the load? (remember - in my case it doesn't handle the RTP, only signalling). Can't it handle so much registrations? (I am using realtime DB, it is has any relevance). My experience has shown that using a dedicated registrar for large installs is more effective; it doesn't tie up resources on the Asterisk box with all those registration refreshes, for one. A product built to be a high-throughput standalone registrar will handle the concurrency requirements and perform better. I've looked at doing various things to chan_sip to improve signaling performance (hash tables for call lookups, etc.) I gave up when I realized that the overhead of handling the RTP was so far above the overhead of processing SIP signaling that it didn't really matter much. The only reason I have ever had to use a SIP registrar (OpenSER in my case) was if I needed to load balance calls across multiple asterisk servers. If most of the phones are not separated by a NAT from Asterisk (as would be the case in something like a University network), the registration timeout could be set to a relatively high value w/o causing any problems which would cut down on some of the SIP traffic from registrations. In fact, I just ran some tests using SIPp and w/o any audio, using realtime w/ 10k accounts I can register 100/second while doing 10 calls/second. If you are looking just at registrations every 15 minutes or so, that is 90k devices that could register to asterisk. This was using 1.6.0.1 on my little HP amd64 development box--not anything near the kind of machine that you would probably install in a large installation. Asterisk just gets faster and faster. Some of the it isn't good at x stuff comes from experiences with older releases. In a HA and/or high volume scenario I worry about stuff like this that has been in tree since 1.0 or earlier and is in 1.6, channel.c lines 3825~3828: /* XXX This is a seriously wacked out operation. We're essentially putting the guts of the clone channel into the original channel. Start by killing off the original channel's backend. I'm not sure we're going to keep this function, because while the features are nice, the cost is very high in terms of pure nastiness. XXX */ That's not something I want in my high-end, high-capacity, high-availability production system! Actually that's exactly the kind of comment I _do_ want to see in an opensource platform. It is honest, open and an encouragement to others to think of a better fix. Discourage poor coding, critique the design etc - but please don't discourage this kind of commenting, it is the kind of thing that helps one find a bug _infinitely_ faster that you could without the clue the original author left for you. Tim. ___ -- Bandwidth and Colocation Provided 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] MozIAX - Mozilla IAX2 soft-phone 3sec delay
On 21 Nov 2008, at 21:12, Joseph wrote: Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried DIAX over dial-up connection and the voice quality was acceptable with very little delay. Sounds to me as if you'll need to tweak the audio settings on the eee . DIAX probably does that for you, it might be worth looking in /proc/snd while DIAX is running to see how it configures the audio device, then getting moziax to do the same. I'm surprised you got reasonable response over a dialup connection - which codec are you using ? Tim. ___ -- Bandwidth and Colocation Provided 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] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
Hi Jeff - I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Thanks, Noah ___ -- Bandwidth and Colocation Provided 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] [SOLVED] Re: Upgrade 1.4.19 to 1.6 = segementation fault
Ronald Wiplinger (Lists) wrote: During compiling I have not seen an error, however, when I start asterisk again it ends with: app_morsecode.so = (Morse code) == Registered custom function 'SYSINFO' func_sysinfo.so = (System information related functions) Segmentation fault (core dumped) How can I figure out what is wrong? I removed all modules, which were left from the 1.4 installation and now it works! ___ -- Bandwidth and Colocation Provided 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] Setting up to reveive faxes.
Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. - Noah ___ -- Bandwidth and Colocation Provided 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] Setting up to reveive faxes.
On Sat, 22 Nov 2008, Noah Miller wrote: Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, with 1.2, I use spandsp (or rather the RxFAX application which uses spandsp) to terminate faxes and save them as TIF files, then outside asterisk email them to the appropriate destination... I'm currently looking at moving to 1.4 and this is something that's essential for me... I guess it's time to look deeper into this. Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint priority with 50 channels
Loic Didelot wrote: Why do we put 80 character limits, computers have GB or memory? Loic Asterisk was written in c and they do have to declare how much memory should be reserved for a variable in c, so the programmer arbitrarily chose a number. They may have put some logic or investigation behind it, but thats pretty much it. If you don't like that chose, edit the definition in the source code and then recompile and voila! you have your longer string handling. ___ -- Bandwidth and Colocation Provided 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] Setting up to reveive faxes.
