Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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] asterisk realtime database structure
Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- _ -- 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] Originate call from cli does not work for SIP line...
Carlos Chavez писал 02.08.2012 20:24: originate SIP/protel-out/0445540881644 application playback tt-monkeys And if you route it via local context? For example, originate local/outgoing/0445540881644 application playback tt-monkeys where outgoing is your context which knows how to route 0445 prefix -- With Best Regards Mikhail Lischuk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime database structure
If you check the contrib/realtime direco 2012/8/3 Daniel-Constantin Mierla mico...@gmail.com Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,** +MySQL+table+structurehttps://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storagehttps://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.** 1.x-asterisk-1.6.2-astdbhttp://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk realtime database structure
If you check the contrib/realtime/mysql directory in the source tree, you'll find scripts for almost all the tables. Leandro 2012/8/3 Daniel-Constantin Mierla mico...@gmail.com Hello, I was wondering if there is a tool that can create the realtime database structure for latest Asterisk version or a web resource/file containing the sql scripts. Hope I haven't missed obvious things, I had no luck searching on the web, in the wiki I found few pages with bits of sql or table structures, like: https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,** +MySQL+table+structurehttps://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storagehttps://wiki.asterisk.org/wiki/display/AST/ODBC+Voicemail+Storage I have several table structures from the Asterisk 1.6, I dug for them in the code or found on the web when I wrote the tutorial about integration with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.** 1.x-asterisk-1.6.2-astdbhttp://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb), but hopefully now it is an easy way to get the db structure. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/**micondahttp://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
It means ... Asterisk don't make any IVR at realtime. It just fire Mysql/Odbc query and get *app and appdata.* On Fri, Aug 3, 2012 at 11:50 AM, Leandro Dardini ldard...@gmail.com wrote: It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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] MixMonitor creating file on non-bridged calls with option b
Hi Ikka,
I'm using asterisk 10.0.0 10.5.2 10.6.0 10.6.1 and they all leave
empty files.
Hope somebody can help.
Regads
Thorben G. Jensen
2012/8/3 Ikka Vertika (Mitra Kreasindo) ikka.vert...@mitrakreasindo.com
HI,
** **
What version is your asterisk ? I’m using 10.2, 10.4. 10.6, there all have
the same problem.
I had read once, that there’s a bug in asterisk 10.4, and fixed with
patch. But if it fixed, why 10.61 still have the same problem ?
I tried to patch it (ver 10.6x), but the patching process was
unsuccessfull. Some error occurs. The problem is still there…
** **
Regards,
Ikka (Jakarta, Indonesia)
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Thorben Jensen
*Sent:* Saturday, July 28, 2012 1:58 PM
*To:* asterisk-users
*Subject:* [asterisk-users] MixMonitor creating file on non-bridged calls
with option b
** **
I am using MixMonitor to record calls and I have set the b option as I
don't want to get files for non-bridged calls.
** **
Mixmonitor always creates a file with 0 bytes even when the call is not
bridged. Is it possible to avoid this somehow?
** **
This is what I do:
** **
Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});**
**
MixMonitor(${CALLFILENAME},b);
** **
Regards
Thorben
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
On Friday 03 August 2012, virendra bhati wrote: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 Well, what you are wanting would be mathematically impossible! A simple text file is read sequentially, so you can infer priorities from the position of lines within the file. A database is truly random-access, and so doesn't have any such natural ordering. You have to indicate the sequence explicitly within a field in each record, and refer to this with an ORDER BY clause. -- AJS Answers come *after* questions. -- _ -- 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] Voicemail full.
Hi I've made a call to our elastix server and the call was redirected to the voicemail, which the user should leave a message. Intead recording the call the service returned a message like Sorry, but the user's mailbox can't accept more messages. I'm a little lost in the configs here, what parameter should I edit to increase the mailbox capacity or what steps I take to 'clean' the mailbox? -- Att.* *** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Leandro I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is many developers when pushing to put features in they don't put on their designers hat and think out side the box first.Heaven knows I have been guilty of this one over the years and had to go back and refactor. It is not so reasonable to think that this limitation has to exist developers have been putting order by fields in db driven systems for years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) so they can add specialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field. As it sits now how do you do n, i, h or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this ability available when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it . Thanks Bryant Zimmerman (ZK Tech Inc.) From: Leandro Dardini ldard...@gmail.com Sent: Friday, August 03, 2012 2:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used 'n' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Thorben Jensen wrote:
From: Thorben Jensen i...@thorben.dk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, August 3, 2012 4:15:58 AM
Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls
with option b
Hi Ikka,
I'm using asterisk 10.0.0 10.5.2 10.6.0 10.6.1 and they all
leave empty files.
