Re: [asterisk-users] Dial() M parameter in 1.6.2.11-rc2
Please note that I don't claim myself a guru, just happened to be working with Asterisk for some good number of years, so probably know some stuff better than others. As for the number of lines, 1800 lines will come down to 1000 lines using AEL but not the opposite. When I'll be back home, hopefully tomorrow, after a beautiful tour (my first) of New York city, I'll start writing some blogs on AEL. I guess an IVR example could be a good point to start, as it is enough complicated in itself. Zeeshan A Zakaria -- Sounds great. A confbridge example would be very welcome to me (just to contribute a personal wish) :-) Enjoy the rest of your trip. Oliver -- _ -- 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] how to place a call on hold and play music on hold using agi
Hi, I have the following problem. I have an xlite client registered with asterisk server. If i dial say 1500 an FAGI script is invoked which plays a greeting message. I now want to hold this call and play music on hold from FAGI. How do i achieve this? Please suggest me. Thanks in Advance, Jahnavi. -- _ -- 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] how to place a call on hold and play music on hold using agi
Hi Jahnavi, try StartMusicOnHold and StopMusicOnHold On Wed, Aug 4, 2010 at 12:45 AM, Janu Mukherjee janu.mu...@gmail.comwrote: Hi, I have the following problem. I have an xlite client registered with asterisk server. If i dial say 1500 an FAGI script is invoked which plays a greeting message. I now want to hold this call and play music on hold from FAGI. How do i achieve this? Please suggest me. Thanks in Advance, Jahnavi. -- _ -- 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
[asterisk-users] How to record a file and play some other file at the same time
Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory path i mentioned i could not see any files being created. But i again tried after some time this time i could see some recorded files but i couldnt hear any file being played at the same time. I need to achieve these both at the same time? Is there a way to do this??Please suggest me. Thanks Regards, Jahnavi. -- _ -- 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] How to record a file and play some other file at the same time
On Wed, Aug 4, 2010 at 12:12 PM, Janu Mukherjee janu.mu...@gmail.com wrote: Hi, Hi, please learn to ask questions. I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running *to threads one to record files and one to play files.* What records the files? Asterisk? Then, how are threads here involved? Do you create your application and use *to* threads for recording and playing? So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. *But in the directory path i mentioned* you didn't mention any directory. i could not see any files being created. But i again tried after some time this time i could see some recorded files but i couldnt hear any file being played at the same time. I need to achieve these both at the same time? Is there a way to do this??Please suggest me. Both situation and what are you trying to achieve are totally unclear. I reading this before asking again: http://catb.org/esr/faqs/smart-questions.html Any case, I think it's system buffer fault that files are not instantly seen. If I understood correctly from the question. Kind regards, Motiejus -- _ -- 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] callerid between 2 asterisk servers
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
been able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
register = serverA:serverapassw...@ip_of_serverb_nic/serverB
users.conf of serverA: users.conf of serverB:
[serverB] [serverA]
type=friend type=friend
fromuser=serverBfromuser=serverA
secret=serverBpassword secret=serverApassword
host=dynamichost=dynamic
etc.etc.
[serverA] [serverB]
type=user type=user
secret=serverApassword secret=serverBpassword
context=serverA_incomingcontext=serverB_incoming
host=dynamichost=dynamic
etc.etc.
serverA extensions.conf:
exten = _8X.,n,Dial(SIP/serverB/${EXTEN},20,r)
With this set up, when I dial from an extension such as 6000 on serverA to
an extension such as 8000 on serverB, instead of sending the callerid info
of 6000 it sends serverB. I cannot seem to find a way around this.
Anyone know of a way to send the 6000 callerid info? Somehow via sending a
user-defined field via the dial statement?
If not via the dial, then a way to transfer via writing to the file system?
Is there a way to use, in extensions.conf, some kind of info transferred
between serverA and serverB such as the tag id so that I can specify a
filename for them to write/read? I cannot seam to find something that each
server sees which I can dynamically read in and use in extensions.conf.
Thanks!!
--
_
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Re: [asterisk-users] callerid between 2 asterisk servers
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
een able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
egister = serverA:serverapassw...@ip_of_serverb_nic/serverB
users.conf of serverA: users.conf of serverB:
[serverB] [serverA]
ype=friend type=friend
romuser=serverBfromuser=serverA
ecret=serverBpassword secret=serverApassword
ost=dynamichost=dynamic
tc.etc.