On Saturday 22 November 2008 09:10:39 am Gordon Henderson wrote: On Sat, 22 Nov 2008, Noah Miller wrote: Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, with 1.2, I use spandsp (or rather the RxFAX application which uses spandsp) to terminate faxes and save them as TIF files, then outside asterisk email them to the appropriate destination... I'm currently looking at moving to 1.4 and this is something that's essential for me... I guess it's time to look deeper into this. with 1.6, i'm using this: http://messinet.com/?page_name=AsteriskFAXGateway to do email-fax gatewaying in both directions. i suppose it could also work with 1.4's tx_fax and rx_fax which i think are in the agx-ast-addons package available at http://sourceforge.net/projects/agx-ast-addons/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint priority with 50 channels
Thanks for that idea. That what I had in mind. Now I just need to figure out where to change and test for side effects. Loic On Sat, 2008-11-22 at 08:19 -0700, Anthony Francis wrote: Loic Didelot wrote: Why do we put 80 character limits, computers have GB or memory? Loic Asterisk was written in c and they do have to declare how much memory should be reserved for a variable in c, so the programmer arbitrarily chose a number. They may have put some logic or investigation behind it, but thats pretty much it. If you don't like that chose, edit the definition in the source code and then recompile and voila! you have your longer string handling. ___ -- Bandwidth and Colocation Provided 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] ISDN Cause codes
I have found that the messages are not played as the hangup cause clears down the channel and passed hangup to the other end should I have progress() before the dial command? Robb Martin Smith wrote: Hi Robert, I'd recommend the following options for Dial() so that you corroborate operator messages w/ cause codes: 1. remove R and r - we've found this can supress operator recordings on early audio 2. likewise, remove m to disable MOH Also, check the values of DIALSTATUS to compare to HANGUPCAUSE. Good luck, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Friday, November 21, 2008 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN Cause codes Thanks for the reply Could you be a little more specific? Thanks Robb Martin Smith wrote: Hi Robert, I'd suggest tweaking the Dial() arguments so that you (1) allow early audio, (2) don't force it play ringing to the calling party, and (3) modify any other options to be as relaxed as possible. if you make those changes, you'll start hearing the operator message recordings and those are sometimes easier to reference against the cause codes. Cheers, Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Boardman Sent: Thursday, November 20, 2008 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN Cause codes Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not completing. there are various other codes including 17 busy 34 channel unavaliable and 44 requested channel unavaliable, which add up to another 1%.* * the telco says there is no problem with the line, I'm trying to understand what the problem could be now alot of calls complete OK so I don't think is my configs Any advice would be appriciated Versions asterisk 1.4.21.1 zaptel 1.4.12.1 Robb ___ -- Bandwidth and Colocation Provided 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 ___ -- Bandwidth and Colocation Provided 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] MozIAX - Mozilla IAX2 soft-phone 3sec delay
On 11/22/08 13:29, Tim Panton wrote: On 21 Nov 2008, at 21:12, Joseph wrote: Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried DIAX over dial-up connection and the voice quality was acceptable with very little delay. Sounds to me as if you'll need to tweak the audio settings on the eee . DIAX probably does that for you, it might be worth looking in /proc/snd while DIAX is running to see how it configures the audio device, then getting moziax to do the same. I'm surprised you got reasonable response over a dialup connection - which codec are you using ? Tim. Over dial up, I used GSM; I'll try to install Ubuntu and try it again. Any idea what to look for in snd file? If there is snd file they must have put it somewhere else as it is not in /proc/snd. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- Bandwidth and Colocation Provided 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] hint priority with 50 channels
On Saturday 22 November 2008 10:31:34 Loic Didelot wrote: Thanks for that idea. That what I had in mind. Now I just need to figure out where to change and test for side effects. In main/pbx.c, search for ast_add_extension2 (in my branch, it's at line 1917, but it may vary slightly). The top line of the function has: char hint[AST_MAX_EXTENSION]. Change AST_MAX_EXTENSION to something larger (say, 512) and recompile. BTW, I'm going to make this change in trunk, so starting in 1.6.1, it will no longer be as much of a problem. -- 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] Setting up to reveive faxes.
