Re: [asterisk-users] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup Try this below dilaplan exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi) same = n,Hangup console output: -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new stack -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024, CHANNEL(language)=en_AU) in new stack -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh -- Playing 'welcome' (escape_digits=1) (sample_offset 0) -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new stack == Spawn extension (AllPhones, 5150, 4) exited non-zero on 'SIP/PBX-0024' But nothing happens...as in even when it says that it's playing the file (as verified in the asterisk 'full' log), I hear nothing on the phone What gives? spent 2 hrs Googling but nothing! :( Thx \A -- _ -- 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 -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.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] AGI STREAM FILE not working?
On Mon, 6 Jun 2011, A E [Gmail] wrote: Hello,using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' What gives? spent 2 hrs Googling but nothing! :( Maybe 1.5 hrs should have been spent reading :) One line does not an AGI make. AGI is an interface. It consists of reading the AGI environment from STDIN and then, writing requests on STDOUT and reading the response from STDIN. If you don't follow these 3 steps in order (steps 2 and 3 can be repeated) then your program has violated the protocol and will not function reliably if at all. Please use an existing AGI library for the language of your choice. Nobody gets it right the first time. -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 12:12 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup console output: -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new stack -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024, CHANNEL(language)=en_AU) in new stack -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh -- Playing 'welcome' (escape_digits=1) (sample_offset 0) -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new stack == Spawn extension (AllPhones, 5150, 4) exited non-zero on 'SIP/PBX-0024' But nothing happens...as in even when it says that it's playing the file (as verified in the asterisk 'full' log), I hear nothing on the phone What gives? spent 2 hrs Googling but nothing! :( Thx \A Bizarre, I found a bunch of other agi scripts in the default directory...modified the agi-test.agi (perl script) so it played my file, no joy! then I used a php script I found somewhere else asa tutorial to writing AGI scripts in php, modified that to play my script and it works. I don't get it. esp. when everything (with agi debug set on) looks exactly the same with my bash script and this php script except that with the php script, I see this ONE line that's extra SIP/PBX-002bAGI Rx STREAM FILE welcome # -- Playing 'welcome' (escape_digits=#) (sample_offset 0) SIP/PBX-002bAGI Tx 200 result=35 endpos=87200 that I don't see with my bash script which does this SIP/PBX-002eAGI Rx STREAM FILE welcome # -- Playing 'welcome' (escape_digits=#) (sample_offset 0) -- SIP/PBX-002eAGI Script streamcontact.sh completed, returning 0 -- Executing [5150@AllPhones:5] Hangup(SIP/PBX-002e, ) in new stack So confused!! -- _ -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 2:06 AM, mahesh katta maheshka...@flexydial.comwrote: On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup Try this below dilaplan exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi) same = n,Hangup No deal. Doesn't find the AGI script -- _ -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, 6 Jun 2011, A E [Gmail] wrote: Hello,using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' What gives? spent 2 hrs Googling but nothing! :( Maybe 1.5 hrs should have been spent reading :) touche ;) One line does not an AGI make. Did you just pull a 'Yoda' Steve? AGI is an interface. It consists of reading the AGI environment from STDIN and then, writing requests on STDOUT and reading the response from STDIN. Right! I did read that, the problem is how do I do this in bash?? I tried read the result in and just post a Noop kind of a thing just to tell that I read something, but it didn't help. I also explicitly did that in the perl script, but doesn't work. It only works in PHP. If you don't follow these 3 steps in order (steps 2 and 3 can be repeated) then your program has violated the protocol and will not function reliably if at all. Please use an existing AGI library for the language of your choice. Nobody gets it right the first time. -- _ -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.com wrote: AGI is an interface. It consists of reading the AGI environment from STDIN and then, writing requests on STDOUT and reading the response from STDIN. On Mon, 6 Jun 2011, A E [Gmail] wrote: Right! I did read that, the problem is how do I do this in bash?? I tried read the result in and just post a Noop kind of a thing just to tell that I read something, but it didn't help. I also explicitly did that in the perl script, but doesn't work. It only works in PHP. Bash would probably be my last choice of language to write an AGI with. Personally, I use C because it is my sharpest tool and because you can execute hundreds of AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. I suspect that the problems you are experiencing with Perl may have something to do with flushing STDOUT or