[asterisk-users] signalling default value
Hi list What is the default value for signalling in /usr/local/etc/asterisk/chan_dahdi.conf file?-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFMBAEBCgA3BQJSSoHxMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJYmRB/USyTbAqhAsnFZSGGjIcLK7 uQ3nsVNGcmE18LaBN/XFicwp5UjVB5Euju+fjKu1FhqAzECsAPMup/1JUytikmYz +32wV5YL1SNKMA/ddi/zvVa9qIbKA9yP1HuBilpD+W0DO3hdnzr2xrdR1S2z5PGZ pnYWsVlXbWYEslOuK1oaMqINoxWbsQulwQi86GPTCwPtZmhcLrvBm1sDFxWb/oPP lsPy33ZH5BeQ/XEf6nWfoiEu4Hk2S0brCH74zsz9uD6PKL1CFdLcpWv/4k5M+Mly At2PC+leZZ/TX3VNqbasslQkyv/QLZIQVtG0qQ7DGflnkrzNi5/pNV7CVT5sdPQ= =5rja -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
On 10/02/2013 09:33 AM, s m wrote: and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? AFAIK, you need one license for each channel that is transcoding from one given codec to g729 (or the other way around). So if at any given time on an asterisk box you would have a maximum of 3 simultaneous calls that are g729 at one end and ulaw at the other, you would need a license key for 3 transcoding channels. Anyone, please correct me if I am wrong on this. Frederic -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
Are these end points Hard IP Phones having g729 codec? If yes then you dont need any license. Just download passthrough g729 license. Mitul On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com wrote: On 10/02/2013 09:33 AM, s m wrote: and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...**pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? AFAIK, you need one license for each channel that is transcoding from one given codec to g729 (or the other way around). So if at any given time on an asterisk box you would have a maximum of 3 simultaneous calls that are g729 at one end and ulaw at the other, you would need a license key for 3 transcoding channels. Anyone, please correct me if I am wrong on this. Frederic -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
thanks guys for your replies. no, these endpoints are soft phones which can have different codec. now, for my scenario, how many license key in needed? On Wed, Oct 2, 2013 at 11:23 AM, Mitul Limbani mi...@enterux.in wrote: Are these end points Hard IP Phones having g729 codec? If yes then you dont need any license. Just download passthrough g729 license. Mitul On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com wrote: On 10/02/2013 09:33 AM, s m wrote: and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...**pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? AFAIK, you need one license for each channel that is transcoding from one given codec to g729 (or the other way around). So if at any given time on an asterisk box you would have a maximum of 3 simultaneous calls that are g729 at one end and ulaw at the other, you would need a license key for 3 transcoding channels. Anyone, please correct me if I am wrong on this. Frederic -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference password and time limitation
So this web-meetme applicationrequires to enable the real time in asterisk? Where I can find documentation about web-meetme application? Regards Bilal On Tuesday, October 1, 2013 6:57 PM, Dan Austin dan_aus...@phoenix.com wrote: Look at Web-MeetMe ( http://sf.net/projects/web-meetme ) If you are on Asterisk 1.6.7 or later you have access to RealTime MeetMe conference storage, otherwise you need to use a script and Asterisk application included with the WMM download. Dan From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, October 01, 2013 12:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme conference password and time limitation Hello; We need to have admin page, so the administrator can create passwords to be used to join the meetme conferences and can determine the allowed time .. Well, the admin interface can be done easy (I do not know if there is something ready), and the password and the time limitation can be added to the database (or even text file), but how asterisk can use it? Do I need to use the AGI to read/write from database and do the meetme conference within the AGI script it self, or there is simpler method? Regards Bilal-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invalid options
Hello, While connecting to my voicemail, I noticed that Asterisk may perform some tasks isn't included in the options levels that currently played! For ex: I am listening to the main menu which asking me to press 1 for new messages, 2 change folders, 0 mailbox options which is OK for these numbers, but if I pressed 7 by mistake for example it will tell me Message deleted which it doesn't make sense!Is there a way to stop this behavior, let's say to run invalid option if any other number was pressed doesn't belong to the current menu level? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi_dummy is more accurate than core timer?
Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Below is some stats (trimmed, but you get the idea). Thanks! /Johan ** With Dahdi 2.7.0.1, and core timers: 99.998% 99.611% 99.615% 99.997% 99.993% 99.997% 99.996% 99.608% 99.999% 99.612% 99.607% 99.613% 99.999% 99.998% 99.994% 99.609% --- Results after 177 passes --- Best: 100.000% -- Worst: 99.604% -- Average: 99.901099% Cummulative Accuracy (not per pass): 99.998 ** With Dahdi 2.7.0.1, and dahdi_dummy loaded: 99.993% 99.998% 99.998% 99.993% 99.996% 99.998% 99.996% 99.998% 99.998% 99.997% 99.999% 99.998% 99.996% 99.998% 99.999% 99.997% --- Results after 177 passes --- Best: 100.000% -- Worst: 99.993% -- Average: 99.997738% Cummulative Accuracy (not per pass): 99.998 ** With Dahdi 2.7.0.1, and Wildcard TE220 providing timing 99.981% 99.983% 99.983% 99.982% 99.983% 99.982% 99.984% 99.981% 99.982% 99.983% 99.984% 99.981% 99.980% 99.984% 99.983% 99.983% --- Results after 177 passes --- Best: 99.996% -- Worst: 99.974% -- Average: 99.982104% Cummulative Accuracy (not per pass): 99.982 Kernel: 2.6.32-5-openvz-amd64 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
On 02/10/13 12:17, Johan Wilfer wrote: Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Its a little different when you are using meetme as its an application built into dahdi itself and not a native asterisk application. It will therefore always use dahdi for its timing. If dahdi doesnt have a hardware interface (sangoma sell a usb based timing source if you want a hardware source) then it will use a software timing source of some form. I dont know what method it uses. Asterisk itself will use dahdi for timing but if res_timerfd is available it will use that itself. With your kernel version timerfd should be available. So if you run the following command asterisk will perform 1024 ticks over 1000ms which is equivalent to the dahdi test of 8192 over 8000ms (if you run it a few times). You can see in my case asterisk is using timerfd and no issues at all with the timing. timing test 1024 Attempting to test a timer with 1024 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 1024 timer ticks We are developing a conferencing feature and rather than use meetme with its dahdi requirement we are using confbridge10 instead so we dont have to have dahdi installed and it will use the better timerfd timing source. More information about timing sources can be found at :- https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
timing test 1024 Attempting to test a timer with 1024 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 1024 timer ticks For a typical virtual machine the values for timerfd can be significantly worse, even on otherwise powerful host systems. For a recent Fedora based test system running under VirtualBox I get typically varying values as low as 550 for timing test 1024, but this does not seem to have any major influence on plain calls. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
2013-10-02 13:55, Gareth Blades skrev: On 02/10/13 12:17, Johan Wilfer wrote: Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? - Is a hardware-card for timing the most efficient way to get timing even if I just use the card for the timing? Its a little different when you are using meetme as its an application built into dahdi itself and not a native asterisk application. It will therefore always use dahdi for its timing. If dahdi doesnt have a hardware interface (sangoma sell a usb based timing source if you want a hardware source) then it will use a software timing source of some form. I dont know what method it uses. Yes, this is for a legacy application that are using Meetme. Maybe I was unclear above but with core timer I meant just modprobe dahdi. dahdi_dummy = modprobe dahdi_dummy and the hw card used wct4xxp. More information about timing sources can be found at :- https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces The wiki states: As of DAHDI Linux 2.3.0 the dahdi_dummy module has been removed and its functionality moved into the main dahdi kernel module. As long as the dahdi module is loaded, it will provide timing to Asterisk either through installed telephony hardware or utilizing the kernel timing facilities when separate hardware is not available. But when I test just dahdi (core timer, no dahdi_dummy) I get distortion and bad quality in the Meetme-conferences. This does not happen with dahdi_dummy. -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invalid options