On Saturday 22 November 2008 09:10:39 Gordon Henderson wrote: On Sat, 22 Nov 2008, Noah Miller wrote: Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, with 1.2, I use spandsp (or rather the RxFAX application which uses spandsp) to terminate faxes and save them as TIF files, then outside asterisk email them to the appropriate destination... I'm currently looking at moving to 1.4 and this is something that's essential for me... I guess it's time to look deeper into this. You missed the part about IP faxes. Yes, you can receive faxes over a Zaptel or DAHDI channel on your system. That feature has not gone away. It is the feature of being able to terminate T.38 faxes that was never in 1.2, isn't in 1.4, but is in 1.6. RxFax has never worked with IP faxes. It requires the use of a Zaptel/DAHDI channel. -- 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] Setting up to reveive faxes.
On Sat, 22 Nov 2008, Tilghman Lesher wrote: On Saturday 22 November 2008 09:10:39 Gordon Henderson wrote: On Sat, 22 Nov 2008, Noah Miller wrote: Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, with 1.2, I use spandsp (or rather the RxFAX application which uses spandsp) to terminate faxes and save them as TIF files, then outside asterisk email them to the appropriate destination... I'm currently looking at moving to 1.4 and this is something that's essential for me... I guess it's time to look deeper into this. You missed the part about IP faxes. Yes, you can receive faxes over a Zaptel or DAHDI channel on your system. That feature has not gone away. It is the feature of being able to terminate T.38 faxes that was never in 1.2, isn't in 1.4, but is in 1.6. OK. We were talking about T.38, and I must have missed that. RxFax has never worked with IP faxes. It requires the use of a Zaptel/DAHDI channel. Er, wrong, and didn't we discuss this some weeks back? Maybe what you need to say is: RxFax will never work reliably with IP faxes. Myself and a colleague have sent several faxes via VoIP in the UK between servers at the end of ADSL lines at opposite ends of the country, and they've worked just fine. It's not something I'd sell or recommend because it's not reliable, but it can work. I also know people who've got FAX machines connected to ATAs VoIP connected to PBXs with ISDN lines and those FAX machines work OK too, so although it's better IP over a LAN than over the public Internet, it's still FAX over IP, as is the case where people are running IAXMODEM on a separate PC to the Asterisk box connected to HylaFax... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint priority with 50 channels
Thanks for the help. Loic. On Sat, 2008-11-22 at 11:41 -0600, Tilghman Lesher wrote: On Saturday 22 November 2008 10:31:34 Loic Didelot wrote: Thanks for that idea. That what I had in mind. Now I just need to figure out where to change and test for side effects. In main/pbx.c, search for ast_add_extension2 (in my branch, it's at line 1917, but it may vary slightly). The top line of the function has: char hint[AST_MAX_EXTENSION]. Change AST_MAX_EXTENSION to something larger (say, 512) and recompile. BTW, I'm going to make this change in trunk, so starting in 1.6.1, it will no longer be as much of a problem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need Recording Solution in Asterisk
One of our client Bank has 900 employees working in different locations. They need to record all internal and external calls. Can any body suggest Call Recording Solution for this requirement. We need to know the Hardware / Bandwidth and all requirements and costing. Few questions first 1. Why are they being recorded (business need)? 2. Does the value of the recording remain constant over time or diminish? 3. What criteria will you be required to retrieve the recording with? 4. Do you expect users to retrieve their own recordings or make requests of a records management operations staff? 5. Does everything need to be on-line or near-line/off-line and do you require a data management and migration solution? 6. Do you need to do word spotting and trend analysis on the content of these recordings (target marketing and customer service analysis typically)? Recording the call is quite easy. Storing it for retrieval which is acceptable to the business under their potentially diverse requirements is the tough part to nail down. There are commercial products like Witness out there which do a good job of this at a premium price. If the business drivers have low impact, you could simply record in asterisk and archive the files with some creative scripting and database work. You said this is a bank so I'm presuming they will have a formal risk analysis methods in place which would guide you through qualifying the requirements. Check out what the IT/CIO folks have to help you out in this manner. -dbc. ___ -- Bandwidth and Colocation Provided 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] MozIAX - Mozilla IAX2 soft-phone 3sec delay
