Re: [asterisk-users] asterisk distributed device state = res_jabber Versus res_ais
11 aug 2010 kl. 15.49 skrev Leif Madsen: On 10-08-10 04:11 AM, Olle E. Johansson wrote: 26 jul 2010 kl. 18.13 skrev Leif Madsen: On Asterisk 1.6.2, your only option for distributing device state is with res_ais. I've used it in a labbing system and it works well -- the caveat is that your machines need to be on a low latency network (i.e. LAN). With Asterisk 1.8 (currently 1.8.0-beta1) you can use XMPP to distribute your device states over the WAN. I've made it work with the Tigase XMPP server. More information about it can be found in the doc/distributed_devstate-XMPP.txt file. This fall, we're going to implement it using SIP for 1.4 and 1.8/trunk. Look for project pinana. Development will start later this month. Sounds very cool! I look forward to playing around with it. Also thanks for picking a branch name that is not related to fruit or frogs. Thanks for the feedback. I guess the name was a mistake and I'll take it under reconsideration :-) /O -- _ -- 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] Monitor asterisk
On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote: Might be worth your time to check out: http://www.humbuglabs.org/ Though they write: ... insight into the enterprise’s telephony infrastructure. Utilizing a set of none-intrusive analytical technologies, Humbug is capable of interfacing directly with your PBX system, analyzing its traffic, plotting it and providing ... It looks (!) like an online-service. Who would give an outsider access to your phone-usage info? And what are these so-called none-intrusive analytical technologies ?? With this scarce amount of info, i wouldn't even bother to sign When looking at their terms of service : Humbug Agent means the open source Humbug Agent which is installed on a PBX or Softswitch for the purpose of collecting Customer Data, together with any fixes, updates and upgrades provided to you (collectively, the Humbug Agent). Servers means the servers controlled by Humbug (or its wholly owned subsidiaries) upon which the Processing Software and Customer Data are stored. So it looks like your CDR-data is collected by an Humbug-program and afterwards sent to on of _THEIR_ servers Scarry!!! Funny thing is though, that they even dare to use the phrase fraud detection. Security is about knowing what to trust and what not. -- _ -- 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 sip peers : musiconhold class
Can anyone help because I don't understand why Asterisk can not read the input file, there is not much info given... 2 files : [r...@asterisk testing]# file testExtended.wav testExtended.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz [r...@asterisk testing]# file testLong.wav testLong.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz to mono : [r...@asterisk testing]# sox testExtended.wav -r 8000 -c1 testExtended2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative [r...@asterisk testing]# sox testLong.wav -r 8000 -c1 testLong2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox effects: resample clipped 2 samples; decrease volume? afterwards : [r...@asterisk testing]# file testLong2.wav testLong2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz [r...@asterisk testing]# file testExtended2.wav testExtended2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz But Asterisk can not open them : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav Any thoughts ?! Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- 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] dial_exec_full problems with TDM400
Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? At the moment I have a terminal open all the time with verbose=10 and debug=10, sadly this log is not written to the logfiles so is lost when the terminal exists (perhaps there is a way round this, I don't know) Shutting down asterisk and restarting dahdi removes the problem for another day. Asterisk is version 1.6.2.5-0ubuntu1 Dahdi is version 2.2.1 Any help appreciated. I am at a complete loss what to do, except go back to the old 1.4 server. Cheers, Jason. -- _ -- 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] MP3Player audio format
Hi, I've successfully installed Asterisk and placed test calls using the MP3Player application. However I notice that my call quality varies drastically depending on which MP3 I use. Since I'm not changing any settings I'm assuming that the encoding of the file makes a difference. I've tried with both the g729 and alaw codec and they both give the same replicable results. I read that the underlying program mpg123 prefers mp3 files without ID3 tags. What is the recommended audio format (bit/sec) for using the MP3Player application or does this not make a difference? Thanks, Andy -- _ -- 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] dial_exec_full problems with TDM400