Hi Ahmed, Seems to be that you have a problem recognizing DTMF digits. Do you have RFC2833 as DTMF protocol in both Asterisk and soft/hard-phone ? Rgds, On Wed, 2013-10-02 at 11:24 +0200, Asmaa Ahmed wrote: Hello, While connecting to my voicemail, I noticed that Asterisk may perform some tasks isn't included in the options levels that currently played! For ex: I am listening to the main menu which asking me to press 1 for new messages, 2 change folders, 0 mailbox options which is OK for these numbers, but if I pressed 7 by mistake for example it will tell me Message deleted which it doesn't make sense! Is there a way to stop this behavior, let's say to run invalid option if any other number was pressed doesn't belong to the current menu level? Thanks. -- Rodrigo Montiel Ing. en Telecomunicaciones GNU/Linux VoIP System Support Cel: +549 351 2581376 Freetech Solutions www.freetechsolutions.com.ar Tel/Fax: +54 351 6387585 attachment: logo.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
In your scenario, all the calls are from endpoints on 181 to endpoints on 183. If the endpoint devices are similar, it seems to me that there should be no need to transcode-you can use a codec common to the endpoints. 729 would not be required. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m Sent: Wednesday, October 02, 2013 2:34 AM To: Dominik George Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] is g729 codec free? or under license??? thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFMBAEBCgA3BQJSSoHxMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJYmRB/USyTbAqhAsnFZSGGjIcLK7 uQ3nsVNGcmE18LaBN/XFicwp5UjVB5Euju+fjKu1FhqAzECsAPMup/1JUytikmYz +32wV5YL1SNKMA/ddi/zvVa9qIbKA9yP1HuBilpD+W0DO3hdnzr2xrdR1S2z5PGZ pnYWsVlXbWYEslOuK1oaMqINoxWbsQulwQi86GPTCwPtZmhcLrvBm1sDFxWb/oPP lsPy33ZH5BeQ/XEf6nWfoiEu4Hk2S0brCH74zsz9uD6PKL1CFdLcpWv/4k5M+Mly At2PC+leZZ/TX3VNqbasslQkyv/QLZIQVtG0qQ7DGflnkrzNi5/pNV7CVT5sdPQ= =5rja -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] iax: unable to transfer - one way audio
On 09/30/2013 12:09 PM, Sean Darcy wrote: On 09/28/2013 11:11 AM, Asghar Mohammad wrote: Hi, If you post your configuration someone may help you. On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy seandar...@gmail.com mailto:seandar...@gmail.com wrote: On 09/27/2013 09:08 PM, Sean Darcy wrote: We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from zoiperipaddr: requested format = speex, requested prefs = (), actual format = ulaw, host prefs = (silk16|ulaw|gsm|g722), priority = mine -- Executing [8447@nz-in:1] Dial(IAX2/n4-270, IAX2/sydney) in new stack -- Called IAX2/sydney -- Call accepted by nyipaddr (format ulaw) -- Format for call is (ulaw) -- IAX2/sydney-8819 is ringing -- IAX2/sydney-8819 answered IAX2/n4-270 -- Channel 'IAX2/n4-270' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer -- Channel 'IAX2/sydney-8819' unable to transfer The NY server: -- Accepting AUTHENTICATED call from sydneyipaddr: -- requested format = ulaw, -- requested prefs = (ulaw|silk16|gsm|g722), -- actual format = ulaw, -- host prefs = (ulaw|gsm|g722), -- priority = mine -- Executing [s@incoming-nz:1] Goto(IAX2/home-2152, incoming,s,nz-in) in new stack -- Goto (incoming,s,5) -- Executing [s@incoming:5] Dial(IAX2/home-2152, DAHDI/g0SIP/250SIP/251,60,__tT) in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called DAHDI/g0 -- Called SIP/250 -- Called SIP/251 -- DAHDI/1-1 is ringing -- SIP/251-001d is ringing -- SIP/250-001c is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 answered IAX2/home-2152 -- Channel 'IAX2/home-2152' unable to transfer -- Hanging up on 'DAHDI/1-1' Any help appreciated. sean FWIW, sydney server is 11.5.1, ny server 11.6.0-rc1. sean Thanks for the reply. Here's sydney iax.conf: [general] bandwidth=medium trunkmtu=1240 disallow=all