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Joseph a écrit : | Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. | http://moziax.mozdev.org/ | | I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! | The delay is about 2sec or 3sec. and very bad echo. | I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. | | As a comparison I've tried DIAX over dial-up connection and the voice quality was acceptable with very little delay. | MozIAX and DIAX share the same IAX2 implementation (libiaxclient, see http://iaxclient.wiki.sourceforge.net/projects), so I doubt this is the problem. I'm not aware of such delay problems; maybe there is a sound daemon running on the EEE PC. Maybe you could ask on the MozIAX mailing list. Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mandriva - http://enigmail.mozdev.org iEYEARECAAYFAkkoXk8ACgkQuu7Rv+oOo/im4gCfUEimQ33BMHsjyNR4fygwdOBm kP8Anj7Ei+FQNLiKslJepE2hV8xRI6fq =voCP -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided 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] CDR Desgin
I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation that is already overly so. I think it's a mistake to try and think about all the different call scenarios and come up with little tricks for the more complicated ones. There will always be something missed; app_shotgun initiates calls to 100 random numbers and as soon as three or more calls are answered it will start randonly transferring them amongst each other at 2 second intervals. I think it's important to clarify at the outset what a CDR should be. The most fundamental requirement for CDRs is that they accurately record the following pieces of information for EVERY call entering or leaving the system (note every means every and not; channel calls but not peer calls). 1. Destination (aka as A Number) 2. AccountCode (aka as B Number) 3. Call Start Time (answer time), 4. Duration. Of course adding extra information can be very useful and I'm not proposing any fields be removed from the current implementation (although for pity's sake one change that should be made it to use a GUID/UUID for the CDR's uniqueid and save endless confusion). People that really do need verbose or enhanced CDRs to do things like tracking a call's flow as it travels in and out of queues, parking lots etc. would be better off using AMI or the new CEL and not CDRs. At the very least if problems arise with their call flow tracking they will still be able to rely on the accuracy of the CDRs to piece it altogether to work out what's going wrong. My proposal of creating a 1-to-1 relationship between CDRs and Asterisk channels already exsits but somewhere along the line it's going awry. As an experiment, and to actually do something instead of continually moaning about it, I started commenting out the blocks of code in res_featrures.c and sip_channel.c that muck around with the channel CDRs when a transfer occurs. The results of that were that the CDRs for blind and attended transfers actually got better! They're still not quite right but are pretty close with only one CDR on each having a wrong detstination. Regards, Greyman. ___ -- Bandwidth and Colocation Provided 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] presence with polycom DND
You might have to look at writing a forward macro on the server that would be dialed by the DND button - that also changed the device status to busy(via the devstate app?). My guess is that it would be less than 10 lines of dialplan code, but maybe 1.6 only PaulH cfh wrote: hi, I have configured asterisk 1.4.21 to control the presence BLF (hint + watch buddy parameter) of Polycom phones (650,550,330) and it works good. But when I set the phones on Do Not Disturb (DND) on the server there arent sip notifications and the presence doesnt change. On the Polycom configuration I have try to use the server based DND option but i dont know how to use this with asterik. What can i do ? Are there some workaround to use the DND button and the BLF on asterisk? thanks cfh ___ -- Bandwidth and Colocation Provided 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] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
Noah Miller wrote: Hi Jeff - I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users from my recent research Exchange2003 does have a master user that can be given write access to all mailboxes. Exchange2007, though removes the MasterUser capability. # Asterisk/Exchange Voicemail http://blog.lithiumblue.com/2007/07/asterixexchange-voicemail.html # Asterisk 1.6.0 + Exchange 2007 SP1 Unified Messaging http://blogs.technet.com/gclark/archive/2008/10/22/asterisk-1-6-0-exchange-2007-sp1-unified-messaging.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Instant message passing with eyebeam
