Re: [asterisk-users] Call Recording and Posting
You can try using NFS. Also you can pay some one to write script that would move the files over on hang up. - Original Message - From: Dan Journo To: asterisk-users@lists.digium.com Sent: Monday, October 12, 2009 01:15 Subject: [asterisk-users] Call Recording and Posting Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. Many thanks Dan Journo -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPRINTF option : format %1$s not supported
Hi, With 1.6.1.7-rc2, doc says: select*CLI -= Info about function 'SPRINTF' =- [Syntax] SPRINTF(format,arg1[,...argN]) [Synopsis] Format a variable according to a format string [Description] Parses the format string specified and returns a string matching that format. Supports most options supported by sprintf(3). Returns a shortened string if a format specifier is not recognized. I'm trying use sprintf option allowing to swap argument display according format string. More precisely, I'm trying to this %1$s specifier (which means use 1st argument). Then, the reply is : ERROR[3185]: func_strings.c:547 acf_sprintf: Format type not supported: '%1$' with argument '1234' Though the message is clear, before giving up, I thought I should ask here if someone could successfully use the %1$s specifier (which is very useful when you want to localize some messages or output some command strings). Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
Joseph wrote: I just double checked the setting of the remote asterisk and it has the same setting as mine. Sip.conf has in Global: dtmfmode = rfc2833 individual extension has no dtmf setting at all, so global setting take precedence. All units Linksys, Sipura have DTMF Tx Method: Auto Linksys has an additional setting: DTMF Tx Mode: Strict My asterisk is using old Sipura units and dtmf tones to access voicemail are recognized. The remote asterisk is using newer Linksys units and dtmf to voicemail does not work, the phone hangs up. The strange part is: PSTN -- Asterisk (voicemail access) works OK on both sytemes. Asterisk (w/Linksys) -- Asterisk (w/Sipura) to Voicemail works OK Asterisk (w/Linksys) -- Asterisk (w/Linksys) to Voicemail DOES NOT work Asterisk (w/Sipura) -- Asterisk (w/Linksys) to Voicemail DOES NOT work You are using the ATAs to access from one Asterisk to the other one? Wouldn't make sense to connect those two asterisk through SIP or IAX via Internet instead of calling via PSTN anyway? Anyway, In this case it seems that this is not asterisk related, try the several Sipura/Linksys settings related to DTMF, also if the asterisk boxes are the only thing your ATA connects to, there is no point on using the auto mode. Check the relaxed dtmf setting on yhe linksys, and also check the Impedance, Rx and Tx gain, As well as the DTMF duration, All this knobs can mess up your DTMF tones if there is something wrong. Just -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote: I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? As said by others, there is no such built-in capability I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. It is really important for you that the recordings are available asap on your FTP destination? I had a similar task and I found problematic to upload the files from inside the dialplan, Is not that it can't be done, but if the FTP is slow and your number of connections limited, you may run into a problem with simultaneous calls ending and asterisk trying to upload 20 files at the same time. In my case the files could be uploaded every hour, so I made a simple bash script that uploads the new files to the FTP using 'curlftpfs', a nice command that mounts the remote FTP on a local mount point using FUSE, then the script just moves the files from the local folder to the FTP, and voila. Asterisk just takes care of moving recordings that ended to the desired path. I can post the bash script if you are interested. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Hi Iván, Thank you for replying. I hadn't thought about the problem of simultaneous calls. It would be a problem if a number of calls ended at the same time. If you can post it, the script would really be helpful as I'm only a beginner with Linux. Many thanks Dan Journo -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: 12 October 2009 12:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recording and Posting Dan Journo wrote: I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? As said by others, there is no such built-in capability I ask because I don't have any experience in Linux programming, so I won't be able to create a monitoring program on my own. It is really important for you that the recordings are available asap on your FTP destination? I had a similar task and