Re: [asterisk-users] ATS X10001P
On Tuesday 18 September 2007 00:51:47 Kevin Kiely wrote: Per the earlier recommendation, I picked up one of the ATS X10001P to evaluate. I was able to configure the LAN for access, however, I don't see where to enter the sip credentials. I have accessed the web interface with root/test and don't see any sip configuration information. I also accessed via Telnet and see more info but no place for the realm or Sip credentials. Am I missing something? -Original Message- http://asterisk.drunkcoder.com/hacks/ats-config/ That is precisely why I created the configuration menu which you see on my site. It is nothing more than a frameset with links to your local phone website. You're welcome to use the frameset as I've published it online, or you may follow the link and download the html/CGI combination frameset for use internally. Note that the frameset exists only for your convenience; my site is in no way accessing your phone and the phone does not need a public IP for you to use the frameset. -- Tilghman ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Useful GUI? [Was: Why does everyone seem to dislike *now?]
s/Trixbox/FreePBX/g Please, Trixbox is a distro, the GUI is FreePBX. Another option might be Destar. Google it up. On 9/18/07, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ken D'Ambrosio wrote: Are there any Asterisk GUIs out there that actually parse the data files, themselves, instead of having some sort of metadata middle-man, which leads to said overwriting? I mean, I, personally, love the CLI -- always have been a fast typist -- but I also know the CLI would scare the living bejeepers out of my boss if/when I try to push hard on an Asterisk solution. What I'd prefer is: Pretty sure that AsteriskNow is reading as well as writing. Also, murf made some changes to clean up rewriting the other day (blank comment lines now get retained or something). - The chance to do CLI stuff as I see fit, BUT - the ability to let users -- even administrative users -- use a GUI, without messing up my beautiful config files. :) I'd say that unless you get a race condition (i.e. GUI reads files, you save your changes from CLI, GUI writes out changes), you should be sweet now, although someone who uses AsteriskNow should be able to confirm/deny. Is this a pipe dream, or is there a GUI out there that might actually do the job? - From what I've heard, AsteriskNow is shaping up pretty nicely. There are options for things like TrixBox too - i.e. the custom extensions.conf stuff, but you need to remember that the machine is running TrixBox and not change the base extensions.conf. This used to be ok because if the extensions.conf-custom (or whatever the filename is) only appeared on machines which had TrixBox. I've lately seen a few machines where the extra config files exist but TrixBox is not running (i.e. someone copied /etc/asterisk from a TrixBox machine). I actually put my extensions.conf stuff into a generate.php file which writes out the extensions.conf file with parameters supplied by the customer stored in separate files. So our software is doing the same thing (overwriting configs) but I don't want the users changing settings too much. I guess this is probably pretty similar to the TrixBox idea but I haven't had a look at how that works under the hood. I think AsteriskNow also lets you edit configuration files from the web page. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG7zliDQNt8rg0Kp4RAmKTAJ9mqmG/j0dw88L0N4g/4R1FFH0KCwCfQKMJ lBE5riG5KZ038I30E3R7liA= =/xas -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Issue with Asterisk realtime
Dear folks, I'm using * realtime with no problems on most of the systems i've setup but rarely i confront this problem that the asterisk doesn't load from database when the systems rebooted and i have to reload it manually or restart it, but it would work fine afterward, no problem how many times you stop and start the *. It seems, there is a missequence of deamon loading at boot time but i have no clue which deamons! Im using FC5, MySQL5, Asterisk 1.2.18 Any help would be highly appreciated. --- M. Shokuie Nia. _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Useful GUI? [Was: Why does everyone seem to dislike *now?]
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Diego Iastrubni wrote: s/Trixbox/FreePBX/g Please, Trixbox is a distro, the GUI is FreePBX. Except we were comparing with AsteriskNow - http://www.asterisknow.com/ (a distro) rather than AsteriskGUI - http://svn.digium.com/svn/asterisk-gui/trunk/ (a GUI). Granted that Trixbox includes a lot more than just control for the PBX. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG74cYDQNt8rg0Kp4RAlrgAJoDXDRW+zuObuaGU3H/j8xf7A8NlwCgj1Xy jrdpmacI6T4tKypSglp2YhE= =9AMz -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Useful GUI? [Was: Why does everyone seem to dislike *now?]
Hi On Tue, Sep 18, 2007 at 02:35:14PM +1200, Matt Riddell wrote: I actually put my extensions.conf stuff into a generate.php file which writes out the extensions.conf file with parameters supplied by the customer stored in separate files. So our software is doing the same thing (overwriting configs) but I don't want the users changing settings too much. I guess this is probably pretty similar to the TrixBox idea but I haven't had a look at how that works under the hood. As Diego wrote: it's FreePBX (or is it freePBX?) Anyway, if you have an automatic script generating your configuration from some data, then there are 3 ways to affect your configuration: 1. change the data 2. change the generated configuration directly 3. change the script I was really surprized to see how few people chose option (3). This is evident by the fact that those scripts in FreePBX do not have their own custom options. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Randomly half-voice at sip/zap
Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The incoming route directed to a ring group. Sometimes the incoming calls - from pstn - are not, the caller do not hear any voice from us. When i call out on the sip line, it happens indirectly, so i can't hear nothing from the other side, especially when i call my sip telco provider. (10 try, 2 wrong) If they're calling me, everything is ok! Please help me! Thanks in advance! _ Peter Toth _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Randomly half-voice at sip/zap
On Tue, Sep 18, 2007 at 10:20:14AM +0200, Péter Tóth wrote: Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The incoming route directed to a ring group. Sometimes the incoming calls - from pstn - are not, the caller do not hear any voice from us. When i call out on the sip line, it happens indirectly, so i can't hear nothing from the other side, especially when i call my sip telco provider. (10 try, 2 wrong) If they're calling me, everything is ok! Is the call a direct call? Can you hear / see the audio in ztmonitor? The next step would probably be to enable 'bri debug span 1' and get traces from a good call and from a bad call. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Randomly half-voice at sip/zap
What do you mean on direct call? The error is more frequently on my sip trunk. Should I make a sip debug? My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem? Anyway i will watch the bri debug, and try to make a wrong and a correct call. Thanks 2007/9/18, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Sep 18, 2007 at 10:20:14AM +0200, Péter Tóth wrote: Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The incoming route directed to a ring group. Sometimes the incoming calls - from pstn - are not, the caller do not hear any voice from us. When i call out on the sip line, it happens indirectly, so i can't hear nothing from the other side, especially when i call my sip telco provider. (10 try, 2 wrong) If they're calling me, everything is ok! Is the call a direct call? Can you hear / see the audio in ztmonitor? The next step would probably be to enable 'bri debug span 1' and get traces from a good call and from a bad call. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Randomly half-voice at sip/zap
On Tue, Sep 18, 2007 at 12:07:20PM +0200, Péter Tóth wrote: What do you mean on direct call? The error is more frequently on my sip trunk. Should I make a sip debug? My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem? Anyway i will watch the bri debug, and try to make a wrong and a correct call. Can you successfully call an echo-test extension? (Echo() ) from SIP? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] stanaphone issues. can someone verify my config?