allow=silk16 allow=ulaw allow=gsm allow=g722 jitterbuffer=yes forcejitterbuffer=no trunktimestamps=yes authdebug=yes tos=ef cos=5 autokill=yes codecpriority=caller [default](!) type=friend auth=md5 host=dynamic context=nz-in qualify=1000 setvar=Protocol=IAX2 [n4](default) secret=n4pw callerid=callerid [sydney](default) secret=pwsydney username=home-sydney home iax.conf: [general] bandwidth=medium disallow=all allow=ulaw allow=gsm allow=g722 jitterbuffer=yes forcejitterbuffer=no tos=0x10 autokill=yes register = sydney:pwsydney@sydneyipaddr [nz](!) type=friend secret=pwhome context=incoming-nz [home-sydney](nz) host=sydneyipaddr username=sydney callerid=House sean Any thoughts? Anybody? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] is g729 codec free? or under license???
When calling between two g729 client endpoints you do not need any licenses as long as no audio prompts or voicemail is evolved. Also as a sip trunk provider we offer g729 as a source and destination codec this allows you to make calls in and out using g729 (most carrier grade providers offer this option) You really only need to buy the number of g729 licenses that you will need for callers that require simultaneous transcoding. This is when a callers stream in or out will need to be converted to another codec format. This occurs when callers are jumping from say g729 to g711 or g729 to g722, g729 to gsm. If you plan things right and make sure any audio prompts your system is using are recorded in g729 as well as g711 and g722 you will reduce the number of g729 license considerable. Process that use a lot of g729 transcodes. ConfBridge uses g722 so all g729 has to be converted to and from g722 so 10 g729 callers to a confbridge would likely require 10 codecs (**See confbridge trick below). If you have prompts that are not pre-encoded in g729 those would use a transcoder license while playing. Voicemail would require a license as g729 has to be transcoded to one of the storage formats. The real number is based on how you are using your system. ConfBridge Trick - Have seen this used for voicemail as well, Make sure you test when using this method. If you can live with using higher bandwidth to the asterisk switch when using confbridges (endpoints also have to support in call reinvites correctly) you can force endpoints to re-invite to g722 before dropping into the conference bridge. This has the upside of not needing to transcode on the server thus improving performance and reducing g729 license requirements. This comes at the cost of needing higher bandwidth between the client endpoints and the phone. Figure about double the bandwidth when using this method. It may or may not be worth it to you depending on your scenario. Please let us know if this information helps you. Thanks Bryant Zimmerman Sr. Systems Architect Grand Dial Communications , A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) From: Don Kelly d...@donkelly.biz Sent: Wednesday, October 2, 2013 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] is g729 codec free? or under license??? In your scenario, all the calls are from endpoints on 181 to endpoints on 183. If the endpoint devices are similar, it seems to me that there should be no need to transcode-you can use a codec common to the endpoints. 729 would not be required. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m Sent: Wednesday, October 02, 2013 2:34 AM To: Dominik George Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] is g729 codec free? or under license??? thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid iQFMBAEBCgA3BQJSSoHxMBxEb21pbmlrIEdlb3JnZSAobW9iaWxlIGtleSkgPG5p a0BuYXR1cmFsbmV0LmRlPgAKCRAvLbGk0zMOJYmRB/USyTbAqhAsnFZSGGjIcLK7 uQ3nsVNGcmE18LaBN/XFicwp5UjVB5Euju+fjKu1FhqAzECsAPMup/1JUytikmYz +32wV5YL1SNKMA/ddi/zvVa9qIbKA9yP1HuBilpD+W0DO3hdnzr2xrdR1S2z5PGZ pnYWsVlXbWYEslOuK1oaMqINoxWbsQulwQi86GPTCwPtZmhcLrvBm1sDFxWb/oPP lsPy33ZH5BeQ/XEf6nWfoiEu4Hk2S0brCH74zsz9uD6PKL1CFdLcpWv/4k5M+Mly At2PC+leZZ/TX3VNqbasslQkyv/QLZIQVtG0qQ7DGflnkrzNi5/pNV7CVT5sdPQ= =5rja -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] is g729 codec free? or under license???