I found problematic to upload the files from inside the dialplan, Is not that it can't be done, but if the FTP is slow and your number of connections limited, you may run into a problem with simultaneous calls ending and asterisk trying to upload 20 files at the same time. In my case the files could be uploaded every hour, so I made a simple bash script that uploads the new files to the FTP using 'curlftpfs', a nice command that mounts the remote FTP on a local mount point using FUSE, then the script just moves the files from the local folder to the FTP, and voila. Asterisk just takes care of moving recordings that ended to the desired path. I can post the bash script if you are interested. -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme and confbridge
Dear All, I read somewhere that app_meetme in 1.2 has a single lock therefore supporting 10+ conferencing is problematic... is this still the case with version 1.6? What is the functional difference between meetme and confbridge?? Thanks in advance for responses. Rgds, Robor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tealtime static
Dear All, Can I mix realtime conf and static configuration files? Thanks for responses. Rgds, Robor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPRINTF option : format %1$s not supported
On Monday 12 October 2009 05:51:31 Olivier wrote: [Description] Parses the format string specified and returns a string matching that format. Supports most options supported by sprintf(3). Returns a shortened string if a format specifier is not recognized. I'm trying use sprintf option allowing to swap argument display according format string. More precisely, I'm trying to this %1$s specifier (which means use 1st argument). Then, the reply is : ERROR[3185]: func_strings.c:547 acf_sprintf: Format type not supported: '%1$' with argument '1234' Though the message is clear, before giving up, I thought I should ask here if someone could successfully use the %1$s specifier (which is very useful when you want to localize some messages or output some command strings). Nope. Dollar ($) format specifiers are not supported at all. Given that the arguments to SPRINTF are specified in a string, you could simply repeat an argument where necessary, which is why I never bothered implementing any of the positional formats. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] libss7 problem with dialing a non numeric string
Hei! I'm trying to send special characters out to ss7 link, but libss7 seems to convert them to zeroes. The challenge is that our service provider demands some of the regional numbers to be sent in format D0+number. When I use D in front of the number in dialplan, libss7 replaces it with 00, So I have a dial string: exten = _[A-Z].,1,Dial(DAHDI/g1/DD0501,,g) But in SS7 trace I see: --VARIABLE LENGTH PARMS[1]-- Called Party Number: Nature of address: 3 NI: 0 Numbering plan: 1 Address signals: 00501# [ 07 03 10 00 00 00 05 f1 ] Do you have any idea how to fix that? My chan_dahci.conf is as follows: [channels] switchtype=euroisdn ;;; linkset 1 context=incoming_ss7 echocancel=yes echotraining=yes echocancelwhenbridged=yes group=1 linkset=1 signalling=ss7 ss7type=itu pointcode=50 adjpointcode=14 defaultdpc=14 ;networkindicator=international networkindicator=national_spare ss7_called_nai=dynamic ss7_calling_nai=dynamic ;ISDN call type ss7_internationalprefix = 00 ;ss7_nationalprefix = ;ss7_subscriberprefix = ;ss7_unknownprefix = DD cicbeginswith = 2 channel = 2-31 sigchan=1 Thanks in advance BR Rennes Neps ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPRINTF option : format %1$s not supported
On Mon, Oct 12, 2009 at 4:51 AM, Olivier oza-4...@myamail.com wrote: Hi, With 1.6.1.7-rc2, doc says: select*CLI -= Info about function 'SPRINTF' =- [Syntax] SPRINTF(format,arg1[,...argN]) [Synopsis] Format a variable according to a format string [Description] Parses the format string specified and returns a string matching that format. Supports most options supported by sprintf(3). Returns a shortened string if a format specifier is not recognized. I'm trying use sprintf option allowing to swap argument display according format string. More precisely, I'm trying to this %1$s specifier (which means use 1st argument). Then, the reply is : ERROR[3185]: func_strings.c:547 acf_sprintf: Format type not supported: '%1$' with argument '1234' Though the message is clear, before giving up, I thought I should ask here if someone could successfully use the %1$s specifier (which is very useful when you want to localize some messages or output some command strings). My initial (imho) impression is that localizations in the code, are in general a bad way to approach localization in general. The localizations should be located neither in Asterisk code nor in dialplan code. I know, I know, that such code already exists, but that's still not making my assertion false... murf -- Steve Murphy ParseTree Corp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] live audio streaming using monitor, mixmonitor or chanspy