Sorry if this comes thru twice, I had the wrong account selected to send the first time... Callers to the number get ringing, I get stuff in my asterisk console, and it calls my softphone and ata, but answering either gets silence, and the caller gets the ringing stop, if they wait ages they get the stanaphone voicemail. I have had the account for ages, and it never has worked, other sip incoming works ok so I don't think its any issues, and the machine is the DMZ of the adsl router so it should be forwarded for everything. These are the relevant snips of the file and the console output. --sip.conf- [general] context=mainmenu allowguest=yes allowoverlap=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xx (our static IP is here) localnet=192.168.0.0/255.255.0.0; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register = 089xyz:[EMAIL PROTECTED]/8800 ; richards italk to ext 8800 register = 64997x:[EMAIL PROTECTED]/8800 --- later down in it. [stanaphone-richard] type=friend username=089x fromuser=089x (all the same, and as stanaphone give in the sip config) authname=089x secret= (as stanaphone give in the sip config host=sip.stanaphone.com allow=all (tried that since the softphoen uses pcm when it works - no change) allow=g729 allow=gsm dtmfmode=rfc2833 insecure=very canreinvite=no qualify=yes nat=yes port=5060 context=richardincoming mohinterpret=better I don't believe that the extensions.conf is a problem since I have other voips going to the same 8800 extension and being handled right. What I get in the console on an incoming call to the stanaphone number is. -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/089x-081c8b08, 9974) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/089x-081c8b08, ) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(SIP/089x-081c8b08, SIP/richardSIP/richardsoftphone|15|tr) in new stack -- Called richard -- Called richardsoftphone -- SIP/richardsoftphone-081d1348 is ringing -- SIP/richard-081cca70 is ringing -- SIP/richard-081cca70 answered SIP/08923542-081c8b08 [Sep 18 22:32:46] NOTICE[22616]: chan_sip.c:14815 do_monitor: Disconnecting call 'SIP/089x-081c8b08' for lack of RTP activity in 5 seconds == Spawn extension (richardincoming, 8800, 3) exited non-zero on 'SIP/089x-081c8b08' [Sep 18 22:32:57] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:02] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) [Sep 18 22:33:09] WARNING[22616]: chan_sip.c:1881 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 200 (Critical Response) Those continue on for quite some time and then stop (will get about 7 or 8 of the critical error) The lack of RTP everywhere makes it look to be a nat issue, but I have done everything I can think of to have that work, and the config is the same other then host, username and password on italk which is working fine. I have googled for the Maximum retries exceeded on transmission - I could only see some stuff related to broken sip phones, not a voip server. Alternativly, since it seems that stanaphone is a bit of a hit and miss from some other reading, is there any other functional US inwards provider for free that doesn't need a credit card that works well with asterisk? The softphone works, but I really need to get it going to my phones in the house instead. Soft client was closed when testing the asterisk. Many thanks. Richard Malcolm-Smith... ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Randomly half-voice at sip/zap
Hi! Yes, the echo test worked perfectly. When i try ztmonitor as follows, it gives strange output... [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* R ###* And so on... Is this normal? Thanks! 2007/9/18, Tzafrir Cohen [EMAIL PROTECTED]: On Tue, Sep 18, 2007 at 12:07:20PM +0200, Péter Tóth wrote: What do you mean on direct call? The error is more frequently on my sip trunk. Should I make a sip debug? My pbx is behind nat, maybe it is a nat problem?! Or a SIP setup problem? Anyway i will watch the bri debug, and try to make a wrong and a correct call. Can you successfully call an echo-test extension? (Echo() ) from SIP? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk crash and core dump
My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51 DEBUG[32650] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:54 DEBUG[32650] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Sep 18 13:42:54 VERBOSE[419] logger.c: -- SIP/4002-082aef20 is ringing ---MESSAGE FROM SAFE_ASTERISK--- Automatically restarting Asterisk. Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf' : Found Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Dynamic Loader loading preload modules: Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/modules.conf' safe_asterisk, ie FreePBX, notifies me that Asterisk exited on signal 11. I have core dumps in /tmp. What can I do to isolate the cause of these segmentation faults? Thank you for your advice, Vieri Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 crash and core dump
On Tue, 2007-09-18 at 05:15 -0700, Vieri wrote: I have core dumps in /tmp. What can I do to isolate the cause of these segmentation faults? You'd have to get an Asterisk developer to look at the backtraces generated from those core files. There's more information on the backtraces either in doc/backtrace.txt in the Asterisk source, or at the following URL: http://svn.digium.com/view/asterisk/branches/1.4/doc/backtrace.txt?revision=46298view=markup If you're able to get some good backtraces, grab one of the Asterisk developers on the #asterisk-bugs channel in IRC (on the Freenode network) and they'd be happy to try to help you figure out the cause of your crashes. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 crash and core dump
On Tuesday 18 September 2007 15:15:38 Vieri wrote: My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51 DEBUG[32650] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:54 DEBUG[32650] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Sep 18 13:42:54 VERBOSE[419] logger.c: -- SIP/4002-082aef20 is ringing ---MESSAGE FROM SAFE_ASTERISK--- Automatically restarting Asterisk. Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf' : Found Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Dynamic Loader loading preload modules: Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/modules.conf' safe_asterisk, ie FreePBX, notifies me that Asterisk exited on signal 11. I have core dumps in /tmp. What can I do to isolate the cause of these segmentation faults? You should compile asterisk with debbuging enabled (and optimization disabled), and then take backtraces from core dumps. Please see http://www.voip-info.org/wiki-Asterisk+debugging Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Bug labs
I thought this would interest a few people on the list - asterisk enabled home security video recording dvr anyone? http://deancollinsblog.blogspot.com/2007/09/bug-labs-opensource-hardware .html I had a really interesting conference call today about a new startup called http://www.buglabs.net http://www.buglabs.net/ They are looking to introduce a modular set of 'devices' that fit together and along with supporting software will allow you to create your own 'opensource hardware mashups'. So the core 'base unit' is a fully programmable and hackable Linux computer, equipped with a fast CPU, 128MB RAM, built-in WiFi, rechargeable battery, USB, Ethernet, and a small LCD with button controls. From there 4 additional modules can be added, gps, video camera, lcd display, accelerometer/motion sensor being the first 4 for release (though 81 have been mocked up so far). The long term concept is if you want a 'weather station with live video feeds and gps location control you can add various modules together to deliver what you are looking to achieve. I have high hopes for the concepts, and wish the guys well as it seems their hearts are in the right place.though it's going to be a long (but interesting) road to travel. Cheers, Dean Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Enabling MySQL UNIQUE from cdr.conf
Thanks for your replies, but the file you mention (cdr_mysql.conf) I have it already configured since I'm already storing the CDRs in a MySQL database. As I understand this file (cdr_mysql) is only for enabling mysql cdr storage and cdr.conf should be used for cdr backend parameterizations. Or there are any directives I should include in it in order to store the UNIQUEID field in the database? One more question, does anynone knows why this feature (storing UNIQUEID field by default) was deprecated from the previous version of asterisk-addons? Thanks Luis Palma ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote: stuff useless. My real concern was the immediate '/ignore' for asking about an issue with the *now ditro that actually had nothing to do with the GUI itself. Truth be told, most of my time today was in the CLI You may be taking what happened on IRC a bit too literally. You got plenty of help on your Polycom and call parking issues on 09/14, from some very knowledgeable people. On 09/17, you asked one question about rPath/Conary and one person did a '/me puts jcanfield on ignore' emote. He probably didn't even ignore you, it was just his way of saying he wasn't interested in answering questions about packaging systems on a Linux distro he doesn't use. The simple answer to your question (how do I get the LDAPGet module) is answered on the Wiki - you download it from the author's site. The question of how do I package some arbitrary source code into a conary package? isn't really germane to #asterisk. As to the second class citizen point, I think you'll find that people who come into #asterisk asking about problems with their GUI-enabled Asterisk install fall into one of two categories: those who are willing to reduce the problem down to it's non-GUI elements and pastebin the configs and output, and those who are incapable or unwilling to do so. The former tend to get help from people on #asterisk; IMHO the latter should find other places to ask for help, or pay for consulting services. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Useful GUI? [Was: Why does everyone seem to dislike *now?]
the GUI does NOT have meta-data in the sense you think of. I can seamlessly operate a full PBX through the GUI and add things myself, it is VERY simple. The [default] context is global for the other numberplans (trunks/users) There is no problem with overwriting files if you do it right. When you visit a page, the GUI loads the information from the proper config files, then when you save, it saves the information. If you do something in between, there is nothing the GUI can do about it. However, it does not overwrite any of your contexts, sip users/iax users, zapata settings, anything! If you are just somewhat careful, it works great. If ANYTHING comes up, report it to bugs.digium.com and I will fix it immediately! -bk - Original Message - From: Ken D'Ambrosio [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 17, 2007 6:59:12 PM (GMT-0600) America/Chicago Subject: [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?] On Mon, September 17, 2007 7:21 pm, Matt Riddell wrote: In the past, you could help someone sort a problem, only for the config files to be overwritten the next time the user did something in the GUI. Are there any Asterisk GUIs out there that actually parse the data files, themselves, instead of having some sort of metadata middle-man, which leads to said overwriting? I mean, I, personally, love the CLI -- always have been a fast typist -- but I also know the CLI would scare the living bejeepers out of my boss if/when I try to push hard on an Asterisk solution. What I'd prefer is: - The chance to do CLI stuff as I see fit, BUT - the ability to let users -- even administrative users -- use a GUI, without messing up my beautiful config files. Is this a pipe dream, or is there a GUI out there that might actually do the job? Thanks, -Ken ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Enabling MySQL UNIQUE from cdr.conf
On 9/17/07, Luís Palma [EMAIL PROTECTED] wrote: Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database backend for storing CDRs without having to recompile asterisk-addons as stated here http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? After version 1.4 it is said in release that it can be done (not sure if it applies to mysql backend) In addons v1.4.2, it's not possible without recompilation. You get one of two versions of code depending on the definition of a compile time constant. If that constant isn't defined, the text of the SQL INSERT statement in the shared module will be: INSERT INTO %s (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) instead of INSERT INTO %s (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) In the -trunk version of cdr_addon_mysql.c, the behaviour of loguniqueid was changed from a compile-time to runtime option, just like userfield already was. The changes to make loguniqueid a runtime option are pretty small, and trivial to backport to the 1.4 branch on their own. You'd have to do more research to see if you can just build the trunk version against 1.4, given that trunk also has added MySQL SSL support. Of course, if your question stems from the fact that you are unable to recompile anything in your installation, none of this is much help. :( -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] T1/PRI pricing
I know this borders on commercial, so I apologize. I will take this off list as soon as possible. Someone a couple months ago claimed to know how to get PRI or T1 voice circuits significantly cheaper than going through the ILEC. I would appreciate that person contacting me (off-list) at this email address. Thanks, David Gomillion [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
James FitzGibbon wrote: On 9/17/07, *Jim Canfield* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: stuff useless. My real concern was the immediate '/ignore' for asking about an issue with the *now ditro that actually had nothing to do with the GUI itself. Truth be told, most of my time today was in the CLI You may be taking what happened on IRC a bit too literally. You got plenty of help on your Polycom and call parking issues on 09/14, from some very knowledgeable people. On 09/17, you asked one question about rPath/Conary and one person did a '/me puts jcanfield on ignore' emote. He probably didn't even ignore you, it was just his way of saying he wasn't interested in answering questions about packaging systems on a Linux distro he doesn't use. The simple answer to your question (how do I get the LDAPGet module) is answered on the Wiki - you download it from the author's site. The question of how do I package some arbitrary source code into a conary package? isn't really germane to #asterisk. As to the second class citizen point, I think you'll find that people who come into #asterisk asking about problems with their GUI-enabled Asterisk install fall into one of two categories: those who are willing to reduce the problem down to it's non-GUI elements and pastebin the configs and output, and those who are incapable or unwilling to do so. The former tend to get help from people on #asterisk; IMHO the latter should find other places to ask for help, or pay for consulting services. -- j. Three cheers, Most of these people you are expecting help from are volunteering their time and assistance, for you to complain that you don't think you got enough service from volunteers is a bit preposterous. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Useful GUI? [Was: Why does everyone seem to dislike *now?]