[serverA] [serverB]
ype=user type=user
ecret=serverApassword secret=serverBpassword
ontext=serverA_incomingcontext=serverB_incoming
ost=dynamichost=dynamic
tc.etc.
serverA extensions.conf:
xten = _8X.,n,Dial(SIP/serverB/${EXTEN},20,r)
With this set up, when I dial from an extension such as 6000 on serverA to
n extension such as 8000 on serverB, instead of sending the callerid info
f 6000 it sends serverB. I cannot seem to find a way around this.
nyone know of a way to send the 6000 callerid info? Somehow via sending a
ser-defined field via the dial statement?
f not via the dial, then a way to transfer via writing to the file system?
s there a way to use, in extensions.conf, some kind of info transferred
etween serverA and serverB such as the tag id so that I can specify a
ilename for them to write/read? I cannot seam to find something that each
erver sees which I can dynamically read in and use in extensions.conf.
Thanks!!
--
Try uncommenting fromuser on both boxes.
Or did you set callerid in your users.conf when you write etc.? If so, also
uncomment it.
Oliver
--
_
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Re: [asterisk-users] mapping of disconnect reasons
Tilghman, thank you for your reply. The mapping in RFC 3398 is logically correct therefore I do not need to submit a suggestion to its editor. The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required is mapped to 16 Normal termination instead of 21 Call Rejected. Could you direct me to the relevant file of code where these mappings are done? Before reporting a bug I would like to confirm whether this issue has been addressed on newer releases. Thanks, Harel -- On Tuesday 03 August 2010 06:21:23 Philipp von Klitzing wrote: Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 Payment Required from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. * if you think the mapping is wrong, then you should open a ticket on the Asterisk bug tracker Actually, much of the mapping is specified by RFC 3398 section 8.2.6.1. Thus, if you think the mapping is wrong, you should submit a suggestion for amendment to the RFC editor. Only for response codes specified differently than in this section should you open an issue in the tracker. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Asterisk not working with Festival
Hello, I am having a Mac 10.6.4 (Snow Leopard). I have compiled and built Asterisk 1.6.2.9 and Festival 2.0.95:beta on my machine. Asterisk is working fine with SIP channels without Festival. I have written following context in extension.conf: [connect-to-me] exten = s,1,Answer Exten = s,n,SayDigits(1¹) exten = s,n,Festival(hello john) exten = s,n,Hangup I use call files to make calls to my mobile and once call is answered then asterisk attaches it to ³connect-to-me² context. But after that, I can hear only a voice saying ³one² but nothing after that. Please find below details on configuration files: festival.conf: ; Festival Configuration [general] host=localhost port=1314 usecache=yes cachedir=/var/lib/asterisk/festivalcache/ festivalcommand=(tts_textasterisk %s 'file)(quit)\n And, festival.scm : (define (tts_textasterisk string mode) (tts_textasterisk STRING MODE) Apply tts to STRING. This function is specifically designed for use in server mode so a single function call may synthesize the string. This function name may be added to the server safe functions. (let ((wholeutt (utt.synth (eval (list 'Utterance 'Text string) (utt.wave.resample wholeutt 8000) (utt.wave.rescale wholeutt 5) (utt.send.wave.client wholeutt))) I have placed the above text before the last line which is (provide 'festival). Below is the debug log shown on asterisk console : [Aug 4 17:50:11] Channel SIP/gafachi1a- was answered. [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Answer' [Aug 4 17:50:11] -- Executing [...@connect-to-me:1] Answer(SIP/gafachi1a-, ) in new stack [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'SayDigits' [Aug 4 17:50:11] -- Executing [...@connect-to-me:2] SayDigits(SIP/gafachi1a-, '1') in new stack [Aug 4 17:50:11] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Aug 4 17:50:11] -- SIP/gafachi1a- Playing 'digits/1.slin' (language 'en') [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (571 requested / 100 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:2488 ast_settimeout: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format ulaw [Aug 4 17:50:12] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'Festival' [Aug 4 17:50:12] -- Executing [...@connect-to-me:3] Festival(SIP/gafachi1a-, hello john) in new stack [Aug 4 17:50:12] == Parsing '/usr/local/etc/asterisk/festival.conf': [Aug 4 17:50:12] DEBUG[17094]: config.c:1330 config_text_file_load: Parsing /usr/local/etc/asterisk/festival.conf [Aug 4 17:50:12] == Found [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:376 festival_exec: Text passed to festival server : hello john [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:446 festival_exec: Cache file exists, strln=10, strlen=10 [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:448 festival_exec: Size OK [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:467 festival_exec: Reading from cache... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:491 festival_exec: Passing data to channel... [Aug 4 17:50:12] DEBUG[17094]: app_festival.c:513 festival_exec: Festival WV command [Aug 4 17:50:12] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a- to write format slin [Aug 4 17:50:34] DEBUG[17094]: chan_sip.c:3562 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 67.216.35.162:5060 And, festival server console looks like following: $ ./bin/festival --server serverWed Aug 4 17:49:04 2010 : Festival server started on port 1314 client(1) Wed Aug 4 17:50:12 2010 : accepted from localhost client(1) Wed Aug 4 17:50:12 2010 : disconnected I have to end the call after sometime. Festival works fine if I got into its console and type SayText(³hello john²) Please let me know how I can fix this. Thanks, Davinder -- _ -- 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