Hi guys. I would also add that in countries which do not recognize software patents (New Zealand for example) there is no need to get a license and the codecs can therefore be downloaded from http://asterisk.hosting.lv/ and used freely. In countries where there is ambiguity about certain software patents, (such as Canada or Germany) well you take your chances. Countries that do recognize them (such as the US or Japan) you'd be smart to get a license. Also I do believe the US patent on g729 expires next year anyway, so again you might want to weigh that costs/risks factor too. cheers, darryl On Wed, 2013-10-02 at 09:55 -0400, Bryant Zimmerman wrote: When calling between two g729 client endpoints you do not need any licenses as long as no audio prompts or voicemail is evolved. Also as a sip trunk provider we offer g729 as a source and destination codec this allows you to make calls in and out using g729 (most carrier grade providers offer this option) You really only need to buy the number of g729 licenses that you will need for callers that require simultaneous transcoding. This is when a callers stream in or out will need to be converted to another codec format. This occurs when callers are jumping from say g729 to g711 or g729 to g722, g729 to gsm. If you plan things right and make sure any audio prompts your system is using are recorded in g729 as well as g711 and g722 you will reduce the number of g729 license considerable. Process that use a lot of g729 transcodes. ConfBridge uses g722 so all g729 has to be converted to and from g722 so 10 g729 callers to a confbridge would likely require 10 codecs (**See confbridge trick below). If you have prompts that are not pre-encoded in g729 those would use a transcoder license while playing. Voicemail would require a license as g729 has to be transcoded to one of the storage formats. The real number is based on how you are using your system. ConfBridge Trick - Have seen this used for voicemail as well, Make sure you test when using this method. If you can live with using higher bandwidth to the asterisk switch when using confbridges (endpoints also have to support in call reinvites correctly) you can force endpoints to re-invite to g722 before dropping into the conference bridge. This has the upside of not needing to transcode on the server thus improving performance and reducing g729 license requirements. This comes at the cost of needing higher bandwidth between the client endpoints and the phone. Figure about double the bandwidth when using this method. It may or may not be worth it to you depending on your scenario. Please let us know if this information helps you. Thanks Bryant Zimmerman Sr. Systems Architect Grand Dial Communications, A ZK Tech Inc. Company 616-299-5607 (mobile) 616-855-1030 Ext. 2003 (office) __ From: Don Kelly d...@donkelly.biz Sent: Wednesday, October 2, 2013 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] is g729 codec free? or under license??? In your scenario, all the calls are from endpoints on 181 to endpoints on 183. If the endpoint devices are similar, it seems to me that there should be no need to transcode-you can use a codec common to the endpoints. 729 would not be required. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m Sent: Wednesday, October 02, 2013 2:34 AM To: Dominik George Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] is g729 codec free? or under license??? thank you Dominik you help me a lot. and the last question is how many license key should i buy? i read that license for g729 is per-channel but i don't understand what channel exactly means here. this is my scenario : 10endpointspbx181...pbx182...pbx183...10endpoints pbx181 and pbx183 has 10 endpoints connected to them. the call between these endpoints are established by pbx182. if i want to buy a license for pbx182, how many license key do i need? just one because i have just one connection on it? or two, because two trunks is defined on it? or as many as endpoints which are connected to each other via pbx182? please help me to clarify channel concept in my mind. thanks in advance SAM On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 Hi, about g729, you mean if it get free g729 and all my systems (PBXs and routers) use g729 codec for setting a call, call is set without any problem? Yes, if all systems use g729 directly, you are ready to go. - -nik -BEGIN PGP SIGNATURE- Version: APG v1.0.8-fdroid
Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
Hi Garet,
ok but since the messages contain my own public IP with this method I'm
banning my public IP not the real attacker IP. Am I wrong?
