Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
PS 42 is the answer, now what is the quesstion. :) What is the difference between a bird? -- _ -- 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] automixmon output file location and exec command options
Hi all, I have 2 quick question regarding the file location and post record command of the recording using automixmon in features.conf. With the normal monitor/mixmonitor applications you can change the location of where the recordings will be stored, by changing the MONITOR_FILENAME variable. I tried changing the TOUCH_MIXMONITOR_OUTPUT variable to include a path but it sill puts the recorded file in /var/spool/asterisk/monitor. Is there any way I can change this? The second question, is, is it possible to execute a command after one touch mixmonitor has completed? With the mixmonitor application this is possible, I was wondering if the option was available for the automixmon feature in features.conf I've been doing some googling around to see if I can find any info on the above, but I seem to be coming up short, or I'm looking in the wrong places. Any tips/suggestions would be appreciated. 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] Correct operation of timout parameter for dial application
I have now logged issue number 0018447 relating to this query.
Thanks all for your responses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: 03 December 2010 22:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Correct operation of timout parameter for dial
application
On Tuesday 30 November 2010 07:14:34 Bruce McAlister wrote:
Hi All,
I'd just like to verify what the correct operation of the timeout
parameter is for the dial application. I'm not sure if I've encountered
a bug or a configuration issue.
When a sip phone is not responding to invites on an outbound call, the
dial application still waits the duration of timeout before continuing
with dialplan execution. I was under the impression that app_dial would
timeout on the signalling prior to the timeout parameter specified in
the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to
voicemail. However, if/when the device is not responding to invites,
for whatever reason, the dial application waits 30 seconds before
setting the DIALSTATUS to NOANSWER. Is this expected behaviour? In
previous versions of asterisk, specifically (v1.2/v1.4) when the device
did not respond to invites the dial application exited prior to the
value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
I seem to recall an issue like that some time back where somebody thought
that if their SIP phone wasn't responding, the Dial app should wait the
full 30 seconds before giving up, but I cannot find the related commit for
that. I'm sure there's arguments on both sides for the behavior. I'd
suggest that you open an issue on issues.asterisk.org, and we can take a
look at how we could accommodate both approaches.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1
Hi RR, I've not tried compiling 1.8.1-rc1 on Solaris yet and I've not come across this issue as of yet. I did build 1.8.0-rc5 on Solaris 10 without any build error's though. I'm not sure if the code has changed that much between 1.8.0-rc5 and 1.8.1-rc1. I'm no coding guru by anyone's standards, but I do build a couple applications for Solaris. What has made my life a hell-of-a-lot easier is JDS-CBE and SFE, check out the following 2 links: http://dlc.sun.com/osol/jds/downloads/cbe/ http://pkgbuild.sourceforge.net/spec-files-extra/ What the above does is setup a common build environment for building applications. The SFE (spec-file-extra) is a framework for create rpm type spec files for solaris. Once you have one setup for asterisk then it is just a one line command to download and build asterisk. This is what I have been using to build asterisk on Solaris 10 for the past 3 years. It keeps the environment identical between versions. Have a look at getting that up and going first and then check out the spec file format and create one for your asterisk version you want to compile. My spec file is far from perfect at the moment, but it does work for what we require at the moment. Disclaimer: This is a little bit of work to setup and get working initially, but once it is setup and working, building subsequent asterisk versions and creating the Solaris SRV4 packages is a breeze :) Thanks Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: 08 December 2010 23:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Error building network library on OpenSolaris and 1.8.1-rc1 On Wed, Dec 8, 2010 at 3:55 PM, Tilghman Lesher tles...@digium.commailto:tles...@digium.com wrote: On Wednesday 08 December 2010 14:21:57 RR wrote: Hi Guys, Any one want to take a stab at helping with this please?? All I have found so far is that the netsock.c file has code that references to taking note when it's being built on a Solaris platform, but since I don't understand this a whole lot, I am not sure where to go from here...this is the excerpt from the netsock.c file: *#if defined (SOLARIS) #include sys/sockio.h #elif defined(HAVE_GETIFADDRS) #include ifaddrs.h #endif * I would've have thought this would have taken care of the issue by making sure 'make' handles this correctly but I guess not. Anyone? Please? http://opensolaris.org/jive/thread.jspa?threadID=116059tstart=105 I suspect we'll have to make a more complex check to verify that the structure elements are all there. Please open an issue on issues.asterisk.orghttp://issues.asterisk.org/ and reference this thread. We can then put up a patch that you can use to verify if better detection fixes your issue. Once verified, the patch will find its way into releases. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.comhttp://www.digium.com/ www.asterisk.orghttp://www.asterisk.org/ G'day Tilghman, Thanks for that thread. I guess a few other things broke because of the change and the consuming application then needs to be a little smarter like you said (and suggested by darrenr) to detect whether you're on OSOL or Solaris. Does that mean I should check this same thing out on Solaris 10 as well and see what happens? I am so lost with the Solaris build environment as (and I whinged about this earlier too) there is no good way of obtaining the standard Solaris packages and dependancies and everything just goes all over the place and then one is left scurrying around to find where the damn library needs to be for it to compile. Anyway, I will open an issue and reference this thread and we'll go from there. BTW, THANK YOU for taking note of this and trying to help. You guys will have bottomless beer pitchers paid for if you guys help me get this working and are ever in the NY area :) Cheers, \R -- _ -- 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] Correct operation of timout parameter for dial application
Hi All,
Just another follow-up, does anyone have any thoughts on the query below?
Thanks
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 01 December 2010 18:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Correct operation of timout parameter for dial
application
Hi All,
Does anyone have any thoughts on the question below, or do you think it may be
a question for the dev list?
Thanks
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Correct operation of timout parameter for dial
application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of timeout before continuing with dialplan
execution. I was under the impression that app_dial would timeout on the
signalling prior to the timeout parameter specified in the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to voicemail.
However, if/when the device is not responding to invites, for whatever reason,
the dial application waits 30 seconds before setting the DIALSTATUS to
NOANSWER. Is this expected behaviour? In previous versions of asterisk,
specifically (v1.2/v1.4) when the device did not respond to invites the dial
application exited prior to the value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
We are currently running v1.6.2.13
Thanks
Bruce
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Re: [asterisk-users] Zaptel / Asterisk on Solaris
Hi RR, As far as I am aware the version of Zaptel on SolarisVoIP is out of date. Aditionally the versions of the packages compiled at SolarisVoIP are only available, as far as I am aware, for the Solaris platform and not the OpenSolaris platform, there may be subtle differences between the two that may be causing your build error. If you have a look at SolarisVoIP there are pre-built packages for SPARC/X86 hardware which you do not need to build yourself. In saying all of the above, your millage may vary with zaptel running in a VM as the timing is virtualized (via usb) and is not, as far as I know, very well supported within a VM. Thanks Bruce From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR Sent: 01 December 2010 00:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zaptel / Asterisk on Solaris Hello nice people :) I have been struggling with trying to get Zaptel from http://www.slimey.org/zaptel-solaris.tar.gz on a Solaris 5.11 VM I obtained from the OpenSolaris Website. I have tried installing all the necessary packages, yet I keep getting errors no matter if I try using the gcc available at sunfreeware.comhttp://sunfreeware.com OR the blastwave CSWgcc packages and GNU 'gmake' (as suggested somewhere on the Internet). I have tried sending emails to the people at SolarisVoIP.com and To Simon, from Slimey.org who built/created this Zaptel Solaris Port, but it's been over 2 weeks and I've not heard anything from anyone. This is EXTREMELY critical for me to work...can anyone kind generous gentleman please help? Thank you so much \RR -- _ -- 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] Correct operation of timout parameter for dial application
Hi All,
Does anyone have any thoughts on the question below, or do you think it may be
a question for the dev list?
Thanks
Bruce
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce McAlister
Sent: 30 November 2010 13:15
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Correct operation of timout parameter for dial
application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of timeout before continuing with dialplan
execution. I was under the impression that app_dial would timeout on the
signalling prior to the timeout parameter specified in the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to voicemail.
