Re: [asterisk-users] Echo Cancellation on VoIP networks
Does anyone have any experience with PBXMate and the quality of the software? Does it cancel echo properly? Can someone also give me an price indication of the software? I can’t find it on their website. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valer Nur Sent: Wednesday, August 27, 2014 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo Cancellation on VoIP networks If the clients are not doing the echo cancellation properly, you can always use a centralized echo cancellation software for VoIP networks. On Wednesday, August 27, 2014 5:25 PM, Dennis Guse dennis.g...@alumni.tu-berlin.demailto:dennis.g...@alumni.tu-berlin.de wrote: On VoIP echo cancellation is basically: hope that the client is doing AND is doing it well. In the best case each client uses a knowledge about his hardware (microphone, speaker, distance etc.). --- Dennis Guse On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.commailto:emilianovazq...@gmail.com wrote: El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.commailto:emilianovazq...@gmail.com Web: http://www.pccentro.com.arhttp://www.pccentro.com.ar/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://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 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 multiple ip
hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- 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 multiple ip
You can achieve your goal with policy based routing ( http://en.wikipedia.org/wiki/Policy-based_routing). You would need to install the iproute2 package and set up ip rules for routing. This would allow you to answer endpoints registering on 192.168.10.30 with the address 192.168.10.30. On Fri, Aug 29, 2014 at 3:26 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- 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 -- Copyright 2014 Derek Andrew (excluding quotations) -- _ -- 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] POSSA CID Superfecta to query a CardDAV server ?
Hi, Would it be hard / anybody tried / any hints how / to add CardDAV server query support to CID Superfecta ? Kind Regards, Lukasz -- _ -- 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] POSSA CID Superfecta to query a CardDAV server ?
Hi, we use OSX CardDAV server and its response is very slow, so we ended up syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query defined in func_odbc.conf. -- marie On 29.08.2014, at 15:03, Lukasz Sokol el.es...@gmail.com wrote: Hi, Would it be hard / anybody tried / any hints how / to add CardDAV server query support to CID Superfecta ? Kind Regards, Lukasz -- _ -- 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 1.6.2.12 segfault
On Fri, Aug 29, 2014 at 12:46 AM, virendra bhati virbh...@gmail.com wrote: we are also facing an issue in Asterisk 11.4.0 as well. What is the route case of this issue is anyone know ? There's not enough information in either your e-mail or the original poster's to know whether or not you are experiencing the same issue - other than something crashed. Since you are running a supported version of Asterisk, please file an issue on the bug tracker at issues.asterisk.org. Make sure that you generate a backtrace from the core file using the instructions on the wiki: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace If you can reproduce the issue, that will help a lot as well. Thanks - Matt -- 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 SugarCrm integration
Unfortunately, my knowledge of SugarCRM is also a little dated. I checked on SugarForge ( http://www.sugarforge.org/softwaremap/trove_list.php?form_cat=407) and there doesn't appear to be an Asterisk integration listed, although there are some tapi dialers (which may allow routing to asterisk via another app). I would recommend filing an issue on the yaai project for 7 support. There may also be some other resources I've missed. On Thu, Aug 28, 2014 at 3:18 PM, Marek Cervenka cerv...@fpf.slu.cz wrote: 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 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 -- [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
Re: [asterisk-users] asterisk multiple ip
it looks like i found solution with chan_pjsip /etc/asterisk/pjsip.conf [transport-udp-net1] type=transport protocol=udp bind=192.168.10.20 [transport-udp-net2] type=transport protocol=udp bind=192.168.10.30 [net1_user1] type=endpoint transport=transport-udp-net1 [net2_user1] type=endpoint transport=transport-udp-net2 can you someone confirm this solution? Dne 29.8.2014 v 11:26 Marek Cervenka napsal(a): hi, i need migrate customers from severeal to one asterisk server with multiple ip aliases like eth0 192.168.10.1 eth0:1 192.168.10.20 eth0:2 192.168.10.30 i must preserve endpoint configuration to these ip adressess the problem is if i register to 192.168.10.30, the answer is from 192.168.10.1 are there some ways for this scenario? 1) chan_pjsip? 2) kamailio in front of asterisk on the same server? 3) iptables magic? 4) ... thanks -- --- 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] RDNIS with tel: vs. sip: header
Looks like this was resolved recently.
https://reviewboard.asterisk.org/r/3349/
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Thursday, August 28, 2014 12:02 PM
To: asterisk-users@lists.digium.com
Subject: 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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[asterisk-users] XMPP + Asterisk integration - a practical and simple example
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Regards, Marcelo H. Terres mhter...@gmail.com Openfire-BR owner list IM: marc...@jabber.mundoopensource.com.br http://www.mundoopensource.com.br http://offtopicsandfun.blogspot.com http://biertasters.blogspot.com http://twitter.com/mhterres -- _ -- 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] XMPP + Asterisk integration - a practical and simple example
http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Wish I had seen this when I was setting it up on my systems. Played around quite awhile using something other than OpenFire and couldn't get it working no matter what I did. Switched to OpenFire and while it wasn't completely smooth sailing, it worked much better.-- _ -- 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
Re-trying again today: both installs worked OK ! Tre trouble is I don't have the slightest idea why it did (and didn't yesterday)) !!! Thanks for helping ! 2014-08-28 19:26 GMT+02:00 Lefteris Zafiris zaf@gmail.com: 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 -- _ -- 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
Hijacking this thread, on a Debian wheezy and with Asterisk 11, default voice (with example bellow) produces a quite rebotic voice. Is this something that can be improved (by tuning espeak config, changing hardware, whatever, ...) or is it simply showing current espeak limits ? exten = 1234,n,Espeak(This is a simple espeak test in english.,any) 2014-08-29 18:58 GMT+02:00 Olivier oza.4...@gmail.com: Re-trying again today: both installs worked OK ! Tre trouble is I don't have the slightest idea why it did (and didn't yesterday)) !!! Thanks for helping ! 2014-08-28 19:26 GMT+02:00 Lefteris Zafiris zaf@gmail.com: 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 -- _ -- 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] POSSA CID Superfecta to query a CardDAV server ?