Hi to all, is it possible to setup a live audio streaming in Asterisk using for source monitor, mixmonitor or chanspy? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules
Hi, I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul. But the phone that is attached to the line does nothing at all. asterisk-CLI shows a lot of -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing Even when I lift up the handset during (and no asterisk-activity) I do not get the usual line tone. CLIdahdi show channel 1 gives: File Descriptor: 17 Span: 1 Extension: Dialing: no Context: DLPN_DialPlan1 Caller ID: 4000 Calling TON: 0 Caller ID name: FXS 1 Mailbox: 4000 Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook What am I missing? Cheers Eckhard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules
- Eckhard Jokisch e.joki...@orange-moon.de wrote: Hi, I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul. But the phone that is attached to the line does nothing at all. asterisk-CLI shows a lot of -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing Even when I lift up the handset during (and no asterisk-activity) I do not get the usual line tone. CLIdahdi show channel 1 gives: File Descriptor: 17 Span: 1 Extension: Dialing: no Context: DLPN_DialPlan1 Caller ID: 4000 Calling TON: 0 Caller ID name: FXS 1 Mailbox: 4000 Destroy: 0 InAlarm: 0 Signalling Type: FXO Kewlstart Radio: 0 Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no DND: no Echo Cancellation: 128 taps currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook What am I missing? Is your cable from the port to the phone good? Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 in asterisk upgrade issue
Hi, I need some help. I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3. For what I understand I can make the backup of the license files in /var/lib/asterisk/licenses/ if I need to reinstall the operating system, but in this case with new O.S and new version can I reuse this licenses? Regards Luis Silva ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 in asterisk upgrade issue
Speaking blindly here, you should be able to unless there is some kind of server architecture involved (probably not with /v/l/a path.) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Silva Sent: Monday, October 12, 2009 12:36 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] G729 in asterisk upgrade issue Hi, I need some help. I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3. For what I understand I can make the backup of the license files in /var/lib/asterisk/licenses/ if I need to reinstall the operating system, but in this case with new O.S and new version can I reuse this licenses? Regards Luis Silva ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 in asterisk upgrade issue
Luis Silva wrote: I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3. For what I understand I can make the backup of the license files in /var/lib/asterisk/licenses/ if I need to reinstall the operating system, but in this case with new O.S and new version can I reuse this licenses? Yes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk to Asterisk access voicemail - not working
On 10/12/09 13:29, Ivan Stepaniuk wrote: You are using the ATAs to access from one Asterisk to the other one? Wouldn't make sense to connect those two asterisk through SIP or IAX via Internet instead of calling via PSTN anyway? Anyway, In this case it seems that this is not asterisk related, try the several Sipura/Linksys settings related to DTMF, also if the asterisk boxes are the only thing your ATA connects to, there is no point on using the auto mode. Check the relaxed dtmf setting on yhe linksys, and also check the Impedance, Rx and Tx gain, As well as the DTMF duration, All this knobs can mess up your DTMF tones if there is something wrong. Just -- Iv?n Stepaniuk Alba Fot?nica S. L. www.albafotonica.com Thanks for the input, I'll try play with Linksys units. The Asterisk on the other end has no Internet connection so I need to use PSTN line. One thing I've noticed on the other end there is a bit echo and TDMF tones playing a bit loud when pressing the buttons, it it PSTN to SPA Gain or SPA to PSTN Impedance is set to standard 600 but I'll try 900; but I've read from some sources that to find the correct match can be a black art http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080477a06.shtml -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to send a digit to a channel??