Re: [asterisk-users] Asterisk Crashes - Segmentation Fault
Hi Dan, I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: sens in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: current_page in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: dateReq in /var/www/html/web-meetme/meetme_control.php on line 573, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include() [a href='function.include'function.include/a]: Failed opening 'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/email_body.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include() [a href='function.include'function.include/a]: Failed opening 'locale.php' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/www/html/web-meetme/lib/email_body.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? Otherwise i am able to record and play the recorded file from the speaker button. --Manmohan Singh On Fri, Jul 30, 2010 at 9:10 PM, Dan Austin dan_aus...@phoenix.com wrote: Manmohan wrote: There was on very silly mistake and i missed to check that properly. Really apologize for that. Following change was done to get the conf-recording into the proper path: chown -R asterisk:asterisk /var/lib/asterisk/sounds/conf-recordings following is the output: [r...@linuxtest sounds]# ll total 6416 drwxrwxr-x 2 asterisk asterisk4096 Jul 30 08:29 conf-recordings [r...@linuxtest sounds]# ll conf-recordings/ total 4060 -rw-r--r-- 1 asterisk asterisk 4150124 Jul 30 08:27 meetme-conf-rec-74438-1280463795.8.wav The only thing now is no speaker icon onto the webpage when i click to past conference link. The web interface cannot find the recording. The reason it cannot is that the name is wrong. By wrong, I mean it contains information that the database and program is not aware of (1280463795.8). To make this clear, if this conference was the 3rd one you ever scheduled on this system the correct file name would be- meetme-conf-rec-74438-3.wav using the format meetme-conf-rec-%PIN%-%BOOKID%.wav The database knows the pin and bookid, so it can construct the file name and test if it exists. Do you say that if i shift from 4.0.1 to 4.0.2 will fix the issue (of getting speaker icon in past conference)? I was not able to get the change into app_meetme to use the bookid in the filename, even though it has access to bookid. I gave up and now store the filename in the database, which app_meetme will use if it exists. Other that a handful of bug-fixes, this is the major difference between 4.0.1 and 4.0.2 Dan -- _ -- 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 -- Thanks Regards Manmohan Singh Jandu -- _ -- 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] callerid between 2 asterisk servers
Thanks Oliver.
I tried those approaches but they did not work.
However, I just found a workaround finally. The SIPAddHeader and SIP_HEADER
functions enabled me to get the callerid working.
Thanks again!!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
unsero...@aol.com
Sent: Wednesday, August 04, 2010 8:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] callerid between 2 asterisk servers
I've got 2 asterisk servers on the same box: ubuntu 10.04 lucid. I have not
been able to send useful callerid info between them (callerid becomes
serverB).
serverA register statement: (serverB has the exact opposite statement)
register = serverA:serverapassw...@ip_of_serverb_nic/serverB
users.conf of serverA: users.conf of serverB:
[serverB] [serverA]
type=friend type=friend
fromuser=serverBfromuser=serverA
secret=serverBpassword secret=serverApassword
host=dynamichost=dynamic
etc.etc.
[serverA] [serverB]
type=user type=user
secret=serverApassword secret=serverBpassword
context=serverA_incomingcontext=serverB_incoming
host=dynamichost=dynamic
etc.etc.
serverA extensions.conf:
exten = _8X.,n,Dial(SIP/serverB/${EXTEN},20,r)
With this set up, when I dial from an extension such as 6000 on serverA to
an extension such as 8000 on serverB, instead of sending the callerid info
of 6000 it sends serverB. I cannot seem to find a way around this.
Anyone know of a way to send the 6000 callerid info? Somehow via sending a
user-defined field via the dial statement?
If not via the dial, then a way to transfer via writing to the file system?
Is there a way to use, in extensions.conf, some kind of info transferred
between serverA and serverB such as the tag id so that I can specify a
filename for them to write/read? I cannot seam to find something that each
server sees which I can dynamically read in and use in extensions.conf.
Thanks!!
--
Try uncommenting fromuser on both boxes.
Or did you set callerid in your users.conf when you write etc.? If so,
also uncomment it.
