Re: [asterisk-users] ODBC freezing Asterisk 13
Saint Michael wrote: Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed that after a few hours of inactivity, any call to func_odbc-defined funcions will block and hang for ever. All we can do at that point is reset Asterisk. I think it was highly rushed a decision to drop all the work done in ODBC inside Asterisk. Maybe unix_odbc pooling is not ready, has bugs, it cannot be used in production. I don't know what the issue is, but I had to downgrade to Asterisk 13.1 and my ODBC problems disappeared. Asterisk did not need to drop the ODBC pooling code. It did work. It should be fixed, made faster, etc. This has already been done[1] and will be released in Asterisk 13.10, which just had an rc3 released. I also sent an email to the list[2] when the fix went in. These fixes have continued to show no problems themselves although they just exposed an issue with func_odbc which was fixed in the rc3 that was just released. There's no issues open currently against that work. As for the res_odbc changes themselves which exposed problems in UnixODBC those went in as of Asterisk 13.8[3], not earlier. Prior to 13.8 there was no pooling at all. [1] http://blogs.asterisk.org/2016/06/15/asterisk-odbc-connections/ [2] http://lists.digium.com/pipermail/asterisk-users/2016-June/289326.html [3] http://blogs.asterisk.org/2016/02/17/odbc_gutting/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.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] ODBC freezing Asterisk 13
Many people are reporting the same issue, so it is not my imagination. Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so. As you know, after that version, the dropped the complexity of Pooling onto unix_odbc itself. Not so simple, it seems. I noticed that after a few hours of inactivity, any call to func_odbc-defined funcions will block and hang for ever. All we can do at that point is reset Asterisk. I think it was highly rushed a decision to drop all the work done in ODBC inside Asterisk. Maybe unix_odbc pooling is not ready, has bugs, it cannot be used in production. I don't know what the issue is, but I had to downgrade to Asterisk 13.1 and my ODBC problems disappeared. Asterisk did not need to drop the ODBC pooling code. It did work. It should be fixed, made faster, etc. -- _ -- 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] 1 way audio but audio+video is fine
Sporadically we get 1 way audio when one party is outside our firewall. The caller is on NAT, and it works fine most of the time. Caller can hear the called party, same thing going the other direction. Caller can hear called party. Asterisk 13.9 on Debian chan_sip with two identical Grandstream GXV3240 SIP phones In the CLI, I can see calls on our PBX running media on port 5004 when the call is audio only. When the caller switches to video calls it opens ports 5004 and 5006 and everything works fine. This does not make sense to me. It seems like audio on port 5004 should fail either way -- adding another media channel for video should not affect audio. I believe I have UDP ports 5000-4 open right now on the firewall. I also don't understand why it varies from day to day. Any ideas on how to debug or what might be happening? -- Brian Wilson, GISP Wildsong: 707-827-0001 Mobile: 707-332-3521 -- _ -- 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 and Yealink T21P E2
No problems with authentication during invite after reboot? I'm using insecure=no in SIP configuration. Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Thu, Jul 14, 2016 at 3:42 PM, Jeff LaCoursiere wrote: > > On 07/14/2016 02:14 PM, Marcelo Terres wrote: >> >> Hello. >> >> Anybody in the list is using this IP phone? >> >> Regards, >> >> Marcelo H. Terres >> IM: mhter...@jabber.mundoopensource.com.br >> https://www.mundoopensource.com.br >> https://twitter.com/mhterres >> https://linkedin.com/in/marceloterres >> > > Sure. Tons of them. > > -- > Jeff LaCoursiere > 312 962 5250 desk > 815 546 6599 cell > > > -- > _ > -- 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 and Yealink T21P E2
On 07/14/2016 02:14 PM, Marcelo Terres wrote: Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres Sure. Tons of them. -- Jeff LaCoursiere 312 962 5250 desk 815 546 6599 cell -- _ -- 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] Compile of smsq.c failed on Ubuntu Xenial (16.04LTS)