reading the complete response from STDIN. I strongly suggest using an existing library for the language of your choice. -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards asterisk@sedwards.com wrote: AGI is an interface. It consists of reading the AGI environment from STDIN and then, writing requests on STDOUT and reading the response from STDIN. On Mon, 6 Jun 2011, A E [Gmail] wrote: Right! I did read that, the problem is how do I do this in bash?? I tried read the result in and just post a Noop kind of a thing just to tell that I read something, but it didn't help. I also explicitly did that in the perl script, but doesn't work. It only works in PHP. Bash would probably be my last choice of language to write an AGI with. Personally, I use C because it is my sharpest tool and because you can execute hundreds of AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. I suspect that the problems you are experiencing with Perl may have something to do with flushing STDOUT or reading the complete response from STDIN. I strongly suggest using an existing library for the language of your choice. Copy that. Not planning to write an AGI script in bash actually...it will be written in C# running on a remote system. I was just doing a quick PoC to figure out how would I use the stream file function to actually read audio files over the network and even though I used to teach Perl 10+ yrs ago, I don't do much scripting/coding for a long time, so the brain doesn't think like a coder anymore. Just needed to try various tricks w.r.t to how can I dynamically bring over audio files from another server, convert them to the codec of my channel and then play/store them locally (cache if you will) but wanted to learn the right way to do it with a local file first before I tried something fancier. Guess I'll continue playing with the php script that worked and once I figure the process out, will give it to the C# dev to implement. Can't believe I wasted more than 2-3 hrs on this :( BTW, I'd raised that a while ago, and got no conclusive response. How / what is the best way to stream audio files (not MOH/Internet Radio/TV and what not) inside a dialplan using AGI without comprising performance/adding latency too much. no examples of shout/ICE I could find that show how to do that simply by allowing me to run a web server remotely and use a shoutcast module to play the audio right into the channel ideas? -- _ -- 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] RealTime Queue Logging in 1.8
On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL And that table wont create in my database... Thanks Ish Can someone at least point me to the source file that I can analyse to find out the table requirements? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] About Asterisk SIP NAT Config
Dear all, I would appreciate it if you could teach me Asterisk SIP NAT Config. I'm trying to capture SIP Register with externip that should set in contact header at External SIP Server as shown below, but I haven't seen it. I need your help. My experiment environment is as follows. [My Environment] X-Lite --- Asterisk(SIP Server+SIP Client) --- External SIP Server * I assume that there is a NAT router between Asterisk and External SIP Server. [IP Configuration] External SIP Server IP: 192.168.100.1 Asterisk: 192.168.100.3 X-Lite IP: 192.168.100.5 [My sip.conf in Asterisk] [general] ; SIP Client Config - Start externip=*.*.*.* localnet=192.168.100.0/255.255.255.0 register = 1000:jrcyagi@192.168.100.1/1234 ; To make a call to a external SIP server [mysipprovider-out] type=friend secret=jrcyagi username=1000 host=jrc.nwt.com fromuser=1000 fromdomain=jrc.nwt.com canreinvite=no insecure=very qualify=yes nat=yes context=from-mysipprovider ; is further defined in extensions.conf ; SIP Client Config - End ; SIP Server Config - Start [1000] type=friend secret=jrcyagi aith=md5 nat=yes host=dynamic reinvite=yes qualify=1000 dtmfmode=inband callerid=yagi1 1000 disallow=all allow=ulaw allow=alaw context=default Regards, Yagishita -- _ -- 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 1.4.41 - Warning and Notice about contact info and stale nonce
Hi, Nobody on this? Le 16/05/2011 23:35, Administrator TOOTAI a écrit : Le 16/05/2011 18:27, Jose P. Espinal a écrit : Administrator TOOTAI wrote: Of course it's 1.4.41. And the result is that devices doesn't register anymore. Thanks for any hint. If you are installing from source, check out if some modules did not load properly due to undefined symbols. # asterisk -gvvc | tee output.txt CLI stop gracefully Then review that output.txt file. Don't think that the problem is here: the devices are working well with previous version of asterisk on the same server. Also, other devices from other manufacturer are still working ok. Question is why auth is OK but registration failed? On 1.4.40 we juste had to change the device local port (eg from 5061 to 5062) and registration was OK. On 1.4.41 this trick is no more working. And stale nonce should have an end of life in our mind, but doesn't. Thanks for your tip. -- Daniel -- _ -- 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] MOH uploading is not working with 1.4