However, if/when the device is not responding to invites, for whatever reason,
the dial application waits 30 seconds before setting the DIALSTATUS to
NOANSWER. Is this expected behaviour? In previous versions of asterisk,
specifically (v1.2/v1.4) when the device did not respond to invites the dial
application exited prior to the value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
We are currently running v1.6.2.13
Thanks
Bruce
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[asterisk-users] Correct operation of timout parameter for dial application
Hi All,
I'd just like to verify what the correct operation of the timeout parameter is
for the dial application. I'm not sure if I've encountered a bug or a
configuration issue.
When a sip phone is not responding to invites on an outbound call, the dial
application still waits the duration of timeout before continuing with dialplan
execution. I was under the impression that app_dial would timeout on the
signalling prior to the timeout parameter specified in the dial parameter.
For example, consider the following dialplan:
exten = 111,1),Dial(SIP/phone1,30,tg)
exten = 111,n,NoOp(DialStatus=${DIALSTATUS})
exten = 111,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = NOANSWER]?unavail)
exten = 111,n,GotoIf($[${DIALSTATUS} = BUSY]?busy)
exten = 111,n,GotoIf($[${DIALSTATUS} = CONGESTION]?busy)
exten = 111,n(unavail), Goto(voice-mail,vmu-phone1,1)
exten = 111,n(busy), Goto(voice-mail,vmb-phone1,1)
Under normal operation the originating caller is passed through to voicemail.
However, if/when the device is not responding to invites, for whatever reason,
the dial application waits 30 seconds before setting the DIALSTATUS to
NOANSWER. Is this expected behaviour? In previous versions of asterisk,
specifically (v1.2/v1.4) when the device did not respond to invites the dial
application exited prior to the value specified by timeout.
Can anyone clarify this issue for me please? Is this expected behaviour?
We are currently running v1.6.2.13
Thanks
Bruce
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Re: [asterisk-users] URGENT Help needed
Hi Michael, With regards the following error: 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache You can fix that one by modifying /etc/asterisk/modules.conf and uncommenting the following 2 lines: preload = res_odbc.so preload = res_config_odbc.so That will ensure the odbc resource is available for any other applications that may require it. Thanks Bruce -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Sent: 22 November 2010 10:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] URGENT Help needed Also, what happens if you do asterisk -c this may help you figure things out. Hi, These are the WARNINGSI found in /var/log/asterisk/messages after running the above command: [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxUdpEC in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [Nov 22 12:10:19] WARNING[2316] udptl.c: T38FaxMaxDatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications. [Nov 22 12:10:19] WARNING[2316] loader.c: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache [Nov 22 12:10:19] WARNING[2316] res_config_ldap.c: No directory user found, anonymous binding as default. [Nov 22 12:10:19] ERROR[2316] res_config_ldap.c: No directory URL or host found. [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] utils.c: trying to reset empty pool [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'SendFAX' [Nov 22 12:10:19] WARNING[2316] pbx.c: Already have an application 'ReceiveFAX' Thanks, Michael -- _ -- 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] PBX Lua with Asterisk ODBC
Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? 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] PBX Lua with Asterisk ODBC
Thanks for the quick response, however, how would I access an odbc dsn from the pbx_lua dialplan that has been defined in res_odbc.conf or related odbc structures? I've not come accross any documentation on that feature yet. Any tips/info/links would be appreciated. On 26/07/10 14:33, Faisal Hanif wrote: you can use all asterisk dial-plan functions and application in lua plus additional complete lua features. so answer is yes. Regards, Faisal Hanif On 7/26/2010 5:34 PM, Bruce McAlister wrote: Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? 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] PBX Lua with Asterisk ODBC
Ahh ok, so I am only able to access the application/functions that are
available to the dialplan.
I was wondering if it would be possible to access the handle of the odbc
connection directly from the lua dialplan.
On 26/07/10 17:10, Leif Madsen wrote:
On 10-07-26 10:34 AM, Faisal Hanif wrote:
You need to create a function is res_odbc for each of required query
and then u can use that function as normal asterisk dialplan function.
So in the dialplan, after you've modified func_odbc.conf you'd be able to do a
query like:
exten = start,1,Set(MyResult=${ODBC_GET_MY_DATA(banana_lollipop)})
How you structure that in pbx_lua I'm not sure, but you create the functions
with func_odbc.conf, which is probably the piece you're missing.
Leif Madsen.
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Re: [asterisk-users] Management interface
On 26/07/10 13:15, Tony LaMear wrote: I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 *Tony * I've been looking at ZenOSS, which appears to have an asterisk zenpack as well. http://www.zenoss.com/ I've not used it as of yet though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi on solaris
Hi Claudio, As far as I am aware, dahdi is not able to compile on Solaris, although I've not attempted to compile it. There may be others out there that may have better experience than I with dahdi on Solaris. Thanks Bruce -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Claudio Furrer Sent: 05 July 2010 22:11 To: asterisk-users@lists.digium.com Subject: [asterisk-users] dahdi on solaris Hello all, Does anybody know if is it possible to install dahdi on solaris 10? I've only found a zaptel modified code for solaris at solarisvoip site. I'd appreciate any comment or experience about asterisk + dahdi/zaptel on solaris.. Best regards, Caio -- _ -- 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 kick/mute using ConfBridge application
Hi All, We are currently evaluating the confbridge application while we prepare to upgrade our environment to asterisk v1.6.2.x. We have run in to two issues using it to kick/mute participants in a bridge and would like to ask for the experience of others running the application for any work-arounds. Firstly for kicking participants, would it be possible to use the softhangup application on a channel to effectively kick a participant from a bridge? Secondly, is it possible to mute a participant in the bridge using the AMI or a CLI. Any tips/suggestions would be greatly appreciated. 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
[asterisk-users] Voice mail maxmessage setting per mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, Is it at all possible to have the maxmessage setting on per user/mailbox value? We have a requirement whereby we want the global maxmessage setting to be 180 seconds per mail box, however, we would like to have certain users to be able to store longer voice mail messages. Is this at all doable in asterisk? Thanks Bruce -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLza18AAoJEI3A8f8Te208X6gH/2HRXwVfEMfSAQqyKK6a4xAY hp4p4AeoqgEK+FjpHgT7wtkLYIoz+/hqIUOUT44W6RBaZ9cEsS6hEPZAhZj+e4qx 7RPQkuxI3UVULWxzCuE/H9lSh6hWH/x95PynvmHVBuxw/Mc5um1XnkC2dmoiwnHZ Jpzlu2SnQ/lkg7BhxDtPxM1g+bYpMbdzNWv64JEaP4MxClpupDJJXJ8ETdUR/P0m f4tsttsoI3XzAuvwrVrlr//ANYA03zk1RjUYY/0Wfpj3tfSnCd0+hDDEIKo1aGJa 8EOSQptiPaISYX5ZUi0xADZ0wreodFYgov7DVQaOqDs4huaDuwYZ+UILpQETTv0= =gTui -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail maxmessage setting per mail box
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Thanks for the tip, although a second instance of asterisk, in our
environment, could very well mean additional hardware to offer this
feature for a select few users. In some cases users want variable
options, leaving 30 minute long VM's, some 10 minutes long, and for the
most part 3 minutes is sufficient. For each defined length of time,
would require a new instance of asterisk to offer the maxmessage setting
requested.
Thanks again for the tip/work around.
Thanks
Bruce
On 20/04/2010 14:58, Danny Nicholas wrote:
The Out of the box answer is no. A simple workaround would be to have a
second instance of Asterisk that you connect to via IAX to let the special
group leave a longer message.
Exten = s,1,noop(voicemail processing)
Exten = s,n,Gotoif(..special..)?longmail
Exten = s,n,Voicemail(${ext...@default)
Exten = s,n,Playback(vm-goodbye)
Exten = s,n,hangup
Exten = s,n(longmail),Dial(IAX2/longmail/${EXTEN},20,m)
On server 2
User 100
Exten = 100,1,Voicemail(1...@default)
Exten = 100,n,playback(vm-goodbye)
Exten = 100,n,hangup
Users 100-199
Exten = _1XX,1,Voicemail(${ext...@default)
Exten = _1XX,n,playback(vm-goodbye)
Exten = _1XX,n,hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce
McAlister
Sent: Tuesday, April 20, 2010 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voice mail maxmessage setting per mail box
Hi All,
Is it at all possible to have the maxmessage setting on per
user/mailbox value?
