Re: [asterisk-users] Queues and penalties
I'm fairly sure the patch to App Queue that was added to Asterisk 13+ should do the job... It causes agent priorities to "float up" over time so that new agents are included without excluding old agents. I can't find it right now but there can't be that many app_queue patches to ast 13 in the last 18 months Steve On Fri, 30 Nov 2018 at 09:18, Paddy Grice wrote: > Thanks Leon > > I will implement and test but I knew there would be a fix for what I > believe is a short coming in app_queue. How do I suggest this as a option > to the base code? > > Paddy > > -- > *From:* Leon Wright [mailto:lwri...@corpcloud.com.au] > *Sent:* 30 November 2018 02:17 > *To:* pa...@wizaner.com; asterisk-users@lists.digium.com > *Cc:* johnkinis...@gmail.com > *Subject:* Re: [asterisk-users] Queues and penalties > > Paddy, > > This appears to be how the queue app works. I ended up patching the queue > app: > > diff --git a/apps/app_queue.c b/apps/app_queue.c > index e3a4e22..72072d0 100644 > --- a/apps/app_queue.c > +++ b/apps/app_queue.c > @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct > callattempt *outgoing, int *bus > struct callattempt *cur; > /* Ring everyone who shares this best metric (for > ringall) */ > for (cur = outgoing; cur; cur = cur->q_next) { > - if (cur->stillgoing && !cur->chan && > cur->metric <= best->metric) { > + if (cur->stillgoing && !cur->chan && > cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty > * 100) { > ast_debug(1, "(Parallel) Trying > '%s' with metric %d\n", cur->interface, cur->metric); > ret |= ring_entry(qe, cur, busies); > } > > So the penalties get calculated during the 'ringall' strategy and allowing > the queue app to exit, looping and raising the max penalty and calling the > queue app again. > > Leon > > On Thu, 29 Nov 2018 at 18:24, Paddy Grice wrote: > >> Hi John >> >> This works fine providing extensions 1001,1002 and 1003 are "Incall" or >> "Paused" - the problem appears to be that is a handset say 1002 is >> "ringing" then the 2xxx then the penalty is not honoured. >> >> This is well described in the History section of the following link >> https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue >> >> As I say this seems to be a real shortcoming in app_queue. >> >> Any ideas, suggestions, anyone want to work with me to sort this ? >> >> Paddy >> >> >> -- >> *From:* John Kiniston [mailto:johnkinis...@gmail.com] >> *Sent:* 28 November 2018 21:17 >> *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial >> Discussion >> *Subject:* Re: [asterisk-users] Queues and penalties >> >> This should work, How are you defining your timeouts in the queues.conf ? >> >> And to verify, in your extensions.conf you are calling Queue with the >> queue name and the ruleset to apply from queuerules.conf? >> >> On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: >> >>> Hi All >>> >>> I have been looking at this problem for a few days/weeks now and after >>> some advice please. >>> >>> I currently have a customer on 11.25.3 and I am in the process of >>> upgrading versions and OS (Debian) and all things that involves mysql -> >>> PDO etc >>> >>> The problem I have is the customer want a simple call distribution like >>> this >>> >>> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't >>> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing >>> extensions and if no one answers after another 20 seconds the add in 3001, >>> 3002, 3003. >>> >>> Seems a simple queue application to me >>> >>> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall >>> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall >>> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall >>> >>> and rules >>> >>> increasing the maxpenalty 1->2 after 20 seconds >>> and increasing maxpenalty 2->3 after another 20 seconds. >>> >>> But this doesn't work if users don't answer!! >>> >>> if user 1002 or (2001 etc) just lets his phone ring - he forgot to >>> logoff or DND then the penalty is ignored. >>> >>> There seems to have been a patch for FreePBX on V13 - LazyMembers - but >>> that is all I can find and later versions have no mention of this >>> >>> I guess I can use autopause and some AMI / Script but this stops phones >>> ringing because of the timeout so the user has a ringing phone and then it >>> stops and then it starts again whereas the penalty just adds handsets into >>> the ringing group. >>> >>> This seems to be a real shortcoming in app_queue. >>> >>> Any ideas, suggestions, anyone want to work with me to sort this ? >>> >>> Paddy Grice >>> >>> >>> >>> >>> >>> >>> >>