Le 29 août 2014 14:30, Marie Fischer ma...@vtl.ee a écrit : Hi, we use OSX CardDAV server and its response is very slow, so we ended up syncing all the CardDAV contacts to MySQL via cron. Asterisk dialplan then runs a query defined in func_odbc.conf. What kind of carddav query did you send to your server ? Finding caller's name from phone number, I presume ? Which client side tools did you then use ? I may be wrong but googling a bit, most examples of carddav I found where for syncing directories, not querying so your experience is very interesting. -- marie On 29.08.2014, at 15:03, Lukasz Sokol el.es...@gmail.com wrote: Hi, Would it be hard / anybody tried / any hints how / to add CardDAV server query support to CID Superfecta ? Kind Regards, Lukasz -- _ -- 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] Copying menuselect options
Le 15 août 2014 11:52, Rainer Piper rainer.pi...@soho-piper.de a écrit : I compile everything and then disable the unwanted modules in modules.conf like: modules.conf: ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes preload = res_odbc.so preload = res_config_odbc.so ;noload = res_odbc.so ;noload = res_config_odbc.so noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so load = res_musiconhold.so Did you try autoload=no ? I once did but gave up because I could find a reliable method to identify modules I needed to load. noload = chan_alsa.so noload = chan_oss.so noload = chan_console.so noload = chan_sccp.so noload = chan_skinny.so noload = chan_mgcp.so noload = pbx_dundi.so noload = chan_iax2.so noload = chan_unistim.so noload = res_corosync.so noload = res_xmpp.so noload = res_ari.so noload = pbx_ael.so noload = chan_sip.so ;noload = chan_pjsip.so noload = res_config_ldap.so noload = chan_motif.so noload = res_fax.so noload = res_fax_spandsp.so noload = res_config_mysql.so noload = bridge_native_rtp.so noload = func_odbc.so noload = res_ari_applications.so noload = res_ari_bridges.so noload = res_ari_device_states.so noload = res_ari_events.so noload = res_ari_model.so noload = res_ari_recordings.so noload = res_ari_sounds.so noload = res_ari_asterisk.so noload = res_ari_channels.so noload = res_ari_endpoints.so noload = res_ari_mailboxes.so noload = res_ari_playbacks.so noload = res_ari.so noload = cel_custom.so noload = cel_manager.so noload = cel_odbc.so noload = cel_pgsql.so noload = cel_radius.so noload = cel_sqlite3_custom.so noload = cel_tds.so noload = cdr_pgsql.so noload = res_config_pgsql.so noload = app_morsecode.so noload = res_phoneprov.so noload = app_ices.so noload = app_macro.so noload = app_festival.so noload = app_page.so noload = app_alarmreceiver.so Am 15.08.2014 um 11:32 schrieb Thorsten Göllner: Am 14.08.2014 17:22, schrieb Mitch Claborn: Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? I am not sure - but I would'nt do that. Make a hardcopy from your console and transcribe the settings to your new installation. It yould take you not more than 10 minutes. -- Rainer Piper Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 P2P: sip:7...@sip.soho-piper.de:5072 (pjsip-test) -- _ -- 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] Feature wishes for Asterisk 14
Hello, As asterisk 13 beta has been released I would be curious to share in this thread the features or improvements you would like to find in next Asterisk version. Let me start with these unordered items: - further improvements in resource list subscriptions (thanks for bringing this in asterisk 13), - being to select voicemail storage type through config files (instead of menuselect), - memcached-like storage, - g729 licence on ARM platforms, What would you like to add ? Regards -- _ -- 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] XMPP + Asterisk integration - a practical and simple example
People ask me about process_xmpp_msg.agi script, so you can find it in my blog: http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Regards, Marcelo H. Terres mhter...@gmail.com IM: marc...@jabber.mundoopensource.com.br http://www.mundoopensource.com.br http://offtopicsandfun.blogspot.com http://biertasters.blogspot.com http://twitter.com/mhterres On Fri, Aug 29, 2014 at 11:51 AM, Marcelo Terres mhter...@gmail.com wrote: http://www.mundoopensource.com.br/en_page_xmpp_asterisk_pratical_example/ Regards, Marcelo H. Terres mhter...@gmail.com Openfire-BR owner list IM: marc...@jabber.mundoopensource.com.br http://www.mundoopensource.com.br http://offtopicsandfun.blogspot.com http://biertasters.blogspot.com http://twitter.com/mhterres -- _ -- 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
O == Olivier oza.4...@gmail.com writes: O Hijacking this thread, [the espeak] default voice ... produces a O quite rebotic voice. Is this something that can be improved...? O exten = 1234,n,Espeak(This is a simple espeak test in english.,any) Try the other voices. Run espeak from the command line, try out the various voices and the other options which also are available in the Asterisk-eSpeak espeak.conf file until you find a combination which sounds OK. Depending on your hardware and settings, you may need to have espeak write its output to a file, and use another applicaion to play that. The option space for espeak has large variability. Flite also needs such tuning for nice output. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6 -- _ -- 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