Oliver
--
_
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[asterisk-users] can't write to queues_additional.conf
Hello, In my Asterisk server when i try to set the value for the queue option Skip Busy Agents in Freepbx GUI it is not being written into the backend file queues_additional.conf. As a result sometimes agents in queue gets calls when they are already busy with another call. So i set ringinuse=no option manually from backend. Is it bug ? Is there any fix for this?. I am providing the details of version of asterisk and freepbx. Asterisk : Asterisk 1.4.33.1 FreePBX version : 2.7.0.5 queue Module version : 2.5.4.8 -- _ -- 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] outboundproxy timeout or qualify
Hi! Let's say I call by SIP/trunk1/number and the proxy server is down, is there a way to getCHANUNAVAIL? *CLI core show application Dial Unfortunatelythe timeout parameter will not do the job for me. I need somethingequivalentto qualify to monitor the outboundproxy. Why not qualify and ChanIsAvail() or SIPPEER(peername|status) if you really do not want to use the DIALSTATUS variable after your first Dial()? Philipp -- _ -- 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] mapping of disconnect reasons
The mapping in Asterisk 1.4.24 is the problem: 402 Payment Required is mapped to 16 Normal termination instead of 21 Call Rejected. Could you direct me to the relevant file of code where these mappings are done? Before reporting a bug I would like to confirm whether this issue has been addressed on newer releases. Look in channels/chan_sip.c and search for 3398 See also: http://www.voip- info.org/wiki/index.php?page=Asterisk+variable+hangupcause Philipp -- _ -- 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] Queue to queue transfer error
Hi all, I have problem when i transfer call from one queue extension to other queue extension. *Scenario *some one call to DID 8833383932 which is assigned to queue1 and pickedup by extension1 of queue1, Now extension1 transfer call to queue2's exntesion2, extension2 picked up the call but no voice and caller only hear queue2 greetings but after picking up by extension2 then no voice. what is the issue.. please help if someone know the answer. Thanks you -- Toqeer Ali Syed Red Hat Certified Engineer mob: +92 321 9059916 -- _ -- 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] Tweaking AMD in Asterisk
Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. 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] Asterisk (1.8-beta2) and SIP IPv4/IPv6 dual-stack possibilities
Dear list, I'm trying to get Asterisk to work dual-stack on Linux and I'm left with a question. Imagine that a user (on the road) connects to Asterisk from various places. Many of them probably don't have IPv6 support yet. However, his house and office do have IPv6 connectivity. I would like to make sure that whenever IPv6 is available, the connection will be made over IPv6, but offer IPv4 as a fallback option. The pitfall, in my opinion, is to create one sip.conf entry for that user which supports the voicecalls over IPv4 and IPv6. However, settings like nat=, directmedia= and/or canreinvite= seem to be addressfamily unrelated. I want to configure it in a way that when I connect using IPv6, no NAT options should be set and the mediapath (almost) always should be directly between the peers and not over the Asterisk server (so, nat=no and canreinvite=yes). But, when a user comes via IPv4, changes are that he's on NAT. When that happens obviously the connections should traverse the NAT using options like nat=yes and canreinvite=no. There's little to no documentation available as far as my google-skills go. There's some in sip.conf, and I couldn't find anything on the website. Does anyone have some pointers for me, either for the configuration of the sip.conf entry or for more documentation on this? Best regards, Wouter Schoot -- _ -- 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] Tweaking AMD in Asterisk
Hi, the basic settings are pretty good ones. What I did to do improve the performance and prevent the false positives, I started to recorded every call, and analyzed every incorrect detection :) Fairly soon I came with optimal set for my environment: initial_silence= 2500 greeting = 1500 after_greeting_silence = 300 total_analysis_time= 5000 min_word_length= 120 between_words_silence = 50 maximum_number_of_words= 4 ; it's usuall to pickup saying Jon Anderssen, hello in here silence_threshold = 384 by the way, for outgoing SIP calls you might want to do this Background trick as it helped me a lot regarding AMD on SIP. exten = _X.,n,Background(blank_audio) exten = _X.,n,AMD On Wed, Aug 4, 2010 at 5:08 PM, Tino t...@sparksupport.com wrote: Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] How to record a file and play some other file atthe same time
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Janu Mukherjee Subject: [asterisk-users] How to record a file and play some other file atthe same time Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory path i mentioned i could not see any files being created. But i again tried after some time this time i could see some recorded files but i couldnt hear any file being played at the same time. I need to achieve these both at the same time? Is there a way to do this??Please suggest me. Thanks Regards, Jahnavi. Since this forum is posted to and read from world-wide, it is arrogant and foolish to think we all speak or type the Queen's English (In Alabama, we think of Freddie Prince in that regard). Rephrasing OP's question: I have an xlite softphone registered to an Asterisk Server. We have a set up where I dial into an AGI that is supposed to play sound on one thread and record on another and combine the two threads into a new recording. I dialed in and began recording (the xlite indicates that I am recording my voice and I can hear the sound from the other thread). But in my expected directory path, I find no files being created. Trying again later I could see some recorded files but couldn't hear any file being played at the same time. I need to be able to get the combined recording file. Is there a way to do this? My suggestion would be to set up a meetme room with recording and pipe the playback thread in as a caller and/or use monitor. There are lots of posts regarding this within the last 6 months. -- _ -- 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] Tweaking AMD in Asterisk