Re: [asterisk-users] RTP Timestamp rewind
Mark, You have cropped the image you inserted above and removed a very important part of the line you highlighted. I think is says ",Mark" after the time value - You can even see the un-cropped comma in your picture. RTP timestamps can be reset mid-stream if needed - It is part of the spec, and most commonly happens when initially (eg Asterisk) generated audio is replaced with audio from an external source once the call is bridged. The early timestamp comes from Asterisk, and the subsequent timestamp is retained from the new source of the RTP. No packets should be dropped though in my experience some jitter buffers can handle it poorly. Hope that helps, Steve On Tue, 29 Aug 2017 at 19:39 Mark Wiater wrote: > Hi folks. > > I have a couple of questions regarding RTP. > > The background of my inquiry is that I have packet captures of SIP and RTP > traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has > a time stamp that rewinds by 480 using g.711u. The Sequence number > continues to increment appropriately, but the timestamp just rewinds. > > > > It doesn't happen on every call, but it's frequent enough to make me want > to understand it better. > > My questions are: > > Is there ever a circumstance where it would be normal or logical to see > the RTP timestamp go backwards during the RTP stream? Consistently by 480, > 3 voice frames? > > Will Asterisk just drop the packets that compromise the rewind? > > Thanks > > Mark > > -- > _ > -- 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] Asterisk sip_autodestruct messages - extensions locked
Based on the line number of that error in chan_sip.c, it looks like you're running Asterisk 1.8 or earlier. AFAIK, The issue you are seeing was fixed years ago, but not THAT many years ago! If I'm right, you should upgrade to fix that issue. Cheers, Steve On Fri, 30 Jun 2017 at 13:39 Stefan Viljoen wrote: > Hi guys > > > > Does anybody have any opinion on what causes tens of thousands of these > messages per hour to pop up in the CLI: > > > > [Jun 30 14:24:59] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct: > Autodestruct on dialog '7e9597ae6ce95fef23374f4b380a9b70@192.168.0.1:5060' > with owner SIP/1148-0005bb2d in place (Method: BYE). Rescheduling > destruction for 1 ms > > [Jun 30 14:25:01] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct: > Autodestruct on dialog '6faefcc24547f1e774864ca87e3ff335@192.168.0.1:5060' > with owner SIP/1028-0005bb3f in place (Method: BYE). Rescheduling > destruction for 1 ms > > [Jun 30 14:25:02] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct: > Autodestruct on dialog '0d64480f4052a9e9054153552f1af7ba@192.168.0.1:5060' > with owner SIP/1412-0005bb5d in place (Method: BYE). Rescheduling > destruction for 1 ms > > > > Symptoms are that an extension will dial, converse and hang up, but then > be unable to dial for up to four minutes after the initial call that > extension made - all the while thousands of the above messages scroll by in > the CLI. Along with then hundreds of “too many calls” warnings for each > extension which are limited to 1 call per extension at a time in sip.conf > > > > I’ve googled intensively, no AGI is being run, hangup literally calls > Hangup(). I’ve already disabled and unloaded CDR-TDS, and ODBC CEL and CDR > logging to MySQL (in case db issues are causing a lock or something when > writing CDRs) > > > > I see on big G that many many people have had this issue, but nowhere is > there any kind of concrete end result or answer to any of the questions. > This has started to happen at our site this Monday... it comes and goes > driven by no factor I can determine, except maybe workload. Server load > average when this is going on is 8 in top for a quad-Core i7 machine with > Centos 7 and 8GB of RAM, but the machine has hit 16 as load avg without > this happening. It also sometimes happens with the load average at 4 or > less... > > > > The particular server has been running undisturbed for about three years > and has handled tens of millions of calls, and only this Monday started > exhibiting this behaviour. The dialplan was last changed about three months > (and about 3 million calls) ago. > > > > Any body got ANY advice or ideas where I can begin to diagnose this? > > > > Thanks, > > > > Stefan > > > > > -- > _ > -- 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] OT: Explain where mailing list bouncing comes from ?
