Re: [asterisk-users] features.conf and mixmonitor stop and start
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
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37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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[asterisk-users] Asterisk 1.6.2.12 segfault
Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect SOME DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- 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] RDNIS with tel: vs. sip: header
Has anyone had success patching chan_sip.c so that Asterisk will recognize
the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, sip:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, Huh? Not an RDNIS SIP header
(%s)?\n, exten);
return -1;
}
Audiocodes Mediant 2000 devices send this header as a tel:...
*[Aug 28 02:25:42] WARNING[1283][C-1574] chan_sip.c: Huh? Not an RDNIS
SIP header (tel:41068558XX)?*
*(number obscured for privacy purposes)*
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Re: [asterisk-users] features.conf and mixmonitor stop and start
On 28 August 2014 07:56, Leandro Dardini ldard...@gmail.com wrote:
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
Do the pause/unpause in a Macro or Gosub and reference that from the
features.conf
Also, make sure you put the filename into a variable and give it full
inheritance so you can resume recording to the same file (using the a
option)
On 27 August 2014 21:20, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I have a recording started in the dialplan with the MixMonitor
application. I want to be able to stop it during a call and maybe restart
it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any idea on how I can stop the MixMonitor application while it is
running?
[featuremap]
automixmon = *1
I tried also to use the [applicationmap]] but it doesn't seem to work.
Pressing #1 do nothing. Here my dialplan:
= {
Set(__DYNAMIC_FEATURE=pauseMonitor);
MixMonitor(test);
Dial(SIP/1000@srv01,30,TtX);
}
[applicationmap]
pauseMonitor = #1,self/both,stopMixMonitor
Any advice?
extensions.conf:
[macro-pause-recording]
exten = s,1,Verbose(Stopping Recording)
exten = s,n,StopMixMonitor()
[macro-unpause-recording]
exten = s,1,Verbose(Resuming Recording)
exten = s,n,MixMonitor(${REC_FILE_NAME},a)
features.conf
StopMixMonitor = #00,peer/both,Macro(pause-recording)
;
MixMonitor = #01,peer/both,Macro(unpause-recording)
Make sure you set REC_FILE_NAME early on with a double underscore and
remember to add Set(__DYNAMIC_FEATURES=MixMonitor#StopMixMonitor) early on
too
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Asterisk 1.6.2.12 segfault
Grant, Perhaps it's time to upgrade? I used to see tons of unexplained segfaults in 1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally found a stable version) You should, also, have you heard of FreeSWITCH? IMO much more stable PBX software. Thanks On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect “SOME” DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- 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] RDNIS with tel: vs. sip: header
On 28-08-14 11:57, Positively Optimistic wrote:
Has anyone had success patching chan_sip.c so that Asterisk will
recognize the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, sip:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, Huh? Not an RDNIS SIP header
(%s)?\n, exten);
return -1;
}
Audiocodes Mediant 2000 devices send this header as a tel:...