Asterisk does not support the MESSAGE method. Max Alex wrote: Hi All, I am searching about asterisk IM message passing with eyebeam. but i am not able to send instant message to another registered users. i am working in asterisk 1.4 branch. i have tested within call and without call but there is no message recieved. and every time i got error user not found in eye beam. and in asterisk i got Method is not implemented. Can anybody helps me in this? If any patches are there then please let me know. Thanks in advance!! Thanks, Max Alex Voip Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided 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] Large Asterisk installations (~10, 000 extensions), preferably at universities
Bourvine, So, why won't we save the big bucks we pay them, hire two professionals (who cost less) and support an open source code by ourselves? This way we depend on ourselves only. Thanks, __Yehavi: I remember hearing University of Pennsylvania have been using Asterisk for sometime. I am not certain where I came across that information, but google confirmed it as a fact. And you may need to ask for more details from Digium as they worked together, or call the school. I am relatively certain they would share their experience. The deployment was of 15,000 extensions, just about what you have in mind. Below is some articles. http://www.networkworld.com/news/2007/071707-open-source-voip.html http://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania William 2008/11/21 Grygoriy Dobrovolskyy [EMAIL PROTECTED] 2008/11/21 Yehavi Bourvine [EMAIL PROTECTED] Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations? Thanks! __Yehavi: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello very interesting project you have, however asterisk is not a registry server, i suggest that you use opensips/opense/kamalio for your registrar, from where you dispatch to you asterisk servers, inside a good environment with a controlled network and nice tagged voip flow you could acheve a good results. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/9a8636b6/attachment-0001.htm -- Message: 9 Date: Fri, 21 Nov 2008 09:46:13 -0500 From: Alex Balashov [EMAIL PROTECTED] Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Jason Aarons (US) wrote: Just switching from Nortel to something else may not eliminate hardware/software failures, or prevent those without experience from pushing the enter key at the wrong time. One also has to keep in mind - Asterisk, like any large open-source project, gets a lot more QA, patches and bug fixes than any commercial product sold in the intra-industrial channel (i.e. excluding consumer mass-market stuff) ever will! It has a massive installed base, many users reporting bugs through an open and easy to understand process, and a large community either directly or derivatively involved in contributing fixes and testing code. How much installed base from which to harness that kind of large-scale technical feedback does Nortel have? Avaya? Cisco? Asterisk has by far the best QA mechanism. In terms of potential bugs that impact mission-critical availability, I would feel better using it than any of these black-box, proprietary vendor solutions any day. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Message: 10 Date: Fri, 21 Nov 2008 15:46:59 +0100 From: Philipp Kempgen [EMAIL PROTECTED] Subject: Re: [asterisk-users] A way to run extenrnotify when IMAP events take place... To:
[asterisk-users] SendImage()
SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. Might be a -dev question though. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided 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] MOH Realtime Problem
Someone?? Any idea?? De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sebastian Enviado el: Friday, November 21, 2008 9:09 PM Para: asterisk-users@lists.digium.com Asunto: [asterisk-users] MOH Realtime Problem Hi, I'm having 2 problems: 1) MOH in realtime is not working, I have configured it but never go to look at the database, no warning or error found and I can do a query using realtime and the family from the cli. 2) I have SIP phones via realtime, if I register one of them and a call to a queue comes the call is never delivered to the phone, I have to make a call from the phone so the phone start getting calls from de queue. (RESOLVED) Any ideas?? Ast 1.6.0.1 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How does IMAP notify Asterisk that I've read a message?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I have an Asterisk box sitting between the PSTN and a legacy PBX. I have successfully configured Asterisk to use IMAP for voicemail and have written the necessary script to turn the MWI indicator (via a .call file to the PBX) on and off. I have two issues still outstanding: 1) When the user listens to his voice mail via the phone, it will be announced that the caller is unknown, in spite of the fact that the email headers show the appropriate Callerid(num) information. I can live with that, but I'll eventually need to get it fixed. 