On Tuesday 17 Aug 2010, Jason Morgan wrote: Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? Have you got any extensions defined that aren't physically connected to anything? I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I built myself, and was getting similar symptoms to what you describe. It seemed not to be freeing up channels it was trying to associate with non-existent devices. I made sure that every entry in sip.conf had a corresponding phone plugged in somewhere, then went through the dialplan and removed all references to anything that wasn't mentioned in sip.conf. (And there were a few.) It seems to have stayed up since then . -- AJS -- _ -- 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] dial_exec_full problems with TDM400
Hi AJ, Surely this is a really bad bug if unconnected SIP devices ( a very likely occurance ) can take out trunk lines. Anyway what you say is true, there are several sip phones defined and not all are physically present all the time. I'll remove definitions of all sip phones for a while and see what that does. Cheers, Jason. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of A J Stiles Sent: 17 August 2010 10:36 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dial_exec_full problems with TDM400 On Tuesday 17 Aug 2010, Jason Morgan wrote: Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24 or so, not nailed it down precisely) incoming calls are not answered and outgoing calls get dial_exec_full. Incoming calls are reported to either A:just ring and ring, or B:get an engaged tone. Strangely when this happens asterisk DOES see the incoming call in situation A, but fails to answer. What tests can I do to resolve this as it is very inconvenient as we are missing a lot of calls? Have you got any extensions defined that aren't physically connected to anything? I replaced an ancient Asterisk version with a bang-up-to-date 1.6 version I built myself, and was getting similar symptoms to what you describe. It seemed not to be freeing up channels it was trying to associate with non-existent devices. I made sure that every entry in sip.conf had a corresponding phone plugged in somewhere, then went through the dialplan and removed all references to anything that wasn't mentioned in sip.conf. (And there were a few.) It seems to have stayed up since then . -- AJS -- _ -- 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] MySQL Connect problem...
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get stuck
here? i don't see anything in cli and core show channels just shows everyone
stuck in state ring on the connect string with no errors.
Cheers
Geraint
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Re: [asterisk-users] realtime sip peers : musiconhold class
Hi. Just to let you know, we record voices with audacity, and export audio as flac, just in case we need to edit it. Then I have the following sh script: o# cat convert.sh #!/bin/sh today=$(date +%F); mkdir -p $today/flac; mkdir -p $today/wav; mkdir -p $today/ul; for i in *.flac; do echo echo Processing $i; echo #$filename= sox $i -r 8000 -c 1 $(echo $i|rev|cut -d . -f2-10|rev).wav; normalize-audio -a 25dB $(echo $i|rev|cut -d . -f2-10|rev).wav; mv $i $today/flac/; sox $(echo $i|cut -d . -f1).wav $(echo $i|rev|cut -d . -f2-10|rev).ul; mv $(echo $i|rev|cut -d . -f2-10|rev).wav $today/wav/; mv $(echo $i|rev|cut -d . -f2-10|rev).ul $today/ul/; echo ; done echo All done; On 17 August 2010 08:07, Jonas Kellens jonas.kell...@telenet.be wrote: Can anyone help because I don't understand why Asterisk can not read the input file, there is not much info given... 2 files : [r...@asterisk testing]# file testExtended.wav testExtended.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz [r...@asterisk testing]# file testLong.wav testLong.wav: RIFF (little-endian) data, WAVE audio, 20294 channels 1414676809 Hz to mono : [r...@asterisk testing]# sox testExtended.wav -r 8000 -c1 testExtended2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative [r...@asterisk testing]# sox testLong.wav -r 8000 -c1 testLong2.wav resample -ql sox sox: effect `resample' is deprecated; see sox(1) for an alternative sox effects: resample clipped 2 samples; decrease volume? afterwards : [r...@asterisk testing]# file testLong2.wav testLong2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz [r...@asterisk testing]# file testExtended2.wav testExtended2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz But Asterisk can not open them : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav Any thoughts ?! Jonas. On 08/14/2010 04:30 PM, Motiejus Jakštys wrote: On Sat, Aug 14, 2010 at 4:24 PM, Jonas Kellens jonas.kell...@telenet.be wrote: intro extended version.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz You need *MONO, 8000Hz* $ man sox -- Motiejus Jakštys -- _ -- 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] sending sms from Asterisk server
Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Convert wav-file to alaw-file
Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav The wav-file is MONO, 8000Hz according to SoX and confirmed by the sox mailinglist (http://sourceforge.net/mailarchive/message.php?msg_name=4C6A4661.9000209%40telenet.be) [r...@asterisk testing]# soxi testLong2.wav Input File : 'testLong2.wav' Channels : 1 Sample Rate: 8000 Precision : 16-bit Duration : 00:01:27.05 = 696427 samples ~ 6529 CDDA sectors Sample Encoding: 16-bit Signed Integer PCM Why then can I not use Asterisk to 'convert' to alaw ?! Kind regards, Jonas. -- _ -- 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] Convert wav-file to alaw-file
On 8/17/2010 9:00 AM, Jonas Kellens wrote: Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav Try using an absolute path to the files: asterisk -rx file convert /path/to/testExtended2.wav /path/to/textExtended2.alaw Sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Convert wav-file to alaw-file
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Subject: [asterisk-users] Convert wav-file to alaw-file Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]# asterisk -rx file convert testLong2.wav testLong2.alaw Unable to open input file: testLong2.wav The wav-file is MONO, 8000Hz according to SoX and confirmed by the sox mailinglist (http://sourceforge.net/mailarchive/message.php?msg_name=4C6A4661.9000209%4 0telenet.be) [r...@asterisk testing]# soxi testLong2.wav Input File : 'testLong2.wav' Channels : 1 Sample Rate: 8000 Precision : 16-bit Duration : 00:01:27.05 = 696427 samples ~ 6529 CDDA sectors Sample Encoding: 16-bit Signed Integer PCM Why then can I not use Asterisk to 'convert' to alaw ?! Kind regards, Jonas. As I interpret what I have read in this forum over the last several months, there is not really an ALAW format. If you do a sox (don't remember if you can do in native Asterisk) convert to RAW (headerless) format, Asterisk is happy to consider that as ALAW. -- _ -- 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 Context
Hi, As nobody has answered it yet, and long time ago I also seeked answer to this exact same question, so I guess there is still no way to do it. What I ended up doing was defining 100 contexts in extensions.conf, all with the switch statements, so I didn't have to do a reload everytime. And then I would add these contexts in the realtime database as per requirement. It was about 3 years ago, and I don't remember more details about it, but just an idea in case it might help you. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-15 2:41 PM, Dan Journo d...@keshercommunications.com wrote: Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch = Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Create File Directory
Hello,
I have to following dial plan.
exten = 5551234,1,Answer()
exten =
5551234,n,Read(ACCOUNTNUMBER|/var/lib/asterisk/clientsounds/company/recordingsystem/welcome_accountnumberplease)
exten = 5551234,n,Read(MENUNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/menu_number_please)
exten = 5551234,n,Read(WHICHMESSAGE|/var/lib/asterisk/clientsounds/ company
/recordingsystem/whichmessage|1)
exten =
5551234,n,Playback(/var/lib/asterisk/clientsounds/kesher/recordingsystem/recordaftertone)
exten =
5551234,n,Record(/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}/${MENUNUMBER}/${WHICHMESSAGE}.wav)
However, Record fails if the directory doesnt exist.
How can I automatically create the directory before (or while) running the
Record command?
Thanks
Dan
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Re: [asterisk-users] Create File Directory
Solved.
exten = 5551234,1,Answer()
exten = 5551234,n,Read(ACCOUNTNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/welcome_accountnumberplease)
exten = 5551234,n,System(mkdir
/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER})
exten = 5551234,n,Read(MENUNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/menu_number_please)
exten = 5551234,n,System(mkdir
/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}//${MENUNUMBER})
exten = 5551234,n,Read(WHICHMESSAGE|/var/lib/asterisk/clientsounds/ company
/recordingsystem/whichmessage|1)
exten = 5551234,n,Playback(/var/lib/asterisk/clientsounds/ company
/recordingsystem/recordaftertone)
exten =
5551234,n,Record(/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}/${MENUNUMBER}/${WHICHMESSAGE}.wav)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: 17 August 2010 15:10
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Create File Directory
Hello,
I have to following dial plan.