Also, update the asterisk GUI to what I have been working on now. http://asterisknow.org/install-related to the asteriskNOW branch. (This latest work includes VOIP Seamless service providers integration, and also digital card detection and setup) -bk - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 18, 2007 3:06:48 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?] -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Diego Iastrubni wrote: s/Trixbox/FreePBX/g Please, Trixbox is a distro, the GUI is FreePBX. Except we were comparing with AsteriskNow - http://www.asterisknow.com/ (a distro) rather than AsteriskGUI - http://svn.digium.com/svn/asterisk-gui/trunk/ (a GUI). Granted that Trixbox includes a lot more than just control for the PBX. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG74cYDQNt8rg0Kp4RAlrgAJoDXDRW+zuObuaGU3H/j8xf7A8NlwCgj1Xy jrdpmacI6T4tKypSglp2YhE= =9AMz -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
James FitzGibbon wrote: On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote: stuff useless. My real concern was the immediate '/ignore' for asking about an issue with the *now ditro that actually had nothing to do with the GUI itself. Truth be told, most of my time today was in the CLI You may be taking what happened on IRC a bit too literally. You are right, judging from the responses I have received both on and off-list. Everyone has been extremely polite and understanding. I partly expected to get flamed on this one. Many thanks to all who responded! You got plenty of help on your Polycom and call parking issues on 09/14, from some very knowledgeable people. On 09/17, you asked one question about rPath/Conary and one person did a '/me puts jcanfield on ignore' emote. He probably didn't even ignore you, it was just his way of saying he wasn't interested in answering questions about packaging systems on a Linux distro he doesn't use. The simple answer to your question (how do I get the LDAPGet module) is answered on the Wiki - you download it from the author's site. The question of how do I package some arbitrary source code into a conary package? isn't really germane to #asterisk. In no way did I mean to discredit the help I have received so far. As a newcomer I saw asterisk/asteriskNOW/Digium as one and the same. I realize now, the ecosystem that is Asterisk, is much more diverse than that. As to the second class citizen point, I think you'll find that people who come into #asterisk asking about problems with their GUI-enabled Asterisk install fall into one of two categories: those who are willing to reduce the problem down to it's non-GUI elements and pastebin the configs and output, and those who are incapable or unwilling to do so. The former tend to get help from people on #asterisk; IMHO the latter should find other places to ask for help, or pay for consulting services. 100% agree. Thanks again. jc References Visible links 1. mailto:[EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Interesting Conference Request - Can this be done ?
Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Dell Power Edge 1900
Does anyone know if the Dell Power Edge 1900 has an issue with multiport E1 cards? We've had this server running for a while now with 2 E1 cards. At first we tried to install an Openvox D210P card with two E1 ports but after a couple of kernel panics we thought that maybe the card was defective and we replaced it with two Digium TE120P cards. Now the customer needs a third E1 port and since the computer pnly has two PCI ports we decided to install a Digium TE411P card. Over the past few days there have been several kernel panics. We have the latest Asterisk 1.4.11, Zaptel 1.4.5.1 on CentOS 5 with all the latest upgrades applied. The E1 links are all R2 using Unicall. The server has a couple PCI-Express slots but I do not know if getting a multiport E1 card for PCIEx will have the same problems. -- 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Here's what I'm showing in the logs... [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format g729, extension(s) g729 [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full license text supplied by the accompanying [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: register utility, or ask for a copy from Digium. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes software developed by the OpenSSL Project [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize G.729 copy protection! [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Any ideas where this points me? Thanks, Scott On 9/17/07, Scott Moseman [EMAIL PROTECTED] wrote: What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman [EMAIL PROTECTED] wrote: I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Interesting Conference Request - Can this be done ?
Quoting Dovid B [EMAIL PROTECTED]: Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). what would they hear then ? if they can't hear anyone else, just an extension that goes nowhere they talk into would do what you need. I am guessing you didn't explain clearly enough though. Any ideas ? Thanks. Dovid Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] ISDN data packets
I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? Please let me know if I am thinking or looking at something wrong. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Dell Power Edge 1900
Carlos Chavez wrote: Now the customer needs a third E1 port and since the computer pnly has two PCI ports we decided to install a Digium TE411P card. Over the past few days there have been several kernel panics. We have the latest Asterisk 1.4.11, Zaptel 1.4.5.1 on CentOS 5 with all the latest upgrades applied. The E1 links are all R2 using Unicall. Please contact Digium technical support - [EMAIL PROTECTED] They are here to help you with these types of issues. -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Interesting Conference Request - Can this be done ?