We have a requirement whereby we want the global maxmessage setting to
be 180 seconds per mail box, however, we would like to have certain
users to be able to store longer voice mail messages.
Is this at all doable in asterisk?
Thanks
Bruce
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Re: [asterisk-users] Voice mail maxmessage setting per mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jared, I'm talking about the maxmessage setting which is the maximum amount of time that a voice message can be, not the maximum number of messages in a mail box. Thanks Bruce On 20/04/2010 15:51, Jared Smith wrote: On Tue, 2010-04-20 at 14:34 +0100, Bruce McAlister wrote: Is it at all possible to have the maxmessage setting on per user/mailbox value? Absolutely, as long as you're talking about the maxmsg setting! In fact, there's an example in the sample voicemail.conf file that comes with Asterisk: ;4200 = 9855,Mark Spencer,marks...@linux-support.net, mypa...@digium.com,attach=no|serveremail=mya...@digium.com|tz=central| maxmsg=10 See how we set this particular mailbox to only have a maximum of ten messages? -- Jared Smith Digium, Inc. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLzcJ+AAoJEI3A8f8Te2081KQH/3Vl1w+J+XbgzoWS2KpmOfQ4 zMiErgXl74euV0kVNtHOetoTV8MzWGo3MfVPQEiqJPZgTk3svSYuH9dBd92o1CyF LVomdyLGqgogz4vyE/uVp6mu04dJpMXNaIqeGEeZ4xasggZL0xn2oEBh+Gxr46cG MuqGpdzOUlpITmh/Xbj4Zcp45l71fCJhhRLpzR3/5/wFz6YE036Sg8UGTRODeE2U 61Z/lGKHeq3Em0Nk+C34IQyb52sp7PUf7p6Y4TUAHyaSZF1olLosyVPqXKuL2F+U F4Kl874H1w4y4OPjFT2v4MOM1VlRfV3fbAm8+0OVkBLZWR72Otuja9n9QSLwreM= =uvRN -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail maxmessage setting per mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Steve, Thats interesting, I had a look through versions 1.6.0, 1.6.1 and 1.6.2 and I didnt see anything mentioned that maxmessage (maxsecs) has been enabled on a per mailbox setting. I did see that the maxmessage setting was renamed to maxsecs so that it would be easier to differentiate. Incidently we are currently running 1.4.x for our voice mail server, however if 1.6.x offers maxsecs (and others) as a configurable, per mailbox, setting then we will look in to upgrading the environment. Can you, or any else, confirm that maxsecs is a tunable per mailbox in 1.6.2.x? Thanks Bruce On 20/04/2010 15:48, Steve Edwards wrote: On Tue, 20 Apr 2010, Bruce McAlister wrote: Is it at all possible to have the maxmessage setting on per user/mailbox value? I'm a 1.2 Luddite, so YMMV... In 1.2, maxmessage is a global setting. In 1.6.1.6 (just what I happened to have on hand), maxmessage has been renamed to maxsecs and is a per user setting. I don't know where between 1.2 and 1.6.1.6 this change was made -- scouring the change logs is left as an exercise for the reader :) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLzcPjAAoJEI3A8f8Te208Wo4IAJsMBbeBzKQegEYk7QlDvng2 8v97f8Me3eMa/zx6Cx2ett/DZ2UcxSTCseqjXA7k/zQeyv6cK+zUMmTysXDMUIOn KQK87MGl91RKuM6UXsuY531CBRHVSx40upl/9d5BWDiWslPXnc0EuyAVGA3nEiSG /qXamLIMGsG40qCSXJbvncxn19QJ5jtXbA0Hq6wX+6LBtOwQoHeS37gdInprJRK/ rH+7+6/OyK6otLih5BqJ8y3RrFyJE9N9Yug1JtKwx9o9sPAt8+0+qYNfqyDm3OoP XtKHwRIOR3NGwOUI3kuttiXGlbucNaaPsp9Jb4suHhiEB7oua7f2se0bdBBEjB0= =r3fn -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice mail maxmessage setting per mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Thanks all for your input, much appreciated. I will investigate this further with The Google :) On 20/04/2010 16:27, Steve Edwards wrote: On Tue, 20 Apr 2010, Bruce McAlister wrote: Incidently we are currently running 1.4.x for our voice mail server, however if 1.6.x offers maxsecs (and others) as a configurable, per mailbox, setting then we will look in to upgrading the environment. Can you, or any else, confirm that maxsecs is a tunable per mailbox in 1.6.2.x? GIYF... On 12/31/2006 tilghman updated voicemail.conf.sample. The revision log notes the name change and that it is per-user. (www.mirrors.docunext.com/websvn/asterisk/view/trunk/configs/voicemail.conf.sample?rev=49075) When it made it into a release will take a little more googling. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEcBAEBAgAGBQJLzdQPAAoJEI3A8f8Te208k0gIAJx+EZ21b2bFNWl602NFqa8i t3R+020BzJQM2qCLK7payMIRgRSljHbTdzsRu94XobIo9qsZB+/kIO/BwbwnvTpz AmmKTKWMgiy+8JKGZ9HyN1kBVJgBJZnzJxOqgBHuKj8eXtvXmvr4y4APFXxIKdiD 2gLYd8zwC+Qa4vowD0qr8GJvbUqhqNe1dzyhjdsT/oGSnPNsw0I0Hk+9+2HZuMVL /7VAFgSvtbF5YDRAnDp1A3MEXkIHTMQCocuifKEJ+amdDgGha+TCgO4/4H4ipUWn rInoQ9v/a0OfZKE7NdcrBFG3M6vYgvIoytlYbkl7iPRRkoNNK3ZJzlu1PnkcVC4= =c80d -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 open source codec and sample size
Andres wrote: codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i386)) This line would seem to indicate the binary loads fine. I would concentrate on the License aspect. Delete the license from the directory and see if you get the same 'copy protection error'. If not it means the License location was correct but the file has a problem. Thanks for the tip the Andres. I will build asterisk 1.4.20 and try it with the v33 codec over the next couple days. I thought that if the codec loaded properly you would be able to issue show g729 from the asterisk CLI. However that command fails as it appears that the module is not loaded and exported its functions. Am I wrong in that assumption? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 open source codec and sample size
Eric ManxPower Wieling wrote: I don't understand why people won't pay $10/channel for a fully licensed, legal, and Asterisk supported G729 codec. I wish I could use $10/channel G729 codec from Digium, however, I've been trying to get that codec working on Solaris since v32 of that codec. The codec fails to load no matter what I do, and troubleshooting information from Digium (and the lists) is severly lacking. I do understand that it is unsupported, however, I wonder if the people who build the codec have successfully loaded the module within asterisk on Solaris themselves. If I can get this working we would be buying the digium codes without any questions at all. Just my 0.02c ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 open source codec and sample size
Jared Smith wrote: I see that Jason Parker from Digium answered your question in both July and August of last year. The issue (at least from what I read in the archives) seems to point to math libraries not being found in the proper location. Maybe there are some Solaris folks lurking on the list that can shed some light -- I'm pretty worthless when it comes to Solaris. Are you still trying on OpenSolaris, and is there anything different about the way it handles dynamic linking? Yes, Jason answered the question saying that the codec was unsupported and the other suggestion that was given was that it could possibly be that the license was in the wrong directory. This is the first time that I've heard of the math library not being in the correct location? Do you have a reference as to what Jason mentioned about the math library? When I first posed the question on the lists and a question via the digium channels I mentioned that I was using Solaris 10 Update 3. Which is what I was told the codec was built on. I've not tried it on OpenSolaris at all. The company I work for will only use the standard Solaris distribution, and not OpenSolaris in production. -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 open source codec and sample size
Jared Smith wrote: The issue (at least from what I read in the archives) seems to point to math libraries not being found in the proper location. Maybe there are some Solaris folks lurking on the list that can shed some light -- I'm pretty worthless when it comes to Solaris. Are you still trying on OpenSolaris, and is there anything different about the way it handles dynamic linking? I forgot to mention, in my previous email, that the math libraries on our boxes reside in the /lib directory, which is where the Solaris installer installs them by default. Looking at my last attempt to try and get this going (which, co-incidently, is the same system that Jason helped me with) I checked to see if the codec has any unresolved libraries: ldd ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 libm.so.2 = /lib/libm.so.2 The math libraries appear to be found OK on the box. The license is located in : /var/lib/asterisk/licenses The license file is in the directory: -rw-r--r-- 1 root root 308 Aug 27 2007 G729-39F0ABB3.lic However, every time I try to load the codec, I get the following in the asterisk console: codec_g726.so = (ITU G.726-32kbps G726 Transcoder) [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:403 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:407 load_module: This module is supplied under a commercial license granted by Digium, Inc. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:408 load_module: Please see the full license text supplied by the accompanying [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:409 load_module: register utility, or ask for a copy from Digium. [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:410 load_module: This product includes software developed by the OpenSSL Project [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:411 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Jun 10 23:16:40] NOTICE[2673]: codec_g729.c:412 load_module: Copyright (C) 1998-2006 The OpenSSL Project [Jun 10 23:16:40] WARNING[2673]: codec_g729.c:420 load_module: Failed to initialize G.729 copy protection! codec_g729a.so = (Annex A/B (floating point) G.729 Codec (optimized for i386)) In this case I am using asterisk v1.4.13, however, I have tried this with asterisk versions: 1.2.17 - 29 1.4.13 - 18 The codec versions I have tried are the i386 32-bit below: unsupported v32 unsupported v33 unsupported trunk v33 I cannot seem to locate version 34 for Solaris on the download site which is apparently the latest version which I have not tried as of yet. When I built asterisk I changed the directory locations to install everything in /opt/asterisk as apposed to spread over multiple directories. This would be the ideal case for us. However, when trying to get it to work as expected, I built asterisk using the default install directories to rule out any weirdness I may have caused by modifying the make file to install to a single top level directory. I've also asked the guys at SolarisVoIP some time ago to see if they had got G729 going, and as far as I am aware, they have not been able to get the codec working either on their Solaris systems. There are multiple posts on that mailing list where people mention large scale rollouts on Solaris being held back because they are unable to get the G729 codec operational under Solaris. I am not alone :) Any suggestions tips/tricks that you may be able to shed on this issue would be *greatly* appreciated. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729-v34 Builds Now Available
Build 32 and 33 do not even load under Solaris, I cannot test build 34 as it's not up on the website yet. Has anyone actually been able to successfully load the G729a codec under a Solaris version of Asterisk (v.1.4.18 for example)? The Asterisk Development Team wrote: Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux builds for both 1.4 and 1.6 have been updated for various API changes. * All of the Linux builds include changes so that an Ethernet interface explicitly named eth0, or eth1, etc., is no longer required. All of the builds are available from the following URL: * http://downloads.digium.com/pub/telephony/codec_g729/ Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729-v34 Builds Now Available
Hi, I have just checked again and the Solaris build of the codec appears to be v33 and not v34 as advertised. Thanks Bruce Bruce McAlister wrote: Hi, The Solaris build still appears to be at v32. Am I being a little hasty :) Thanks Bruce The Asterisk Development Team wrote: Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux builds for both 1.4 and 1.6 have been updated for various API changes. * All of the Linux builds include changes so that an Ethernet interface explicitly named eth0, or eth1, etc., is no longer required. All of the builds are available from the following URL: * http://downloads.digium.com/pub/telephony/codec_g729/ Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] codec_g729-v34 Builds Now Available
Hi, The Solaris build still appears to be at v32. Am I being a little hasty :) Thanks Bruce The Asterisk Development Team wrote: Greetings, The software G.729 codec module from Digium has been updated for all platforms. There are x86_32 and x86_64 versions optimized for specific processors available for both Asterisk 1.6 and 1.4 for the following platforms. * Linux * Solaris 10 * FreeBSD 7.0 * FreeBSD 6.1 Changes: * For Asterisk trunk / 1.6, builds have been updated for CLI API changes. * All non-Linux builds for both 1.4 and 1.6 have been updated for various API changes. * All of the Linux builds include changes so that an Ethernet interface explicitly named eth0, or eth1, etc., is no longer required. All of the builds are available from the following URL: * http://downloads.digium.com/pub/telephony/codec_g729/ Thank you for your support! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 requirements
Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hi, Thanks for the information, I will keep this for reference. Thanks Bruce Mindaugas Kezys wrote: This can help (script for Debian): apt-get install flex bison #dirty hack to prevent error from missing file cd /usr/include/linux touch compiler.h #PWLIB cd /usr/src wget http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz tar zxvf pwlib-v1_10_0-src-tar.gz cd pwlib_v1_10_0/ ./configure make make install make opt PWLIBDIR=/usr/src/pwlib_v1_10_0 export PWLIBDIR #OpenH323 cd /usr/src wget http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz tar zxvf openh323-v1_18_0-src-tar.gz cd openh323_v1_18_0/ ./configure make make opt make install OPENH323DIR=/usr/src/openh323_v1_18_0/ export OPENH323DIR cd /usr/src/asterisk/channels/h323/ make make opt cd /usr/src/asterisk ./configure make make install echo /usr/local/lib /etc/ld.so.conf ldconfig #or similar way #cp /usr/local/lib/* /usr/lib Regards, Mindaugas Kezys http://www.kolmisoft.com MOR PRO - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Thursday, February 21, 2008 10:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] chan_h323 requirements Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_h323 requirements
Hi, Thank you for the details of which versions to get. I will be building these two versions on Solaris to test chan_h323. Did your patch for building with OpenH323+ make it into the 1.4 edition of Asterisk? Thanks Bruce Vlasis Hatzistavrou (KTI) wrote: Hello, To compile chan_h323 as is distributed you need to download OpenH323 v1.18.0 and PwLib v1.10.0 from: http://www.voxgratia.org Some months ago I had made a patch to compile the 1.4.x version and the trunk version (which evolved to 1.6.x) with H323+. Sadly, the patch was not included in the 1.6.x version which is being released soon. So, for the time being you need to use the above versions from Voxgratia. Best regards, Vlasis Hatzistavrou. Bruce McAlister wrote: Hi All, I would just like to clarify the requirements of the h323 channel within asterisk. Can I use a recent edition of PTLib and OpenH323, for example, the editions located at OpenH323+: http://www.h323plus.org/source/ OpenH323+ v1.20.2 PTLib v2.0.1 Or do I need to use the versions at the original, now defunct, OpenH323 website: http://www.openh323.org/ OpenH323 v1.12.2 PWLib v1.5.2 I am hoping to build this for Asterisk 1.4.18 running on Solaris 10. Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +---+ | Bruce McAlister Blueface Ltd | | [EMAIL PROTECTED] http://www.blueface.ie | +---+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_h323 build failure - `IPTOS_MINCOST' undeclared
Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: -- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared identifier is reported only once chan_h323.c:2863: error: for each function it appears in.) gmake[1]: *** [chan_h323.o] Error 1 gmake: *** [channels] Error 2 -- I have downloaded PWLIB v1.10.0 and OpenH323 v1.18.0 and they are both built and installed properly. Has anyone come across this issue, or do I have to log a bug report at Digiums bug tracker? Thanks Bruce ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Steve Totaro wrote: I would suggest building it yourself (http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt http://www.readytechnology.co.uk/open/ipp-codecs/doc-svn6.txt). It is not that difficult and ensures that it should be compatible with your machine. Just a little work. Has anyone tried building this on Solaris, I just had a look at the link and it looks like the Intel IPP stuff is only released for Windows, Linux and MAC. And the v32 G729 codec from Digium does not load within asterisk on Solaris, sooo, the Solaris users out there dont have much support when it comes to G729 codecs, a real pity really, this stops some large scale roll-outs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.729 pre-compiled binaries and Asterisk 1.2.x.
Andrew Joakimsen wrote: They used to have solaris on the Digium FTP site but they seem to be gone now :( On the free codec site they have some complied with icc and others with gcc4 so I don't see why you can't get this working with gcc on solaris. Digium do still have the Solaris version of their codec on their download site and the following url: http://downloads.digium.com/pub/telephony/codec_g729/unsupported/ This codec is at version 32, whereas the latest is at 33. We tried this codec with valid licenses too, but the codec just fails to load in Asterisk. I was under the impression that the free codec required the Intel IPP libraries to be available on the system, or, statically linked into the codec. How would one build the codec if you could not link in a Solaris version of the IPP libraries, or am I missing something fundamental here? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.12 Release?