exten = 5551234,1,Answer()
exten =
5551234,n,Read(ACCOUNTNUMBER|/var/lib/asterisk/clientsounds/company/recordingsystem/welcome_accountnumberplease)
exten = 5551234,n,Read(MENUNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/menu_number_please)
exten = 5551234,n,Read(WHICHMESSAGE|/var/lib/asterisk/clientsounds/ company
/recordingsystem/whichmessage|1)
exten =
5551234,n,Playback(/var/lib/asterisk/clientsounds/kesher/recordingsystem/recordaftertone)
exten =
5551234,n,Record(/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}/${MENUNUMBER}/${WHICHMESSAGE}.wav)
However, Record fails if the directory doesnt exist.
How can I automatically create the directory before (or while) running the
Record command?
Thanks
Dan
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Re: [asterisk-users] Create File Directory
Un-top-posting...
On Tue, 17 Aug 2010, Dan Journo wrote:
I have to following dial plan.
[snip]
However, Record fails if the directory doesnt exist.
How can I automatically create the directory before (or while) running
the Record command?
(Seems like a reasonable feature to add to the record application...)
On Tue, 17 Aug 2010, Dan Journo wrote:
Solved.
exten = 5551234,n,Read(ACCOUNTNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/welcome_accountnumberplease)
exten = 5551234,n,System(mkdir
/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER})
exten = 5551234,n,Read(MENUNUMBER|/var/lib/asterisk/clientsounds/ company
/recordingsystem/menu_number_please)
exten = 5551234,n,System(mkdir
/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}//${MENUNUMBER})
exten = 5551234,n,Read(WHICHMESSAGE|/var/lib/asterisk/clientsounds/ company
/recordingsystem/whichmessage|1)
exten = 5551234,n,Playback(/var/lib/asterisk/clientsounds/ company
/recordingsystem/recordaftertone)
exten =
5551234,n,Record(/var/lib/asterisk/clientsounds/features/${ACCOUNTNUMBER}/${MENUNUMBER}/${WHICHMESSAGE}.wav)
Ugly*, but if it works for you...
If you add the --parents command line option, the first mkdir is
unnecessary.
*) Personally, I would wrap all of this into a simple AGI
(record-client-message?) where you could handle errors better and make it
easier to re-use the logic for multiple extensions.
--
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] Realtime Context
We use a script on a cron which runs from a table/queue that only has an entry placed in it if a new context is added. If that is the case the script writes a new extenions.conf generating all the contexts and switch statements by doing a distinct select of contexts in the table. It also executed the dialplan reload after the new conf file had been generated. If your shell scripting isn't up to much I can give you an example Ish On 15/08/10 19:35, Dan Journo wrote: Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch = Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -- 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
Re: [asterisk-users] Realtime Context
The whole point of using real-time, as the word 'real' says, is to avoid the reloads. If one is ok with the reloads, then why bother with real-time at all. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-17 10:50 AM, Ishfaq Malik i...@pack-net.co.uk wrote: We use a script on a cron which runs from a table/queue that only has an entry placed in it if a new context is added. If that is the case the script writes a new extenions.conf generating all the contexts and switch statements by doing a distinct select of contexts in the table. It also executed the dialplan reload after the new conf file had been generated. If your shell scripting isn't up to much I can give you an example Ish On 15/08/10 19:35, Dan Journo wrote: Hi, I'd like to be able to create contexts in r... -- 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
Re: [asterisk-users] IXJ Quicknet PhoneJack issues
On Friday 13 August 2010 15:13:53 Infra wrote: -- dialplan works, ext. 1265 rings, has two-way audio, call progress tones heard, sends dtmf, can dial out, but has no dialtone. Look for a line in chan_phone.c that looks like: wait = ast_tv(3, 0); Change that line to: wait = ast_tv(0, 3); -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Context