Dovid B wrote: I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). There are listen-only and talk-only options to the MeetMe application. -- Russell Bryant Software Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 agents w/ DUNDi
All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. The problem is that since the incoming context is the same context as the normal incoming call context, they get sent to voicemail if they don't answer. I thought the solution would be as simple as defining a separate queue for the AgentCallbackLogin, and dumping calls to that queue. The problem that I see with that is I can no longer use DUNDi to route the call. If I do that and don't have their explicit extension in the [agentqueue] context, I get 'Extension is not valid for automatic login of agent ' when they try to login. If I put a huge _ match in the [agentqueue] then I can login, but it can't route the call properly because routing the call needs a DUNDi lookup which puts them back into stdexten. I *think* my solution is to basically create two DUNDi networks, one that I can route calls to from the agentqueue context, and the other from the inbound context. The agentqueue DUNDi network would then just not include the stdexten dialing and would only be called by queue calls. If anyone has a more simple solution I'm all ears. Hopefully I'm making this more complicated that it needs to be. :) My ideal setup is: 1. If a call comes into the queue, go to agent but don't ever go to voicemail (Just Dial(SIP/)) 2. If a call comes in for that same agent to their DID, route to stdexten macro 3. All calls routed w/ DUNDi so the system doesn't care about which server they log into -- Kyle Sexton http://www.mocker.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] ISDN PRI debug in Asterisk
Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? Thanks -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Scott Moseman wrote: Here's what I'm showing in the logs... [Sep 18 09:52:09] VERBOSE[2786] logger.c: == Registered file format g729, extension(s) g729 [Sep 18 09:52:09] VERBOSE[2786] logger.c: format_g729.so = (Raw G729 data) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Please see the full license text supplied by the accompanying [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: register utility, or ask for a copy from Digium. [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: This product includes software developed by the OpenSSL Project [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Sep 18 09:52:09] NOTICE[2786] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Sep 18 09:52:09] VERBOSE[2786] logger.c: == G.729 Host-ID: x:x:x:etc [Sep 18 09:52:09] WARNING[2786] codec_g729.c: Failed to initialize G.729 copy protection! [Sep 18 09:52:09] VERBOSE[2786] logger.c: codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Any ideas where this points me? I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. Matthew Fredrickson Thanks, Scott On 9/17/07, Scott Moseman [EMAIL PROTECTED] wrote: What's the best way to debug what's going on within Asterisk? I turned up the 'core debug', but that did not give me what I was hoping to find. I'm hoping to see some kind of error that explains why it will not pass through the g729 codec. Thanks, Scott On 9/14/07, Scott Moseman [EMAIL PROTECTED] wrote: I have a fresh 1.4.10.1 installation that appears to have a problem with g729 pass-through. I can see the gateway in question sending an INVITE using g729b. However, the Asterisk is only sending g711 in the INVITE to my Polycom phone. [gateway] disallow=all allow=g729 [phone] disallow=all allow=ulaw allow=alaw allow=g729 There's the codec configs for the gateway and the phone in question. I even attempted to setup the phone with only the allow=g729, but in that instance it won't even complete the call. We had to add g711 support to the gateway in question for now to get it up and running, but we want to get it back to using only g729. CLI show modules like g729 Module Description Use Count format_g729.so Raw G729 data 0 codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 2 modules loaded I downloaded the pre-compiled g729 module from Digium. The sip.conf references g729 and the codec module is loaded. Unless there's anything else I need to do that I'm missing? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 data packets
Thanks for the reply. I was not looking for a visualizer. I justed wanted to see the data packets flowing in the asterisk CLI (for example something similar to the rtp packets that flow when making a voip call). I can see the various messages like CONNECT, SETUP etc. I am a newbie regarding ISDN and I might be looking at things wrongly. Thanks Regards Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? pri intense debug is used to see signalling that happens on the PRI. There is not a visualizer for b channel voice data. The closest thing you could try to use is ztmonitor or a record() in your dialplan. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. -- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 data packets
Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? pri intense debug is used to see signalling that happens on the PRI. There is not a visualizer for b channel voice data. The closest thing you could try to use is ztmonitor or a record() in your dialplan. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 PRI debug in Asterisk
On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? To get the equivalent of a packet sniffer, you'll need to go to a lower-level tool than asterisk. For sangoma cards, you can use the `wanpipemon` command to do a packet dump. I'm not sure what the equivalent for Digium cards is, but I'm sure it's possible. -erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
You cannot set up your dialplan with the CLI or am I missing something? Creating relatively simple dialplans manually can be quite time consuming. A GUI takes care of all that grunt work. -Original Message- From: SIP [mailto:[EMAIL PROTECTED] Sent: Monday, September 17, 2007 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why does everyone seem to dislike *now? Not at all relevant to your query, but I still use the mysql CLI for any mysql task... and while most OSs have nice, functional tools to add users (command-line tools), there are SOME (*cough* Irix *cough*) where there are no CLI tools and VI is your only option (especially if you're remotely logged in via a term window and have no X). GUIs have their place. But it's often a trade-off between abstracting the details to make things user-friendly and hiding the power that is available via the CLI from someone who knows it. If you're comfortable with the CLI, why learn another tool? If you're NOT comfortable with the CLI, by all means use a GUI, but don't expect people who never use it to be of much help when you ask questions. That being said, I like AsteriskNow's GUI. They've obviously spent a lot of work on it (prettier than the stuff that comes with Trixbox). However, for me, I learned using vi and the cli, so I can never quite find what I'm looking for in AsteriskNow. N. Jim Canfield wrote: Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few quesions in #asterisk...my biggest mistake was mentioning *now and I was quickly marked as the GUI idiot. Not entirely untrue at this point but not helpful for someone who is getting started. Here are my first impressions: * The Devs have spent a LOT of time on *now and seem to be doing a fantastic job. * *now is not just a GUI...it's a complete base/reference system - I like that the MOST. * *now is a great starting point for someone new (Me). * *now needs documentation! I know it's in beta, but having links to a down site, is not cool. (Sign me up for help if needed). * *now could be more geared for use as a universal tool. The default contexts and files were quickly replaced with more standard configs. * *now could be very helpful in tracking issues with links to Report a problem or search the WIKI from the app. I understand the tendency to love the CLI, but I honestly think there a place for a GUI in Asterisk. How many of us still use the mysql CLI? I can't expect my helpdesk guy to know emacs or vi just to add a user. Curiously, jc ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
On Tue, Sep 18, 2007 at 10:03:50AM -0700, shadowym wrote: You cannot set up your dialplan with the CLI or am I missing something? Creating relatively simple dialplans manually can be quite time consuming. A GUI takes care of all that grunt work. You write a dialplan with a text editor. Or copy from an existing sample / template. And you use decent automation and proper includes and patterns. GUIs often tend to force you to do time-consuming work over and over again and get in the way of automation, which is usually trivial in a decent command-line interface. GUIs automate certain things. As long as you are within the supported flow. But once you leave it, you often have to do more work. (Those are generic observasions. Let's not go over the GUI flamewars again) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 PRI debug in Asterisk
Arpit Mehta wrote: Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? No, like I said in response to your other question, the only thing you can directly see in pri intense debug is the signalling packets. Data with TDM is not packetized as its native format, so that is why there isn't a way to see tdm voice packets like you can see RTP packets. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Why does everyone seem to dislike *now?
If the #AsteriskNOW channel is dead on IRC that does not mean you can bring your problems to a channel dedicated to Asterisk (i.e. no GUI). Go ahead and use AsteriskNOW, but don't pester the people in #asterisk, most of which have never used it and many have never even heard of it. All the GUIs make incredibly complex dialplans, macros, and AGIs to do what they do. If you are using AsteriskNOW and build your config files from scratch then most everyone on #asterisk is willing to help you -- you are using simple easy to understand, troubleshoot, and diagnose config files. The problem with this is that the GUI will no longer work. AsteriskNOW is not Asterisk. Asking questions about AsteriskNOW on #asterisk is like asking for support for your linux based cell phone on #linux. The environment, built, paths, libraries, and software are all totally different on a linux based cellphone and people that know Linux really can't help you. All you end up doing is wasting people's time. There is a similar problem with #asterisk-dev. You do not go to #asterisk-dev to ask user type questions. That is not what the channel is for. #asterisk-dev is for development questions related to Asterisk. I'm sure that AsteriskNOW is a great product. Use the correct support methods, that is all we ask. SIP wrote: Not at all relevant to your query, but I still use the mysql CLI for any mysql task... and while most OSs have nice, functional tools to add users (command-line tools), there are SOME (*cough* Irix *cough*) where there are no CLI tools and VI is your only option (especially if you're remotely logged in via a term window and have no X). GUIs have their place. But it's often a trade-off between abstracting the details to make things user-friendly and hiding the power that is available via the CLI from someone who knows it. If you're comfortable with the CLI, why learn another tool? If you're NOT comfortable with the CLI, by all means use a GUI, but don't expect people who never use it to be of much help when you ask questions. That being said, I like AsteriskNow's GUI. They've obviously spent a lot of work on it (prettier than the stuff that comes with Trixbox). However, for me, I learned using vi and the cli, so I can never quite find what I'm looking for in AsteriskNow. N. Jim Canfield wrote: Greetings, Last week I began researching Asterisk for the first time. I did what most noobs would do; downloaded an image that seemed simple and straightforward and had some credibility (*now). I also downloaded the TFOT version 1 as a guide. As questions arose, I tossed a few out in #asterisckNOW channel..and found it to be a ghost town. Only later did i start to ask a few quesions in #asterisk...my biggest mistake was mentioning *now and I was quickly marked as the GUI idiot. Not entirely untrue at this point but not helpful for someone who is getting started. Here are my first impressions: * The Devs have spent a LOT of time on *now and seem to be doing a fantastic job. * *now is not just a GUI...it's a complete base/reference system - I like that the MOST. * *now is a great starting point for someone new (Me). * *now needs documentation! I know it's in beta, but having links to a down site, is not cool. (Sign me up for help if needed). * *now could be more geared for use as a universal tool. The default contexts and files were quickly replaced with more standard configs. * *now could be very helpful in tracking issues with links to Report a problem or search the WIKI from the app. I understand the tendency to love the CLI, but I honestly think there a place for a GUI in Asterisk. How many of us still use the mysql CLI? I can't expect my helpdesk guy to know emacs or vi just to add a user. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote: But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Wouldn't that be the same as paging/intercom, then? http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 PRI debug in Asterisk