I am also getting this, three or four times in the last month after years of no problems. I agree that Gmail is the likely common factor, but I would love to have access to these bounce messages to know whether it is actually an overly-paranoid list server! Steve On Mon, 12 Jun 2017 at 09:09 Andrew Furey wrote: > Ditto; a Gmail issue? > > Andrew > > On 12 June 2017 at 16:00, Marcelo Terres wrote: > >> It is happening the same with me. >> >> 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 12 June 2017 at 08:07, Olivier wrote: >> > Hello, >> > >> > I'm a faithful reader of this mailing list, for several years now. >> > >> > Lately, I'm receiving emails asking me to re-enable my list >> subscription due >> > to "excessive bouncing". >> > >> > What does this exactly mean and why am I receiving this ? >> > Beside re-enabling my subscription, what can I do to improve things ? >> > >> > Regards >> > >> > -- >> > _ >> > -- 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 >> > > > > -- > Linux supports the notion of a command line or a shell for the same > reason that only children read books with only pictures in them. > Language, be it English or something else, is the only tool flexible > enough to accomplish a sufficiently broad range of tasks. > -- Bill Garrett > -- > _ > -- 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] Using queue priorities to add agents
Hi, Thanks for this suggestion - I believe it does not quite fit the requirement as follows: - When you move up to priority 2, you will stop ringing 'Receptionist' as they are out of scope penalty 1 < 2. - Changing the penalty from penaltychange => 20,2,2 to penaltychange => 20,1,2 in order to include Receptionist does not work either, as it will still not treat 'Receptionist', 'Kelli', 'Traci' equally as required. In THEORY (I've not tried this disgusting hack yet), I could use: [mrule] penaltychange => 20,2,2 penaltychange => 40,3,3 penaltychange => 80,4,4 penaltychange => 120,5,5 penaltychange => 150,6,6 penaltychange => 180,1,1 penaltychange => 200,2,2 penaltychange => 220,3,3 penaltychange => 240,4,4 penaltychange => 260,5,5 penaltychange => 280,6,6 penaltychange => 300,1,1 [myqueue] member => SIP/100.2,1,Receptionist member => SIP/100.2,2,Receptionist member => SIP/101.2,2,Kelli member => SIP/102.2,2,Traci member => SIP/100.2,3,Receptionist member => SIP/101.2,3,Kelli member => SIP/102.2,3,Traci member => SIP/103.2,3,Debi member => SIP/100.2,4,Receptionist member => SIP/101.2,4,Kelli member => SIP/102.2,4,Traci member => SIP/103.2,4,Debi member => SIP/104.2,4,Debbie member => SIP/105.2,4,Luci ...and so on... :) See my problem? Cheers, Steve On Thu, 11 May 2017 at 16:44 John Kiniston wrote: > I have a real ugly queue that has this in it's rules > > [mrule] > penaltychange => 20,2,2 > penaltychange => 40,3,3 > penaltychange => 80,4,4 > penaltychange => 120,5,5 > penaltychange => 150,6,6 > penaltychange => 180,1,1 > penaltychange => 200,2,2 > penaltychange => 220,3,3 > penaltychange => 240,4,4 > penaltychange => 260,5,5 > penaltychange => 280,6,6 > penaltychange => 300,1,1 > > [myqueue] > member => SIP/100.2,1,Receptionist > member => SIP/101.2,2,Kelli > member => SIP/102.2,2,Traci > member => SIP/103.2,3,Debi > member => SIP/104.2,4,Debbie > member => SIP/105.2,4,Luci > member => SIP/106.2,5,Sheila > member => SIP/107.2,6,Mike > > So every 20 seconds it jumps up to the next Penalty and every few minutes > it resets the penalty back down to 1 and starts again. > > > On Thu, May 11, 2017 at 4:17 AM, Steve Davies wrote: > >> Hi, >> >> I have a scenario that I am failing to implement using the Queue app, but >> which I had thought would be commonplace... >> >> 1) (this bit works fine) I want a queue caller to have access to the >> basic set of agents initially, with an overflow to additional agents if >> they are busy - This is done using penalty: >> >> queues.conf: >> member => SIP/dev1,0,Agent1 >> member => SIP/dev2,0,Agent2 >> member => SIP/dev3,1,Agent3 is overflow >> >> 2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and >> 2 are busy or not. None of the queuerules options seem to achieve this >> because regardless of which agents are included or not, the penalty used to >> group them is also penalising them. >> >> Help? Is what I want possible? >> >> PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it >> actually increases lower penalties to it's current value, thus putting them >> on an even footing, instead of blocking out agents. >> >> Thanks, >> Steve >> >> > -- _ -- 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] Using queue priorities to add agents