Hi Aurimas, Thanks for your thoughts on this. Can you please let me know how playing a silent audio file before AMD will help to tweak the parameter values. On Wed, Aug 4, 2010 at 8:30 PM, Aurimas Skirgaila a.skirga...@gmail.comwrote: Hi, the basic settings are pretty good ones. What I did to do improve the performance and prevent the false positives, I started to recorded every call, and analyzed every incorrect detection :) Fairly soon I came with optimal set for my environment: initial_silence= 2500 greeting = 1500 after_greeting_silence = 300 total_analysis_time= 5000 min_word_length= 120 between_words_silence = 50 maximum_number_of_words= 4 ; it's usuall to pickup saying Jon Anderssen, hello in here silence_threshold = 384 by the way, for outgoing SIP calls you might want to do this Background trick as it helped me a lot regarding AMD on SIP. exten = _X.,n,Background(blank_audio) exten = _X.,n,AMD On Wed, Aug 4, 2010 at 5:08 PM, Tino t...@sparksupport.com wrote: Hello , I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My current values are AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx 25-30 % of all answering machines. Anybody have any suggestion to improve the accuracy of AMD. 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] Tweaking AMD in Asterisk
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino Subject: Re: [asterisk-users] Tweaking AMD in Asterisk Hi Aurimas, Thanks for your thoughts on this. Can you please let me know how playing a silent audio file before AMD will help to tweak the parameter values. Just a WAG - playing the file gives AMD a few more seconds to properly do it's thing. -- _ -- 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] Asterisk and RAID
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro -- _ -- 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] Asterisk and RAID
Alejandro Cabrera Obed wrote: Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro Either RAID1 with a couple of spare drives or RAID5 across 3 discs with a hot spare. I assume disc capacity is not an issue. If the system supports RAID 6 that would be ideal as you will have two drives and two parity sets so could cope with 2 simultaneous drive failures compared to 1 for raid1 and raid5. -- _ -- 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] Asterisk and RAID
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alejandro Cabrera Obed Subject: [asterisk-users] Asterisk and RAID Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Regards Alejandro Not really an Asterisk question, but in R1 you would only use 2 of your 4 drives; R5 would use 3 out of 4. I use R5 on my Dell boxes. -- _ -- 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] Asterisk and RAID
On Wed, Aug 4, 2010 at 11:57 AM, Alejandro Cabrera Obed aco1...@gmail.com wrote: Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? Not really an asterisk question. Asterisk will run well regardless of what you choose. -- _ -- 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] Asterisk and RAID
On Wed, 4 Aug 2010, Alejandro Cabrera Obed wrote: Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with four HD's available, using CentOS as the OS. What's the best RAID type recommendation ??? RAID 1 or RAID 5 ??? RAID-10 If your controller supports it. If not, do it with Linux software RAID. RAID-1 will give you 1 x the drive size with data being written to all 4 disks at the same time, but being read from one - very redundant, but slow writes. RAID-5 will give you 3x your single disk capacity with one disk acting as a parity drive - reasonable performance, but one day you'll lose a drive and then find that a 2nd drive has sector errors when reconstructing the array and it's then game over - unless it's Linux software RAID and you're a guru - which you're not as you'd not be posting this question here. RAID-6 will give you 2x your drive capacity with the ability to survive 2 drive failling - hopefully you can replace one and not have bad sectors on another. RAID-10 will also give you 2x your drive capacity but has more performance than RAID-6. Not really an asterisk question though... Gordon -- _ -- 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] Femtocell to VoIP?
Steve, Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming up very dry. OpenBTS sounds like it would work, but is way too expensive to roll out to residential homes. On Mon, Aug 2, 2010 at 6:53 PM, Steve Kennedy steve-aster...@gbnet.netwrote: On Mon, Aug 02, 2010 at 03:36:59PM -0400, Matt wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? Most people seem to be concentrating on 3G femtocells (there are various companies making designs based on picoChip soft radios). OpenBTS can be used (and there have been some successful quite large installations). Hay Systems were meant to be producing a 2G (GSM/GPRS) femtocell, but they seem to have gone quiet. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] Asterisk Crashes - Segmentation Fault
Manmohan wrote: I had tried the new version of webmeetme i.e., 4.0.2 The recording works very well. Great! I see following php errors whenever i try to add in conference. [Wed Aug 04 16:29:02 2010] [error] [client 32.93.17.40] PHP Notice: Undefined variable: order in /var/www/html/web-meetme/meetme_control.php on line 278, referer: http://10.1.1.30/web-meetme/meetme_control.php?s=4 You can ignore the Notices. They are fairly harmless, and only mean that variable is not set by the code or being passed in on the URL. You can turn off notices in /etc/php.ini if they bother you. Also the Reports link doesnt display anything and in httpd error logs it gives me following php errors: [Wed Aug 04 16:30:22 2010] [error] [client 32.93.17.40] PHP Warning: include(locale.php) [a href='function.include'function.include/a]: failed to open stream: No such file or directory in /var/www/html/web-meetme/lib/defines.php on line 3, referer: http://10.1.1.30/web-meetme/daily.php? In lib/defines.php, either comment out the 3rd line or add ../ before locale.php- include(../locale.php); But that is not likely why you do not get the reports. The most likely cause is A PHP notice is being thrown while the GD code is rendering the graph, resulting in a corrupt image which your browser cannot display. Check these settings /etc/php.ini- error_reporting = E_ALL display_errors = Off Dan -- _ -- 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] Tweaking AMD in Asterisk