Hope somebody can help.
Regads
Thorben G. Jensen
2012/8/3 Ikka Vertika (Mitra Kreasindo)
ikka.vert...@mitrakreasindo.com
HI,
What version is your asterisk ? I’m using 10.2, 10.4. 10.6, there all
have the same problem.
I had read once, that there’s a bug in asterisk 10.4, and fixed with
patch. But if it fixed, why 10.61 still have the same problem ?
I tried to patch it (ver 10.6x), but the patching process was
unsuccessfull. Some error occurs. The problem is still there…
Regards,
Ikka (Jakarta, Indonesia)
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com ] On Behalf Of Thorben
Jensen
Sent: Saturday, July 28, 2012 1:58 PM
To: asterisk-users
Subject: [asterisk-users] MixMonitor creating file on non-bridged
calls with option b
I am using MixMonitor to record calls and I have set the b option
as I don't want to get files for non-bridged calls.
Mixmonitor always creates a file with 0 bytes even when the call is
not bridged. Is it possible to avoid this somehow?
This is what I do:
Set(CALLFILENAME=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(num)});
MixMonitor(${CALLFILENAME},b);
Regards
Thorben
--
MixMonitor creates the file before it starts recording. The b option simply
waits until the bridge event to take audio frames and add them to the stream. I
don't really see why this is a problem. It isn't like you are going to run out
of space for zero length files, and more to the point if you really dislike the
clutter generated by having zero length files you could always run a script
after recording to check if you added a zero length file and remove it if you
did.
Are there any actual bug reports (in JIRA) you could reference though? If not,
please create one and we'll look into it.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but if I set it to wav it will create a 64 byte file. Off course I can delete the files in a script, but I thought I was doing something wrong. I have not made a bug report. Regards Thorben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
I am kissing every inch of land where each one of the asterisk's developer is putting his feet. In the last 10 years I have worked thanks to the availability of the asterisk code. Most of my income was possible just thanks to asterisk, so I am pretty biased when trying to evaluate if the asterisk code is good or not. You can understand if I love the way asterisk has been coded. Nevertheless things can be better and they can be better thanks to you. Asterisk is open source and Mark is a very kind person when you submit patches, so put your ideas in new code and send to him. If you don't know how to code, hire some developer and have him to code your view of a better RT code. If it will be accepted by the core developer, all us will be happy. if it will not accepted, you'll be happy with you own personal branch. I run for a small period of time my personal asterisk tree because the italian telephony system is flawed and clients want services not suitable for the general asterisk audience, so there is nothing to worry to have your personal asterisk code. Leandro PS I think your idea of extension RT can be accomplished with some triggers and replacing the extension table with a view on your own n-enabled extension table 2012/8/3 Bryant Zimmerman brya...@zktech.com Leandro I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is many developers when pushing to put features in they don't put on their designers hat and think out side the box first.Heaven knows I have been guilty of this one over the years and had to go back and refactor. It is not so reasonable to think that this limitation has to exist developers have been putting order by fields in db driven systems for years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) so they can add specialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field. As it sits now how do you do n, i, h or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this ability available when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it . Thanks Bryant Zimmerman (ZK Tech Inc.) -- *From*: Leandro Dardini ldard...@gmail.com *Sent*: Friday, August 03, 2012 2:18 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, virendra bhati virbh...@gmail.com ha scritto: Hi Team, I want to used *'n*' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
I have read an earlier posts in this forum that said it was a bug in asterisk 10.4 and there's a patch to fix it. So it already reported as bug. But i use asterisk 10.61 and that bug is still there, unfixed. We upgrade to the newest version for the new feature that we can use, and also hoping for fixed bugs. But we absolutly not hoping to get an extra job fixiing the bug that has supposely has been fix... Sent from Samsung Mobile Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but if I set it to wav it will create a 64 byte file. Off course I can delete the files in a script, but I thought I was doing something wrong. I have not made a bug report. Regards Thorben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Not to bash on the developer who did this I get that we don't always think out side the box all the timeYou can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers. Sucks? yes. Is the realtime limitation going to stop me from doing what I want? no way.Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: Leandro Dardini ldard...