Thanks for the suggestion - That is what is currently in place, but it allows queue-jumping as Asterisk does not know that one queue should be serviced (drained) before the other. That can be improved upon by doing a Waiting count on the 2nd queue etc etc, but there is always a q-jumping scenario unless the whole thing is managed inside a single queue. Cheers, Steve On Thu, 11 May 2017 at 16:36 Alexander Lopez wrote: > If after 60 seconds you mean ’60 seconds of caller hold time’ then set up > another queue as overflow, > > > > Set the first queue to timeout after 60 secs. Then send to the overflow > queue with all agents/members as same priority. > > > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Davies > *Sent:* Thursday, May 11, 2017 7:18 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Subject:* [asterisk-users] Using queue priorities to add agents > > > > Hi, > > > > I have a scenario that I am failing to implement using the Queue app, but > which I had thought would be commonplace... > > > > 1) (this bit works fine) I want a queue caller to have access to the basic > set of agents initially, with an overflow to additional agents if they are > busy - This is done using penalty: > > > > queues.conf: > > member => SIP/dev1,0,Agent1 > > member => SIP/dev2,0,Agent2 > > member => SIP/dev3,1,Agent3 is overflow > > > > 2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and > 2 are busy or not. None of the queuerules options seem to achieve this > because regardless of which agents are included or not, the penalty used to > group them is also penalising them. > > > > Help? Is what I want possible? > > > > PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it > actually increases lower penalties to it's current value, thus putting them > on an even footing, instead of blocking out agents. > > > > Thanks, > > Steve > > -- _ -- 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] Using queue priorities to add agents
Hi, I have a scenario that I am failing to implement using the Queue app, but which I had thought would be commonplace... 1) (this bit works fine) I want a queue caller to have access to the basic set of agents initially, with an overflow to additional agents if they are busy - This is done using penalty: queues.conf: member => SIP/dev1,0,Agent1 member => SIP/dev2,0,Agent2 member => SIP/dev3,1,Agent3 is overflow 2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and 2 are busy or not. None of the queuerules options seem to achieve this because regardless of which agents are included or not, the penalty used to group them is also penalising them. Help? Is what I want possible? PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it actually increases lower penalties to it's current value, thus putting them on an even footing, instead of blocking out agents. Thanks, Steve -- _ -- 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] ** in extensions.conf
On Wed, 26 Apr 2017 at 20:29 Jerry Geis wrote: > I just tried this in my extensions.conf > > exten => **,1,Noop(Testing) > exten => **,n,Playback(demo-congrats) > > Did a reload... and the above does not happen. > I created as 12 instead of the ** and that works fine. > > Is there anyway to get the ** to work? I also am using a polycom phone if > that affects things. I'm using asterisk 13.15.0 > > Thanks > > Jerry > > On a Polycom handset, dialling '**' will by default be translated into '+' before it is dialled. You could: 1) dial *..pause..* which will overcome that AFAIK. 2) Configure call.InternationalDialing.enabled="0" on the handset to stop it. 3) Put a pattern of _[+],1,... into your dialplan. That would be by guess anyway :) Steve -- _ -- 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] Cisco IP 8841 asterisk integration
If you need to know what the provisioning XML should look like for a 3PCC build of a Cisco 78xx or 88xx phone, then let it boot without provisioning, and then log in to its web interface. Select admin mode and log-in if necessary. Then edit the URL in the browser from: http://ip-address/admin/ to http://ip-address/admin/cfg.xml To see the handset's current configuration. Note that these phones are VERY fussy about being sent clean, compliant XML for provisioning. Hope that helps, Steve On Tue, 13 Dec 2016 at 12:37 Gopalakrishnan N wrote: > Thank you for the information. Actually the phone came with sip firmware. > I tried with TFTP with SEPMAC.CNF.XML and other relevant xml files. But the > phone stuck with blank screen while bootup. > > And Cisco TAC support says, the phones part number is with enterprise > firmware and it can't work with Asterisk. > > Regards. > > > -- _ -- 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] Cisco IP 8841 asterisk integration