Giorgio
On 10/01/2013 05:26 PM, Gareth Blades wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com
wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It
seems someone is attacking my asterisk PBX.
Is there a way to fix this problem?
in sip.conf I have guest connections permitted and have them going to
the default context which contains :-
[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion
Then in fail2ban I have it match the following :-
failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:
--
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Re: [asterisk-users] Dahdi_dummy is more accurate than core timer?
On Wed, Oct 02, 2013 at 01:17:15PM +0200, Johan Wilfer wrote: Hi, I have some servers that are dedicated to do meetme conferencing. From some previous test i concluded that I need to use dahdi_dummy as it is more accurate. If I did use the core timers in dahdi (not loading dahdi_dummy) I got bad quality in the conferences and dahdi_test showed 99.6% as worst. Hmm...this is the first report I've heard of dahdi_dummy being more performant than the core timer. I wonder if this has something to do with the fact that you're running under 2.6.32-5-openvz-amd64 which might be doing more work in the system timer (which is where the standard core timer work is processed). If you update to the latest 2.6.32-openvz kernel do you still have the audio problems in conferneces? After a quick search I found this stackoverflow question which talks about too much time spent in software interrupt context on 2.6.32-5-openvz being resolved by a kernel update. This could definitely be related: http://serverfault.com/questions/399837/software-interrupts-cpu-time-is-high-and-keeps-growing I thought maybe the issue as bad hardware for the timing or something else. But today I re-ran these tests on another server showing the same thing. - Can anybody comment on why DAHDI with core timers drop down to 99.6% occasionally? This is because when using the core timer, the timer is only scheduled to fire ever 4ms. The differences in each *individual* measurement you see is due to timer jitter + the increased interval leaking more of the slight jitter up to userspace. However, this isn't typically a problem when mixing audio in 20ms chunks by default as is typically done when you're using meetme conferences. The number that is generally more interesting is the Cummulative Accuracy which shows over the entire dahdi_test how close DAHDI was to processing the expected amount of audio. If you run dahdi_test with the -vv flag you can see how sometimes it's a little over and sometimes a little under. This is running under virtual box on a system configured with 4ms ticks and NO_HZ: # grep HZ /boot/config-$(uname -r) CONFIG_RCU_FAST_NO_HZ=y CONFIG_NO_HZ=y # CONFIG_HZ_100 is not set CONFIG_HZ_250=y # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=250 CONFIG_MACHZ_WDT=m # dahdi_test -vv -c 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8191.312 system clock sample intervals (99.992%) 8192 samples in 8198.448 system clock sample intervals (99.921%) 8192 samples in 8184.913 system clock sample intervals (99.913%) 8192 samples in 8191.640 system clock sample intervals (99.996%) 8192 samples in 8191.720 system clock sample intervals (99.997%) 8192 samples in 8192.128 system clock sample intervals (99.998%) 8192 samples in 8190.824 system clock sample intervals (99.986%) 8192 samples in 8192.256 system clock sample intervals (99.997%) 8192 samples in 8191.576 system clock sample intervals (99.995%) 8192 samples in 8191.631 system clock sample intervals (99.995%) --- Results after 10 passes --- Best: 99.998% -- Worst: 99.913% -- Average: 99.978985% Cummulative Accuracy (not per pass): 99.996 On another system with 10 ms timer ticks the jitter is increased, but even this system does not have any problems mixing audio in meetme conferences: $ zcat /proc/config.gz | grep HZ # CONFIG_RCU_FAST_NO_HZ is not set CONFIG_NO_HZ=y CONFIG_HZ_100=y # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set CONFIG_HZ=100 CONFIG_MACHZ_WDT=m $ sudo dahdi_test -vv -c 10 Opened pseudo dahdi interface, measuring accuracy... 