Re: [asterisk-users] Copying received and sent RTP packets due legal obligations
I'd suggest you take a look at Voipmonitor, it may have what you need in the community version. It's built for monitoring SIP/VoIP traffic. Pretty inexpensive to license the reporting tool as well, if needed. Also, you may need to have a switch/router that supports port mirroring. Thanks, *Glenn* On Wed, Jul 12, 2017 at 2:43 PM, Mark Wiater wrote: > On 7/12/2017 5:30 PM, Holger Freyther wrote: > > I have to copy/mirror/forward the RTP streams for some selected call > > to an external address/port > I'd think that what you want to do might be best done outside of > Asterisk. If you're working with SIP, I'd suggest packet capture tools. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copying received and sent RTP packets due legal obligations
On 7/12/2017 5:30 PM, Holger Freyther wrote: > I have to copy/mirror/forward the RTP streams for some selected call > to an external address/port I'd think that what you want to do might be best done outside of Asterisk. If you're working with SIP, I'd suggest packet capture tools. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Copying received and sent RTP packets due legal obligations
Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End indication once the call has been cleared I tried to use app_chanspy and ran into three different problems: - It is always transcoding - No support for input only on the audiohooks used by app_chanspy (send input, output to two different dest ports) - End of call only through hangup A post mentioned the framehooks but from what I see: - It is the audio payload but not the RTP header - I considered to rebuild the RTP header but: ast_rtp_instance can't give me the rx SSRC, rx SQNO,... Currently we are extending ast_rtp_instance to have a mirror_audio function pointer with dest ip/dest port and a custom protocol to send the rtp frames. And we have extended ast_channel_tech to add a start_mirror function pointer and have implemented it for chan_sip and chan_pjsip. They find the ast_rtp_instance and then call the mirror_audio method on it. Finally there is an AMI handler to initiate the mirroring for a given Call-Id on the right ast_channel. We would like to avoid modifying asterisk and wonder how we could move forward? Is there any hope to get our modifications included? Could there be a framehook for the rtp frame (with remote peer info)? Extend ast_rtp_instance to export more info about the RX info? Anything else? looking forward for feedback and hints holger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script
Please open a Ticket (https://issues.asterisk.org), to let them know that they need to update the documentation in Wiki and also handle this situation when using Alembic in Debian 9 (could happens in other Distros too). Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 12 July 2017 at 13:11, Floimair Florian wrote: > Nevermind guys! > > I just found out the solution myself: > > MariaDB in Debian uses utf8mb4 as default character set (see here: > https://mariadb.com/kb/en/mariadb/differences-in- > mariadb-in-debian-and-ubuntu/). > > I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf > in the following files: > > 50-client.cnf (1 line) > 50-mysql-clients.cnf (1 line) > 50-server.cnf (2 lines) > > > > With best regards > > Florian Floimair > > > -Ursprüngliche Nachricht- > Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] Im Auftrag von Floimair Florian > Gesendet: Mittwoch, 12. Juli 2017 13:50 > An: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Betreff: [asterisk-users] Asterisk realtime - Error with index length in > alembic script > > Hi! > > I just tried setting up Asterisk realtime database following the wiki > article https://wiki.asterisk.org/wiki/display/AST/Setting+up+ > PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to > MariaDB). > > One has to install mariadb-plugin-connect, python-mysqldb and alembic > packages (alembic does not work when installed via pip). > Additionally - since MariaDB by default does not have a root user password > set and running mysql -u root requires sudo as well - you need to execute > the following: > sudo mysql_secure_installation > sudo mysql_upgrade -p --force > > So far so good. I run into problems when running alembic when I get to the > following change: > https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_ > increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjs > CUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_ > tDVCtQerGg6RZ&typo=1 > mariadb fails this operation with error "Specified key was too long; max > key length is 767 bytes" when it tries to increase some fields to > varchar(255). > > Any idea how to solve this? Do I maybe have to switch to a different > encoding for this to work? > > Thanks in advance > > > > With best regards > > Florian Floimair > > COMMEND INTERNATIONAL GMBH > A-5020 Salzburg, Saalachstraße 51 > https://linkprotect.cudasvc.com/url?a=http://www.commend.com&c=E,1, > AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_ > t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,&typo=1 > > Security and Communication by Commend > > FN 178618z | LG Salzburg > > -Ursprüngliche Nachricht- > Von: https://linkprotect.cudasvc.com/url?a=https://asterisk- > users-boun...@lists.digium.com&c=E,1,7s7_D_Myc9BrXsqexg- > b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jq > olYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,&typo=1 > https://linkprotect.cudasvc.com/url?a=https://[mailto: > asterisk-users-boun...@lists.digium.com&c=E,1, > b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_ > PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,&typo=1] Im > Auftrag von Thomas > Gesendet: Montag, 10. Juli 2017 14:07 > An: https://linkprotect.cudasvc.com/url?a=https://asterisk- > us...