[asterisk-users] Double queue calls being delivered to agents
I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We’ve been having a strange issue with a customer’s queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they’re already speaking with a client. This in turn causes a few issues: - Agent stats are no longer accurate, as it gets marked down as a ‘missed call’. - Cannot use ‘autopause’ feature any longer, as the second queue call goes unanswered and pauses the agents. The basic queue setup is as follows: Autofill = yes Ringinuse = no Wrapuptime = 5 Strategy = fewestcalls (tried ‘random’ also) Timeout = 15 We’re on Asterisk 11.21.2 currently. In talking to a few colleagues, they seem to recall there being an old patch for the Asterisk queues application that inserted a short 100ms delay between delivering first and second calls. I’ve scoured the web today, and found some old forums posts of people looking for something exactly like this, but haven’t found the actual patch, if one even exists. I’m hoping someone may have some suggestions on some options we can try to eliminate this issue. Thanks for taking the time to read this. -Derek B -- _ -- 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] Double queue calls being delivered to agents
Sorry for last post -- forgot to wipe out the digest contents :/ Derek B -- _ -- 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] Double queue calls being delivered to agents
Awesome. Thanks again Richard. On May 4, 2016, at 10:59 PM, Richard Mudgett <rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote: On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett <rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote: On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <de...@empire-team.com<mailto:de...@empire-team.com>> wrote: I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We've been having a strange issue with a customer's queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they're already speaking with a client. This in turn causes a few issues: - Agent stats are no longer accurate, as it gets marked down as a 'missed call'. - Cannot use 'autopause' feature any longer, as the second queue call goes unanswered and pauses the agents. The basic queue setup is as follows: Autofill = yes Ringinuse = no Wrapuptime = 5 Strategy = fewestcalls (tried 'random' also) Timeout = 15 We're on Asterisk 11.21.2 currently. In talking to a few colleagues, they seem to recall there being an old patch for the Asterisk queues application that inserted a short 100ms delay between delivering first and second calls. I've scoured the web today, and found some old forums posts of people looking for something exactly like this, but haven't found the actual patch, if one even exists. I'm hoping someone may have some suggestions on some options we can try to eliminate this issue. Thanks for taking the time to read this. This issue has been around a long time and was just recently fixed and I think it was just released in the latest v11 version. See https://issues.asterisk.org/jira/browse/ASTERISK-16115 Looks like it will be in the next release as the issue does not have a target release set. Richard -- _ -- 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] Double queue calls being delivered to agents
I took a look through Asterisk 11 and 13 change logs but didn't see any mention of that patch/fix. Am I missing something? Derek B > On May 4, 2016, at 8:50 AM, "asterisk-users-requ...@lists.digium.com" > <asterisk-users-requ...@lists.digium.com> wrote: > > Send asterisk-users mailing list submissions to >asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to >asterisk-users-requ...@lists.digium.com > > You can reach the person managing the list at >asterisk-users-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: Asterisk 13 Realtime Voicemail frustratingissue > (John Kiniston) > 2. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Michael Maier) > 3. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Joshua Colp) > 4. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Eric Wieling) > 5. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Joshua Colp) > 6. Call a subroutine via Originate? (John Kiniston) > 7. Re: Call a subroutine via Originate? (Bruce Ferrell) > 8. Double queue calls being delivered to agents (Derek Bolichowski) > 9. Execute an app on the master channel from inside a Macro on > the called channel (Saint Michael) > 10. Re: Double queue calls being delivered to agents (Richard Mudgett) > 11. Re: Migrating asterisk 11 to 13: some callers get no ringback > tone any more (Michael Maier) > 12. Re: T.38 with Audiocodes gateway [SOLVED] (Olivier) > 13. Asterisk registers with TLS,but sends out calls via UDP > (Sebastian Damm) > 14. Compatibilty between agi for asterisk 13.8.0 andphp5.6 > (Mamadou NGOM) > > > -- > > Message: 1 > Date: Tue, 3 May 2016 11:39:44 -0700 > From: John Kiniston <johnkinis...