Thanks Danny, What should be the length of audio file ? On Wed, Aug 4, 2010 at 9:21 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino *Subject:* Re: [asterisk-users] Tweaking AMD in Asterisk Hi Aurimas, Thanks for your thoughts on this. Can you please let me know how playing a silent audio file before AMD will help to tweak the parameter values. Just a WAG – playing the file gives AMD a few more seconds to properly do it’s thing. -- _ -- 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] Tweaking AMD in Asterisk
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tino Subject: Re: [asterisk-users] Tweaking AMD in Asterisk Thanks Danny, What should be the length of audio file ? I'm supposing that 3 to 5 seconds should be ok. -- _ -- 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] Tweaking AMD in Asterisk
in my case it's 0.1 second and I can confirm, that on SIP channels it really helps. On Wed, Aug 4, 2010 at 8:51 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino *Subject:* Re: [asterisk-users] Tweaking AMD in Asterisk Thanks Danny, What should be the length of audio file ? I’m supposing that 3 to 5 seconds should be ok. -- _ -- 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 -- Mvh, Aurimas Skirgaila -- _ -- 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] Femtocell to VoIP?
On Wed, Aug 04, 2010 at 01:13:56PM -0400, Matt wrote: Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming up very dry. OpenBTS sounds like it would work, but is way too expensive to roll out to residential homes. Pretty much all Femtocells use 3G locally and send stuff back over VoIP (in some form or other). In the UK Vodafone sell a 3G femtocell (which has an internal 2G radio too, to ensure it's being used in the UK). ATT sell their own. Try contacting PicoChip or Ubiquisys who both have femtocells. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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] Identify remote prompts: Partial audio matching?
Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and This user is temporarily unavailable. Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio stream)? Recently there was an at least slightly related posting on this list, if I recall that correctly, but I have simply not been able to turn this up. Philipp P.S.: This is all about audio analysis, not about cause codes. -- _ -- 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] Identify remote prompts: Partial audio matching?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Subject: [asterisk-users] Identify remote prompts: Partial audio matching? Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and This user is temporarily unavailable. Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio stream)? Recently there was an at least slightly related posting on this list, if I recall that correctly, but I have simply not been able to turn this up. Philipp P.S.: This is all about audio analysis, not about cause codes. You might be able to record these snippets then pass them through the Vestec or Lumenvox Speech engine to get what you want. -- _ -- 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] Identify remote prompts: Partial audio matching?
You might be able to record these snippets then pass them through the Vestec or Lumenvox Speech engine to get what you want. Unfortunately that won't work because: * the containing recordings/feeds can be quite long, can be embedded/surrounded by silence, ringing tones, music or special tones, and the ASR engines are not really designed to handle this situation. * next to this both LumenVox and Vestec do not cover the language(s) that I need this for, since both companies are focused on the American market (and yes, I am aware of Loquendo and Nuance). So that is why I am looking for something like partial audio fingerprinting; this is a bit like these find duplicate mp3 songs in my huge media library tools, only that in this case it is 1. not about an exact duplicate, and 2. the audio quality can vary, and 3. this is about finding contained parts instead of comparing full songs with each other. Philipp -- _ -- 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] Identify remote prompts: Partial audiomatching?
Ot Nuance for linux? -Ursprungligt meddelande- Från: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] För Philipp von Klitzing Skickat: den 4 augusti 2010 22:29 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] Identify remote prompts: Partial audiomatching? You might be able to record these snippets then pass them through the Vestec or Lumenvox Speech engine to get what you want. Unfortunately that won't work because: * the containing recordings/feeds can be quite long, can be embedded/surrounded by silence, ringing tones, music or special tones, and the ASR engines are not really designed to handle this situation. * next to this both LumenVox and Vestec do not cover the language(s) that I need this for, since both companies are focused on the American market (and yes, I am aware of Loquendo and Nuance). So that is why I am looking for something like partial audio fingerprinting; this is a bit like these find duplicate mp3 songs in my huge media library tools, only that in this case it is 1. not about an exact duplicate, and 2. the audio quality can vary, and 3. this is about finding contained parts instead of comparing full songs with each other. Philipp -- _ -- 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] Femtocell to VoIP?