Hi All, I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2 weeks, a month? Thanks Bruce begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 on 1.4.10.1
I am experiencing the exact same problem on solaris, and we do have licenses purchased. I will log a bug at digium in the next day or two about my particular instance. Scott Moseman wrote: On 9/18/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: I hate to ask what may be a silly question, but have you purchased any G.729 licenses to use with the g.729 codec you downloaded? If you haven't registered codec_g729 yet, that would be why you are seeing this problem with codec_g729. My understanding was that it's not required for pass-through. PSTN Phone - g729 Gateway - Asterisk - g729 Phone Does this not equate to pass-through? Maybe I misunderstood? Thanks, Scott ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Bruce McAlister wrote: Moises Silva wrote: Open a bug in http://bugs.digium.com/ including all the information you provided here. OK, bug id 0010734 created: http://bugs.digium.com/view.php?id=10734 Interesting, this bug was deleted without any notification (that I'm aware of). I have subsequently logged another one 0010737: http://bugs.digium.com/view.php?id=10737 Also remember to read the bugs guidelines before openning the bug, this might be already reported. Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Hi Joshua, My bad, I thought that when you monitor the bug id that you would get an email when there were any additional notes added to the bug. I have been receiving email messages from [EMAIL PROTECTED], but, there is nothing in the message body, ie, an empty message. If it has indeed been fixed, then that's great! Is there a patch I can apply to 1.4.11 to get this working, or do I have to wait until 1.4.12 is released? Thanks Bruce Joshua Colp wrote: Hi Bruce, It was not deleted, it was closed automatically when the commit to 1.4 to fix it happened and then an additional note was added for the commit to trunk. If you didn't get an email detailing this as you should have I will test and pass it off to get fixed. Joshua Colp Software Developer Digium, Inc. - Original Message - From: Bruce McAlister [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Mon, 17 Sep 2007 09:19:06 -0300 Subject: Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault Bruce McAlister wrote: Moises Silva wrote: Open a bug in http://bugs.digium.com/ including all the information you provided here. OK, bug id 0010734 created: http://bugs.digium.com/view.php?id=10734 Interesting, this bug was deleted without any notification (that I'm aware of). I have subsequently logged another one 0010737: http://bugs.digium.com/view.php?id=10737 Also remember to read the bugs guidelines before openning the bug, this might be already reported. Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Bruce McAlister n:McAlister;Bruce org:Blueface Ltd adr:;;8 Clanwilliam Terrace;Dublin;Dublin;Dublin 2;Ireland email;internet:[EMAIL PROTECTED] tel;work:+353 1 524 2009 x-mozilla-html:FALSE url:http://www.blueface.ie version:2.1 end:vcard ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Moises Silva wrote: Open a bug in http://bugs.digium.com/ including all the information you provided here. OK, bug id 0010734 created: http://bugs.digium.com/view.php?id=10734 Also remember to read the bugs guidelines before openning the bug, this might be already reported. Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.11, res_features.so, SegFault
Hi All,
I have a really strange issue occuring where if I run show dialplan or
dialplan show or dialplan show parkedcalls, then asterisk dumps core.
It only appears to happen with contexts that are created within
res_features. I am able to display all my other dialplans, but, every
time I try to just do a normal dialplan show asterisk core dumps
(Segmentation Fault).
My environment is as follows:
Asterisk v 1.4.11
Solaris 10 update 3 (11/06), patched current
gcc v3.4.3
example console output
--
*CLI dialplan show
[ Context 'default' created by 'pbx_config' ]
Include ='demo'
[pbx_config]
[ Context 'page' created by 'pbx_config' ]
'_X.' = 1. Macro(page|SIP/${EXTEN})
[pbx_config]
[ Context 'demo' created by 'pbx_config' ]
SNIP
[ Context 'ael-dundi-e164-local' created by 'pbx_ael' ]
Include ='ael-dundi-e164-canonical'[pbx_ael]
Include ='ael-dundi-e164-customers'[pbx_ael]
Include ='ael-dundi-e164-via-pstn' [pbx_ael]
[ Context 'parkedcalls' created by 'res_features' ]
Segmentation Fault (core dumped)
--
Here are the traces:
--
(gdb) bt
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
#5 0x0808e1f0 in ast_cli_command (fd=1, s=0x0) at cli.c:1979
#6 0x08074167 in main (argc=135703622, argv=0x8047a5c) at asterisk.c:1388
--
--
(gdb) bt full
#0 0xfebe4d0c in strlen () from /lib/libc.so.1
No symbol table info available.
#1 0xfec3a386 in _ndoprnt () from /lib/libc.so.1
No symbol table info available.
#2 0xfec3d144 in snprintf () from /lib/libc.so.1
No symbol table info available.
#3 0x080babea in show_dialplan_helper (fd=1, context=0x0, exten=0x0,
dpc=0x8047840, rinclude=0x0, includecount=0,
includes=0x8047640) at pbx.c:6156
p = (struct ast_exten *) 0x81865b9
c = (struct ast_context *) 0x8186808
old_total_exten = 0
__PRETTY_FUNCTION__ = show_dialplan_helper
#4 0x080bb1b7 in handle_show_dialplan (fd=1, argc=2, argv=0x80478e0) at
pbx.c:3663
exten = 0x0
context = 0x0
counters = {total_context = 40, total_exten = 67, total_prio =
134, context_existence = 1, extension_existence = 1}
incstack = {0x1 Address 0x1 out of bounds, 0x0, 0x0, 0x80a8537
\215eô[^_ÉÃÇ\003\200, 0x0, 0x8120b95 logger.c,
0x37c Address 0x37c out of bounds, 0x811596c ast_verbose, 0x0,
0x0, 0x80476e8 `\025\025\b\001v\004\b\210\026\b,
0x80e3ee6 \203Ä0\215eô[^\211ø_ÉÃ\220©\200, 0x8047890
çrÄþp¯\027\b\002, 0x100 Address 0x100 out of bounds, 0x81a8830 Ò,
0x1b Address 0x1b out of bounds, 0xfec8c640 , 0x0, 0xfec8c640 ,
0xfeba2000 , 0xfec88000 \034\213\f, 0x0,
0x811d611 *CLI , 0x8151566 , 0x80476b4
[EMAIL PROTECTED]@\206µþøv\004\bDÑÃþ\027Ö\021\b\fw\004\bàv\004\b,
0xfec5a75a \203Ä\004\205Àt,Pè9], 0xfec8c640 , 0xfeba2000 ,
0xfec88000 \034\213\f, 0x80476d4 àv\004\b,
0xfec504ae \203Ä\0043É\213Eü\211\b_^[\213å]Ãj, 0xfec8c640 ,
0xfeb58640 , 0x80476f8 \230x\004\bÎ\006\a\b`\025\025\bÈ,
0xfec3d144 \203Ä\020\213L$\bÆ\001, 0x811d617 , 0x804770c ,
0x80476e0 Á, 0x0, 0x8163e88 ´\005\a\b\006,
0xc1 Address 0xc1 out of bounds, 0x8151566 , 0x8151560 *CLI ,
0x8047601 , 0x8163e88 ´\005\a\b\006, 0x0,
0x8047898 \002, 0x80706ce
\203Ä\020\215eô[^¸`\025\025\b_ÉÃPh\030Ö\021\bëÙ\211ö\213µ\204þÿÿ\205ötßj\024j\036j%\215uÈV1öèu8\a,
0x8151560 *CLI , 0xc8 Address 0xc8 out of bounds, 0x811d611
*CLI , 0x0, 0xfeba2000 , 0xfec88000 \034\213\f,
0x8047958 øy\004\b, 0x0, 0xfec8b800 , 0xfda18200 @\202¡ý, 0x0,
0xfec88000 \034\213\f, 0x4 Address 0x4 out of bounds,
0x0, 0x0, 0x804788c t\035\025\bçrÄþp¯\027\b\002, 0x8047888
àx\004\bt\035\025\bçrÄþp¯\027\b\002,
0x2f Address 0x2f out of bounds, 0x1 Address 0x1 out of bounds,
0x0, 0x1 Address 0x1 out of bounds, 0x0, 0x0,
0x43 Address 0x43 out of bounds, 0x0, 0x0, 0x0, 0xfbebdff8 , 0x0,
0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0,
0x5 Address 0x5 out of bounds, 0x5 Address 0x5 out of bounds, 0x0,
0x0,
0x812a83c No such command '%s' (type 'help' for help)\n, 0x81bedc3
s' ]\n, 0xfeba2000 , 0xfec88000 \034\213\f,
0x5f00796f Address 0x5f00796f out of bounds, 0xfec8c800 ,
0xfec8c800 , 0x80477d8 »ÔÃþ\020\225Èþ,
0xfec5a75a \203Ä\004\205Àt,Pè9], 0xfec8c800 , 0xfec8c800 ,
0x8047808 x\004\b2/Àþ\020\225Èþ,
0xfec3d4bb \203Ä\020\213L$\bÆ\001, 0xfec89510 , 0x0, 0xfec02e74
Re: [asterisk-users] G729 copy protection
Bruce McAlister wrote: Jason Parker wrote: Bruce, Please see my response to some of these questions on July 23rd. http://lists.digium.com/pipermail/asterisk-users/2007-July/192473.html I'm not entirely certain of what libraries we statically link in, but if you see any problems with the output of `ldd codec_g729.so`, those will of course need to be installed. Hi Jason, Thanks for the information, it appears then that you have built the codec on Solaris 10 (not OpenSolaris). Do you know if the build was done against the libraries that come with Solaris 10, or did you have newer libraries installed that would be required? An ldd codec_g729.so yields the following: ldd ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 libm.so.2 = /lib/libm.so.2 however an ldd -r codec_g729a.so yields the following: ldd -r ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 symbol not found: ast_cli_unregister (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: pow (./codec_g729a.so) symbol not found: cos (./codec_g729a.so) symbol not found: acos (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_module_register (./codec_g729a.so) symbol not found: connect (./codec_g729a.so) symbol not found: ast_cli_register (./codec_g729a.so) symbol not found: ast_cli (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_module_unregister (./codec_g729a.so) symbol not found: ast_trans_frameout(./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) libm.so.2 = /lib/libm.so.2 The strange thing here is that one would have thought that the following symbols would be part of the math library (libm), however, they are undefined/not found here: cos, acos, sqrt The rest