Hi when I Upload MOH file from Asterisk GUI ,it is getting success and even not getting any error,But if check the destination path the file is not showing , even the source file and destination path and formate are correct .I am not getting any error log from asterisk console too. I read in google that there is some known issue in asterisk 1.4 regarding this ,I suspect this als o same .Can anyone explain what is expecting to happen in back end when upload MOH file from asterisk GUI ,then I am will able to debug more on this . Thanks Nikhil On 06/03/2011 07:25 PM, Steve Edwards wrote: On Fri, 3 Jun 2011, Nikhil wrote: I am using 1.4 asterisk and asterisk GUI. If I do moh upload its is not working .help me on this Not unless you provide some details. What's not working? Is the file not being uploaded? Is the file not being uploaded to the correct directory? Is the file in the wrong format and Asterisk refuses to play it? Please reply with a snipped of the console output showing what error you are receiving. -- _ -- 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 Online Training
Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? 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
Re: [asterisk-users] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.com wrote: I strongly suggest using an existing library for the language of your choice. On Mon, 6 Jun 2011, A E [Gmail] wrote: Copy that. Not planning to write an AGI script in bash actually...it will be written in C# running on a remote system. How / what is the best way to stream audio files (not MOH/Internet Radio/TV and what not) inside a dialplan using AGI without comprising performance/adding latency too much. Well, C# means you're getting your data from a Windows host, so I'd fix that first :) Without knowing all the details, the options I see are: ) Transfer the file using HTTP, FTP, SCP, etc. You'll have to wait until the entire file is transferred before you can start playing. ) 'Stream' the file using a shared file system like NFS or Samba. If the 'source' and 'target' hosts are on different continents this may not be practical. If they are in the same rack... ) Stream the file using a custom application. app_playback.c is only about 550 lines (1.8.0) which includes all the standard application 'boilerplate' for help, cli interface, loading, unloading, etc. as well as all of playback's little buddies like SayAlpha, SayDigits, SayNumber, etc. so a custom application cribbed from app_playback.c should only be 100 lines or so. Sometimes you can be a little bit 'sneaky' and hide uncomfortable waits in your 'caller experience.' Many years ago, I built an Asterisk system that needed to authorize a credit card transaction before delivering content (aka adult chat). The boss was a real picky SOB (we're still good friends) and always complained about the smallest interruption in the 'caller experience.' My solution was a multi-threaded AGI where one thread did the credit card thing while another thread played 'please hold while we authorize this transaction -- and get ready for a great time.' By the time the prompt finished, the response was back from the credit card processor and the process appeared (to the caller) to be instantaneous. -- 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] AGI STREAM FILE not working?
In article banlktikpxc_jk5xvhyx9akakltvl0v6...@mail.gmail.com, A E [Gmail] all.efor...@gmail.com wrote: Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup console output: -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new stack -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024, CHANNEL(language)=en_AU) in new stack -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh -- Playing 'welcome' (escape_digits=1) (sample_offset 0) -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new stack == Spawn extension (AllPhones, 5150, 4) exited non-zero on 'SIP/PBX-0024' But nothing happens...as in even when it says that it's playing the file (as verified in the asterisk 'full' log), I hear nothing on the phone What gives? spent 2 hrs Googling but nothing! :( Firstly, you need to check that you can successfully play files outside of the AGI environment. Replace the AGI command with: same = n,Playback(welcome) If that doesn't work, the problem is nothing to do with AGI. However, I think what else is happening is that your AGI script is sending the STREAM FILE command and then immediately exiting. This goes back to the dialplan and executes a hangup when only a tiny fraction of the welcome file has been played. You could test this theory in two different ways, as I'm not sure whether it's the exiting of the AGI or the subsequent hangup that is aborting the playback. a) Put a sleep 5 in your agitest.sh after the echo. As others have said, you should really use a proper library that reads responses to AGI commands, but for testing, a sleep will keep the AGI script alive while the message plays. b) Put a same = n,Wait(5) after the AGI command. If the AGI leaves the message playing, this would give it some time to play before you hang up the line. Hope this helps! Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Bridged Call
I have a Bridged call with 2 parties. I want to redirect one party to a conference room and the other party to an outside number. I tried doing that with a dialplan. I used ChannelRedirect in the dialplan and redirected the first channel to the conference room. however, the second channel disconnects. Reading thru the mailing list i understand that this expected. However, I don't understand how I can connect the second channel to an outside number. Can someone give a hand? -- _ -- 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] broken SVN asterisk 1.8 ?