2) If I listen to the voicemail using my email client, the MWI on the phone is not turned off, which isn't surprising given that my script needs to be called to generate the .call file. What I don't know is how, exactly, Asterisk is notified that I've listened to my voicemail via email. Does Asterisk poll the server? If so, where is the frequency of the poll set? Can Asterisk be configured to call the script again when the messages are read and the MWI should be turned off? The docs don't say anything about this and I've not found anything in my googling that has given me any leads? I'm currently using Asterisk 1.4.22. Thanks for any information that you can provide. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJKNlyCFu3bIiwtTARAo4XAJwMlW8ylyY1JEu1H1qsth73m2cwzACdFzuI EwIJuEkqSyJdnVRJVeVA5Zk= =DmZw -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided 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] SendImage()
Philipp Kempgen wrote: SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. Might be a -dev question though This is typical of the criticism that has been levelled at Digium time and time and time again - making changes that don't really add any functionality, but break compatibility. I had a hell of a time migrating a couple of systems from 1.2 to 1.4 - so much that I have no plans at all in the near future of migrating them from 1.4 to 1.6. Even the comments made at the time suggesting a parsing tool be provided to point out where changes to dialplan code would be required got a nice idea response, but nothing has been forthcoming. This habit of breaking functionality for limited or no reason, plus making the results from functions far /less/ useful (note my previous complaints about the REALTIME() function) and more difficult to use is the biggest problem with Asterisk bar none. ___ -- Bandwidth and Colocation Provided 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] database queries from extensions.conf
Klaus Darilion wrote: Wolfgang Pichler schrieb: Hi, you yould also use DBQuery (does only support mysql) - take a look at http://www.voip-info.org/wiki/view/Asterisk+cmd+DBQuery (it does also contain a cdr backend to write customzied cdr entries to the database) hi wolfgang! Have you programmed this yourself? Do you know how it compares to MYSQL function and func_odbc? regards klaus regards, Wolfgang Klaus Darilion schrieb: Hi! What is the preferred way to make database lookups from within the dialplan? I only know the MYSQL function from asterisk-addons. Are the other methods too? (e.g. for postgresql, unixodbc) thanks klaus ___ -- Bandwidth and Colocation Provided 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 func_odb only allows a SINGLE database statement Ergo you cannot do Transactions or Multi-statement SQL It is a MAJOR Backstep in DB Access. The MYSQL add-on is the BEST way to access DB from Dial Plan Digium should support and ADD to this rather than non putting a SINGLE mention of it in the last book and making no mention of it at Astricon. With this Add-on, and if DIGIUM would fix the brain dead implement ion of REAL-TIME for Exstensions.conf, things would/could be Soo Sweet. [ Can I get a Amen for having LABELS for steps in exstensions.conf when it is in Real Time ? Why the Heck do I have to use Different Format for Applications in exstensions.conf ] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] database queries from extensions.conf
Klaus Darilion wrote: Hi Jared! Thanks for the info - looks very flexible - you only have to edit 4 configuration files for a simple query :-) just a few questions: The ODBC library is unixodbc? How does it compare to the other solutions in terms of performance? e.g. (I have to make several queries for each call (caller preferences, LNP, LCR...) regards klaus Jared Smith schrieb: On Thu, 2008-11-13 at 15:16 +0100, Klaus Darilion wrote: What is the preferred way to make database lookups from within the dialplan? The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. I also presented on func_odbc at AstriCon, and you can download my presentation: http://www.astricon.net/2008/glendale/web/presentations/DatabaseDriven_JSmith.pdf Quote The preferred method is to use func_odbc, which takes SQL queries and builds custom dialplan functions from them. I've used it quite a bit, and am very happy with it. How can you be VERY HappY with something that allows ONLY single statemts of SQL, ipso-facto you CANNOT do Begin Transaction SQL Statement SQL Statement End Transaction This isSOOO much more limited that the MYSQl add-on ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