Re: [asterisk-users] G729 copy protection
Jason Parker wrote: Bruce, Please see my response to some of these questions on July 23rd. http://lists.digium.com/pipermail/asterisk-users/2007-July/192473.html I'm not entirely certain of what libraries we statically link in, but if you see any problems with the output of `ldd codec_g729.so`, those will of course need to be installed. Hi Jason, Thanks for the information, it appears then that you have built the codec on Solaris 10 (not OpenSolaris). Do you know if the build was done against the libraries that come with Solaris 10, or did you have newer libraries installed that would be required? An ldd codec_g729.so yields the following: ldd ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 libm.so.2 = /lib/libm.so.2 however an ldd -r codec_g729a.so yields the following: ldd -r ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 symbol not found: ast_cli_unregister (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: pow (./codec_g729a.so) symbol not found: cos (./codec_g729a.so) symbol not found: acos (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_module_register (./codec_g729a.so) symbol not found: connect (./codec_g729a.so) symbol not found: ast_cli_register (./codec_g729a.so) symbol not found: ast_cli (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_module_unregister (./codec_g729a.so) symbol not found: ast_trans_frameout(./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) libm.so.2 = /lib/libm.so.2 The strange thing here is that one would have thought that the following symbols would be part of the math library (libm), however, they are undefined/not found here: cos, acos, sqrt The rest of the symbols I can only assume are exported by the asterisk binary. I'm just wondering if there are any particular
Re: [asterisk-users] G729 copy protection
Jason Parker wrote: Bruce, Please see my response to some of these questions on July 23rd. http://lists.digium.com/pipermail/asterisk-users/2007-July/192473.html I'm not entirely certain of what libraries we statically link in, but if you see any problems with the output of `ldd codec_g729.so`, those will of course need to be installed. Hi Jason, Thanks for the information, it appears then that you have built the codec on Solaris 10 (not OpenSolaris). Do you know if the build was done against the libraries that come with Solaris 10, or did you have newer libraries installed that would be required? An ldd codec_g729.so yields the following: ldd ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 libm.so.2 = /lib/libm.so.2 however an ldd -r codec_g729a.so yields the following: ldd -r ./codec_g729a.so libgcc_s.so.1 = /usr/sfw/lib/libgcc_s.so.1 libc.so.1 = /lib/libc.so.1 symbol not found: ast_cli_unregister (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: ast_translator_activate (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: log10 (./codec_g729a.so) symbol not found: pow (./codec_g729a.so) symbol not found: cos (./codec_g729a.so) symbol not found: acos (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: sqrt (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_verbose (./codec_g729a.so) symbol not found: ast_module_register (./codec_g729a.so) symbol not found: connect (./codec_g729a.so) symbol not found: ast_cli_register (./codec_g729a.so) symbol not found: ast_cli (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: ast_config_AST_VAR_DIR (./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: socket(./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: __ast_register_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_unregister_translator (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_log (./codec_g729a.so) symbol not found: ast_module_unregister (./codec_g729a.so) symbol not found: ast_trans_frameout(./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: ast_translator_deactivate (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: shutdown (./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) symbol not found: option_verbose(./codec_g729a.so) libm.so.2 = /lib/libm.so.2 The strange thing here is that one would have thought that the following symbols would be part of the math library (libm), however, they are undefined/not found here: cos, acos, sqrt The rest of the symbols I can only assume are exported by the asterisk binary. I'm just wondering if there are any particular
Re: [asterisk-users] G729 copy protection
Bruce McAlister wrote: Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! Hi, Could anyone from Digium please shed some light on the build environment for the solaris 10 g729 codec? Was it build on Solaris or OpenSolaris? Are there any specific versions of libraries required? I'm still having this issue, and still cannot get the codec working. I've had a few tips/pointer from Joe at Solaris VoIP, but now we need to know a little more about the build environment to see if we can actually get this codec working. i have tried to run the codec with asterisk 1.2.17, 1.2.20. 1.2.24, 1.4.4, 1.4.10, 1.4.10.1 and 1.4.11, they all fail with the same messages. Asteris has been built on Solaris 10 Update 3 patched up as of friday last week. Our focaus now is to try and get the codec working with asterisk 1.4.x on Solaris 10. I've also tried i386, i586 - pentium4 32bit, opteron 32bit, on physical Opteron 285's and intel Xeon (Nacona's), all faile with the same message. The codec version is v32. This message comes up whether I have a valis g729 license from Digium or not, I have tried both. In either case, I would assume that codec would at least load, and a show g729 at the cli would work with and without a license. Has anyone been able to test this codec with asterisk? Any tips/suggestions would be greatly appreciated. Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Does anyone out there know what version/release of solaris the g729 (v32) codec is built on? Is it built on Solaris 10 GA, Solaris 10 U1, Solaris 10 U2, Solaris 10 U3, OpenSolaris (Nevada), which build? I'm just trying to find out if my problem with the codec may be due to a release difference, possibly a version of a library that the codec requires is not there? I will give it a test with the same version that the codec is built on just to see if it works. Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Jason Parker wrote: I'd wager that you're using the wrong path for the licenses. I believe the correct path is something like /var/opt/asterisk/licenses/ - it's whatever Asterisk has ast_config_AST_VAR_DIR set to, with /licenses/ at the end. The easiest way to tell, is to find the sounds dir (usually at /var/lib/asterisk/sounds/ on Linux), and go up a directory, and then from there create the licenses/ directory. When I register the codec using the register facility, it goes ahead a stores the license file in: /var/lib/asterisk/licenses When I check my asterisk.conf file the location astcarlibdir is as follows: astvarlibdir = /usr/local/asterisk/var/lib I have now tried to symlink the /var/lib/asterisk/licenses to /usr/local/asterisk/var/lib/licenses, and I have also tried to manually create the directory, with the same permissions as the original and copy the license file into the /usr/local/asterisk/var/lib/licenses directory. In each case the asterisk console still comes up with the following error when trying to initialize the codec on startup: Jul 20 08:40:01 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I'm beginning to think that the issue is not the license file, because the above error/warning occurs even when I have not registered the codec. Although, if anyone has more comments/suggestions, please feel free to offer them, I'm willing to try anything twice :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 copy protection