@gmail.com Date: Fri, August 03, 2012 10:11 am To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com I am kissing every inch of land where each one of the asterisk's developer is putting his feet. In the last 10 years I have worked thanks to the availability of the asterisk code. Most of my income was possible just thanks to asterisk, so I am pretty biased when trying to evaluate if the asterisk code is good or not. You can understand if I "love" the way asterisk has been coded. Nevertheless things can be better and they can be better thanks to you. Asterisk is open source and Mark is a very kind person when you submit patches, so put your ideas in new code and send to him. If you don't know how to code, hire some developer and have him to code your view of a better RT code. If it will be accepted by the core developer, all us will be happy. if it will not accepted, you'll be happy with you own personal branch. I run for a small period of time my personal asterisk tree because the italian telephony system is flawed and clients want services not suitable for the general asterisk audience, so there is nothing to worry to have your personal asterisk code. LeandroPSI think your idea of extension RT can be accomplished with some triggers and replacing the extension table with a view on your own n-enabled extension table 2012/8/3 Bryant Zimmerman brya...@zktech.com Leandro I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is manydevelopers when pushingto put featuresin they don't put on their designers hat and think out side the box first.Heavenknows I have been guilty of this oneover the years and had togo back and refactor. It is not so reasonable to think that this limitation has to existdevelopers have been putting order by fields in dbdriven systemsfor years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) sothey can addspecialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field. As it sits now how do you do n, i, h or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this abilityavailable when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it . Thanks BryantZimmerman (ZK Tech Inc.) From: "Leandro Dardini" ldard...@gmail.com Sent: Friday, August 03, 2012 2:18 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority It is reasonable 'n' is not usable as priority number. How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries. Leandro Il giorno 03/ago/2012 06:27, "virendra bhati" virbh...@gmail.com ha scritto: Hi Team, I want to used 'n' as priority in asterisk realtime but asterisk don't support n as next priority I am using Asterisk 1.4.41 -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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
Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Ikka.vertika wrote: From: Ikka.vertika ikka.vert...@mitrakreasindo.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, August 3, 2012 9:20:03 AM Subject: Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b I have read an earlier posts in this forum that said it was a bug in asterisk 10.4 and there's a patch to fix it. So it already reported as bug. But i use asterisk 10.61 and that bug is still there, unfixed. We upgrade to the newest version for the new feature that we can use, and also hoping for fixed bugs. But we absolutly not hoping to get an extra job fixiing the bug that has supposely has been fix... Mailing lists (and particularly the asterisk-users list) aren't the proper place for bugs to be reported. If an issue doesn't get reported to JIRA, it doesn't go into the evaluation process and at the end of the day, developers are less likely to notice them and even more unlikely to be assigned to them. I don't doubt that someone noticed the topic and created a patch for it, but that doesn't mean the patch was applied to any build in SVN. In fact, our license agreement for writing patches can't be verified if the patch isn't uploaded to JIRA (and maybe reviewboard I think). Without that we can't even legally use the patch. Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but if I set it to wav it will create a 64 byte file. Off course I can delete the files in a script, but I thought I was doing something wrong. I have not made a bug report. Regards Thorben That's totally different. When you invoke MixMonitor before a call, you are setting up MixMonitor immediately. The call itself is incidental to that operation. When you use the MixMonitor option with Queue, the Queue Application is responsible for starting MixMonitor, so it won't start the MixMonitor until the call is starting. What is going into the empty wav file by the way is a header describing the contents of the file... I'm not an expert on the format, but I would assume it would be things like sample rate, number of channels, etc. For all Asterisk recordings through MixMonitor, this should be the same, so you can use that assumption for creating your deletion script as well. There are probably also utilities available which could be used to read the length of a wav file so that you can delete them based on that. If you think it's really a problem though, feel free to submit a JIRA report. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Jonathan Rose wrote: Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw, but if I set it to wav it will create a 64 byte file. Off course I can delete the files in a script, but I thought I was doing something wrong. I have not made a bug report. Regards Thorben That's totally different. When you invoke MixMonitor before a call, you are setting up MixMonitor immediately. The call itself is incidental to that operation. When you use the MixMonitor option with Queue, the Queue Application is responsible for starting MixMonitor, so it won't start the MixMonitor until the call is starting. I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start MixMonitor through dialplan features or something rather than actually doing an automatic MixMonitor. Could you clarify what you mean by 'the MixMonitor option with Queue'? Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