*[Aug 28 02:25:42] WARNING[1283][C-1574] chan_sip.c: Huh? Not an
RDNIS SIP header (tel:41068558XX)?*
*
*
*(number obscured for privacy purposes)*
Not a dev but have you tried something like this (hope the formatting
stays sane):
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, sip:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, tel:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, Huh? Not an RDNIS SIP header (%s)?\n, exten);
return -1;
}
HTH,
Patrick
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Re: [asterisk-users] Asterisk 1.6.2.12 segfault
Hello, Yes, we use FreeSWITCH primarily for our main platform. Works like a charm! But we also have some applications running on Asterisk (older versions) which can’t be upgraded without careful planning and testing. Anyways, thanks for the response! Grant From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vik Killa Sent: Thursday, August 28, 2014 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.2.12 segfault Grant, Perhaps it's time to upgrade? I used to see tons of unexplained segfaults in 1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally found a stable version) You should, also, have you heard of FreeSWITCH? IMO much more stable PBX software. Thanks On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl wrote: Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect “SOME” DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- 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 SugarCrm integration
hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- --- Marek Cervenka === -- _ -- 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 SugarCrm integration
I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- --- Marek Cervenka === -- _ -- 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 -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029 Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-eSpeak and Asterisk 12
Hi, I'm giving a look at [1] with this: cd /tmp git clone https://github.com/zaf/Asterisk-eSpeak.git cd Asterisk-eSpeak ln -s path-to-asterisk-folder/include/asterisk.h ln -s path-to-asterisk-folder/include/asterisk make I'm getting this: gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_espeak.o app_espeak.c In file included from asterisk.h:21:0, from app_espeak.c:34: asterisk/autoconfig.h:7:32: fatal error: asterisk/buildopts.h: File Not found (above line translated) I can't find any buildopts.h anywhere in Asterisk 12 source files though it exists in Asterisk 11. Did I miss something ? Regards PS: If possible, I would prefer to keep asterisk external modules in seperate folder. Is there a smarted way to get (smater than the above) ? [1] http://zaf.github.io/Asterisk-eSpeak/ -- _ -- 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-eSpeak and Asterisk 12
On a side note, with Asterisk 11, I'm getting this : gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_espeak.o app_espeak.c app_espeak.c: In function ‘espeak_exec’: app_espeak.c:219:13: error: dereferencing pointer to incomplete type app_espeak.c:221:47: error: dereferencing pointer to incomplete type (My plaftform is still Debian Wheezy). 2014-08-28 16:50 GMT+02:00 Olivier oza.4...@gmail.com: Hi, I'm giving a look at [1] with this: cd /tmp git clone https://github.com/zaf/Asterisk-eSpeak.git cd Asterisk-eSpeak ln -s path-to-asterisk-folder/include/asterisk.h ln -s path-to-asterisk-folder/include/asterisk make I'm getting this: gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_espeak.o app_espeak.c In file included from asterisk.h:21:0, from app_espeak.c:34: asterisk/autoconfig.h:7:32: fatal error: asterisk/buildopts.h: File Not found (above line translated) I can't find any buildopts.h anywhere in Asterisk 12 source files though it exists in Asterisk 11. Did I miss something ? Regards PS: If possible, I would prefer to keep asterisk external modules in seperate folder. Is there a smarted way to get (smater than the above) ? [1] http://zaf.github.io/Asterisk-eSpeak/ -- _ -- 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-eSpeak and Asterisk 12
On Thu, 28 Aug 2014 17:22:54 +0200 Olivier oza.4...@gmail.com wrote: On a side note, with Asterisk 11, I'm getting this : gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_espeak.o app_espeak.c app_espeak.c: In function ‘espeak_exec’: app_espeak.c:219:13: error: dereferencing pointer to incomplete type app_espeak.c:221:47: error: dereferencing pointer to incomplete type (My plaftform is still Debian Wheezy). 2014-08-28 16:50 GMT+02:00 Olivier oza.4...@gmail.com: Hi, I'm giving a look at [1] with this: cd /tmp git clone https://github.com/zaf/Asterisk-eSpeak.git cd Asterisk-eSpeak ln -s path-to-asterisk-folder/include/asterisk.h ln -s path-to-asterisk-folder/include/asterisk make I'm getting this: gcc -pipe -fPIC -Wall -Wextra -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -g -O2 -c -o app_espeak.o app_espeak.c In file included from asterisk.h:21:0, from app_espeak.c:34: asterisk/autoconfig.h:7:32: fatal error: asterisk/buildopts.h: File Not found (above line translated) I can't find any buildopts.h anywhere in Asterisk 12 source files though it exists in Asterisk 11. Did I miss something ? Regards PS: If possible, I would prefer to keep asterisk external modules in seperate folder. Is there a smarted way to get (smater than the above) ? [1] http://zaf.github.io/Asterisk-eSpeak/ Hello, please make sure that you are using the latest trunk code and not some older 'stable' release. You can get it from here: http://github.com/zaf/Asterisk-eSpeak/tarball/master Regards, Lefteris Zafiris -- _ -- 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 SugarCrm integration