Hi All, I have been trying to get the Solaris version of the G729 codec to work with asterisk 1.2.17 and 1.2.22. However, I come up against the very same error every time I try to install it. Has anyone out there seen this error, taken from the asterisk console straight from startup: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i386)) Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 14:11:23 NOTICE[28243]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 14:11:23 WARNING[28243]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I have tried it with all the available v32 architectures and every one of them comes back with the very same error. I have done a search to see if anyone else came accross this error, there was one reference to the FreeBSD codec doing this, but apparently a new version of the codec came out that fixed it, the link for the FreeBSD error reference is here: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000699.html I'm at wits end at the moment, so, if anyone has any suggestions whatsoever, please feel free to put them forth, I'm willing to try anything at the moment. Oh, and the hardware we're running it with is: Solaris 10 Update 3 The CPU's are Opterons, but I have forced Solaris to boot in 32bit mode as the target server for the asterisk package I'm making is 32bit Solaris. Hopefully the i386 version of the codec should work on Opteron processors? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Jared Smith wrote: I'm probably asking the obvious here, but were you able to successfully register your codec with the Digium registration server? Hase your ethernet MAC address changed since you registered the codec? Hi Jared, I tried to run the register utility and I get as far as this (entering the Key-ID): # ./register Digium Product Registration - Version Copyright (C) 2004-2007, Digium, Inc. Use the '-l' option to see license information for software included in this program. Please select a product category. 1 - Digium Products 2 - Cepstral Products 0 - Quit Your Choice: 1 Please select a Product. 1 - Asterisk Business Edition 2 - G.729 Codec 3 - High Performance Echo Can 0 - Quit Your Choice: 2 Please enter the Key-ID: How do I know what the Key-ID is that it's asking for? If I run the asthostid app that accompanies the register utility and enter that ID in the above questions, then I get the following: The license key for this product should begin with G729! Am I doing something wrong? The README files dont quite explain how to get the Key-ID? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Jared Smith wrote: On Thu, 2007-07-19 at 16:20 +0100, Bruce McAlister wrote: Am I doing something wrong? The README files dont quite explain how to get the Key-ID? You should have received a key from Digium when you bought your license to use the G.729 codec. If you haven't yet bought any G.729 licenses, you can buy them from Digium's website at http://www.digium.com/en/products/voice/g729codec.php OK, I got hold of the G729 Key that was issued to us by digium recently and have now successfully registered the codec on the host. However, it still comes back with the following warning on the console after a restart: [codec_g729a.so] = (Annex A/B (floating point) G.729 Codec (optimized for i686)) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:465 load_module: G.729 transcoding module version 32, Copyright (C) 1999-2007 Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:469 load_module: This module is supplied under a commercial license granted by Digium, Inc. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:470 load_module: Please see the full license text supplied by the accompanying Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:471 load_module: register utility, or ask for a copy from Digium. Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:472 load_module: This product includes software developed by the OpenSSL Project Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:473 load_module: for use in the OpenSSL Toolkit. (http://www.openssl.org/) Jul 19 17:07:27 NOTICE[20591]: codec_g729.c:474 load_module: Copyright (C) 1998-2006 The OpenSSL Project Jul 19 17:07:27 WARNING[20591]: codec_g729.c:481 load_module: Failed to initialize G.729 copy protection! I can see the licence there (10 channel), but it looks like the codec does not want to inititalize properly. Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Darryl Dunkin wrote: Make sure there are no other files in the license path other than your valid license for this server. Hi, I have just checked this, and there is only the 1 license file in the /var/lib/asterisk/licenses directory. Is that what you meant? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 copy protection
Darryl Dunkin wrote: Correct, if you have multiple licenses in there (say a single storage location for a cluster of servers), it won't load. If you've tried other architectures of the codec and still had no luck, I'd say contact Digium support on it. Hmm, one caveat tho', these are the Solaris 10 32bit g729 codecs, and according to the FTP directory structure, are unsupported. This is why i emailed the list, hoping to bounce some ideas of you lot, to see if someone could help out :) Thanks for all the suggestions thus far, any more would be greatly appreciated. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 on Solaris SPARC/x86/x64 Codec
Jason Parker wrote: There already are x86 Solaris builds for codec_g729 - ftp.digium.com/pub/telephony/codec_g729/unsupported/ Excellent, thank you for the link. Now, next question, what happened to the SPARC versions, I do recall seeing them some time ago, but I cannot find them now :/ - Bruce McAlister [EMAIL PROTECTED] wrote: Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or unsupported). Does anyone know if Digium plan on releasing a SPARC *and/or* Intel/AMD G729 codec on Solaris? I would have thought with the availability of Solaris and Open Solaris that a little more enthusiasm would have been forthcomming in getting the codecs running on those environments? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 on Solaris SPARC/x86/x64 Codec
Hi All, Does anyone know what the current status is of the G729 codec on Solaris? According to the following link: http://www.asteriskvoipnews.com/asterisk_releases/_digium_g729_codec_now_available_forsolarissparc.html there is a version available for SPARC processor's. However, I have just had a quick look around Digium's FTP server and cannot seem to find these codecs (supported or unsupported). Does anyone know if Digium plan on releasing a SPARC *and/or* Intel/AMD G729 codec on Solaris? I would have thought with the availability of Solaris and Open Solaris that a little more enthusiasm would have been forthcomming in getting the codecs running on those environments? Thanks Bruce ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisk On Solaris 10
You would probably be better off getting support from the SolarisVoIP mailing list. Kapil Dhawan wrote: Any help is appreciated. Kapil Dhawan wrote: Hi List Whats the best way to run * on Solaris 10 with x86 architecture. I am following solarisvoip.com using svn, but came across issues like 1. app_lookupcnam compilation issue - Wrong format of ELF. Is this the correct way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk to record CDR in DB Oracle
Hi Tim, You will need an Oracle ODBC driver that Asterisk can use to connect to an oracle instance (local/remote). As far as I am aware, Oracle don't have unix/linux ODBC driver as of yet, but you can get one from EasySoft. They have an eval version you can try out to see if it works, have a look here: http://www.easysoft.com/products/data_access/odbc_oracle_driver/index.html I have tested this in the past and have managed to get asterisk to connect to a remote oracle instance using this driver, so it does work :) Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton Sent: 08 May 2007 09:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to record CDR in DB Oracle On 7 May 2007, at 17:27, Florian Overkamp wrote: Hi Everton, Everton Goularth wrote: I had success to do my asterisk to record CDR in a databese MYSQL... Now, I need to do it to record CDR in Oracle... Does Anybody knows how to do this?? Every hints are welcome There is no native Oracle driver available to my knowledge, but if you can install an ODBC driver for Oracle, Asterisk will happily use that. If anyone gets this to work, especially against an oracle instance on a separate machine, I'd love to know how you did it. I spent a day or so failing to get it to work, then gave up and had a perl script written that regularly posts the new CDR records to oracle over http(s). Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no)
Hi Gavin, I don't know if this will help, but can you check to see if you have libtool installed? I had a similar issue with unixodbc, and once I installed libtool, it rectified the issue. Once libtool is installed, re-run configure and it should hopefully work. Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: 05 May 2007 22:31 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checkingfor PQexec in -lpq... no) Dear All, Why does my configure fail like so: checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config checking for PQexec in -lpq... no configure: *** configure: *** The PostgreSQL installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** including --without-postgres Configure options are: env CC=/usr/local/bin/gcc ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 configure has found pg_config, what more does it need? I even tried: env CC=/usr/local/bin/gcc CPPFLAGS=-I/usr/local/pgsql/8.2.4/include \ LDFLAGS=-L/usr/local/pgsql/8.2.4/lib \ LD_LIBRARY_PATH=/usr/local/pgsql/8.2.4/lib ./configure --with-ssl=/usr/local/ssl --with-postgres=/usr/local/pgsql/8.2.4 Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help
Hi Remco Post,
Thank you for the tip. I have verified that the permissions are correct for
the table and procedure.
However, I think I may have got to the bottom of the issue now.