On 11-06-05 12:18 PM, satish patel wrote: Hey guys! I have just download latest SVN Revision 322051 and compile and install but my asterisk -V showing still old version :( is it broken ? /usr/sbin/asterisk -V Asterisk SVN-branch-1.8-r321926 No, are still running the old binary? Also, did you make clean; make distclean before ./configure? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://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 GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x
Hello Folks; Perhaps I am chasing my tail here. Before I go any further, is this compatible/supported in Asterisk 1.6x? If so, I would be willing to post any manager.conf or http.conf snippets needed. When I attempt to open the Asterisk Web GUI, I get a 'page not found'. I am sure this is something really minor - something silly that I missed. Any words of wisdom? Glen -- _ -- 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 GUI - the one from Diguim/Asterisk - issues on Asterisk 1.6x
On Mon, Jun 6, 2011 at 11:55 AM, Silver Thorne szilvertho...@gmail.com wrote: Hello Folks; Perhaps I am chasing my tail here. Before I go any further, is this compatible/supported in Asterisk 1.6x? If so, I would be willing to post any manager.conf or http.conf snippets needed. When I attempt to open the Asterisk Web GUI, I get a 'page not found'. I am sure this is something really minor - something silly that I missed. Any words of wisdom? Glen I would get used to using the command line interface or use PBX In a Flash, FreePBX or something like that if you want / need what they offer. -M -- _ -- 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] issues.asterisk.org
On 06/06/2011 12:08 AM, Jeremy Kister wrote: i'm trying to review issues that i'm monitoring and/or have reported at http://issues.asterisk.org when I click on 'My View' or 'View Issues' I get an error: APPLICATION ERROR #401 Database query failed. Error received from database was #1142: DELETE command denied to user 'mantisreadonly'@'localhost' for table 'mantis_tokens_table' for the query: DELETE FROM mantis_tokens_table WHERE '2011-06-06 00:03:56' expiry. Please try this again. Are tickets that I had set up for monitoring on mantis going to be automatically monitored in jira ? No. We migrated as much as we could. This was one minor thing that was not migrated over. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.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] DAHDI - skipping sound on voicemail
Hi, I'm seeing some voicemails with skipping sound. Specifically, when somebody is giving their phone number sometimes you can hear that a digit is missing, but it was clearly said.it sounds like somebody just removed sloppily the digit using a wav editor. It might happen on live calls, but because voicemails are recorded as wav files I actually have some of those in archive. Where should I be looking first? All those voicemail calls I have archived are DAHDI-inbound. And I do hear some small clicking noise on live calls, so it might be related. echotraining seems an old Zapata thing, but I can`t find this setting anywhere on my server anyhow. Mike -- _ -- 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] AGI STREAM FILE not working?
On Mon, Jun 6, 2011 at 10:39 AM, Tony Mountifield t...@mountifield.orgwrote: In article banlktikpxc_jk5xvhyx9akakltvl0v6...@mail.gmail.com, A E [Gmail] all.efor...@gmail.com wrote: Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten = 5150,1,Answer() same = n,Set(CHANNEL(language)=en_AU) same = n,AGI(testagi.sh) same = n,Hangup console output: -- Executing [5150@AllPhones:1] Answer(SIP/PBX-0024, ) in new stack -- Executing [5150@AllPhones:2] Set(SIP/PBX-0024, CHANNEL(language)=en_AU) in new stack -- Executing [5150@AllPhones:3] AGI(SIP/PBX-0024, testagi.sh) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/testagi.sh -- Playing 'welcome' (escape_digits=1) (sample_offset 0) -- SIP/PBX-0024AGI Script testagi.sh completed, returning 0 -- Executing [5150@AllPhones:4] Hangup(SIP/PBX-0024, ) in new stack == Spawn extension (AllPhones, 5150, 4) exited non-zero on 'SIP/PBX-0024' But nothing happens...as in even when it says that it's playing the file (as verified in the asterisk 'full' log), I hear nothing on the phone What gives? spent 2 hrs Googling but nothing! :( Firstly, you need to check that you can successfully play files outside of the AGI environment. Replace the AGI command with: same = n,Playback(welcome) Yes, had done that as first order of business before even trying AGI If that doesn't work, the problem is nothing to do with AGI. However, I think what else is happening is that your AGI script is sending the STREAM FILE command and then immediately exiting. This goes back to the dialplan and executes a hangup when only a tiny fraction of the welcome file has been played. You could test this theory in two different ways, as I'm not sure whether it's the exiting of the AGI or the subsequent hangup that is aborting the playback. I thought so too, didn't know what to do about it a) Put a sleep 5 in your agitest.sh after the echo. As others have said, you should really use a proper library that reads responses to AGI commands, but for testing, a sleep will keep the AGI script alive while the message plays. This works! b) Put a same = n,Wait(5) after the AGI command. If the AGI leaves the message playing, this would give it some time to play before you hang up the line. This doesn't work Thanks for the help. I just need some sort of a wait loop there (as I don't really know how to pick up the 200 result=0) till the prompt finishes. -- _ -- 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 Online Training
2011/6/6 Antonio Modesto mode...@isimples.com.br: Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. I have not bought the course nor will I. I am self taught everything in IT and Telephony. I know each person learns differently, that is why when I train someone, I don't show them, I make them do it and coach until they don't need help. In college, I studied for business administration and accounting. It was too easy. My classes at WVU had 300 students, so attendance was impossible to keep track of. I aced all the tests. Then one summer I was a book keeper at t a travel agency. I hated it. I hated waking up to go to work, I hated the commute, and I hated the job. Then it dawned on me that I was always good at computers and figuring things out, ever since I got my VIC 20 and later my Commodore 64. I even had a Timex Sinclare(sp?) I got into a bit of trouble but since I was a kid and there was no malicious intent, the legal anchorites went away, just had to deal with my parents. Total tangent, I apologize. Classes slow me down and then become super boring. I fell asleep in the 3Com NBX course a few times, but got a 98% on the cert exam. I learned Cisco on the job, walked into a Prometric and passed with 80 something % and got my CCNA. Same for M$ products. I am planning to take the Red Hat cert soon. Anyways, I find structured courses to be a waste. Usually they hamper your creativity and are filled with fluff. If you cannot get a solid grasp of Asterisk in six months on your own lab using Google, voip-info, and howtos, then the course is probably for you you. Thanks, Steve Totaro -- _ -- 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] Issues with Digum Repos / AsteriskNOW Bad Packages