So that we can create a custom interface for our customers to be able to do simple stuff like adding extensions, changing dialplans without our (for our read my!) intervention. The DB storage is the main thing for us. On 17/08/10 16:09, Zeeshan Zakaria wrote: The whole point of using real-time, as the word 'real' says, is to avoid the reloads. If one is ok with the reloads, then why bother with real-time at all. Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-08-17 10:50 AM, Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk wrote: We use a script on a cron which runs from a table/queue that only has an entry placed in it if a new context is added. If that is the case the script writes a new extenions.conf generating all the contexts and switch statements by doing a distinct select of contexts in the table. It also executed the dialplan reload after the new conf file had been generated. If your shell scripting isn't up to much I can give you an example Ish On 15/08/10 19:35, Dan Journo wrote: Hi, I'd like to be able to create contexts in r... -- 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 -- 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
Re: [asterisk-users] Realtime Context
Ishfaq, do you use the asterisk real-time architecture? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-17 11:45 AM, Ishfaq Malik i...@pack-net.co.uk wrote: So that we can create a custom interface for our customers to be able to do simple stuff like adding extensions, changing dialplans without our (for our read my!) intervention. The DB storage is the main thing for us. On 17/08/10 16:09, Zeeshan Zakaria wrote: The whole point of using real-time, as the word 'rea... -- _ -- 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 Hardwares
On 16/08/10 11:46, Tino wrote: Hello, Can antbody recommend devices that can be used along with my Asterisk server Paging Amplifier SIP enabled Paging Gateway VOIP SIP loudspeaker Also , please recommend video phone sets that suppot paging, intercom (autoanswer) Thanks A Snom PA-1 should cover all those requirements. cheers, Paul. -- _ -- 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] colored CLI with reattach
Tilghman Lesher wrote: Eric Smith wrote: Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Yes, but you'll need to upgrade to the latest 1.4 release. This also only works if you do not explicitly disable colors (-n) in the main daemon. Eric, You could also run the main Asterisk daemon within a screen http://www.manpagez.com/man/1/screen/ session. It will require some changes to how Asterisk is started at boot, but they should be simpler and less prone to introducing any side effects than upgrading to the latest release. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Add play moh-files without reload
Hello list, is it normal that when adding new moh-files to the directory /var/lib/asterisk/moh/, asterisk does not see these new files ?! When I do a moh reload, then Asterisk is aware of the new files... Is there a solution that does not need a moh reload ?! Kind regards, Jonas. -- _ -- 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] Add play moh-files without reload
Jonas Kellens wrote: Hello list, is it normal that when adding new moh-files to the directory /var/lib/asterisk/moh/, asterisk does not see these new files ?! When I do a moh reload, then Asterisk is aware of the new files... Is there a solution that does not need a moh reload ?! Kind regards, Jonas. Not an official Asterisk Solution, but I use icecast, and Liquidsoap to stream MOH into Asterisk. Works pretty well, I can change MOH on the fly, based off playlists, and time of the day. I can't think of anything though that would be instantly aware of new files in the directory. ~Seann smime.p7s Description: S/MIME Cryptographic Signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Add play moh-files without reload
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Subject: [asterisk-users] Add play moh-files without reload Hello list, is it normal that when adding new moh-files to the directory /var/lib/asterisk/moh/, asterisk does not see these new files ?! When I do a moh reload, then Asterisk is aware of the new files... Is there a solution that does not need a moh reload ?! Kind regards, Jonas. This is a reach, but if you make your moh real-time that would probably resolve it. Also, if you use mpg123 or some external player as opposed to the built-in player, that might do it. If you are doing out-of-the-box reload is the only option. -- _ -- 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] IXJ Quicknet PhoneJack issues
On 17 Aug 2010 at 10:29, Tilghman Lesher wrote:
On Friday 13 August 2010 15:13:53 Infra wrote:
-- dialplan works, ext. 1265 rings, has two-way audio, call
progress
tones heard, sends dtmf, can dial out, but has no dialtone.