On Friday 03 August 2012, C. Savinovich wrote: Not to bash on the developer who did this I get that we don't always think out side the box all the time You can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers. Sucks? yes. Is the realtime limitation going to stop me from doing what I want? no way. That is the sort of thing that might actually be worth submitting upstream. There must be loads of dialplans out there that use same, n and labels all over the place. The only reason mine don't, is because I've been using Asterisk since before these features were introduced and I got used to the old ways. -- AJS Answers come *after* questions. -- _ -- 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] No audio playing back voicemail from odbc
Well, it gets even stranger I've installed version 10.2.1, instead of 10.7.1, and copied the configuration files from another identical server that is running 10.2.1 is working correctly. I STILL can't get voicemail to play back. I can hear the password prompts Theses are, what I think to be, the relevant settings in voicemail.conf: ;minmessage=3 maxsilence=10 silencethreshold=128 When I set silencethresholdo either 500 or 64, I still didn't hear anything. (But I did hear several seconds of actual silence. The .wav file contains nothing but silence. So, fiddling with the silencethreshold in both directions, doesn't seem to change the symptoms. Where else should I look? TIA Mike. On Saturday 28 July 2012 6:45:15 pm Matthew Jordan wrote: - Original Message - From: Support mdi...@diehlnet.com To: asterisk-users@lists.digium.com Cc: Matthew Jordan mjor...@digium.com Sent: Saturday, July 28, 2012 2:38:09 PM Subject: Re: [asterisk-users] No audio playing back voicemail from odbc CLI core show translation paths slin --- Translation paths SRC Codec slin sample rate 8000 --- slin To g723 : No Translation Path slin To gsm : (slin)-(gsm) slin To ulaw : (slin)-(ulaw) I'm using the ulaw audio codec and wav for storage, so this SHOULD work If you're getting a duration of 0, I have to wonder if your silencethreshold is playing a factor here. Asterisk may be treating the entire recording as silence. What happens if you set the maxsilence to some valid integer value? Does it end the voicemail messages while you're leaving it? If so, that might indicate that it isn't detecting any sound. If you set minduration/maxsilence and Asterisk starts killing recordings and not saving files, that will also tell you if Asterisk believes the recordings are mostly silence. I don't, but this configuration worked before I upgraded from 1.6.x to 10.x. I should have mentioned that this was part of an upgrade, but it was late, and I was tired. So, is there something I'm missing? I'm not sure. I'd be curious to see your voicemail.conf. File storage is the only mechanism to have video voicemail (with audio) at this time. Is there any interest in fixing this situation? It doesn't seem like it would be too difficult. I wouldn't mind helping if there is an effort already underway. There has been some interest expressed from users, but no development plans have been put into place for this feature. There are a couple of reasons for that: while it would be possible to have multiple formats stored in ODBC/IMAP backends, that doesn't solve all of the problems with associating an audio file with a video file. For example, some soft phones allow you to start the video media stream after the audio media stream has already begun. This will work fine during the video call; however, if the video/audio is stored as a voicemail message, Asterisk has no way to associate the beginning of the video file with some arbitrary point in the audio file. Hence, when the video message is played back, the video will be out of sync with the audio - both are played back starting at the same time, but the soft phone didn't start sending the video at the beginning of the audio. The solution to this would be to store the audio/video as a single file in a media container (such as Matroska). Not only does this solve the audio/video sync issue, but now you don't need to store more then a single file in a storage backend. Unfortunately, this is an extremely non-trivial effort. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
AJ, You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle?Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: A J Stiles asterisk_l...@earthshod.co.uk Date: Fri, August 03, 2012 11:45 am To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com On Friday 03 August 2012, C. Savinovich wrote: Not to bash on the developer who did this I get that we don't always think out side the box all the time You can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?. All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers. Sucks? yes. Is the realtime limitation going to stop me from doing what I want? no way. That is the sort of thing that might actually be worth submitting upstream. There must be loads of dialplans out there that use "same", "n" and labels all over the place. The only reason mine don't, is because I've been using Asterisk since before these features were introduced and I got used to the old ways. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