Anyone have an update as to when Digium will ship a working package? -- Forwarded message -- From: Andrew Joakimsen joakim...@gmail.com Date: Wed, Mar 23, 2011 at 23:53 Subject: Issues with Digum Repos / AsteriskNOW Bad Packages To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] Issues with Digum Repos / AsteriskNOW Bad Packages
On 06/06/2011 08:07 PM, Andrew Joakimsen wrote: Anyone have an update as to when Digium will ship a working package? According to https://issues.asterisk.org/view.php?id=18748 new packages should already have been pushed. If not perhaps you could join #asterisk or #asterisk-dev on irc.freenode.net and ask Qwell (aka Jason Parker) about this. Regards, Patrick -- _ -- 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] Issues with Digum Repos / AsteriskNOW Bad Packages
I have used those packages: [Apr 7 01:09:51] WARNING[27966]: loader.c:434 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/asterisk/modules/app_voicemail_imapstorage.so: undefined symbol: copy [Apr 7 01:09:51] WARNING[27966]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. You're now apparently running into issue #18718 (https://issues.asterisk.org/view.php?id=18718), which was a regression introduced in Asterisk 1.4.39 or so. This specific issue won't be fixed in a normal Asterisk release until 1.4.41. However, packages are in a slightly better position, since we can sometimes apply a fix and just rebuild the packages. I'll do that today, and you will see 1.4.40-3 shortly (I'll also send you a note when they're available). I can only imagine how frustrating this is for you... Unfortunately, app_voicemail.c is written in an overly complicated way, and it's difficult to catch issues like these. I've talked to the person that manages our test platform, and we'll be taking steps to watch for these types of issues in the future. On Mon, Jun 6, 2011 at 14:31, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 06/06/2011 08:07 PM, Andrew Joakimsen wrote: Anyone have an update as to when Digium will ship a working package? According to https://issues.asterisk.org/view.php?id=18748 new packages should already have been pushed. If not perhaps you could join #asterisk or #asterisk-dev on irc.freenode.net and ask Qwell (aka Jason Parker) about this. Regards, Patrick -- _ -- 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 -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?)
On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards asterisk@sedwards.comwrote: On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards asterisk@sedwards.com wrote: I strongly suggest using an existing library for the language of your choice. On Mon, 6 Jun 2011, A E [Gmail] wrote: Copy that. Not planning to write an AGI script in bash actually...it will be written in C# running on a remote system. How / what is the best way to stream audio files (not MOH/Internet Radio/TV and what not) inside a dialplan using AGI without comprising performance/adding latency too much. Well, C# means you're getting your data from a Windows host, so I'd fix that first :) now now. It works pretty well actually, can implement extremely complicated logic, multi-threaded, can run as a service, and integrates with the web-app which is all in asp.net etc. anyway, moving on Without knowing all the details, the options I see are: ) Transfer the file using HTTP, FTP, SCP, etc. You'll have to wait until the entire file is transferred before you can start playing. ) 'Stream' the file using a shared file system like NFS or Samba. If the 'source' and 'target' hosts are on different continents this may not be practical. If they are in the same rack... ) Stream the file using a custom application. app_playback.c is only about 550 lines (1.8.0) which includes all the standard application 'boilerplate' for help, cli interface, loading, unloading, etc. as well as all of playback's little buddies like SayAlpha, SayDigits, SayNumber, etc. so a custom application cribbed from app_playback.c should only be 100 lines or so. Right. Had thought about all of those, but looking for something along the lines of an application that can be invoked from inside the AGI socket connection i.e. picking a file over the network from a fast/lite http server (ala lighthttpd/nginx) and streaming it into the channel. So kind of like a 'Playback/Background over the network' kind of an app so one doesn't have to worry about bringing the file over, using NFS/SAMBA fileshares, caching and thus avoiding excessive file i/o. Does the MP3Player application do that? We could do that but ideally I'd like to avoid any transcoding etc. so we can create and save files in a ulaw/g729 etc formats and then just stream them avoiding all latency, file i/o, CPU issues. You're right, playback/background could be modified, unfortunately I'm not a C developer, so I might not be able to do it. But if someone knows of something that does the above from inside an AGI connection, that'd be awesome. Thanks so much -- _ -- 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 issue with polycom dialplan
Hi all, I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from _71XX. now what happen if i dial any 711X number my polycom just dial 711 and say busy number look like my phone doing some regex itself. like 911 number.. Did you get what i am trying to say ? it was working before with 1.2 but after upgrade 1.8 it started issue. why its just going with 711* 611* 511* etc... -S -- _ -- 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 1.8 issue with polycom dialplan
look like we found issue in phone configuration files [2-9]xx From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:43:22 + Subject: [asterisk-users] asterisk 1.8 issue with polycom dialplan Hi all, I have just upgrade asterisk 1.2 to 1.8 and we have numbers starting from _71XX. now what happen if i dial any 711X number my polycom just dial 711 and say busy number look like my phone doing some regex itself. like 911 number.. Did you get what i am trying to say ? it was working before with 1.2 but after upgrade 1.8 it started issue. why its just going with 711* 611* 511* etc... -S -- _ -- 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] half sip registration at 1.8.3
Hi all,
I've got something strange, that got me searching for quite awhile.