On 2016-07-13 17:09, Ernie Dunbar wrote: Hi everyone. I'm trying to compile Asterisk with the smsq utility on Ubuntu 16.04 LTS, and while most things are compiling fine, smsq fails with the following output: root@test25:/usr/src/asterisk-certified-13.1-cert7/utils# make smsq [CC] smsq.c -> smsq.o [LD] smsq.o strcompat.o -> smsq strcompat.o: In function `_ast_malloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:535: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_strdup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:624: undefined reference to `ast_log' strcompat.o: In function `_ast_strndup': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:654: undefined reference to `ast_log' strcompat.o: In function `_ast_vasprintf': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:694: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `_ast_realloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:596: undefined reference to `ast_log' strcompat.o: In function `_ast_calloc': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `_ast_mem_backtrace_buffer' /usr/src/asterisk-certified-13.1-cert7/include/asterisk/utils.h:559: undefined reference to `ast_log' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1014: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_append_va': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1032: undefined reference to `__ast_str_helper' strcompat.o: In function `ast_str_set_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1039: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_substr': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1046: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_set_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1053: undefined reference to `__ast_str_helper2' strcompat.o: In function `ast_str_append_escapecommas': /usr/src/asterisk-certified-13.1-cert7/include/asterisk/strings.h:1060: undefined reference to `__ast_str_helper2' collect2: error: ld returned 1 exit status ../Makefile.rules:163: recipe for target 'smsq' failed make: *** [smsq] Error 1 Years and years of installing binary packages have made my make-fu weak, but I've surmised that it's having trouble loading the asterisk.h library. To get this far, I modified smsq.h to specify the path of asterisk.h to say: #include "../include/asterisk.h" But now I get the output we see above. Perhaps there's an easier way to make it find the include files it needs? Through trial and error, I've found the solution b
[asterisk-users] Asterisk and Yealink T21P E2
Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres -- _ -- 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 Mailboxes + Cassandra
Hi List I have two questions: 1- Mailbox on the Asterisk Voicemail Server are created automatically? 2- Is there any support on the code to put the voice records on a Cassandra NoSQL database? BR Joaquin This email is confidential and may be subject to privilege. If you are not the intended recipient, please do not copy or disclose its content but contact the sender immediately upon receipt. -- _ -- 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] Certified Asterisk 13.8-cert1 Now Available
The Asterisk Development Team has announced the release of Certified Asterisk 13.8-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New Features made in this release: --- * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write contents to file (Reported by Ray Crumrine) * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel Journo) * ASTERISK-25480 - [patch]Add field PauseReason on QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders) * ASTERISK-25549 - Confbridge: Add participant timeout option (Reported by Mark Michelson) * ASTERISK-24922 - ARI: Add the ability to intercept hold and raise an event (Reported by Matt Jordan) * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID to something more palatable (Reported by Mark Michelson) * ASTERISK-25252 - ARI: Add the ability to manipulate log channels (Reported by Matt Jordan) * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by Joshua Colp) * ASTERISK-25238 - ARI: Support push configuration (Reported by Matt Jordan) * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an Asterisk module (Reported by Matt Jordan) * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation (Reported by Dwayne Hubbard) * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a channel (Reported by Matt Jordan) Bugs fixed in this release: --- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua Colp) * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported by George Joseph) * ASTERISK-26099 - res_pjsip_pubsub: Crash when sending request due to server timeout (Reported by Ross Beer) * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George Joseph) * ASTERISK-26089 - Invalid security events during boot using PJSIP Realtime (Reported by Scott Griepentrog) * ASTERISK-25885 - res_pjsip: Race condition between adding contact and automatic expiration (Reported by Joshua Colp) * ASTERISK-26074 - res_odbc: Deadlock within UnixODBC (Reported by Ross Beer) * ASTERISK-26054 - Asterisk crashes (core dump) (Reported by B. Davis) * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early nosignal packet (Reported by George Joseph) * ASTERISK-26030 - call cut because of double Session-Expires header in re-invite after proxy authentication is required (Reported by George Joseph) * ASTERISK-26004 - res_pjsip: The transport/method parameter is ignored (Reported by George Joseph) * ASTERISK-25978 - res_pjsip_authenticator_digest: Should not use source port in nonce verification (Reported by Mark Michelson) * ASTERISK-25998 - file: Crash when using nativeformats (Reported by Joshua Colp) * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive sometimes two calls (Reported by nik600) * ASTERISK-25951 - res_agi: run_agi eats frames it shouldn't (Reported by George Joseph) * ASTERISK-25947 - Protocol transfers to stasis applications are missing the StasisStart with the replace_channel object. (Reported by Richard Mudgett) * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis fails to get app name (Reported by John Bigelow) * ASTERISK-24782 - StasisEnd event not present for channel that was swapped out for another after completing attended transfer (Reported by John Bigelow) * ASTERISK-25942 - res_pjsip_caller_id: Transfer results in mixed ConnectedLine information (Reported by George Joseph) * ASTERISK-25928 - res_pjsip: URI validation done outside of PJSIP thread (Reported by Joshua Colp) * ASTERISK-25929 - res_pjsip_registrar: AOR_CONTACT_ADDED events not raised (Reported by Joshua Colp) * ASTERISK-25796 - res_pjsip: DOS/Crash when TCP/TLS sockets exceed pjproject PJ_IOQUEUE_MAX_HANDLES (Reported by George Joseph) * ASTERISK-25707 - Long contact URIs or hostnames can crash pjproject/Asterisk under certain conditions (Reported by George Joseph) * ASTERISK-25854 - No audio after HOLD/RESUME - incorrect a=recvonly in SDP from Asterisk (Reported by Robert McGilvray) * ASTERISK-25882 - ARI: Crash can occur due to race condition when attempting to operate on a hung up channel (Part 2) (Reported by Richard Mudgett) * ASTERISK-25849 - chan_pjsip: transfers with direct media
Re: [asterisk-users] Force out-bond call to specific CIC
Yes, as far as I remember, in your dial string, simply use a Dial(DAHDI/X/1234567) where X is the dahdi device channel number. Hope that helps. Matthew Fredrickson On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi wrote: > Hi > > How is it possible to use Dial application to force out-bond call use > "specified channel" number in one E1 or specified CIC (SS7) ? > > Regards > M.Shirazi > > > > > -- > _ > -- 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 -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- 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] CDR replacement with CEL
hi,
i'm trying replace CDR with CEL
reasons:
- minimize Stasis listeners (CDR)
- CEL, CDR produces "similar" data
- own logic of CDR meaning like "calldate,src,dst,direction,.." dst is
always first connected point in PBX - real user or IVR/queue etc.,
numbers are only attributes of object "user"
do you have any tips/logic/comments for this goal?
my custom cdr table
CREATE TABLE `cdr` (
`id` bigint(20) NOT NULL AUTO_INCREMENT,
`user_id` int(11) NOT NULL COMMENT 'user id',
`tenant_id` int(11) NOT NULL COMMENT 'tenant id',
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) NOT NULL DEFAULT '',
`src` varchar(80) NOT NULL DEFAULT '',
`dst` varchar(80) NOT NULL DEFAULT '',
`duration` int(11) NOT NULL DEFAULT '0',
`billsec` int(11) NOT NULL DEFAULT '0',
`disposition` varchar(45) NOT NULL DEFAULT '' COMMENT 'asterisk hangup cause',
`way` enum('loc','in','out') NOT NULL DEFAULT 'loc' COMMENT 'call direction
(loc - local, in - incoming, out - outgoing)',
`trunk` varchar(80) NOT NULL COMMENT 'used SIP trunk',
`hangupcause` varchar(10) NOT NULL COMMENT 'hangup cause',
`hangupside` varchar(10) NOT NULL COMMENT 'hangup on which side',
`uniqueid` varchar(64) NOT NULL DEFAULT '',
`linkedid` varchar(64) NOT NULL,
`data` json NOT NULL COMMENT 'metadata',
`stamp` timestamp NOT NULL DEFAULT CURRENT_TIMESTAMP ON UPDATE
CURRENT_TIMESTAMP COMMENT 'creation date',
PRIMARY KEY (`id`),
KEY `dst` (`dst`),
KEY `uniqueid` (`uniqueid`),
) ENGINE=InnoDB DEFAULT CHARSET=utf8;
CEL pairing for simple call scenario
user_id = own variable (from cel userfield or CELGenUserEvent app)
tenant_id = own variable (from cel accountcode or CELGenUserEvent app)
calldate = eventtime
src = cid_num
dst = exten
duration = eventtime(event HANGUP) - eventtime(eventtype BRIDGE_ENTER) (no
eventtype PICKUP,FORWARD,*TRANSFER)(or howto identify event RINGING?)
billsec = eventtime(event HANGUP) - eventtime(eventtype ANSWER) (no eventtype
PICKUP,FORWARD,*TRANSFER)
way = own variable (CELGenUserEvent app)
disposition = extra:
{"hangupcause":16,"hangupsource":"SIP/siptrunk-0a80","dialstatus":"ANSWER"}
trunk = own variable (CELGenUserEvent app)
hangupcause = extra:
{"hangupcause":16,"hangupsource":"SIP/siptrunk-0a80","dialstatus":"ANSWER"}
hangupside = ???