I have done an OpenBTS research and try project and OpenBTS is working great. A complete set to roll out OpenBTS is not cheap but as far as I know all femtocell kind of solutions need serious investments and OpenBTS seems to be the cheapest among them. Asterisk is actually one of the lego pieces the OpenBTS solution is made of. On 2 aug 2010, at 21:36, Matt wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -- _ -- 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] Asterisk and RAID
Wednesday, August 4, 2010, 6:02:45 PM, Danny wrote: R5 would use 3 out of 4. You can have R5 across 10 drives too. Yes, the writes will be slow, but it possible. -- Best regards, Gergomailto:csi...@gmail.com -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =1. Wait(1)[pbx_config] 2. Answer() [pbx_config] 3. Background(welcome)[pbx_config] 4. Background(and)[pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations)[pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: Extension is unavailable. Please leave your message after the tone. sip.conf: [general] register = NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(welcome) exten = s,n,Background(and) exten = s,n,Background(thank-you-for-calling) exten = s,n,Background(conference-reservations) exten = s,n,Waitfor() exten = s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood sch...@gmail.com wrote: Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' =1. Wait(1) [pbx_config] 2. Answer() [pbx_config] 3. Background(welcome) [pbx_config] 4. Background(and) [pbx_config] 5. Background(thank-you-for-calling) [pbx_config] 6. Background(conference-reservations) [pbx_config] 7. Waitfor() [pbx_config] 8. Hangup() [pbx_config] Unfortunately, no matter how I configure extensions.conf or sip.conf, the phone call always ends up saying: Extension is unavailable. Please leave your message after the tone. sip.conf: [general] register = NPANXX:passw...@service_provider_ip registertimeout=29 registerattempts=0 defaultexpiry=60 [DID_NUMBER] type=peer context=default host=SERVICE_PROVIDER_IP authuser=DID_NUMBER fromuser=DID_NUMBER fromdomain=SERVICE_PROVIDER_REALM remotesecret=SERVICE_PROVIDER_PASSWD secret=SERVICE_PROVIDER_PASSWD dtmfmode=rfc2833 disallow=all allow=ulaw qualify=yes I am attempting just to get the starting point where I can direct users through my asterisk box, but it won't direct users to the 's' extention, only to some voicemail box. I've removed the voicemail config. My extensions.conf is tiny: [globals] [general] [default] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(welcome) exten = s,n,Background(and) exten = s,n,Background(thank-you-for-calling) exten = s,n,Background(conference-reservations) exten = s,n,Waitfor() exten = s,n,Hangup() What am I doing wrong here? Thanks for any help you can give. Joe You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'default' (0xb77980c0) in local table 0xb77960c0; registrar: pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 1 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 2 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 3 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 4 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 5 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 6 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 7 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '_X.' priority 8 to default (0xb77980c0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Registered extension context 'parkedcalls' (0xb7797ee0) in local table 0xb77960c0; registrar: features [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Added extension '700' priority 1 to parkedcalls (0xb7797ee0) [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to scan old dialplan and merge leftovers back into the new: 0.89 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to restore hints and swap in new dialplan: 0.02 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Time to delete the old dialplan: 0.11 sec [Aug 4 19:16:42] VERBOSE[12287] pbx.c: -- Total time merge_contexts_delete: 0.000102 sec [Aug 4 19:17:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:19:04] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 [Aug 4 19:21:39] VERBOSE[12255] netsock.c: == Using SIP RTP CoS mark 5 I get the same error. Same random voicemail when no voicemail is configured. I was under the impressing that s was the catchall for all incoming trunks. What has changed? Joe -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the start extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] CDR: MySQL query
Thx Rudi. but this query results in *Empty set (0.32 sec) src AND dst like number *seems to be the problem area. * * Also, how can I get the hold time talk time as separate values OR may be total call connect time talk time (the difference of the 2 will be hold time). Thx Sans On Wed, Aug 4, 2010 at 6:18 PM, Rudi Oosthuizen rudi.oosthui...@nha.co.zawrote: Mysql Use asteriskcdrdb; Select calldate,src,dst,disposition,duration,billsec,uniqueid from cdr where src like 'NUMBER' and dst like 'NUMBER' order by calldate; Rudi Oosthuzen Hi, Can someone help me formulate MySQL Query(s) which will help me extract the following details for a given DID (date range can be excluded for simplicity). Date-Time DNID (I am recording this is `userfield`) CLID time-1 (when call was received) time-2 (when call was answered by agent) time-3 (when call was hung-up) My Call flow is as follows: - Caller dials a DNID - Call enters queue - Call rings in round-robin format to all logged in agents - Agent answers call - Both parties hand-up Any help with MySQL queries or pointers are deeply appreciated. Thx Sans -- _ -- 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] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: I don't see any On Wed, Aug 4, 2010 at 7:04 PM, Warren Selby wcse...