Look for a line in chan_phone.c that looks like:
wait = ast_tv(3, 0);
Change that line to:
wait = ast_tv(0, 3);
Thanks so much, dialtone is now present:
--- chan_phone.c.0 Mon Aug 10 14:12:35 2009
+++ chan_phone.cTue Aug 17 11:41:06 2010
@@ -1029,7 +1029,7 @@
if (tonepos = sizeof(DialTone))
tonepos = 0;
if (ast_tvzero(tv)) {
- tv = ast_tv(3, 0);
+ tv = ast_tv(0, 3);
}
res = ast_select(n + 1, rfds, NULL, efds,
tv);
} else {
--
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Re: [asterisk-users] Add play moh-files without reload
On 08/17/2010 08:36 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Subject:* [asterisk-users] Add play moh-files without reload
Hello list,
is it normal that when adding new moh-files to the directory
/var/lib/asterisk/moh/, asterisk does not see these new files ?!
When I do a moh reload, then Asterisk is aware of the new files...
Is there a solution that does not need a moh reload ?!
Kind regards,
Jonas.
This is a reach, but if you make your moh real-time that would
probably resolve it. Also, if you use mpg123 or some external player
as opposed to the built-in player, that might do it. If you are doing
out-of-the-box reload is the only option.
For realtime music-on-hold I have this :
CREATE TABLE IF NOT EXISTS `RTmusiconhold` (
`name` varchar(80) NOT NULL,
`directory` varchar(255) NOT NULL default '',
`application` varchar(255) NOT NULL default '',
`mode` varchar(80) NOT NULL default '',
`digit` char(1) NOT NULL default '',
`sort` varchar(16) NOT NULL default '',
`format` varchar(16) NOT NULL default '',
PRIMARY KEY (`name`)
) ENGINE=MyISAM DEFAULT CHARSET=latin1;
--
-- Dumping data for table `RTmusiconhold`
--
INSERT INTO `RTmusiconhold` (`name`, `directory`, `application`, `mode`,
`digit`, `sort`, `format`) VALUES
('106002', '/var/lib/asterisk/moh/106002', '', 'files', '', '', '');
But Asterisk does not see this moh-class, even after a complete reload :
[Aug 17 21:27:36] == Parsing '/etc/asterisk/extconfig.conf': [Aug 17
21:27:36] Found
[Aug 17 21:27:36] == Binding voicemail to mysql/asterisk/voicemail_users
[Aug 17 21:27:36] == Binding sipusers to mysql/asterisk/sip_buddies
[Aug 17 21:27:36] == Binding sippeers to mysql/asterisk/sip_buddies
[Aug 17 21:27:36] == Binding queues to mysql/asterisk/queues
[Aug 17 21:27:36] == Binding queue_members to mysql/asterisk/queue_members
[Aug 17 21:27:36] == Binding meetme to mysql/asterisk/conference
[Aug 17 21:27:36] == Binding musiconhold to mysql/asterisk/RTmusiconhold
asterisk*CLI moh show classes
Class: default
Mode: files
Directory: /var/lib/asterisk/moh
In musiconhold.conf I have just the default class defined :
[default]
mode=files
directory=/var/lib/asterisk/moh
random=yes
Kind regards,
Jonas.
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Re: [asterisk-users] Create File Directory
Un-top-posting... Sorry, been a while since I posted and forgot that Outlook top posts by design. -- _ -- 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] Polycom 331 freezes connecting to FreePBX
Sorry, I should clarify - we have had a similar setup (IPSEC VPN, Polycom 331) working at a different location with a different handset for this same firm. We've never gotten the phone/VPN to work at this particular site. I was just trying to explain that it DOES appear to be successfully connecting to the TFTP server that provisions the phones. The TFTP server is sitting right next to the Asterisk server so if it can connect to one it should be able to connect to the other - basic connectivity, it appears, is working between the sites. We currently have 57 other Polycom phones (most of them 331s) working in this system, with the current application, just fine, including maybe 10 that connect over an IPSEC VPN from a 3rd location. That does give me an idea though...we could take the handset from the failing location to the 3rd location (barely a mile away) and plug it in and see if it works there. If it does then the problem must be somewhere in the connection and not with the handset itself. If it doesn't work in the other location either then the problem is probably with the phone and/or it's configuration. Make sense? Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Monday, August 16, 2010 11:04 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote: We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the CFG files for it. The phone starts up, downloads SIP and the new application and otherwise seems to be booting normally. Then it gets to the LAN Properties screen that shows the phone's IP address, MAC address and firmware version and then...nothing. It just sits there frozen. I have a suggestion... Put back the 'old application', and determine whether the 'new application' broke your phone boot. Since you don't mention changing anything else, survey says it's probably the last thing you changed that broke things. -- _ -- 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] Directory routing to wrong extension if dial tones are pressed too quick.
Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension. The problem seems to revolve around how quickly the keys are pressed. So if 645 for Mikell is pressed very quickly, they end up being routed to Sarah Fish. But if they take their time, say 2 seconds between each keystroke, everything works ok. Are they any settings that can be adjusted for this? Thanks, Eddie Mikell -- _ -- 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 with Motorola Canopy
Hi Everyone, Can anyone share their experience with Motorola Canopy solution deployment and Asterisk? Is this a good fit? Thanks, Bruce -- _ -- 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 with Motorola Canopy
On Tue, 17 Aug 2010, bruce bruce wrote: Hi Everyone, Can anyone share their experience with Motorola Canopy solution deployment and Asterisk? Is this a good fit? I worked for the UK company that developed the canopy product before they were bought out by Motorola.. Based on my experience there, I'd say there were a good fit. What I found annoying (and the engineers, I suspect :) was that telco's wanted E1 interfaces on them, rather than use VoIP natively... What a waste of 300Mb/sec of bandwidth by crippling them with E1 interfaces. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sending sms from Asterisk server
On 08/17/2010 09:00 AM, Tino wrote: Hello, I would like to send sms to some external phone numbers from my asterisk server. Is it possible to send sms via softphones like X-Lite ? . Any tips regarding this will be helpful thanks This is easy to do by using email to SMS gateways. A list of them is on wikipedia (http://en.wikipedia.org/wiki/List_of_SMS_gateways). For the Asterisk side, you have an extension that sends the email. I personally use an AGI script for this part, but you could use a System() call as well. --johann -- _ -- 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] Polling DND status of a Linksys SPA9xx/5xx phone?
Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes on the phone, then dialing the feature code doesn't display 'Do Not Disturb' on the phone. What I need is an indication on the phone that it is on DND, AND an indication through BLF to other users that a particular phone is on DND. Any ideas? Thanks. -- James -- _ -- 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] Pfsense and IAX2 - What is the proper firewall NAT setup?
Hi Everyone, Just trying to connect the Zoiper Communicator to connect to Asterisk which is behind Pfsense. Here is what I get at debug and it doesn't register. Error code 16. Can someone please let me know their firewall, NAT, outbound 1-to-1 pfsense settings as it seems to me I am doing something wrong on the firewall? *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00426 DCall: 0 [44.55.66.77:4569]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00427 DCall: 0 [44.55.66.77:4569]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK * * Timestamp: 3ms SCall: 00427 DCall: 1 [44.55.66.77:4569]* 44.55.66.77 is my client IP. I see no Tx packets. What is happening? Thanks for sharing, Bruce -- _ -- 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] Monitor asterisk
On 17/08/10 6:34 PM, Hans Witvliet wrote: On Mon, 2010-08-16 at 13:35 -0400, Jamie A. Stapleton wrote: Might be worth your time to check out: http://www.humbuglabs.org/ Though they write: ... insight into the enterprise’s telephony infrastructure. Utilizing a set of none-intrusive analytical technologies, Humbug is capable of interfacing directly with your PBX system, analyzing its traffic, plotting it and providing ... It looks (!) like an online-service. Who would give an outsider access to your phone-usage info? :) Take it you don't use Google Analytics, Facebook insights, Feedburner, Amazon EC3 etc etc. Sure you have to decide who you want to trust (personally I trust the humbuglabs guys) and what their level of protection is (are they looking after their own security), but it seems to be the way things are going at the mo. -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Convert wav-file to alaw-file
On 18/08/10 1:15 AM, Danny Nicholas wrote: As I interpret what I have read in this forum over the last several months, there is not really an “ALAW” format. If you do a sox (don’t remember if you can do in native Asterisk) convert to RAW (headerless) format, Asterisk is happy to consider that as ALAW. That would be signed linear (SLIN). -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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