Configuration as followed:
Linphone on a laptop, that is connected via openvpn to a proxy.
That proxy is connected with iax to another asterisk.
On the second one i have several hard and softphones.
Behaviour at first glance:
From the softphone i can allways set up a connection,
But the otherway round fails 9 out-of 10 times.
However, if i stop-and-start linphone, the connections is allways
succesful.
First conclusion was, that if i got a diffrent (dynamic) ip-adress from
openvpn, i got to restart linphone, to force a re-registration.
Sounds reasonable, but why is linphone able to place calls, but not able
to accept them? (guests are off) I mean, if the phone is registered with
different values, also the outgoing call should fail. Not?
To avoid this behaviour, should i drastically drop the registration
duration at the softphone side? I still uses the default one (3600s).
Or should i tweak the min/max/default expiry-timers at asterisk?
Currently they are (also the default) 60/3600/120 seconds.
Hans
ps these are the lines from the console:
-- Executing [0277611@from_iax:1] noop(IAX2/kc3004-6511,
,0277611)
-- Executing [0277611@from_iax:2] answer(IAX2/kc3004-6511, )
-- Executing [0277611@from_iax:3] dial(IAX2/kc3004-6511,
SIP/0277611 ) [Jun 6 19:03:32] WARNING[23015]: app_dial.c:2039
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [0277611@from_iax:4] hangup(IAX2/kc3004-6511, )
== Spawn extension (from_iax, 0277611, 4) exited non-zero on
'IAX2/kc3004-6511'
-- Hungup 'IAX2/kc3004-6511'
corresponding lines from the ARA-dialplan:
| 118 | from_iax | 0212676 |1 | noop | ${CALLERID},${EXTEN} |
| 119 | from_iax | 0212676 |2 | answer | |
| 120 | from_iax | 0212676 |3 | dial | SIP/0212676 |
| 121 | from_iax | 0212676 |4 | hangup | |
--
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Re: [asterisk-users] Asterisk Online Training
Way to go Steve. That's the best way to learn. Steve Totaro stot...@asteriskhelpdesk.com wrote: 2011/6/6 Antonio Modesto mode...@isimples.com.br: Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. I have not bought the course nor will I. I am self taught everything in IT and Telephony. I know each person learns differently, that is why when I train someone, I don't show them, I make them do it and coach until they don't need help. In college, I studied for business administration and accounting. It was too easy. My classes at WVU had 300 students, so attendance was impossible to keep track of. I aced all the tests. Then one summer I was a book keeper at t a travel agency. I hated it. I hated waking up to go to work, I hated the commute, and I hated the job. Then it dawned on me that I was always good at computers and figuring things out, ever since I got my VIC 20 and later my Commodore 64. I even had a Timex Sinclare(sp?) I got into a bit of trouble but since I was a kid and there was no malicious intent, the legal anchorites went away, just had to deal with my parents. Total tangent, I apologize. Classes slow me down and then become super boring. I fell asleep in the 3Com NBX course a few times, but got a 98% on the cert exam. I learned Cisco on the job, walked into a Prometric and passed with 80 something % and got my CCNA. Same for M$ products. I am planning to take the Red Hat cert soon. Anyways, I find structured courses to be a waste. Usually they hamper your creativity and are filled with fluff. If you cannot get a solid grasp of Asterisk in six months on your own lab using Google, voip-info, and howtos, then the course is probably for you you. Thanks, Steve Totaro -- _ -- 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 Online Training
+1...I am an autodidact myself, never took any courses in IT or Telephony other a computing course in tge late 80s that was actually a typing class that used computers. Slainte, Sherwood McGowan Sent from my iPhone On Jun 6, 2011, at 5:58 PM, Amadu alsta...@gmail.com wrote: Way to go Steve. That's the best way to learn. Steve Totaro stot...@asteriskhelpdesk.com wrote: 2011/6/6 Antonio Modesto mode...@isimples.com.br: Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. I have not bought the course nor will I. I am self taught everything in IT and Telephony. I know each person learns differently, that is why when I train someone, I don't show them, I make them do it and coach until they don't need help. In college, I studied for business administration and accounting. It was too easy. My classes at WVU had 300 students, so attendance was impossible to keep track of. I aced all the tests. Then one summer I was a book keeper at t a travel agency. I hated it. I hated waking up to go to work, I hated the commute, and I hated the job. Then it dawned on me that I was always good at computers and figuring things out, ever since I got my VIC 20 and later my Commodore 64. I even had a Timex Sinclare(sp?) I got into a bit of trouble but since I was a kid and there was no malicious intent, the legal anchorites went away, just had to deal with my parents. Total tangent, I apologize. Classes slow me down and then become super boring. I fell asleep in the 3Com NBX course a few times, but got a 98% on the cert exam. I learned Cisco on the job, walked into a Prometric and passed with 80 something % and got my CCNA. Same for M$ products. I am planning to take the Red Hat cert soon. Anyways, I find structured courses to be a waste. Usually they hamper your creativity and are filled with fluff. If you cannot get a solid grasp of Asterisk in six months on your own lab using Google, voip-info, and howtos, then the course is probably for you you. Thanks, Steve Totaro -- _ -- 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 -- _ -- 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] PRI issue its BUSY
sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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] Pops clicks at the end of sound files
Hi all, We recently decided to get a professionally recorded set of prompts for our asterisk based IVRs and received these as the following: Bit Rate: 1536Kbps Sample Size: 16bit Channels: Stereo Sample Rate: 48kHz Format: PCM I use Wavepad to convert it to: Bit Rate:64Kbps Sample Size: 8bit Channels: Mono Sample Rate: 48kHZ Format: CCIT-ALAW I copied these files to an asterisk server and then used asterisk -rx to convert the files to g729. The problem I have is that at the end of every file there is a pop / distortion after playback. Anyone have the same issue before? -- _ -- 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] PRI issue its BUSY
This is wired.. If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound calls.. But its not working with asterisk 1.8 :( ( i can call in but not out) From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:11:28 + Subject: Re: [asterisk-users] PRI issue its BUSY sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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 -- _ -- 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] sccp problem
Dear I installed chan-sccp-b v3 on a powerful virtual machine, with 4 cpu cores and 16GB RAM(enabled in kernel by PAE) about 1,200+ clients are going to register in this machine. all data of clients are saved in ORACLE. The asterisk (1.6.2.18) connected to the database throw odbc(unixodbc). all logging are disabled( verbose, debug and sccp debug) . the asterisk was crashed every few minutes. here some question 1-does anyone have experience doing it by 1200+ clients? 2-is chan-sccp-b v2 more stable than v3?what is the best sccp channel for this project? 3-does chan_sccp_b have rtcache for disabling, like IAX and SIP in realtime mode?I had the same experience with IAX, when our online users grew up, asterisk was crashed. but by disabling rtcache, we had better condition best -- Pezhman Lali -- _ -- 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] [SOLVED]PRI issue its BUSY
Solution: pridialplan=unknow From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:33:44 + Subject: Re: [asterisk-users] PRI issue its BUSY This is wired.. If i connect my old asterisk 1.2 box my PRI working great! all inbound outbound calls.. But its not working with asterisk 1.8 :( ( i can call in but not out) From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 02:11:28 + Subject: Re: [asterisk-users] PRI issue its BUSY sometime i am getting Span 1: Channel 0/23 got hangup request, cause 16 but my call doesn't get completed == Primary D-Channel on span 1 up -- Restart requested on entire span 1 == Using SIP RTP CoS mark 5 -- Executing [7076941815@from-sip:1] Dial(SIP/7328-0004, DAHDI/G1/17076941815) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/17076941815 -- DAHDI/i1/17076941815-4 is proceeding passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 is ringing -- DAHDI/i1/17076941815-4 is making progress passing it to SIP/7328-0004 -- DAHDI/i1/17076941815-4 answered SIP/7328-0004 -- Span 1: Channel 0/23 got hangup request, cause 16 -- Executing [h@from-sip:1] Hangup(SIP/7328-0004, ) in new stack == Spawn extension (from-sip, h, 1) exited non-zero on 'SIP/7328-0004' From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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 -- _ -- 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] Pops clicks at the end of sound files
Hi, I had a similar issue converting wav files one time. Ended up using sox to convert to .sln as that ended up being the sounding conversion. I used the below command on a directory of files to convert: for a in *.wav; do sox $a -t raw -r 8000 -s -w -c 1 `echo $a|sed s/.wav/.sln/` resample -ql; done S. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand Sent: Monday, June 06, 2011 7:12 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Pops clicks at the end of sound files Hi all, We recently decided to get a professionally recorded set of prompts for our asterisk based IVRs and received these as the following: Bit Rate: 1536Kbps Sample Size: 16bit Channels: Stereo Sample Rate: 48kHz Format: PCM I use Wavepad to convert it to: Bit Rate:64Kbps Sample Size: 8bit Channels: Mono Sample Rate: 48kHZ Format: CCIT-ALAW I copied these files to an asterisk server and then used asterisk -rx to convert the files to g729. The problem I have is that at the end of every file there is a pop / distortion after playback. Anyone have the same issue before? -- _ -- 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] Pops clicks at the end of sound files
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj
Chand Sent: Monday, June 06, 2011 7:12 PM
We recently decided to get a professionally recorded set of prompts for
our asterisk based IVRs and received these as the following:
[snip]
The problem I have is that at the end of every file there is a pop /
distortion after playback.