@selbytech.com wrote: You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. I don't see any call attempt/logs when I bump up the verbosity, and when I check my verbose logs I show: The next step would be to enable sip debug on the peer you're trying to receive calls from (sip set debug peer PEERNAME or sip set debug ip IPADDRESS). Then send another call inbound and see what's happening. As far as the 's' extension, that's the start extension, it's used when no other extension information is presented. Pretty much every SIP peer I've ever seen presents an extension when entering a context, and thus the 's' extension doesn't catch it. I've typically only seen 's' used in Macros and with inbound analog lines. My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: --- SIP read from UDP:209.221.186.98:5060 --- INVITE sip:s...@209.221.186.50 SIP/2.0 Record-Route: sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 Max-Forwards: 16 From: 2538544199 sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4 To: sip:2063161...@209.221.186.98 Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 CSeq: 200 INVITE Contact: Anonymous sip:2538544...@209.221.186.98:5071 Expires: 300 User-Agent: Sippy Softswitch v2.0.80 cisco-GUID: 1225641884-3786690633-966044271-4144140181 h323-conf-id: 1225641884-3786690633-966044271-4144140181 Content-Length: 321 Content-Type: application/sdp v=0 o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 s=- c=IN IP4 209.221.186.98 t=0 0 m=audio 60304 RTP/AVP 0 a=fmtp:4 bitrate=6300;annexa=no a=rtpmap:96 iLBC/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=fmtp:18 annexb=no a=rtpmap:98 telephone-event/8000 a=oldmediaip:208.76.155.20 a=nortpproxy:yes - [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 0 [ 35]: INVITE sip:s...@209.221.186.50 SIP/2.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 1 [ 75]: Record-Route: sip:209.221.186.98;ftag=f7093e2d7e16a927d0816f6f5ed7aba4;lr [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 2 [ 85]: Via: SIP/2.0/UDP 209.221.186.98;branch=z9hG4bK6b95.ae84c9620e19761029084a0f85255c29.0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 3 [ 94]: Via: SIP/2.0/UDP 209.221.186.98:5071;branch=z9hG4bKf0a046579186bf058ba7783ed98a5a66;rport=5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 4 [ 16]: Max-Forwards: 16 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 5 [ 85]: From: 2538544199 sip:2538544...@209.221.186.98;tag=f7093e2d7e16a927d0816f6f5ed7aba4 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 6 [ 35]: To: sip:2063161...@209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 7 [ 51]: Call-ID: ba711218-1ae3-122e-41bd-18a905721e58-b2b_1 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 8 [ 16]: CSeq: 200 INVITE [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 9 [ 55]: Contact: Anonymous sip:2538544...@209.221.186.98:5071 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 10 [ 12]: Expires: 300 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 11 [ 36]: User-Agent: Sippy Softswitch v2.0.80 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 12 [ 54]: cisco-GUID: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 13 [ 56]: h323-conf-id: 1225641884-3786690633-966044271-4144140181 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 14 [ 19]: Content-Length: 321 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 15 [ 29]: Content-Type: application/sdp [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request: Header 16 [ 0]: [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 0 [ 3]: v=0 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:Body 1 [ 53]: o=- 1280279699622 1280279699622 IN IP4 209.221.186.98 [Aug 4 20:21:40] DEBUG[12487]: chan_sip.c:7857 parse_request:
Re: [asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 10:25 PM, Joe Wood sch...@gmail.com wrote: On Wed, Aug 4, 2010 at 7:49 PM, Warren Selby wcse...@selbytech.com wrote: On Wed, Aug 4, 2010 at 9:25 PM, Joe Wood sch...@gmail.com wrote: My experience with Asterisk in the past has been with inbound analog lines so that would make sense :) See if you spot anything weird here: Try adding insecure=invite to the DID_NUMBER peer, reload SIP and try your call again. By the way, it looks like your SIP provider has a built-in auto-failover to voicemail setup. You may want to get them to disable that once you get everything working on your end. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- 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] No Mailbox Subscription in SIP Users Suddenly
Suddenly the other day we noticed MWI stopped working for SIP clients. A sip show peer X returns this: ast01*CLI sip show peer 719XXX * Name : 719XXX Realtime peer: Yes, cached Secret : Set MD5Secret: Not set Remote Secret: Not set Context : peakinternet-outbound Subscr.Cont. : Not set Language : en AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk As you can see, there is nothing for Mailbox We are using MySQL Realtime Addon. In the sip_buddies table, the mailbox column is filled with 719...@default -- I thought maybe it was something weird with the SQL table so I backed it up, deleted it, recreated it, added a single entry and registered that SIP client. Same thing. Everything else in the table (caller ID, pickupgroup, etc.) will show correctly on sip show peer but just not Mailbox. So I figured maybe it was a bug in 1.6.2.10 so I upgraded to 1.6.2.11-rc2, but the same thing is happening. Is this something silly that I'm just overlooking? I hope someone can help. Thanks in advance for your help!! Jayson -- _ -- 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