What look like was happening is that asterisk was trying to delete any
matching row prior to an insert operation. So, when a user left a message,
for example, message 1, asterisk would attempt to delete message 1 before
inserting it for that user. However, message 1 does not exist at that time
and thus the ODBC driver returns SQL_NO DATA.
The same happens when a user checks their voicemail, once an message has
been listened to asterisk moves it to the Old directory, that way it can
distinguish between new/old messages. When a user listens to the voicemail,
asterisk then tries to insert the message into the Old tree, prior to
doing the insert, asterisk tries to delete the last available message
returned from a select count(*) operation. This message does not exist and
the odbc driver returns SQL_NO_DATA.
The delete_file function in app_voicemail.c does not accommodate for this
return code SQL_NO_DATA and thus spits out the warning on the console.
I thus changed the following condition in function delete_file in
app_voicemail.c from:
if ((res != SQL_SUCCESS) (res != SQL_SUCCESS_WITH_INFO)) {
ast_log(LOG_WARNING, SQL Execute error!\n[%s]\n\n, sql);
SQLFreeHandle (SQL_HANDLE_STMT, stmt);
ast_odbc_release_obj(obj);
goto yuck;
}
To:
if ((res != SQL_SUCCESS) (res != SQL_SUCCESS_WITH_INFO) (res !=
SQL_NO_DATA)) {
ast_log(LOG_WARNING, SQL Execute error!\n[%s]\n\n, sql);
SQLFreeHandle (SQL_HANDLE_STMT, stmt);
ast_odbc_release_obj(obj);
goto yuck;
}
This seems to have fixed the problem.
Thanks
Bruce
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RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help
]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] [May 1 07:50:47] WARNING[30791]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] [May 1 07:50:47] WARNING[30791]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/bruce.mcalister-096da288, ) in new stack == Spawn extension (base-out, 171, 4) exited non-zero on 'SIP/bruce.mcalister-096da288' Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: 01 May 2007 00:06 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help Importance: High Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, Caller VoiceMail Extension = 3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3031/temp' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-intro' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /usr/local/asterisk/var/spool/voicemail/users/3031/tmp/hGkNG0 format: wav49, 0x90539c8 -- User ended message by pressing # -- SIP/bruce.mcalister-09051118 Playing 'auth-thankyou' (language 'en') [Apr 30 23:56:03] WARNING[30123]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3031/INBOX/msg0002.txt': Found Length is 20600 -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/bruce.mcalister-09051118, ) in new stack == Spawn extension (base-out, 170, 4) exited non-zero on 'SIP/bruce.mcalister-09051118' --- Here is an example when someone listens to their voicemail messages without deleting any: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, voicemail extension=3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing 'vm-password' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-youhave' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/19' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-and' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/20' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-Old' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-onefor' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-first' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-message
RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help
Hi All, I tried to send this email this morning, but I think it has been moderated due to size issue's, so I'll resend it again in 3 parts: PART 2 Database Table Definition (taken from asterisk readme's) CREATE FUNCTION loin (cstring) RETURNS lo AS 'oidin' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION loout (lo) RETURNS cstring AS 'oidout' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION lorecv (internal) RETURNS lo AS 'oidrecv' LANGUAGE internal IMMUTABLE STRICT; CREATE FUNCTION losend (lo) RETURNS bytea AS 'oidrecv' LANGUAGE internal IMMUTABLE STRICT; CREATE TYPE lo ( INPUT = loin, OUTPUT = loout, RECEIVE = lorecv, SEND = losend, INTERNALLENGTH = 4, PASSEDBYVALUE ); CREATE CAST (lo AS oid) WITHOUT FUNCTION AS IMPLICIT; CREATE CAST (oid AS lo) WITHOUT FUNCTION AS IMPLICIT; CREATE TRUSTED LANGUAGE plpgsql; CREATE FUNCTION vm_lo_cleanup() RETURNS trigger AS $$ declare msgcount INTEGER; begin --raise notice 'Starting lo_cleanup function for large object with oid %',old.recording; --If it is an update action but the BLOB (lo) field was not changed, dont do anything if (TG_OP = 'UPDATE') then if ((old.recording = new.recording) or (old.recording is NULL)) then raise notice 'Not cleaning up the large object table, as recording has not changed'; return new; end if; end if; if (old.recording IS NOT NULL) then SELECT INTO msgcount COUNT(*) AS COUNT FROM voicemailmessages WHERE recording = old.recording; if (msgcount 0) then raise notice 'Not deleting record from the large object table, as object is still referenced'; return new; else perform lo_unlink(old.recording); if found then raise notice 'Cleaning up the large object table'; return new; else raise exception 'Failed to cleanup the large object table'; return old; end if; end if; else raise notice 'No need to cleanup the large object table, no recording on old row'; return new; end if; end$$ LANGUAGE plpgsql; CREATE TABLE public.voicemailmessages ( id BIGSERIAL PRIMARY KEY USING INDEX TABLESPACE bf_service_idx, msgnum SMALLINT NOT NULL DEFAULT 0, dir VARCHAR(80) DEFAULT '', context VARCHAR(80) DEFAULT '', macrocontext VARCHAR(80) DEFAULT '', callerid VARCHAR(40) DEFAULT '', origtime VARCHAR(40) DEFAULT '', duration VARCHAR(20) DEFAULT '', recordingloDEFAULT NULL, mailboxuser VARCHAR(80) DEFAULT '', mailboxcontext VARCHAR(80) DEFAULT '' ) WITHOUT OIDS; CREATE INDEX idx_voicemailmessages_msgnum_dir ON voicemailmessages(msgnum,dir) TABLESPACE bf_service_idx; CREATE TRIGGER trg_vm_cleanup AFTER DELETE OR UPDATE ON voicemailmessages FOR EACH ROW EXECUTE PROCEDURE vm_lo_cleanup(); Thanks Bruce -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: 01 May 2007 00:06 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help Importance: High Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, Caller VoiceMail Extension = 3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3031/temp
[asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help
Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] This warning occurs whenever a user leaves a message for an extension. It also occurs when someone dials in to listen to their messages when they hang up. These messages do actually exist within the database, and asterisk does extract them from the database when playing back or recording messages. Here is an example when someone leaves a message for someone: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, Caller VoiceMail Extension = 3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3031/temp' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-intro' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /usr/local/asterisk/var/spool/voicemail/users/3031/tmp/hGkNG0 format: wav49, 0x90539c8 -- User ended message by pressing # -- SIP/bruce.mcalister-09051118 Playing 'auth-thankyou' (language 'en') [Apr 30 23:56:03] WARNING[30123]: app_voicemail.c:1280 delete_file: SQL Execute error! [DELETE FROM voicemailmessages WHERE dir=? AND msgnum=?] == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3031/INBOX/msg0002.txt': Found Length is 20600 -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/bruce.mcalister-09051118, ) in new stack == Spawn extension (base-out, 170, 4) exited non-zero on 'SIP/bruce.mcalister-09051118' --- Here is an example when someone listens to their voicemail messages without deleting any: --- -- Executing [EMAIL PROTECTED]:1] AGI(SIP/bruce.mcalister-09051118, agi://10.7.0.136:4573?app=getvoicemailexten) in new stack -- AGI Script agi://10.7.0.136:4573?app=getvoicemailexten completed, returning 0 -- Executing [EMAIL PROTECTED]:2] NoOp(SIP/bruce.mcalister-09051118, voicemail extension=3031) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMailMain(SIP/bruce.mcalister-09051118, [EMAIL PROTECTED]) in new stack -- SIP/bruce.mcalister-09051118 Playing 'vm-password' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-youhave' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/19' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-and' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/20' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-Old' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-onefor' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-INBOX' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-messages' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-first' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-message' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg.txt': Found -- SIP/bruce.mcalister-09051118 Playing 'vm-unknown-caller' (language 'en') -- SIP/bruce.mcalister-09051118 Playing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'vm-message' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/2' (language 'en') == Parsing '/usr/local/asterisk/var/spool/voicemail/users/3204/INBOX/msg0001.txt': Found -- SIP/bruce.mcalister-09051118 Playing 'vm-from-phonenumber' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/4' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/4' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/2' (language 'en') -- SIP/bruce.mcalister-09051118 Playing 'digits/0' (language 'en')