it's old. sugarcrm v7 is not supported Dne 28.8.2014 v 14:54 Scott Griepentrog napsal(a): I've used this before, and it appears to still be an active project. https://github.com/blak3r/yaai On Thu, Aug 28, 2014 at 7:29 AM, Marek Cervenka cerv...@fpf.slu.cz mailto:cerv...@fpf.slu.cz wrote: hello, can you recommend good asterisk-SugarCrm integration plugin? i googled a lot, but i want something what is used on daily basis thank you -- _ -- 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 and UniMRCP Licensing
Hey all - In some previous conversations on the Asterisk mailing lists, we noticed that some users of Asterisk were using UniMRCP [1] with Asterisk, as well as some modules made and distributed by that project. Unfortunately, there were some licensing concerns with using UniMRCP with Asterisk. As such, we contacted the UniMRCP project regarding the licensing issues and, after discussing the issue with them, we believe we have found a good path forward such that users of Asterisk and UniMRCP can use both projects together without violating the license of Asterisk. As you may know, Asterisk is licensed under the GPLv2. When Asterisk is statically or dynamically linked with a library, this creates an overall 'derivative work' as referred to in the GPL. Barring an exception, this means that any library Asterisk dynamically links with must be licensed under a GPLv2 compatible license. Unfortunately, UniMRCP is not licensed with a GPLv2 compatible license, as the Apache 2.0 license is not compatible with the GPLv2 [2]. This makes distribution of modules that link with Asterisk and UniMRCP problematic, as those modules technically should not be licensed under the GPLv2 - and hence should not be used with Asterisk under the GPLv2 license. That being said, we really like the UniMRCP project, and think it a great library for providing complex speech services. In the past, when faced with similar situations, we've added specific disclaimers to the licensing of the Asterisk project such that users are allowed to link Asterisk with specific libraries and distribute the resulting files. As such, we've modified the Asterisk license [3] to read the following: Specific permission is also granted to link Asterisk with OpenSSL, OpenH323, UniMRCP, and/or the UW IMAP Toolkit and distribute the resulting binary files. This should make it easier for participants of both projects to use Asterisk with UniMRCP without violating the licenses of either project. Hopefully this e-mail and the exception in the LICENSE file clears up any ambiguity that people may have had regarding Asterisk and the UniMRCP project. Thanks - Matt [1] http://www.unimrcp.org/ [2] http://www.gnu.org/licenses/license-list.html [3] http://svn.asterisk.org/svn/asterisk/trunk/LICENSE -- Matthew Jordan Digium, Inc. | Engineering Manager 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
Re: [asterisk-users] Asterisk 1.6.2.12 segfault
we are also facing an issue in Asterisk 11.4.0 as well. What is the route case of this issue is anyone know ? On Thu, Aug 28, 2014 at 5:32 PM, Grant Bagdasarian g...@cm.nl wrote: Hello, Yes, we use FreeSWITCH primarily for our main platform. Works like a charm! But we also have some applications running on Asterisk (older versions) which can’t be upgraded without careful planning and testing. Anyways, thanks for the response! Grant *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Vik Killa *Sent:* Thursday, August 28, 2014 1:56 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk 1.6.2.12 segfault Grant, Perhaps it's time to upgrade? I used to see tons of unexplained segfaults in 1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally found a stable version) You should, also, have you heard of FreeSWITCH? IMO much more stable PBX software. Thanks On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian g...@cm.nl wrote: Hello, Could someone explain to me what this means? asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp 40a75910 error 4 Also, would this segfault crash the whole Asterisk process or will Asterisk continue to run? Is it possible this would affect/disconnect “SOME” DAHDI channels, but not all? At this point, upgrading is not an option, even though I agree we should. Regards, Grant -- _ -- 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 -- Virendra Bhati +91-9718500594 +91-9250078532 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn] http://in.linkedin.com/pub/virendra-bhati/6/a30/755 -- _ -- 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