Hello! I need to send a digit to a channel of an established call, from outside of Asterisk, I suppose it must be from the AMI. I want to send a * for example, but in addition to reproducing the sound of that digit (I dont care thatl), I need that the digit sent actually performs an action. For example if I have configured that the attended transfer in Asterisk is #2, I need somehow to be able to send the # and the 2 for the transfer menu begins. Any help with this??? I have proved with PlayDTMF, but all it does is play the sound of the digit, but nothing happens... Please need help! Thank you very much. Pablo Bernasconi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 with TDM2400 and 2 FXS modules
On Mon, Oct 12, 2009 at 05:59:16PM +0200, Eckhard Jokisch wrote: Hi, I set up my asterisk so it does dial out on DAHDI/1 which is on an FXS modul. But the phone that is attached to the line does nothing at all. asterisk-CLI shows a lot of -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing -- DAHDI/1-1 is ringing Hi, Just in case ... Did you plug a molex connector on the card ? The card need external power source to ring a phone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] video support with voicemail question
Hi, I'm trying to get voicemail to record video as well as audio. So far, only audio is recorded. I'm using files instead of the ODBC or IMAP. I'm running 1.6.1 beta code (I've tried several versions here). I've also tried running the video branch code and get similar results. I can run the echo application with both video and audio, so I'm confident that I have my sip and other configuration files set up for video. I can also make video SIP calls fine using H.263 or H.264 My voicemail.conf file looks like (pardon my text editor, it seems to demand that the first letter of each line gets changed to caps) [general] Format=wav|gsm|h263|h264 [default] 3131 = 1234, test mailbox, jonathan.gallme...@polycom.com, tz=central|attach=yes My extensions.conf has a simple test configuration that looks like: [general] Static=yes Autofallthrough=yes [default] Exten = 3131,1,Voicemail(3...@default,u) Exten = 3131,n,Hangup() Exten = 6060,1,VoiceMailMain() Like I said, everything works fine for leaving an audio voicemail. Video is just not recorded even though I'm calling in with a video call to my voicemail extension (3131). Does the voicemail app record video as I've seen stated in several places? Am I missing something big here? Thanks in advance Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Dan Journo wrote: Thank you for replying. I hadn't thought about the problem of simultaneous calls. It would be a problem if a number of calls ended at the same time. If you can post it, the script would really be helpful as I'm only a beginner with Linux The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. You have to set up a cron so the script is run every hour, normally putting the script or a link to it inside '/etc/cron.hourly' is enough. There is also a /etc/crontab file you can use to setup something more complicated if needed (ie: runing it every 2 hours, running at tea time...). read 'man cron', and 'man crontab'. You also need to install the magic part, 'curlftps'. in Debian that's the name of the package too. I use version curlftpfs 0.9.1 libcurl/7.18.2 fuse/2.5 Be careful that in *nix, file.WAV and file.wav are different files. Here is the script: #!/bin/sh MOUNT_POINT=/mnt/remote_ftp FTP_HOST=www.ftphost.com/htdocs/recordings FTP_USER=ftpusername:difficultpassword RECORDINGS=/var/spool/asterisk/monitor/upload echo Starting upload `date` echo Connecting to $FTP_HOST... curlftpfs -o user=$FTP_USER $FTP_HOST $MOUNT_POINT echo Transfering `du -hc $RECORDINGS/*.wav | grep total` ... mv -vf $RECORDINGS/*.wav $MOUNT_POINT echo Disconnecting. umount $MOUNT_POINT exit 0 -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Thanks for that. I really appreciate it! Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: 12 October 2009 22:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recording and Posting Dan Journo wrote: Thank you for replying. I hadn't thought about the problem of simultaneous calls. It would be a problem if a number of calls ended at the same time. If you can post it, the script would really be helpful as I'm only a beginner with Linux The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. You have to set up a cron so the script is run every hour, normally putting the script or a link to it inside '/etc/cron.hourly' is enough. There is also a /etc/crontab file you can use to setup something more complicated if needed (ie: runing it every 2 hours, running at tea time...). read 'man cron', and 'man crontab'. You also need to install the magic part, 'curlftps'. in Debian that's the name of the package too. I use version curlftpfs 0.9.1 libcurl/7.18.2 fuse/2.5 Be careful that in *nix, file.WAV and file.wav are different files. Here is the script: #!/bin/sh MOUNT_POINT=/mnt/remote_ftp FTP_HOST=www.ftphost.com/htdocs/recordings FTP_USER=ftpusername:difficultpassword RECORDINGS=/var/spool/asterisk/monitor/upload echo Starting upload `date` echo Connecting to $FTP_HOST... curlftpfs -o user=$FTP_USER $FTP_HOST $MOUNT_POINT echo Transfering `du -hc $RECORDINGS/*.wav | grep total` ... mv -vf $RECORDINGS/*.wav $MOUNT_POINT echo Disconnecting. umount $MOUNT_POINT exit 0 -- Iván Stepaniuk Alba Fotónica S. L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk dialplan to share fax line