8192 samples in 8159.872 system clock sample intervals (99.608%) 8192 samples in 8159.400 system clock sample intervals (99.602%) 8192 samples in 8239.680 system clock sample intervals (99.418%) 8192 samples in 8159.800 system clock sample intervals (99.607%) 8192 samples in 8239.576 system clock sample intervals (99.419%) 8192 samples in 8159.752 system clock sample intervals (99.606%) 8192 samples in 8159.848 system clock sample intervals (99.608%) 8192 samples in 8239.601 system clock sample intervals (99.419%) 8192 samples in 8159.624 system clock sample intervals (99.605%) 8192 samples in 8239.144 system clock sample intervals (99.425%) --- Results after 10 passes --- Best: 99.608% -- Worst: 99.418% -- Average: 99.531611% Cummulative Accuracy (not per pass): 99.995 When you explictly load the dahdi_dummy module, your results can change in a couple of ways. 1) dahdi_dummy tries to always schedule the system timer to fire at 1ms intervals (which it only will if the system is configured for CONFIG_HZ=1000). 2) If on a newer kernel, dahdi dummy will use kernel high resolution timers to increase the precision of the timer. However this shouldn't be necessary since the jitter in the normal kernel timer should be small compared to all the other jitter in a
Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
Hi Asghar,
surely this can improve security but what I'm looking for is something
to find the real attacker IP address and ban it. Fail2ban bans my own
public ip address.
Thank you
Giorgio
On 10/01/2013 05:53 PM, Asghar Mohammad wrote:
Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject
1st invite.
On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades
mailinglist+aster...@dns99.co.uk
mailto:mailinglist+aster...@dns99.co.uk wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com wrote:
Hi,
I get a lot of these messages on my Asterisk CLI:
Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9
as if my PBX machine is trying to authenticate to itself. It
seems someone is attacking my asterisk PBX.
Is there a way to fix this problem?
in sip.conf I have guest connections permitted and have them going
to the default context which contains :-
[default]
; all unauthenticated connection attempts from the internet come
in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion
Then in fail2ban I have it match the following :-
failregex = Registration from .* failed for \'HOST\' - Wrong
password
Unauthenticated call attempt .*\@HOST\:
--
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Re: [asterisk-users] signalling default value
On Wed, Oct 2, 2013 at 1:41 AM, Nomad Esst noname.e...@yahoo.com wrote: Hi list What is the default value for signalling in /usr/local/etc/asterisk/chan_dahdi.conf file? You should always be explicit in setting that value. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9
On 02/10/13 16:13, gincantalupo wrote: Hi Garet, ok but since the messages contain my own public IP with this method I'm banning my public IP not the real attacker IP. Am I wrong? Giorgio No the asterisk dialplan entry is pulling the IP address out of the SIP Contact: header which in the attacks we have seen always seems to be the correct IP address. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invalid options
Hello, I don't have a problem in recognizing DTMF digits. My problem is breaking the hierarchy level.Let's say I am in the main menu and I have three options ( 1 for new messages, 2 change folders, 0 mailbox options ) all of them are fine I can go through them by pressing the mentioned numbers, but if I pressed another number like 5, 6,... will hear something out of the three options I have, something in different level. if I pressed 7 for example, will hear something like (message deleted) which should be OK only if I am inside option (1 new messages) and asked to delete one of it, so I will press 7... but I was expecting to hear something like invalid option or do nothing at all if I am not inside (new messages) menu! Hope I am more clear! Thanks. From: rodrigo.mont...