Erik Anderson wrote: On 9/18/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, Does Asterisk contain a full fledged ISDN packet sniffer. By giving the command pri intense debug span 1 , does it debug every packet received (control and voice/data packets) ? To get the equivalent of a packet sniffer, you'll need to go to a lower-level tool than asterisk. For sangoma cards, you can use the `wanpipemon` command to do a packet dump. I'm not sure what the equivalent for Digium cards is, but I'm sure it's possible. You can basically use ztmonitor to get a B-channel data dump. That should also work on the Sangoma cards. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 data packets
Arpit Mehta wrote: Thanks for the reply. I was not looking for a visualizer. I justed wanted to see the data packets flowing in the asterisk CLI (for example something similar to the rtp packets that flow when making a voip call). I can see the various messages like CONNECT, SETUP etc. I am a newbie regarding ISDN and I might be looking at things wrongly. Unfortunately, there isn't a way of seeing ISDN TDM data flowing into and out of asterisk like RTP. Matthew Fredrickson Thanks Regards Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Arpit Mehta wrote: I have a ISDN PRI(T1) line coming into my TE110 Asterisk card. When I use pri intense debug span 1 It is supposed to show every packet that is received on the PRI line. I wanted to know in ISDN Pri when a call connects how are the data (voice) packets (for PRI) shown in Asterisk. Or if there is some other command to see these kind of data packets ? pri intense debug is used to see signalling that happens on the PRI. There is not a visualizer for b channel voice data. The closest thing you could try to use is ztmonitor or a record() in your dialplan. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. -- -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
What about using trixbox pro and forcing auto answer with the hud server configuration? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Joao Pereira Sent: Tuesday, 18 September 2007 1:14 PM To: Ed Pastore; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Answer I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote: But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Wouldn't that be the same as paging/intercom, then? http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Chan_SCCP vs. Chan_Skinny
On 9/17/07, Dan Austin [EMAIL PROTECTED] wrote: Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers working on the chan_skinny code. In the next week or so, I'll try to take a look at chan_skinny again before making any comparisons. It would be great if chan_skinny worked as well as chan_sccp. I'm not interested in re-igniting the flame-wars of the past about these channel drivers, but I would like to know what else needs to be addressed in chan_skinny before it users of chan_sccp would consider using it. Thanks, Dan ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy Moore Somewhere I wish I wasn't ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
I am experiencing the exact same problem on solaris, and we do have licenses purchased. I will log a bug at digium in the next day or two about my particular instance. Scott Moseman wrote: On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
Zoiper can do it when you use the provisioning, contact me offlist on [EMAIL PROTECTED] Zoa Joao Pereira wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu So, if someone knows a nice softphone for an Asterisk Call Center, please advice me. Thanks Regards Joao Pereira Ed Pastore wrote: On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote: But still, the user can choose not to answer the phone. I want to force the users to accept the calls. Wouldn't that be the same as paging/intercom, then? http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Does G.729 phone - asterisk - G.729 phone work with reinvite turned off? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Watson Sent: Tuesday, September 18, 2007 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Fyi... [myphone] disallow=all allow=g729 canreinvite=no [otherphone] disallow=all allow=g729 canreinvite=no I attempted this setup and it works. Media routed through the Asterisk. Thanks, Scott On 9/18/07, Jeremy Mann [EMAIL PROTECTED] wrote: Does G.729 phone - asterisk - G.729 phone work with reinvite turned off? -Original Message- From: [EMAIL PROTECTED] Sent: Tuesday, September 18, 2007 1:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman Sent: September-18-07 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
The gateway is transcoding the PSTN into g729 and passing it to Asterisk. The Asterisk never sees the PSTN from the outside. I have watched the INVITE requests, they contain a request for a g729 only call. But the INVITE to the phone does not include g729. However, as previously stated, I did get a g729 phone to talk to another g729 phone. So I assume that means pass-through *can* work, but something is not working right? Thanks, Scott On 9/18/07, Matt Watson [EMAIL PROTECTED] wrote: PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. -- Matt -Original Message- From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] g729 on 1.4.10.1 On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Matt Watson wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Moseman My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? PSTN - g729 requires transcoding at that point. You can however do: G.729 phone - asterisk - G.729 phone without license (from my understanding). But as soon as you introduce a non-g729 hop (ie. Analog PSTN line) it requires a license to preform transcoding. Matt, Look at his path. He's going from a PSTN phone to a g729 gateway. As long as the gateway is there, Asterisk doesn't really know about the PSTN phone. Therefore, yes, this should equate to pass through. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Scott Moseman wrote: The gateway is transcoding the PSTN into g729 and passing it to Asterisk. The Asterisk never sees the PSTN from the outside. I have watched the INVITE requests, they contain a request for a g729 only call. But the INVITE to the phone does not include g729. However, as previously stated, I did get a g729 phone to talk to another g729 phone. So I assume that means pass-through *can* work, but something is not working right? Thanks, Scott If you have anything in Asterisk trying to handle the audio, you cannot pass it through. For instance, if you are trying to record the call in ulaw, or trying to playback prompts that aren't available in g729. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
You have to increase the amount of available file descriptors per process: http://hausheer.osola.com/docs/11%C2%A0%C2%A0 On Tue, 18 Sep 2007, Wai Wu wrote: Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Comfort noice sample (gsm/mp3)
Where do I get sound file for comfort noice. GSM or MP3 is fine. Many thanks. Jim ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Follow me on this, it seems odd (or maybe I don't undertand)... Test #1 [src_phone] disallow=all allow=g729 [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #2 [src_phone] disallow=all allow=g729 allow=ulaw [dest_phone] disallow=all allow=g729 I can make the call (src to dest) and it will work using g729. Both the call handling and media are going through Asterisk. Test #3 [src_phone] disallow=all allow=ulaw allow=g729 [dest_phone] disallow=all allow=g729 The above call attempt will fail, and this is what I'm seeing: chan_sip.c:2944 sip_call: No audio format found to offer. In every test, the source INVITE includes ulaw, alaw and 729. That is the codecs that I configured on the phone themselves. However, in Test #3 the call will fail. Why? This does not necessarily have to do with my g729 gateway, but I'm curious what's wrong with this scenario, maybe using this situation to understand will help me with my gateway... (Although I tried setting only g729 on the gateway and the gateway's peer in the Asterisk and it did not appear to help.) Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Enabling MySQL UNIQUE from cdr.conf
Thanks for the explanation and your cues. I've been able to activate this feature by recompiling again asterisk-addon source code (version 1.4.2). If a runtime option is already undergoing in trunk, that's good news but for now I prefer to stick to version 1.4.2. I'm trying to working with rpm packages only and for now I will try to present to the rpm packager admin from ATrpms to include this compile option in the rpm build. I'm trying to keep things in the asterisk framework uniformized as much as I can. My attention is mainly in another asterisk front-end I'm developing, and running in bleeding edge stuff is not my intention. Just for curiosity in trunk, you setup de uniqueid option in cdr_mysql.conf right? Humm... Why I'm hearing look de source Luke? Thanks again Luis Palma ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
Thnx. That did it. I also reduce the stock size to 512K instead of 8M. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Tuesday, September 18, 2007 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linux limits You have to increase the amount of available file descriptors per process: http://hausheer.osola.com/docs/11%C2%A0%C2%A0 On Tue, 18 Sep 2007, Wai Wu wrote: Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
On 9/18/07, Wai Wu [EMAIL PROTECTED] wrote: Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx Hi Wai, I had this issue once (different software, unrelated to asterisk), and I used this guide to increase file handles: http://confluence.atlassian.com/display/DOC/Fix+'Too+many+open+files'+error+on+Linux+by+increasing+filehandles Cheers, AR -- Alex Robar [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
On Tue, Sep 18, 2007 at 04:22:29PM -0400, Alex Balashov wrote: On Tue, 18 Sep 2007, Wai Wu wrote: Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. [ top posting fixed so I can comment as well ] You have to increase the amount of available file descriptors per process: http://hausheer.osola.com/docs/11%C2%A0%C2%A0 These days, I beleve the typical place to fix that is actually in /etc/sysctl.conf, in most distros: http://www.cs.wisc.edu/condor/condorg/linux_scalability.html That page notes it for RedHat derived distros, but I'm pretty sure SuSe puts it there as well. 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] g729 on 1.4.10.1
Scott Moseman wrote: However, in Test #3 the call will fail. Why? Because Asterisk will attempt to use ulaw in preference to G.729 if possible, and the other endpoint offered to support ulaw. The format(s) supported by the eventual call destination are not relevant, because at the time Asterisk is making a format decision for the incoming call leg, it has no clue what the destination is going to be or what formats it will support. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Enabling MySQL UNIQUE from cdr.conf
Yes it is supported on cdr_mysql.conf. I just have been looking to the example file (cdr_mysql.conf.sample) in http://svn.digium.com/view/asterisk-addons/trunk/configs/ and it has this option clearly stated. Regards Luis Palma ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu Actually i believe you can do it yourself. X-Lite is windows, right? There are a bunch of programs, allowing to edit internal resources of executable files. So, just grab a resource editor (i prefer XN Resource Editor), open .exe file, edit the menu - disable (and hide) items you want to forbid changing for users, and give them the executable. I'm not certain that X-Lite's executable is not packed/crypted, but editing SJPhone was very successful some time ago. Of course, there's always an option for user - to take another softphone, but whatever softphone you choose - they will have the same chance. Regards, Atis ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Limiting Simultaneous calls
You mean in sip.conf? Look at adding to your voip providers peer/user config incominglimit, outgoinglimit or call-limit: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf --- Forrest Beck www.shift8.biz On Sep 18, 2007, at 4:26 PM, Jim Boykin wrote: Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim ___ Sign up now for AstriCon 2007! September 25-28th. http:// www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Limiting Simultaneous calls
On Wed, 2007-09-19 at 01:56 +0530, Jim Boykin wrote: Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. There are lots of ways to skin that particular cat, but my favorite is to use the GROUP() and GROUP_COUNT() functions to artificially limit the number of calls. -- Jared Smith Community Relations Manager Digium, Inc. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Limiting Simultaneous calls
On Wed, Sep 19, 2007 at 01:56:42AM +0530, Jim Boykin wrote:
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
I think you can do this sort of thing with the Set(GROUP) and GROUPCOUNT to
monitor number of calls placed in a call 'group' which in this context does
not mean a pickup group or a caller group, it means 'a group of calls set up
in group $foo' (where $foo is some variable)
Take a look at:-
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
and:-
http://www.voip-info.org/wiki/index.php?page=Superdial%20macro
To see how it is used to limit the number of outgoing calls to a PSTN
carrier.