I tried... repeatedly... I failed. I bought some 3PCC phones, and they just worked. If you have the relevant Cisco telephony server product you might be able to trick it into doing what you want, as that has the proper upgrader for that model of phone. I previously had experience of upgrading the Cisco build to the SIP build on Cisco 7641 handsets, which have 2 similar builds, but none of the techniques seemed to apply this time around. Cheers, Steve On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N wrote: > Can't I upload the 3PCC firmware ? available from the Cisco website? > > Actually it came with sip88xx firmware. > > Regards . > > > On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, wrote: > > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC handset and it is useless, and as far as I > can tell they cannot be re-flashed. > > Cheers, > Steve > > > > On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N > wrote: > > Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have > the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to > upload woth TFTP due to some reason it's getting failed. Do I need to load > 3pcc firmware or anyway to Configure from the phone itself or from the > GUI? > > I have the SEPMAC.cnf.xml as well. > > Any suggestions would be appreciated. > > Regards . > > -- > _ > -- 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 > > -- > _ > -- 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] Cisco IP 8841 asterisk integration
Hi, You have to buy the 3PCC version for this to work. Once you have this, they work very much like the Cisco SPA handsets. I also ended up with a non-3PCC handset and it is useless, and as far as I can tell they cannot be re-flashed. Cheers, Steve On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N wrote: > Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have > the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to > upload woth TFTP due to some reason it's getting failed. Do I need to load > 3pcc firmware or anyway to Configure from the phone itself or from the > GUI? > > I have the SEPMAC.cnf.xml as well. > > Any suggestions would be appreciated. > > Regards . > -- > _ > -- 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] pedantic=yes in sip.conf
Alan, A little more context would be useful. Where are you putting the '#' and why? ( If all else fails, print it out and mail it to them ;-) ) %23 is the correct encoding for a hash '#' symbol in many SIP contexts, and should be decoded by a properly functioning far-end. Regards, Steve On Wed, 30 Sep 2015 at 19:20 sysad...@reed-media.com < sysad...@reed-media.com> wrote: > Hi guys > i'm using asterisk 11.18.0. > I need to send the pound # sign to my SIP provider, but each time it's > reencoded in %23. > I try to put pedantic=yes in the sip.conf as advised here: > http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic > > but nothing's changed. > > Have someone already met this issue please ? > > thanks a lot, > > > regards, > > Alan > > -- > _ > -- 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] "Retransmission Timeout" results in dropped calls after 32 seconds
Hi, In my experience, all Yealink phones work just fine with Asterisk, we have hundreds (perhaps even low-thousands) out there with customers on Asterisk 1.2, 1.6.2, 1.8 and 11. If you are accurately representing the SIP trace on the phone and the SIP trace on Asterisk, then I would strongly suggest a SIP ALG exists in the network between the two devices and that SIP ALG does not understand SIP properly. The two halves simply do not match, so something must surely be interfering. In my experience it is often an innocent looking Cisco router. Cisco's SIP implementation is "SIP By Cisco" rather than "RFC compliant SIP". If that is the case Cisco call it a "SIP fixup" and you just need to disable it. Hope that helps, Steve On Wed, 13 May 2015 at 16:59 Andrew Martin wrote: > > > - Original Message - > > From: "Joshua Colp" > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > > Sent: Wednesday, May 13, 2015 10:50:02 AM > > Subject: Re: [asterisk-users] "Retransmission Timeout" results in > dropped calls after 32 seconds > > > > Andrew Martin wrote: > > > Since some packet loss is a possibility, I assume the protocol has > > > mechanisms > > > for dealing with it. What should be happening differently in the > > > communication > > > when packet loss occurs? Should the phone just be re-sending the OK, > > > instead of > > > printing "<0> | ERROR | receive a request with same cseq??" to its > log? Or > > > should > > > Asterisk be starting with a new cseq on each INVITE retry? > > > > The 200 OK should be retransmitted until an ACK is received. It honestly > > looks like the phone can't talk to Asterisk and it's just generally > > screwing up signaling. > > > > Thanks for the clarification and help debugging this problem. I will work > with the phone vendor to see if they can resolve this from their end. If > you > have any other ideas about how to disable re-INVITEs on the asterisk side, > beyond what I have done already, please let me know. > > Thanks, > > Andrew > > -- > _ > -- 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] hangup call gw FXO