Hi I look after a site which is using asterisk and a vsp for its primary telco needs, so I am on holiday for a week and of course some jack arse has decided to reboot the server and something has gone wrong with the remote access. Now they don't have any internet and i can't fix it remotely. Bad for them because they don't have any inbound call capabilities, they do have outbound via the fallback pstn line connected to the asterisk box, which is also a fax line - its primary useage. So what i would like to to when i get back is to setup the pstn line to be a backup inbound line as well. shared fax and normal phone line. How can I work the dial plan to 1) pickup the call 2) if its a fax hold the line till the fax machine takes over or will now that asterisk has picked up the line I have a tdm410 card with fxo + fxs ports. do I attach the fax machine to the asterisk server and just treat it as a extension, but letting all its outbound calls go through the pstn line on the tdm card - will the fax work that way ? the question is still how to I deal with inbound calls/faxes ? Alex signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
On Behalf Of Ivan Stepaniuk The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. What happens if the script is run while a recording is being written? Will it copy the incomplete file and then delete it? The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. If you used sshfs you could use public keys and eliminate the password hassles. Personally, I'd still vote for uploading the file at the completion of the recording via an AGI. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
Hi, To avoid the problem of deleting/copying calls that are still being recorded, I could record the call into a temp directory. Then using the dial plan, I could copy the temp recording into the ftp root directory once the call has ended. Dan Thank you for contacting Kesher Communications Ltd. IT Maintenance Clients can now receive a faster response by using our Live Chat and Support Service: Click Here This email and any files transmitted with it are confidential and intended solely for the recipient(s). If you are not the named addressee you should not disseminate, copy or alter this email. Under no circumstances may this email be distributed without written permission from the sender. Warning: Although the Company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments. All prices exclude VAT unless otherwise stated. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: 12 October 2009 23:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Recording and Posting On Behalf Of Ivan Stepaniuk The script is very simple and far from complete, it just moves the content into the mounted FTP directory. It has some verbose output as it is run from inside another script that redirects the output to a log file. What happens if the script is run while a recording is being written? Will it copy the incomplete file and then delete it? The script has a password inside so remember to 'chown root' and 'chmod 700' the file to protect it from other users. If you used sshfs you could use public keys and eliminate the password hassles. Personally, I'd still vote for uploading the file at the completion of the recording via an AGI. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [SOLVED] Asterisk to Asterisk access voicemail - not working
On 10/12/09 13:29, Ivan Stepaniuk wrote: You are using the ATAs to access from one Asterisk to the other one? Wouldn't make sense to connect those two asterisk through SIP or IAX via Internet instead of calling via PSTN anyway? Anyway, In this case it seems that this is not asterisk related, try the several Sipura/Linksys settings related to DTMF, also if the asterisk boxes are the only thing your ATA connects to, there is no point on using the auto mode. Check the relaxed dtmf setting on yhe linksys, and also check the Impedance, Rx and Tx gain, As well as the DTMF duration, All this knobs can mess up your DTMF tones if there is something wrong. Just -- Iv?n Stepaniuk Alba Fot?nica S. L. www.albafotonica.com Solved! The problem was caused by echo, changing impedance to 900 and decreasing SPA to PSTN and PSTN to SPA gain to -3 solved the problem. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Recording and Posting
On Tue, 13 Oct 2009, Dan Journo wrote: To avoid the problem of deleting/copying calls that are still being recorded, I could record the call into a temp directory. Then using the dial plan, I could copy the temp recording into the ftp root directory once the call has ended. True, but if you need to execute a process at the end of the call, why not make it an AGI and hide all the ugly details and keep your dialplan nice and clean and shiny and maintainable? Your recordings will be instantly available and the correct operation of your system does not depend on an externally scheduled external process involving clear-text passwords and obscure packages. Your successor will thank you :) -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