@gmail.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion ><asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Asterisk 13 Realtime Voicemail >frustratingissue > Message-ID: ><cafjqogc8syl_fsl8pmr+p6f6p1-nzk-_3rayrakw4kzjev8...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Have you tried using the table definition that comes with the Asterisk > source? > > the file mysql_config.sql is located in contrib/realtime/mysql and defines > a very different voicemail table than what you have in your configuration. > > On Tue, May 3, 2016 at 3:10 AM, Michele Pinassi <michele.pina...@unisi.it> > wrote: > >> Hi all, >> >> i'm experiencing a really frustrating issue with my Asterisk 13.7.2 with >> realtime configuration on MySQL and Voicemail. >> >> Here's res_config_mysql.conf: >> >> *[default]* >> *dbhost = 192.168.1.1* >> *dbname = asterisk* >> *dbuser = asterisk* >> *dbpass = [x]* >> *dbport = 3306* >> *requirements=warn ; or createclose or createchar* >> >> extconfig.conf: >> >> *[settings]* >> *sipusers => mysql,default,sipusers* >> *sippeers => mysql,default,sipusers* >> *sipregs => mysql,default,sipregs* >> *voicemail => mysql,default,vmusers* >> *meetme => mysql,default,meetme* >> >> on Asterisk console: >> >> *asterisk*CLI> realtime mysql status * >> *default connected to asterisk@192.168.1.1 <asterisk@192.168.1.1>, port >> 3306 with username asterisk for 56 minutes.* >> *asterisk*CLI> * >> >> "vmusers" table on MySQL: >> >> >> uniqueid >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60uniqueid%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494> >> customer_id >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60customer_id%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494> >> context >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk=vmusers_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60context%60+ASC_max_rows=25=81771f45cae5714ad1fac75365e0e494> >> mailbox >> <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk
Re: [asterisk-users] Double queue calls being delivered to agents
Looks like it missed 13.9.0 ☹ Thanks, Derek B. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, May 04, 2016 10:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Double queue calls being delivered to agents On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett <rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote: On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski <de...@empire-team.com<mailto:de...@empire-team.com>> wrote: I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We’ve been having a strange issue with a customer’s queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they’re already speaking with a client. This in turn causes a few issues: - Agent stats are no longer accurate, as it gets marked down as a ‘missed call’. - Cannot use ‘autopause’ feature any longer, as the second queue call goes unanswered and pauses the agents. The basic queue setup is as follows: Autofill = yes Ringinuse = no Wrapuptime = 5 Strategy = fewestcalls (tried ‘random’ also) Timeout = 15 We’re on Asterisk 11.21.2 currently. In talking to a few colleagues, they seem to recall there being an old patch for the Asterisk queues application that inserted a short 100ms delay between delivering first and second calls. I’ve scoured the web today, and found some old forums posts of people looking for something exactly like this, but haven’t found the actual patch, if one even exists. I’m hoping someone may have some suggestions on some options we can try to eliminate this issue. Thanks for taking the time to read this. This issue has been around a long time and was just recently fixed and I think it was just released in the latest v11 version. See https://issues.asterisk.org/jira/browse/ASTERISK-16115 Looks like it will be in the next release as the issue does not have a target release set. Richard -- _ -- 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] Calls are dropped after 15 minutes
Set session-timers=refuse in sip.conf and do a sip reload. We had this problem with a handful of devices and this ultimately stopped the issue. Thanks, Derek B. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Stapleton Sent: Tuesday, August 02, 2016 10:52 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Calls are dropped after 15 minutes SIP re-invite (http://www.voip-info.org/wiki/view/SIP+method+invite+re-invite) may be an issue as well. -- _ -- 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] Registration server with PJSIP