--
_
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Re: [asterisk-users] PJSIP defaults for endpoints when using realtime
On Thu, Jul 14, 2016 at 6:45 AM, A J Stiles wrote: > On Thursday 14 Jul 2016, Joshua Colp wrote: > > Carlos Chavez wrote: > > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > > (language, dtmf, vmexten, etc) and just leave many fields in the > > > database as NULL. What would be the proper way to do this for Asterisk > > > 13 and PJSIP? > > > > Kia ora, > > > > PJSIP doesn't have the ability in it to override built-in defaults for > > everything. You have to specify it yourself for realtime. If using > > config files then config file templates can be used to do this. > > If the database you are using is MariaDB or MySQL, then you should be able > to > set default values for columns in the table definition. Then when you do > an > INSERT into only some columns, the rest will be populated with the default > values. > > To alter the structure of an already-created table, use something like > > ALTER TABLE stuff CHANGE COLUMN foo foo VARCHAR(20) NOT NULL DEFAULT > "wibble"; > > (Yes, the column name should be there twice: you might want to rename it, > so > you have to specify an old and a new name even if the two are the same.) > > You will then need to use something like > > UPDATE stuff SET foo="wibble" WHERE foo IS NULL; > While this will work, your defaults may not survive the next time alembic is run to upgrade the database. It's rare but we do occasionally drop and re-create columns to change their types. You could also play with insert triggers which are attached to the table instead of specific columns. These might be easier to manage. > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- 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 > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.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] PJSIP defaults for endpoints when using realtime
On Thursday 14 Jul 2016, Joshua Colp wrote: > Carlos Chavez wrote: > > Until Asterisk 11 I could use sip.conf to set defaults for all phones > > (language, dtmf, vmexten, etc) and just leave many fields in the > > database as NULL. What would be the proper way to do this for Asterisk > > 13 and PJSIP? > > Kia ora, > > PJSIP doesn't have the ability in it to override built-in defaults for > everything. You have to specify it yourself for realtime. If using > config files then config file templates can be used to do this. If the database you are using is MariaDB or MySQL, then you should be able to set default values for columns in the table definition. Then when you do an INSERT into only some columns, the rest will be populated with the default values. To alter the structure of an already-created table, use something like ALTER TABLE stuff CHANGE COLUMN foo foo VARCHAR(20) NOT NULL DEFAULT "wibble"; (Yes, the column name should be there twice: you might want to rename it, so you have to specify an old and a new name even if the two are the same.) You will then need to use something like UPDATE stuff SET foo="wibble" WHERE foo IS NULL; -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] PJSIP defaults for endpoints when using realtime
Carlos Chavez wrote: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? Kia ora, PJSIP doesn't have the ability in it to override built-in defaults for everything. You have to specify it yourself for realtime. If using config files then config file templates can be used to do this. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.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 13 MWI
On Wed, Jul 13, 2016 at 3:44 PM, Carlos Chavez wrote: > On 7/12/16 9:27 PM, George Joseph wrote: > > > > On Tue, Jul 12, 2016 at 11:55 AM, Carlos Chavez > wrote: > >> I am still a little confused about how to activate MWI with PJSIP on >> Asterisk 13.9.1. I use realtime for configuration. So far I have tried >> setting both the mailboxes field on ps_endpoints and the mailboxes field in >> ps_aors but I cannot get the indicator lamp to blink on any of my phones >> (Digium, Aastra and Yealink). I have tried just the number of the mailbox >> and also adding the context. >> >> Any pointers? > > > It should work with "mailboxes = @". Are your phones > set to subscribe for MWI or do they expect unsolicited NOTIFYs for MWI? If > you set mailboxes on the endpoint, then you'll be sending unsolicited > NOTIFYs but if the phone does subscribe, the subscription request will > still work. If specified on the aor, the phone MUST subscribe. > "allow_subscribe = yes" must also be set on the endpoint for the > subscribes to succeed. > > Can you tell if the NOTIFY messages are actually going out but maybe being > ignored by the phones? > > Since you're using realtime, did you do a "module reload res_pjsip" or > restart asterisk after making the database changes? Just changing the > fields in the database won't trigger the MWI code to start sending > unsolicited NOTIFYs. > > > Ok, if I go to the phone configuration and explicitly enable mwi > subscription it works now (while using mailboxes in the aor). Obviously > this is a bit of a pain because it means that I have to change all the > phones configurations (ok, not a pain because of provisioning but I am lazy > ;) . All my pre Asterisk 13 installations just work without any specific > mwi parameter on the phones and I was trying to reproduce that same > behaviour. Up until Asterisk 11 all my phones blink when there is a VM > waiting and since we moved to 13 and PJSIP that is the only thing we were > missing. > > Thank you for the solution. > > > Well, I wouldn't call this a solution because you SHOULD be getting unsolicited MWI just by using mailboxes on the endpoint. Can you post a sample endpoint and aor config? > -- > > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez > +52 (55)9116-91161 > > > -- > _ > -- 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 > -- George Joseph Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.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] PJSIP defaults for endpoints when using realtime
with templates. Regards El 13/07/2016 a las 23:49, Carlos Chavez escribió: Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- _ -- 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