On Fri, Aug 3, 2012 at 9:35 AM, C. Savinovich c.savinov...@itntelecom.comwrote: AJ, You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? Some might say that you should never do that. I mean, not in one context anyway, where the line numbers matter. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
The structured way of thinking (that cursed philosophy where you write 100 lines of code to avoid a goto) says you should have your contexts small enough to not need ns. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Friday, August 03, 2012 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority On Fri, Aug 3, 2012 at 9:35 AM, C. Savinovich c.savinov...@itntelecom.com wrote: AJ, You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? Some might say that you should never do that. I mean, not in one context anyway, where the line numbers matter. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] MixMonitor creating file on non-bridged calls with option b
I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start MixMonitor through dialplan features or something rather than actually doing an automatic MixMonitor. Could you clarify what you mean by 'the MixMonitor option with Queue'? See this web page under the heading Monitor Format http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Kind Regards Thorben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] MixMonitor creating file on non-bridged calls with option b
MixMonitor creates the file before it starts recording. The b option simply waits until the bridge event to take audio frames and add them to the stream. I don't really see why this is a problem. It isn't like you are going to run out of space for zero length files, and more to the point if you really dislike the clutter generated by having zero length files you could always run a script after recording to check if you added a zero length file and remove it if you did. Are there any actual bug reports (in JIRA) you could reference though? If not, please create one and we'll look into it. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 I just found that this has already been reported to JIRA: https://issues.asterisk.org/jira/browse/ASTERISK-20156 Hope this helps. Regards Thorben -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
No, numbers have to be in sequence. Leandro I am typing from my mobile phone... Il giorno 03/ago/2012 20:28, Raj Mathur (राज माथुर) r...@linux-delhi.org ha scritto: On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Using n with labels is what most people do. A dialplan isn't javascript, you don't need two hundred 3 line functions. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: Friday, August 03, 2012 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; r...@linux-delhi.org Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority No, numbers have to be in sequence. Leandro I am typing from my mobile phone... Il giorno 03/ago/2012 20:28, Raj Mathur (राज माथुर) r...@linux-delhi.org ha scritto: On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b
Thorben Jensen wrote: I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start MixMonitor through dialplan features or something rather than actually doing an automatic MixMonitor. Could you clarify what you mean by 'the MixMonitor option with Queue'? See this web page under the heading Monitor Format http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf Kind Regards Thorben -- Ok, in that case what I said before stands. In this case, app_queue is responsible for starting the mixmonitor application, so it simply doesn't do so until the calls are connected. I checked the issue report you mentioned in the other message, so that seems to be in order. You might want to chime in on that issue as well since issues with more activity tend to be prioritized over others. That said, I do feel the severity of this is pretty minor and it might not even really be considered a bug at all. I think getting the file created and ready before recording might even be considered an optimization. Even then though, getting rid of it if we never start recording is pretty simple. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority
Basic?... no man, I am kid!Christian SavinovichVoIP Telephony Consultant646-982-3572 Original Message Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority From: "Raj Mathur (राज माथुर)" r...@linux-delhi.org Date: Fri, August 03, 2012 2:21 pm To: asterisk-users@lists.digium.com On Friday 03 Aug 2012, C. Savinovich wrote: You don't use 'n's in your dialplan?, you number it yourself? man, what if you have a 300 line dialplan and then you decide to insert a new line in the middle? If you ever used BASIC you'd remember the trick is to increment line numbers (priorities) by 10. I presume a dialplan would work fine even if the priorities aren't sequential, as long as they're increasing monotonically. Could someone confirm? Having said that, I use n with abandon. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail full.