Hi Leandro, I believe if you check /usr/local/src/astersisk-13.9.1/contrib/mysql you will find a .SQL file that would build the default tables for you. Looking in the file, it appears there is a table created called `sippeers` which has a column `regserver VARCHAR(20),` It will also create the other PJSIP-related tables such as ps_endpoints, ps_aors, etc. I could be mistaken but perhaps `sippeers` is the table you’re looking for. Thanks Derek B. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini Sent: July 2, 2016 2:59 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Registration server with PJSIP Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -- _ -- 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] queue show - extensions in call going from (in use) to (not in use)
Running Asterisk 11.23.0 realtime currently, I've noticed some odd queue behavior only starting today. We are using queues in conjunction with FOP2 to show agent status, etc. As of today, we noticed that an agent will be on a call but show as available. On an inbound queue call, when an agent answers the output of `queue show` will look like this: 3103 (Local/AG-000-NF-1174@fromotherpbx/n from Custom:3103-tenant) (ringinuse disabled) (realtime) (in call) (In use) has taken 1 calls (last was 69 secs ago) Sometimes, and not always, the call will turn into this, causing FOP to show the agent as available again: 3103 (Local/AG-000-NF-1174@fromotherpbx/n from Custom:3103-tenant) (ringinuse disabled) (realtime) (in call) (Not in use) has taken 1 calls (last was 282 secs ago) Out of curiosity, is there something that would make the queue app think that the extension is no longer in use? My thought process would be that if the agent is in a call, that it should always be 'in use', but perhaps I'm missing some pertinent information here. Any tips would be appreciated. Thanks, Derek B. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference 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] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael Maier Sent: Wednesday, November 30, 2016 12:43 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Dropped call after 900s: 481 call/transaction does not exist and another anomaly during re-invite in timer - full anonymized trace attached Hello all! I can see a strange problem during invite in dialog in the context of timer handling. Given is the following incoming call from provider at 8.195.88.234 (2@2) to my asterisk at 28.19.57.152 (1@1): After 900s suddenly *asterisk* starts the timer reinvite - I would have expected the reinvite started by the provider as usual. The expected reinvite by the provider is started during authentication of the reinvite started by asterisk and is answered immediately by asterisk with sip 481. The answer of the provider after the resend of the reinvite came about 0.5s later and is sip 481, too. => The session obviously isn't known on both sides! Asterisk therefore now drops the call (bye). Does anybody has any idea about the reason why both members don't recognize the existing session any more? I hope the attached sip trace can shed some light on the problem. Thanks, Michael HI Michael, You can set this in sip.conf: session-timers=refuse -- _ -- 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 1.8.32.3 : no video (h.264)
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Friday, April 21, 2017 10:18 AM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Asterisk 1.8.32.3 : no video (h.264) Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. Why not try removing all codecs from the SIP Peer (deny all, allow only H264), unregister the peer, and try a video call again? If it works, try adding G711 back but keep H264 at the top of the priority. -- _ -- 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] Hack attempt sequential config file read looking for valid files.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Friday, April 21, 2017 12:28 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Hack attempt sequential config file read looking for valid files. I "justed" happened to look at /var/log/messages... I saw: Apr 21 12:18:40 in.tftpd[22719]: RRQ from 69.64.57.18 filename 0004f2034f6b.cfg Apr 21 12:18:40 in.tftpd[22719]: Client 69.64.57.18 File not found 0004f2034f6b.cfg Apr 21 12:18:40 in.tftpd[22720]: RRQ from 69.64.57.18 filename 0004f2034f6c.cfg Apr 21 12:18:40 in.tftpd[22720]: Client 69.64.57.18 File not found 0004f2034f6c.cfg Apr 21 12:18:40 in.tftpd[22721]: RRQ from 69.64.57.18 filename 0004f2034f6d.cfg Apr 21 12:18:40 in.tftpd[22721]: Client 69.64.57.18 File not found 0004f2034f6d.cfg Apr 21 12:18:40 in.tftpd[22722]: RRQ from 69.64.57.18 filename 0004f2034f6e.cfg so basically an sequential read of polycom MAC address config files. Some is trying to read to determine if I have any polycom files just sequential read after read. And if so - it would get any extension and password at that time. Luckily I have none. However - how does one block attempts like this ? Thanks! Jerry Jerry, Can you change to FTP Provisioning, or HTTPS etc? Atleast with FTP you can set a user/pass to your directory with mac.cfg to prevent open access. -- _ -- 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