'group' could be a global setting you give it, or the extension number of
the user (to limit globally or per extension)
Specifically:-
${ARG6} - Max. group number (maximum number of concurrent calls you want to
allow for that group)
exten = s,1,Set(GROUP()=${ARG5})
exten = s,2,Set(GROUPCOUNT=${GROUP_COUNT(${ARG5})})
exten = s,3,GotoIf($[${GROUPCOUNT} ${ARG6}]?104)
exten = s,104,Goto(s-CHANUNAVAIL,1)
etc.
Rob
--
Robert Lister - London Internet Exchange - http://www.linx.net/
sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510
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Re: [asterisk-users] Call Center SoftPhone with Auto Answer
On 9/18/07, Atis Lezdins [EMAIL PROTECTED] wrote: On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu Actually i believe you can do it yourself. X-Lite is windows, right? There are a bunch of programs, allowing to edit internal resources of executable files. So, just grab a resource editor (i prefer XN Resource Editor), open .exe file, edit the menu - disable (and hide) items you want to forbid changing for users, and give them the executable. I'm not certain that X-Lite's executable is not packed/crypted, but editing SJPhone was very successful some time ago. Of course, there's always an option for user - to take another softphone, but whatever softphone you choose - they will have the same chance. I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 agents w/ DUNDi
On Tue, Sep 18, 2007 at 11:27:36AM -0500, Kyle Sexton wrote: All, I'm trying to configure queue agents w/ a DUNDi setup so that an agent can login to whatever server they please w/o any custom setup. In general this seems to work, agents login w/ AgentCallbackLogin into the incoming context (not a special queue context) and can receive queue calls. Don't use AgentCallbackLogin() it's odd in some interesting ways (The whole agent stuff isn't very flexible in many ways if your users have multiple ways to get called outside of the Agent.) For example if you have users in queues represented as Agents with also direct numbers respresented as SIP/xxx elsewhere, you will have problems with call waiting and busy detection not working properly, i.e, when the user is making an outgoing call on their SIP extn, the agent stuff does not detect them as being busy, so you cannot use call waiting. An 'agent' can only accept one call at a time but SIP/xxx may have several calls. About your situation, you might be able to solve it by using Local/[EMAIL PROTECTED] to route the call to where you need it to go when a call comes in for an agent that you want to locate in the dialplan somewhere else. The thing you route to using Dial(Local/xxx must be something in the dialplan routable by the current context.) AgentCallbackLogin as I understand it, deprecated as of 1.4.x, and 1.2.x is no longer being actively developed, so I'm trying to get off it, however some stuff I do is not possible now without that feature that they don't seem all that concerned about fixing right now. :-( Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? Makes Vista look like a picnic. -- j. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 SoftPhone with Auto Answer
James FitzGibbon wrote: On 9/18/07, *David Gomillion* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? Makes Vista look like a picnic. -- j. Easy solution == pay by performance. -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] (Getting OT) Re: Call Center SoftPhone with Auto Answer
Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon: On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? I do not know American work law, but if you tell your people to NOT turn off auto answer, and they do for having a break, would that not count as work refusal? As long as they get all the breaks they are supposed to take, of course. If for example a cashier in a supermarket here in Germany would just leave her position for a few minutes for a smoke, small-talk or whatever, outside her assigned break times, she could afaik get a written warning, and at the second occasion the full wad of papers (aka been fired). On the other hand, if you count only times while they are on-a-call, with appropriate logging software, adding a few seconds per-call for overhead, as their worktime, they pretty soon will keep auto answer on to get the required number of work minutes during their shift, I would expect. But this is not as much a technical problem as a social one: If your agents are unmotivated, they might spend time talking off-business to any caller/callee on the phone that seems to be interested in small-talk, and _that_ you could hardly find out technically. So you might get an upgrade without paying for the deinstallation of the previous meatware, but the installation process of course has costs. BTW putting too much pressure on your agents might do bad things to their effiency, motivation, even mental health. Getting the balance between control and good atmosphere right is not easy, and something that cannot be generalized but must be tailored to the situation. The value of a human asset (imagine me vomiting my way through those words) can materialize in the number of sales, calls, ... and also in the customer experience he creates, which is hard to be counted in numbers. For example, I recently bought some music instrument and accessories at a phone-order company. The people there were relaxed, friendly, helpful and made the effort of giving me competent, quick information that I needed. All contact with them was extremely positive. As I needed some more stuff that I knew was a bit cheaper at another store (which only deals with customers in a matter-of-fact way), I decided to honour that effort. I also recommended the company to friends, which they probably will never know about and as such cannot count in as a bonus for their sales personell. *Just my loose change. Man, there were lots of coins in that purse.* Makes Vista look like a picnic. IMO Vista is an apple short of one ;-) To get the Asterisk relevant topics: You could - count on-call minutes to rate agent performance - track off-call intervals on a certain line and so track the turned-off auto answer - do some social engineering or policy work to get this sorted in a non-technical way (work contract terms, etc) - pay some softphone manufacturer to implement needed changes BTW what would hinder your agents from shutting down the softphone app when they do not want to answer calls? What would hinder them from just not talking to the caller when they do not want to? Best regards, Anselm ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. Thnx ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Limiting Simultaneous calls
Try: http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit On Wed, 19 Sep 2007, Jim Boykin wrote: Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] (Getting OT) Re: Call Center SoftPhone with Auto Answer
Anselm Martin Hoffmeister wrote: Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon: On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? I do not know American work law, but if you tell your people to NOT turn off auto answer, and they do for having a break, would that not count as work refusal? As long as they get all the breaks they are supposed to take, of course. If for example a cashier in a supermarket here in Germany would just leave her position for a few minutes for a smoke, small-talk or whatever, outside her assigned break times, she could afaik get a written warning, and at the second occasion the full wad of papers (aka been fired). On the other hand, if you count only times while they are on-a-call, with appropriate logging software, adding a few seconds per-call for overhead, as their worktime, they pretty soon will keep auto answer on to get the required number of work minutes during their shift, I would expect. But this is not as much a technical problem as a social one: If your agents are unmotivated, they might spend time talking off-business to any caller/callee on the phone that seems to be interested in small-talk, and _that_ you could hardly find out technically. So you might get an upgrade without paying for the deinstallation of the previous meatware, but the installation process of course has costs. BTW putting too much pressure on your agents might do bad things to their effiency, motivation, even mental health. Getting the balance between control and good atmosphere right is not easy, and something that cannot be generalized but must be tailored to the situation. The value of a human asset (imagine me vomiting my way through those words) can materialize in the number of sales, calls, ... and also in the customer experience he creates, which is hard to be counted in numbers. For example, I recently bought some music instrument and accessories at a phone-order company. The people there were relaxed, friendly, helpful and made the effort of giving me competent, quick information that I needed. All contact with them was extremely positive. As I needed some more stuff that I knew was a bit cheaper at another store (which only deals with customers in a matter-of-fact way), I decided to honour that effort. I also recommended the company to friends, which they probably will never know about and as such cannot count in as a bonus for their sales personell. *Just my loose change. Man, there were lots of coins in that purse.* Makes Vista look like a picnic. IMO Vista is an apple short of one ;-) To get the Asterisk relevant topics: You could - count on-call minutes to rate agent performance - track off-call intervals on a certain line and so track the turned-off auto answer - do some social engineering or policy work to get this sorted in a non-technical way (work contract terms, etc) - pay some softphone manufacturer to implement needed changes BTW what would hinder your agents from shutting down the softphone app when they do not want to answer calls? What would hinder them from just not talking to the caller when they do not want to? Best regards, Anselm By American (I assume you mean the USA and leaving out the rest of North and South America) employment laws vary state by state. Maryland is an At will state which means you can be fired or quit at will (unless you have a contract in place that says differently). Agents are the absolute best at finding bugs and ways of beating the call center system. You can lock them down but they will find other ways around every time. I think holding a company wide meeting about the issue and creating an official policy on what is acceptable and what is not. Make it a zero tolerance policy. Then wait a few days, the first one you catch messing around with the system, make a very public and open dismissal of the employee and make sure the other agents know exactly why their co-worker was dismissed. There will be quite a bit of gossip and possibly turnover depending on the rep that was dismissed but you will see much less of the offending behavior. Don't stop at one, make sure you let the agents know that they are being monitored and will receive no warning if in violation. Sounds harsh, but you have to lay down the law. Even if you catch your best rep doing it, immediate termination, that will really get the point across. Thanks, Steve
Re: [asterisk-users] Interesting Conference Request - Can this be done ?