Looking at my voicemail.conf I note this snippet: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is . ;maxmsg=100 So in my case max messages is . Assuming you are storing your message in files instead of a Database, you would look at /var/spool/asterisk/voicemail/default/xxx/INBOX where xxx is replaced by the mailbox number. As I understand it, you would have to renumber the files from msg to the maximum number and there are 2 to 4 files per message depending on your setup. I think there was a thread on this in July 2012. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis H. Forchesatto Sent: Friday, August 03, 2012 7:10 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Voicemail full. Hi I've made a call to our elastix server and the call was redirected to the voicemail, which the user should leave a message. Intead recording the call the service returned a message like Sorry, but the user's mailbox can't accept more messages. I'm a little lost in the configs here, what parameter should I edit to increase the mailbox capacity or what steps I take to 'clean' the mailbox? -- Att. -- _ -- 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] Unplanned Asterisk community service outage
Starting around twenty minutes ago, we have begin having some network difficulties affecting community services. The issue is being investigated and will be resolved as promptly as possible. These technical issues appear to be causing an outage to at least the following services: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unplanned Asterisk community service outage
On 8/3/2012 3:24 PM, Asterisk Development Team wrote: Starting around twenty minutes ago, we have begin having some network difficulties affecting community services. The issue is being investigated and will be resolved as promptly as possible. These technical issues appear to be causing an outage to at least the following services: bamboo.asterisk.org code.asterisk.org downloads.digium.com downloads.asterisk.org git.asterisk.org issues.asterisk.org packages.asterisk.org reviewboard.asterisk.org svn.asterisk.org svnview.digium.com wiki.asterisk.org The issues appear to have been resolved for the moment. Those working on the problem are still investigating the root cause, so there is a small chance the issue could re-occur at an unknown time. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scheduled Maintenance for Asterisk Project community services
On Saturday, August 4th, 2012, the Asterisk community services listed below will be undergoing maintenance (software upgrades and updates). The services will be shut down at approximately 9:00 PM CDT (3:00 AM August 5th UTC), and will return no later than 10:00 PM CDT. We apologize in advance for any inconvenience this may cause. The affected services are: issues.asterisk.org/jira -- _ -- 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] Voicemail full.
Un-top-posting... On Fri, 3 Aug 2012, Luis H. Forchesatto wrote: I've made a call to our elastix server and the call was redirected to the voicemail, which the user should leave a message. Intead recording the call the service returned a message like Sorry, but the user's mailbox can't accept more messages. I'm a little lost in the configs here, what parameter should I edit to increase the mailbox capacity or what steps I take to 'clean' the mailbox? On Fri, 3 Aug 2012, Danny Nicholas wrote: Looking at my voicemail.conf I note this snippet: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is . ;maxmsg=100 So in my case max messages is . (After a quick glance at the source...) If not specified, the limit would be MAXMSG (100) not MAXMSGLIMIT (). -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Talk detection during call
I am looking for ways to detect if there is some person talking on the other side of the line and trigger some events based on that.. is there any possible way through which this could be done in asterisk ? Thanks, Sathiish -- _ -- 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] Talk detection during call
Look for AMD (Answering machine detection). On Fri, 2012-08-03 at 14:42 -0700, sathiish kumar wrote: I am looking for ways to detect if there is some person talking on the other side of the line and trigger some events based on that.. is there any possible way through which this could be done in asterisk ? Thanks, Sathiish -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Suggestion of Server Specifications for Asterisk
What the minimum Server Specifications do I need to run 200 concurrent channels at the time with .WAV recording (MixMonitor)? It will be connected via VOIP sip account. Codec will be ulaw. Which UK dedicated server provider do you recommend and how much bandwidth do I need? 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] app_swift 3 and asterisk 1.8.13.0 fails with undefined symbol: swift_port_close
Solving my own issue: The order of linking was incorrect. Change the Makefile as follows to solve this issue. diff --git a/Makefile b/Makefile index dec892d..90d79df 100644 --- a/Makefile +++ b/Makefile @@ -73,7 +73,7 @@ all: banner $(AST_INC_CHECK) $(AST_VER_CHECK) @echo $(NAME).so : $(NAME).o - $(CC) $(SOLINK) -o $@ $(LDFLAGS) $ + $(CC) $(SOLINK) -o $@ $ $(LDFLAGS) banner: @echo -- _ -- 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