On Mon, 6 Jun 2011, Skyler wrote:
I had a similar issue converting wav files one time. Ended up using sox
to convert to .sln as that ended up being the sounding conversion.
I used the below command on a directory of files to convert:
for a in *.wav; do sox $a -t raw -r 8000 -s -w -c 1 `echo $a|sed
s/.wav/.sln/` resample -ql; done
You can save a couple of 'process creations' by replacing '`echo $a|sed
s/.wav/.sln/`' with '${a/.wav/.sln}'
I use sox to transcode my files. I've had occasional issues with pops
at the end of files. I suspected it was because of the DC offset my
voice talent introduces when they use their portable gear instead of their
studio gear.
I use Audacity to trim off the pop. It's manual, but it's faster than
reordering the prompts.
FWIW, I use the following commands to convert to WAV:
sox ${INPUT} -c 1 -s -w -r 8000 /tmp/$$.wav
normalize ${FILENAME}.wav
Sox has a bunch of obtuse (IMNSHO) commands. There may be one that could
automagically trim the pop for you.
--
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] Pops clicks at the end of sound files
On Jun 6, 2011, at 10:51 PM, Steve Edwards asterisk@sedwards.com wrote: Sox has a bunch of obtuse (IMNSHO) commands. There may be one that could automagically trim the pop for you. The argument is question is the trim command. If the OP wishes to find an automagic method, they would need to determine the length of the file in seconds, then feed the length - durationofthepop to the sox trim command -- _ -- 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] Pops clicks at the end of sound files
On Jun 6, 2011, at 10:51 PM, Steve Edwards asterisk@sedwards.com wrote: Sox has a bunch of obtuse (IMNSHO) commands. There may be one that could automagically trim the pop for you. On Mon, 6 Jun 2011, Sherwood McGowan wrote: The argument is question is the trim command. If the OP wishes to find an automagic method, they would need to determine the length of the file in seconds, then feed the length - durationofthepop to the sox trim command Or, you could use the 'reverse' command, trim the first xxx milliseconds, and then reverse again. Then you don't need to know how long the file is. Or, (and this is the one I've always thought was really cool but never mastered) the 'silence' command can trim leading, trailing, and excessive silence between words. It has 'thresholds' that may be of use to eliminate the trailing pop. But since I've never mastered it, I'm just guessing :) Veering further off topic... If somebody really knows there stuff, I suspect the highpass and lowpass filters could also help produce more intelligible prompts. Anyone, anyone? -- 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] RealTime Queue Logging in 1.8
I use following for MySQL... CREATE TABLE queue_log( id int(11) NOT NULL auto_increment, time datetime not null, queuename VARCHAR(50), agent VARCHAR(50), callid varchar(32), event VARCHAR(100), data1 VARCHAR(100), data2 VARCHAR(100), data3 VARCHAR(100), data4 VARCHAR(100), data5 VARCHAR(100), PRIMARY KEY (id) ) ENGINE=InnoDB ; Check the link to know the meaning of data1,data2... for Events. http://www.voip-info.org/wiki/view/Asterisk+log+queue_log [SATISH] On Mon, Jun 6, 2011 at 1:49 PM, Ishfaq Malik i...@pack-net.co.uk wrote: On Thu, 2011-06-02 at 16:03 +0100, Ishfaq Malik wrote: Hi Does anyone know of an accurate resource I could refer to for this? The best I can find is http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL And that table wont create in my database... Thanks Ish Can someone at least point me to the source file that I can analyse to find out the table requirements? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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 get DTMF in Konference module in Asterisk
Hi List, I am trying to get DTMF into conference room. for conference I am using Konference module. Konference don't have an option of DTMF gets. Is there any way by which I can get DTMF within conference room? - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] Is this feature or Bug of all Asterisk versions ?
Hi List, I am facing an issue of automatic DTMF created by Asterisk(1.4,1.6,1.8). Issue is that when conference goes more then 10 minutes then we gets more DTMF which is generated by asterisk. The reason of starting these DTMF is loud volume, more noise area, Baby voice and lady voice. It's creating more issue into conference when you are working on DTMF base services. - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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
