[asterisk-users] connect call to queue to specified agent
hi, is it possible connect call to queue to specified agent? like Mr. Neo called helpdesk queue, call picked by agent Smith Mr. Neo is calling again and i want connect him with agent Smith -- --- 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] Is Asterisk a Linux only system?
On Thu, 12 Feb 2015 16:39:55 +0200 Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Thu, Feb 12, 2015 at 09:25:39AM -0500, D'Arcy J.M. Cain wrote: I know that it runs on other systems but do other ports get the same attention? I have been running it on a NetBSD server for about a year now and while it mostly works it just crashes from time to time with no explanation or core dump. Use the option -g to get core dumps. Did that and it stopped again but still no core file and nothing in the logs. It did stay up for a whole week this time. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:da...@vex.net VoIP: sip:da...@vex.net -- _ -- 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 13 - publish handler
Hi list, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? The trace looks like: ## PHONE - ASTERISK ## PUBLISH sip:1...@example.com SIP/2.0 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u;rport From: 1001 sip:1...@example.com;tag=98slbhbn16 To: 1001 sip:1...@example.com Call-ID: 54ddf28f87c7-ak5cx3jtjc8c CSeq: 12480 PUBLISH Max-Forwards: 70 Event: presence Authorization: Digest username=1001,realm=newsip,nonce=1423831696/f03cd493c485d56261cbcc0648f97e54,uri= sip:1...@example.com ,qop=auth,nc=0001,cnonce=6e4402d8,response=34dcaab743eb5f0837529fbb54105f5a,opaque=20c8892f17d503b0,algorithm=MD5 Expires: 3600 Content-Type: application/pidf+xml Content-Length: 480 ?xml version=1.0 encoding=UTF-8? presence xmlns=urn:ietf:params:xml:ns:pidf xmlns:im=urn:ietf:params:xml:ns:pidf:im entity=pres:1...@example.com tuple id=snom300-0004133D6914 status basicopen/basic im:imAvailable/im:im /status contact priority=1.00sip:1...@example.com/contact note xml:lang=enAvailable/note /tuple /presence ## ASTERISK - PHONE ## SIP/2.0 489 Bad Event To: 1001sip:1...@example.com ;tag=z9hG4bK-d87543-ff0f99061907dd6eaa3b-1--d87543- From: 1001sip:1...@example.com;tag=98slbhbn16 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u Call-ID: 54ddf28f87c7-ak5cx3jtjc8c CSeq: 12480 PUBLISH Server: Asterisk PBX 13.2.0 Content-Length: 0 And the debug logging is reporting: [2015-02-11 12:28:26] DEBUG[32693] res_pjsip_pubsub.c: Event presence does not match asterisk-devicestate [2015-02-11 12:28:26] DEBUG[32693] res_pjsip_pubsub.c: Event presence does not match asterisk-mwi [2015-02-11 12:28:26] WARNING[32693] res_pjsip_pubsub.c: No registered publish handler for event presence -- _ -- 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] connect call to queue to specified agent
When the call comes in, before sending it into the queue, you could consult a database of last agent who helped the user, then check availability of that agent, and send the call directly to the agent instead of putting it into the queue. You can use QueueLog to record that action so that any queue monitoring data is not unaware of it, but otherwise you would need to understand it won't show up in your queue metrics. On Fri, Feb 13, 2015 at 8:49 AM, Marek Cervenka cerv...@fpf.slu.cz wrote: hi, is it possible connect call to queue to specified agent? like Mr. Neo called helpdesk queue, call picked by agent Smith Mr. Neo is calling again and i want connect him with agent Smith -- --- 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: +1 256 580 6090 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 13 - publish handler
Sunny wrote: Hi list, Kia ora, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? Noone has implemented this functionality. The only way to have it handle it would be to write an implementation. 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
[asterisk-users] asterisk -r spammy
when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir -- _ -- 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 -r spammy
Use the -m option to mute console logging. On Fri, Feb 13, 2015 at 12:47 PM, thufir hawat.thu...@gmail.com wrote: when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. thanks, Thufir -- _ -- 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: +1 256 580 6090 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] Is Asterisk a Linux only system?
On Thu, Feb 12, 2015 at 11:40 AM, Matthew Jordan mjor...@digium.com wrote: On Thu, Feb 12, 2015 at 8:52 AM, D'Arcy J.M. Cain da...@vex.net wrote: On Thu, 12 Feb 2015 09:43:33 -0500 Ron Wheeler rwhee...@artifact-software.com wrote: Why not just bite the bullet and move to a supported Linux? If all I had was a phone switch that might be an option but this is just part of a multi-server system that needs to be able to move services back and forth so the underlying OS has to be the same for everything. Besides, I am a NetBSD developer and so I am also interested in making every package rock solid on it. - you can be assured that it works - updates are tested I would be willing to make a NetBSD machine (not my production server) available for running unit tests. Are there already unit tests in the distribution? Yes there are. In addition to unit tests, there are also the functional tests in the Asterisk Test Suite [1]. To enable them as well as set up Asterisk for the Test Suite: 1. Configure Asterisk for development mode: $ ./configure --enable-dev-mode 2. In menuselect, enable the TEST_FRAMEWORK Compiler Flag 3. Also in menuselect, enable the Test Modules. These provide the unit tests. 4. Build/install Asterisk 5. Run Asterisk 6. Execute the unit tests (or a subset thereof) using the CLI: *CLI test execute [category|all] Note that some unit tests require a particular configuration or certain subsystems to be enabled. You can examine the CI build agent scripts used for test runs here: http://svn.asterisk.org/svn/testsuite/bamboo/trunk/bin/ Specifically, the build-asterisk-only.sh script and run-asterisk-unittests.sh. Setting up [2] and running [3] the Asterisk Test Suite is documented on the wiki, and generally covers a lot more functionality than the unit tests. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation [2] https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite [3] https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite It should be noted, we did have a FreeBSD and Ubuntu systems running the testsuite back in 2010. FreeBSD was donated to the project. I personally had a PowerPC system running asterisk / testsuite, on debian. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -r spammy
On Fri, 13 Feb 2015 18:47:02 + (UTC) thufir hawat.thu...@gmail.com wrote: when running asterisk -r, is there a way to turn off the messages? I didn't find the answer in the man page. logger mute It toggles the messages on and off. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] SLA (Shared Line Appearance) and realtime
Leandro Dardini ldardini at gmail.com writes: Hello,do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? Leandro Are there any updates here? This would be a great to implement. -- _ -- 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] call failed... but why? What means SIP_ALREADYGONE?
Hi, I have watched a phenomen, that I can not explain... maybe one of you can see the reason why the call failed, and if the cause is the Snom Hardphone, or the asterisk, or the SIP-Provider... the debug log given below is all I have... What does Setting SIP_ALREADYGONE on dialog.. mean? thanks for watching, yves SIP Phone 110 (callerid 061444018110) tried to call the external Phone Number 0616677823 and gets an hangup after 2 seconds. Another try immediately after the failed call goes fine. The failed call did not arrive at the destination. [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Begin: parsing SIP Supported: timer, 100rel, replaces, from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -timer- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: timer [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -100rel- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: 100rel [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -from-change- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.0.165:3072 [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' AND h ost = 'dynamic' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Stopping retransmission on '9a6bdc548d19-goay25ioz0nd' of Response 1: Match Found [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Allocated port 19528 for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: RTP instance '0x7f2a74158788' is setup and ready to go [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Setting NAT on RTP to On [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP o=root 871055034 871055034 IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 9 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 0 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 8 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 99 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 108 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 18 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 101 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e]