@freetechsolutions.com.ar To: asterisk-users@lists.digium.com Date: Wed, 2 Oct 2013 10:15:58 -0300 Subject: Re: [asterisk-users] Invalid options Hi Ahmed, Seems to be that you have a problem recognizing DTMF digits. Do you have RFC2833 as DTMF protocol in both Asterisk and soft/hard-phone ? Rgds, On Wed, 2013-10-02 at 11:24 +0200, Asmaa Ahmed wrote: Hello, While connecting to my voicemail, I noticed that Asterisk may perform some tasks isn't included in the options levels that currently played! For ex: I am listening to the main menu which asking me to press 1 for new messages, 2 change folders, 0 mailbox options which is OK for these numbers, but if I pressed 7 by mistake for example it will tell me Message deleted which it doesn't make sense! Is there a way to stop this behavior, let's say to run invalid option if any other number was pressed doesn't belong to the current menu level? Thanks. -- Rodrigo Montiel Ing. en Telecomunicaciones GNU/Linux VoIP System Support Cel: +549 351 2581376 Freetech Solutions www.freetechsolutions.com.ar Tel/Fax: +54 351 6387585 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users inline: logo.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invalid options
On Wed, 2 Oct 2013, Asmaa Ahmed wrote:
My problem is breaking the hierarchy level. Let's say I am in the main
menu and I have three options ( 1 for new messages, 2 change folders, 0
mailbox options ) all of them are fine I can go through them by pressing
the mentioned numbers, but if I pressed another number like 5, 6,...
will hear something out of the three options I have, something in
different level. if I pressed 7 for example, will hear something like
(message deleted) which should be OK only if I am inside option (1 new
messages) and asked to delete one of it, so I will press 7... but I was
expecting to hear something like invalid option or do nothing at all
if I am not inside (new messages) menu!
A better subject would also help. 'Invalid options' leaves the reader with
no clue what you need. Better bait, better fish.
Without the dialplan* and the console output**, I can't give any
suggestions.
I would suggest annotating each extension (using priority 1) in each
context with something like:
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}])
this will make it easier to follow along and figure out what's wrong.
*) From the Asterisk CLI, use 'dialplan show context-name' and paste
that into your reply for each of the relevant contexts. I'm not interested
in what you think your dialplan looks like, only what Asterisk thinks it
looks like :)
**) Paste, don't re-type.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] Invalid options
On Wed, 2 Oct 2013, Steve Edwards wrote: My bad. I re-read your initial post and see you're using the voicemail() application, not some dialplan you cobbled up. I guess a better subject might have enticed me to read the original :) I have nothing to offer. I think that's just the way voicemail() works. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Invalid options
I think what Asmaa is indicating here is that inside app_voicemail (s)he's able to select options that are not relevant to the current menu (s)he is playing. That wouldn't be a dialplan problem or a DTMF issue, It would be inside the app_voicemail.c source. It looks like there is a big switch statement starting at line 10180 in my app_voicemail.c source that deals with what option the user is pressing. It looks like at line 10458 there is logic for if delete is not a valid option at the moment, You could try inserting a playback there stating so. On Wed, Oct 2, 2013 at 2:24 AM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, While connecting to my voicemail, I noticed that Asterisk may perform some tasks isn't included in the options levels that currently played! For ex: I am listening to the main menu which asking me to press 1 for new messages, 2 change folders, 0 mailbox options which is OK for these numbers, but if I pressed 7 by mistake for example it will tell me Message deleted which it doesn't make sense! Is there a way to stop this behavior, let's say to run invalid option if any other number was pressed doesn't belong to the current menu level? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