@lists.digium.com&c=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW- > 0IaAjeofhcEGlqHiUa9FX1v_0Z61fjn6Cglc1LwJESdZ5CsnB1ZeUM > Pn7gV2z5agkz3kh8onHV0Oxnmn9c4DjH6CBU&typo=1 > Betreff: [asterisk-users] ConfBridge increase talking volume as standard > > Hello, > > is it possible to increase talking volume for caller in ConfBridge as > standard without need to press buttons after joining an conference room. > > best regards > Thomas > > -- > _ > -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc. > com/url?a=http://www.api-digital.com&c=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8 > R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_ > fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,&typo=1 -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >https://linkprotect.cudasvc.com/url?a=http://lists.digium. > com/mailman/listinfo/asterisk-users&c=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak > 6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYV > FcBMK4FJNKC_QOHqI9aSjt&typo=1 > > -- > _ > -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc. > com/url?a=http://www.api-digital.com&c=E,1,VVFR1vt-VTtY8TkmHZUWfvj_ > qWyv
[asterisk-users] Asterisk 14.6.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 14.6.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.6.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka) * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) * ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell) * ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) * ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson) * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) * ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) * ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex) * ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) * ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier) * ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) * ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) * ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot) * ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp) * ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) * ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen) * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton) * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou) * ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield) * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) * ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters) * ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard) * ASTERISK-26715 - app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel (Reported by David Brillert) * ASTE
[asterisk-users] Asterisk 13.17.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-27108 - Crash using 'data get' CLI command (Reported by Sean Bright) * ASTERISK-27106 - [patch] autodomain (SIP Domain Support): Add only really different domain with TLS. (Reported by Alexander Traud) * ASTERISK-27100 - channel: ast_waitfordigit_full fails to clear flag in an error branch. (Reported by Corey Farrell) * ASTERISK-27090 - PJSIP: Deadlock using TCP transport (Reported by Richard Mudgett) * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY events (Reported by Ove Aursand) * ASTERISK-27065 - call hangup after leaving app_queue (Reported by Marek Cervenka) * ASTERISK-26978 - rtp: Crash in ast_rtp_codecs_payload_code() (Reported by Ross Beer) * ASTERISK-27074 - core_local: local channel data not being properly unref'ed and unlocked (Reported by Kevin Harwell) * ASTERISK-27075 - bridge: stuck channel(s) after failed attended transfer (Reported by Kevin Harwell) * ASTERISK-24052 - app_voicemail reloads result in leaked IMAP sockets. (Reported by Louis Jocelyn Paquet) * ASTERISK-27060 - Comment typo format_g729.c (Reported by Matthew Fredrickson) * ASTERISK-27026 - res_ari: Crash when no ari.conf configuration file exists (Reported by Ronald Raikes) * ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan execution and application unregistration (Reported by Frederic LE FOLL) * ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at sorcery.c (Reported by Ryan Smith) * ASTERISK-27024 - nat/external_media settings ignored in 14.4.1 (Reported by Christopher van de Sande) * ASTERISK-27046 - res_pjsip_transport_websocket: segfault in get_write_timeout (Reported by Jørgen H) * ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for RTCP component (Reported by Michael Walton) * ASTERISK-26923 - bridging: T.38 request is lost when channels are added to bridge (Reported by Torrey Searle) * ASTERISK-27053 - res_pjsip_refer/session: Calls dropped during transfer (Reported by Kevin Harwell) * ASTERISK-27052 - Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network (Reported by alex) * ASTERISK-27039 - chan_pjsip: Device state is idle when channel from endpoint is in early media (Reported by Joshua Colp) * ASTERISK-26996 - chan_pjsip: Flipping between codecs (Reported by Michael Maier) * ASTERISK-26281 - chan_pjsip would send INVITE to 'Unreachable' endpoints (Reported by Jacek Konieczny) * ASTERISK-26973 - bridge: Crash when freeing frame and snooping (Reported by Michel R. Vaillancourt) * ASTERISK-19291 - Background in realtime (Reported by Andrew Nowrot) * ASTERISK-27025 - channel / meetme: Fix missing parentheses (Reported by Joshua Colp) * ASTERISK-27021 - GET /recordings/stored returns 500 Internal Server Error (Reported by Tim Morgan) * ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec (Reported by Frankie Chin) * ASTERISK-23951 - Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded (Reported by Tzafrir Cohen) * ASTERISK-25294 - srtp's crypto_get_random deprecated (Reported by Tzafrir Cohen) * ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't describe BEEP argument (Reported by Rusty Newton) * ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set variable" command without args (Reported by Antoine Pitrou) * ASTERISK-25662 - Malformed AGI 520 Usage response (Reported by Tony Mountifield) * ASTERISK-25101 - DTLS configuration can not be specified in the general section - documentation (Reported by Ben Langfeld) * ASTERISK-27008 - res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space (Reported by John Harris) * ASTERISK-26399 - app_queue: Agent not called when caller is parked (Reported by wushumasters) * ASTERISK-26400 - app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime (Reported by Etienne Lessard) * ASTERISK-26715 - app_queue: Member will not receive
Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script
Nevermind guys! I just found out the solution myself: MariaDB in Debian uses utf8mb4 as default character set (see here: https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/). I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the following files: 50-client.cnf (1 line) 50-mysql-clients.cnf (1 line) 50-server.cnf (2 lines) With best regards Florian Floimair -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian Gesendet: Mittwoch, 12. Juli 2017 13:50 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Asterisk realtime - Error with index length in alembic script Hi! I just tried setting up Asterisk realtime database following the wiki article https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to MariaDB). One has to install mariadb-plugin-connect, python-mysqldb and alembic packages (alembic does not work when installed via pip). Additionally - since MariaDB by default does not have a root user password set and running mysql -u root requires sudo as well - you need to execute the following: sudo mysql_secure_installation sudo mysql_upgrade -p --force So far so good. I run into problems when running alembic when I get to the following change: https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ&typo=1 mariadb fails this operation with error "Specified key was too long; max key length is 767 bytes" when it tries to increase some fields to varchar(255). Any idea how to solve this? Do I maybe have to switch to a different encoding for this to work? Thanks in advance With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 https://linkprotect.cudasvc.com/url?a=http://www.commend.com&c=E,1,AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,&typo=1 Security and Communication by Commend FN 178618z | LG Salzburg -Ursprüngliche Nachricht- Von: https://linkprotect.cudasvc.com/url?a=https://asterisk-users-boun...@lists.digium.com&c=E,1,7s7_D_Myc9BrXsqexg-b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jqolYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,&typo=1 https://linkprotect.cudasvc.com/url?a=https://[mailto:asterisk-users-boun...@lists.digium.com&c=E,1,b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,&typo=1] Im Auftrag von Thomas Gesendet: Montag, 10. Juli 2017 14:07 An: https://linkprotect.cudasvc.com/url?a=https://asterisk-users@lists.digium.com&c=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW-0IaAjeofhcEGlqHiUa9FX1v_0Z61fjn6Cglc1LwJESdZ5CsnB1ZeUMPn7gV2z5agkz3kh8onHV0Oxnmn9c4DjH6CBU&typo=1 Betreff: [asterisk-users] ConfBridge increase talking volume as standard Hello, is it possible to increase talking volume for caller in ConfBridge as standard without need to press buttons after joining an conference room. best regards Thomas -- _ -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com&c=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,&typo=1 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users&c=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYVFcBMK4FJNKC_QOHqI9aSjt&typo=1 -- _ -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com&c=E,1,VVFR1vt-VTtY8TkmHZUWfvj_qWyvKuVj3j3kEWWRkDTqIoK5793fkmzz9AnT3OcPsNe5K5B3f4dXsXnP4GaT1rCJ3ejq9IG69CQTwZrDqsi13OegC5RQ8hdx&typo=1 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users&c=E,1,8XODZp3VpPc6qb0tcihfIZdGrdIrkd-VCre0cSW0BegTWE_FQwvkwDTHGm8U3zoh7SeoujNXlQu50jrPRfD-eWh-imQTu3o3wFA6ag-rUcunf1TZRY11&typo=1 -- _ -- Bandwidth and Colocation Provided by
[asterisk-users] Asterisk realtime - Error with index length in alembic script
Hi! I just tried setting up Asterisk realtime database following the wiki article https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to MariaDB). One has to install mariadb-plugin-connect, python-mysqldb and alembic packages (alembic does not work when installed via pip). Additionally - since MariaDB by default does not have a root user password set and running mysql -u root requires sudo as well - you need to execute the following: sudo mysql_secure_installation sudo mysql_upgrade -p --force So far so good. I run into problems when running alembic when I get to the following change: e96a0b8071c_increase_pjsip_column_size.py mariadb fails this operation with error "Specified key was too long; max key length is 767 bytes" when it tries to increase some fields to varchar(255). Any idea how to solve this? Do I maybe have to switch to a different encoding for this to work? Thanks in advance With best regards Florian Floimair COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg -Ursprüngliche Nachricht- Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Thomas Gesendet: Montag, 10. Juli 2017 14:07 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] ConfBridge increase talking volume as standard Hello, is it possible to increase talking volume for caller in ConfBridge as standard without need to press buttons after joining an conference room. best regards Thomas -- _ -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com&c=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,&typo=1 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users&c=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYVFcBMK4FJNKC_QOHqI9aSjt&typo=1 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