Dovid B wrote: Hi List, I have a client that has an interesting request. He wants to have people call in to a conference room and all be able to talk however they should not hear each other. There should be admin access to for one user to call in and be able to listen in to everyone that is talking (they may want this admin to be able to talk to). Any ideas ? Thanks. Dovid WHAT? I don't get it. What good is a conference if nobody can hear each other? Is this to spy on offices and other locations with phones or something? Thanks, Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Astribank and caller ID from PSTN
On Tue, 2007-09-18 at 19:33 -0500, Guillermo Salas M. wrote:
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
[..]
One thing I suspect is not waiting enough.
Try adding the following to your dialplan:
[pstn-test]
exten = s,1,Wait(1)
exten = s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten = s,n,Playback(tt-monkeys)
exten = s,n,Goto(from-zaptel,s,1)
And then set in zapata.conf: context=pstn-test
Other than that, there are two obvious sanity checks:
1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected
I've connected one phone to the line that was connected on the port 4 of
the astribank. Called from my mobile and the caller id is displayed on
the phone.
2. boot the same system from our live CD and see if caller ID is
detected there.
Booted with the version Xorcom Rapid LiveCD (1.0.2.4131) and configured
the following:
- Created one trunk called g1;
- creted one SIP extension called 666 ;
- edited /etc/asterisk/zapata-channels.conf with:
;;; line=4 XPP_FXO/00/00/3 (no pcm)
signalling=fxs_ks
callerid=asreceived
group=1
context=from-zaptel
channel = 4
context=default
- created one incoming route with freebpx,
- all the calls that are coming on the port 4 of the astribank will be
redirected to the sip extension 666;
Now, dialing from my mobile phone again to the line connected to the
port 4 of the astribank is showing me the called ID on the 666 extension
as Unknown:
[..]
Please check the Zaptel hardware listing from the live cd:
http://pastebin.ca/702694
Regards,
--
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esq
Edificio Barre #2 Primer Piso
Telefono : +593 5 262 8071
Celular : +593 9 985 5138
e-mail : [EMAIL PROTECTED]
www : http://www.manta.telconet.net
http://www.telcocarrier.net
SIP : [EMAIL PROTECTED]
FWD : 558563
Linux User: 255902
Beat me, whip me, make me use Windows!
Please avoid sending me Word or PowerPoint attachments.
See http://www.fsf.org/philosophy/no-word-attachments.html
Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting
___
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--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] (Getting OT) Re: Call Center SoftPhone with Auto Answer
If you have to resort to such measures to get people to work for you in a motivated fashion, you're doing something very, VERY wrong. On Tue, 18 Sep 2007, Steve Totaro wrote: Anselm Martin Hoffmeister wrote: Am Dienstag, den 18.09.2007, 17:33 -0400 schrieb James FitzGibbon: On 9/18/07, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? I do not know American work law, but if you tell your people to NOT turn off auto answer, and they do for having a break, would that not count as work refusal? As long as they get all the breaks they are supposed to take, of course. If for example a cashier in a supermarket here in Germany would just leave her position for a few minutes for a smoke, small-talk or whatever, outside her assigned break times, she could afaik get a written warning, and at the second occasion the full wad of papers (aka been fired). On the other hand, if you count only times while they are on-a-call, with appropriate logging software, adding a few seconds per-call for overhead, as their worktime, they pretty soon will keep auto answer on to get the required number of work minutes during their shift, I would expect. But this is not as much a technical problem as a social one: If your agents are unmotivated, they might spend time talking off-business to any caller/callee on the phone that seems to be interested in small-talk, and _that_ you could hardly find out technically. So you might get an upgrade without paying for the deinstallation of the previous meatware, but the installation process of course has costs. BTW putting too much pressure on your agents might do bad things to their effiency, motivation, even mental health. Getting the balance between control and good atmosphere right is not easy, and something that cannot be generalized but must be tailored to the situation. The value of a human asset (imagine me vomiting my way through those words) can materialize in the number of sales, calls, ... and also in the customer experience he creates, which is hard to be counted in numbers. For example, I recently bought some music instrument and accessories at a phone-order company. The people there were relaxed, friendly, helpful and made the effort of giving me competent, quick information that I needed. All contact with them was extremely positive. As I needed some more stuff that I knew was a bit cheaper at another store (which only deals with customers in a matter-of-fact way), I decided to honour that effort. I also recommended the company to friends, which they probably will never know about and as such cannot count in as a bonus for their sales personell. *Just my loose change. Man, there were lots of coins in that purse.* Makes Vista look like a picnic. IMO Vista is an apple short of one ;-) To get the Asterisk relevant topics: You could - count on-call minutes to rate agent performance - track off-call intervals on a certain line and so track the turned-off auto answer - do some social engineering or policy work to get this sorted in a non-technical way (work contract terms, etc) - pay some softphone manufacturer to implement needed changes BTW what would hinder your agents from shutting down the softphone app when they do not want to answer calls? What would hinder them from just not talking to the caller when they do not want to? Best regards, Anselm By American (I assume you mean the USA and leaving out the rest of North and South America) employment laws vary state by state. Maryland is an At will state which means you can be fired or quit at will (unless you have a contract in place that says differently). Agents are the absolute best at finding bugs and ways of beating the call center system. You can lock them down but they will find other ways around every time. I think holding a company wide meeting about the issue and creating an official policy on what is acceptable and what is not. Make it a zero tolerance policy. Then wait a few days, the first one you catch messing around with the system, make a very public and open dismissal of the employee and make sure the other agents know exactly why their co-worker was dismissed. There will be quite a bit of gossip and possibly turnover depending on the rep that was dismissed but you will see much less of the offending behavior. Don't stop at one, make sure you let the agents know that they are being monitored and will receive no warning if in violation. Sounds harsh,
Re: [asterisk-users] Astribank and caller ID from PSTN
Hi all,
On Sat, 2007-09-15 at 22:05 +0300, Tzafrir Cohen wrote:
Hi Guillermo,
On Sat, Sep 15, 2007 at 01:18:49PM -0500, Guillermo Salas M. wrote:
Hello,
I've one astribank with 8 FXO unit and 8 pstn lines connected to the
astribank. When I receive calls on my ipphone I get always Unknown
callerid.
[..]
One thing I suspect is not waiting enough.
Try adding the following to your dialplan:
[pstn-test]
exten = s,1,Wait(1)
exten = s,n,NoOp(Got number ${CALLERID(all)} on ${CHANNEL})
; If you're just testing:
;exten = s,n,Playback(tt-monkeys)
exten = s,n,Goto(from-zaptel,s,1)
And then set in zapata.conf: context=pstn-test
Other than that, there are two obvious sanity checks:
1. Connect an analog phone with with caller ID display to the same port
and see that caller ID is indeed detected
I've connected one phone to the line that was connected on the port 4 of
the astribank. Called from my mobile and the caller id is displayed on
the phone.
2. boot the same system from our live CD and see if caller ID is
detected there.
Booted with the version Xorcom Rapid LiveCD (1.0.2.4131) and configured
the following:
- Created one trunk called g1;
- creted one SIP extension called 666 ;
- edited /etc/asterisk/zapata-channels.conf with:
;;; line=4 XPP_FXO/00/00/3 (no pcm)
signalling=fxs_ks
callerid=asreceived
group=1
context=from-zaptel
channel = 4
context=default
- created one incoming route with freebpx,
- all the calls that are coming on the port 4 of the astribank will be
redirected to the sip extension 666;
Now, dialing from my mobile phone again to the line connected to the
port 4 of the astribank is showing me the called ID on the 666 extension
as Unknown:
-- Starting simple switch on 'Zap/4-1'
-- Executing NoOp(Zap/4-1, Entering from-zaptel with DID == ) in
new stack
-- Executing Ringing(Zap/4-1, ) in new stack
-- Executing Set(Zap/4-1, DID=s) in new stack
-- Executing NoOp(Zap/4-1, DID is now s) in new stack
-- Executing GotoIf(Zap/4-1, 1?zapok:notzap) in new stack
-- Goto (from-zaptel,s,8)
-- Executing NoOp(Zap/4-1, Is a Zaptel Channel) in new stack
-- Executing Set(Zap/4-1, CHAN=4-1) in new stack
-- Executing Set(Zap/4-1, CHAN=4) in new stack
-- Executing Macro(Zap/4-1, from-zaptel-4|s|1) in new stack
-- Executing NoOp(Zap/4-1, Entering macro-from-zaptel-4 with DID
= s) in new stack
-- Executing Gosub(Zap/4-1, app-blacklist-check|s|1) in new
stack
-- Executing LookupBlacklist(Zap/4-1, ) in new stack
-- Executing GotoIf(Zap/4-1, 0?blacklisted) in new stack
-- Executing Return(Zap/4-1, ) in new stack
-- Executing Set(Zap/4-1, __FROM_DID=s) in new stack
-- Executing Goto(Zap/4-1, ext-local|666|1) in new stack
-- Goto (ext-local,666,1)
== Channel 'Zap/4-1' jumping out of macro 'from-zaptel-4'
-- Executing Macro(Zap/4-1, exten-vm|666|666) in new stack
-- Executing Macro(Zap/4-1, user-callerid) in new stack
-- Executing NoOp(Zap/4-1, user-callerid: ) in new stack
-- Executing GotoIf(Zap/4-1, 0?report) in new stack
-- Executing GotoIf(Zap/4-1, 0?start) in new stack
-- Executing Set(Zap/4-1, REALCALLERIDNUM=) in new stack
-- Executing NoOp(Zap/4-1, REALCALLERIDNUM is ) in new stack
-- Executing Set(Zap/4-1, AMPUSER=) in new stack
-- Executing Set(Zap/4-1, AMPUSERCIDNAME=) in new stack
-- Executing GotoIf(Zap/4-1, 1?report) in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp(Zap/4-1, TTL: ARG1: 666) in new stack
-- Executing GotoIf(Zap/4-1, 0?continue) in new stack
-- Executing Set(Zap/4-1, __TTL=64) in new stack
-- Executing GotoIf(Zap/4-1, 1?continue) in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp(Zap/4-1, Using CallerID ) in new stack
-- Executing Set(Zap/4-1, FROMCONTEXT=exten-vm) in new stack
-- Executing Set(Zap/4-1, VMBOX=666) in new stack
-- Executing Set(Zap/4-1, EXTTOCALL=666) in new stack
-- Executing Set(Zap/4-1, CFUEXT=) in new stack
-- Executing Set(Zap/4-1, CFBEXT=) in new stack
-- Executing Set(Zap/4-1, RT=15) in new stack
-- Executing Macro(Zap/4-1, record-enable|666|IN) in new stack
-- Executing GotoIf(Zap/4-1, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(Zap/4-1, recordingcheck|20070919-002336|
asterisk-5150-1190161411.3) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck|20070919-002336|asterisk-5150-1190161411.3: Inbound
recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(Zap/4-1, No recording needed) in new stack
-- Executing Macro(Zap/4-1, dial|15|tr|666) in new stack
-- Executing DeadAGI(Zap/4-1, dialparties.agi) in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing '/etc/asterisk/manager.conf':
Re: [asterisk-users] Comfort noice sample (gsm/mp3)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jim Boykin wrote: Where do I get sound file for comfort noise. GSM or MP3 is fine. What kind of comfort noise do you mean? Like background static or music? If you just want noise (as in pink or white noise), I could make you up an MP3 or ulaw/alaw file. Any idea how loud you want it? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG8IoPDQNt8rg0Kp4RAhCrAJ0Wx+v26VUbyvJsAWGwJJ5jFjxaEwCeKHZA b5NdDXfkQD36EAXEKpmZUHU= =Y4p5 -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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.conf best practices?
All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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.conf best practices?
Use the extension, and use grep to determine which account uses which phone. For example I provision my spa9xx phones from a subdirectory on apache called spa which on slackware is at: /var/www/htdocs/spa/ doing: grep 123 /var/www/htdocs/spa/* will tell you which phone it is. On 9/18/07, Erik Anderson [EMAIL PROTECTED] wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik -- Erik Anderson http://andersonfam.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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.conf best practices?
Realtime and sip_buddies in mysql works well for very large installations. PaulH On Tue, 2007-09-18 at 22:11 -0500, Erik Anderson wrote: All - I've been wrestling with how to best structure the sip device accounts on a new asterisk server I'm deploying. All of the sip devices (currently only Linksys SPA941s) will reside on the same subnet as the server, and I have already set up a decent automatic provisioning system for the phones. When the rollout is complete, there will be about 100 SIP devices authenticating and routing calls through this server. The question is what to use for the username portion of the SIP account. Part of me says that I should standardize on using each phone's MAC address as the sip account UID, like so: ; Joe Smith, x123 [000E08DA0409] secret = blahblah ... and so on and so forth Doing it that way is nice for standardization's sake, but it makes the dialplan quite a bit more complex. The obvious alternative is to use the extension as the sip UID: ; Joe Smith, x123 [123] secret = blahblah ... This makes the dialplan *much* more simple, but when looking through sip.conf, it's not as immediately obvious what device should be authenticating with that account. Since this is my first large-ish asterisk deployment, I'm seeking the advice of those who have gone before me. What tactic (one of the above options or otherwise) is best to keep your sip.conf sane? Thanks! -Erik ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Linux limits
safe_asetrisk bundled with the package, does increase the file limits in quite a neat way, with some other good setups. Edit MAXFILES or SYSMAXFILES as required. Also, I've read posts online, advising not to use safe_asterisk. Any experiences on this one, anyone? cheers - Ben. Jay R. Ashworth wrote: On Tue, Sep 18, 2007 at 04:22:29PM -0400, Alex Balashov wrote: On Tue, 18 Sep 2007, Wai Wu wrote: Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for asterisk1/700 Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in Linux itself. [ top posting fixed so I can comment as well ] You have to increase the amount of available file descriptors per process: http://hausheer.osola.com/docs/11%C2%A0%C2%A0 These days, I beleve the typical place to fix that is actually in /etc/sysctl.conf, in most distros: http://www.cs.wisc.edu/condor/condorg/linux_scalability.html That page notes it for RedHat derived distros, but I'm pretty sure SuSe puts it there as well. Cheers, -- jra 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. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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.conf best practices?
On 9/18/07, C F [EMAIL PROTECTED] wrote: Use the extension, and use grep to determine which account uses which phone. For example I provision my spa9xx phones from a subdirectory on apache called spa which on slackware is at: /var/www/htdocs/spa/ doing: grep 123 /var/www/htdocs/spa/* will tell you which phone it is. That's a great idea - probably seems like the most simple option. Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Voicemail.conf
Hi Paul, The way to specify the email_id is as follows 8000 = 8000,ajay,[EMAIL PROTECTED] Bye and take care. On 9/17/07, Paul Hales [EMAIL PROTECTED] wrote: Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '', but that didn't work...) later, PaulH ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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.conf best practices?
The obvious alternative is to use the extension as the sip UID: Use the extension as the UID and add the mac address as a comment. Like so: [123] ; Joe Smith ;mac=000E08DA0409 secret = blahblah ... and so on and so forth This will give the best of both worlds. The mac is readily available and the dialplan is clear. I usually try to go one further and setup dhcp to set the last octet of the IP address to the extension number. This makes it easy to point a browser to the phone for configuration as well. John ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] Comfort noice sample (gsm/mp3)
Thanks Matt, just minimal volume to suite comfort noise. Jim On 9/19/07, Matt Riddell [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jim Boykin wrote: Where do I get sound file for comfort noise. GSM or MP3 is fine. What kind of comfort noise do you mean? Like background static or music? If you just want noise (as in pink or white noise), I could make you up an MP3 or ulaw/alaw file. Any idea how loud you want it? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG8IoPDQNt8rg0Kp4RAhCrAJ0Wx+v26VUbyvJsAWGwJJ5jFjxaEwCeKHZA b5NdDXfkQD36EAXEKpmZUHU= =Y4p5 -END PGP SIGNATURE- ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided 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] VoIP Provider for business
Can someone suggests a good and resonable cost voip provider with business unlimited plan in USA and allows simultaneous outgoing calling. Thanks ~Jim ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