Looking at the pastebin, the Vega device sends a CANCEL with reason: Reason: Q.850 ;cause=16. Cause 16 is normal clearing and suggests that the original caller has disconnected. I would take a look at the Vega's logs Regards, Steve On Thu, 5 Mar 2015 at 11:41 ricky gutierrez wrote: > > > On Wednesday, March 4, 2015, ricky gutierrez > wrote: > >> I'm having some problems with a vega sangoma, if a call comes into my >> ivr and hangs up, the call continues to ring and leaves hanging the >> channel, I have to restart Asterisk and everything works Ok >> >> my sangoma is a vega 50 , 4 FXO . >> >> I tried different tone of countries and does not work, >> >> this is the trace of which is for hanging up the channel: >> >> http://pastebin.com/y410Rhzt >> >> I was thinking that might help rpt timeout , I have put in 30s, but >> does not work >> >> any advice? >> >> regardss >> >> >> >> something strange, I have some extensions not connected to Asterisk and > if I call, I get the message busy, the version I'm using is asterisk 11.15 > > > -- > rickygm > > http://gnuforever.homelinux.com > -- > _ > -- 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] Connected Line presentation in 1.8.x upwards
On 29 July 2013 16:55, Kevin Larsen wrote: > > > From: Steve Davies > To:Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com>, > Date:07/29/2013 10:53 AM > Subject:[asterisk-users] Connected Line presentation in 1.8.x > upwards > Sent by:asterisk-users-boun...@lists.digium.com > -- > > > > Hi, > > I've searched the *asterisk.org* <http://asterisk.org/> and voip-info > wiki sites, but not found an answer that seems to match. > > Hopefully this is a simple question. COLP is working very well on our > system - Unfortunately it is working a bit TOO well in some circumstances. > We have some "untrusted" trunks. On these trunks, an initial CallerID can > be used, but any redirected caller numbers, COLP updates etc are not safe > to accept. Sadly I cannot find how to cause COLP updates to be ignored for > a trunk. > > I need solutions for SIP, IAX and DAHDI, what options do I have? This > applies to both in- and out-bound calls. > > Are there some variables that I can set just before dialling an outbound > call, and immediately on receiving an inbound call to determine what the > callerID values will be for the entire duration of the call? (ie. old-style > pre-COLP behaviour for specific trunks) > > Thanks for any pointers. > > Regards, > Steve > > > > I believe what you are looking for in Dial is the 'I' option. > > > Ah. Many thanks. It appears that the normally reliable voip-info wiki is out of date and does not include that option. I should probably have just used Asterisk's built-in documentation anyway :) I guess on an inbound call I will have to conditionally set 'I' on the Dial command based on the originating channel? I will also have to go and check what affect this has when a call is SIP REFER'ed as that might result in an asymmetric requirement. The internal SIP handset will want updating, but the external SIP trunk call will not. Regards, Steve -- _ -- 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] Connected Line presentation in 1.8.x upwards
Hi, I've searched the asterisk.org and voip-info wiki sites, but not found an answer that seems to match. Hopefully this is a simple question. COLP is working very well on our system - Unfortunately it is working a bit TOO well in some circumstances. We have some "untrusted" trunks. On these trunks, an initial CallerID can be used, but any redirected caller numbers, COLP updates etc are not safe to accept. Sadly I cannot find how to cause COLP updates to be ignored for a trunk. I need solutions for SIP, IAX and DAHDI, what options do I have? This applies to both in- and out-bound calls. Are there some variables that I can set just before dialling an outbound call, and immediately on receiving an inbound call to determine what the callerID values will be for the entire duration of the call? (ie. old-style pre-COLP behaviour for specific trunks) Thanks for any pointers. Regards, Steve -- _ -- 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] ignore 183 session progress in parallel call scenarios
I am sure I submitted the following "alternative" behaviour to the bug-tracker in the past, but cannot find any reference to it. Here is the patch I use to IMHO improve this behaviour. In case it is not officially uploaded, I will state here that this code is "disclaimed" and "unencumbered" as if uploaded to JIRA. Regards, Steve On 15 July 2013 17:20, Hristo Trendev wrote: > I think I have found the answer to my questions in the source code of Dial: > > case AST_CONTROL_PROGRESS: > ast_verb(3, "%s is making progress passing it to %s\n", > ast_channel_name(c), ast_channel_name(in)); > /* Setup early media if appropriate */ > if (single && !caller_entertained > && CAN_EARLY_BRIDGE(peerflags, in, c)) { > ast_channel_early_bridge(in, c); > } > if (!ast_test_flag64(outgoing, OPT_RINGBACK)) { > if (single || (!single && !pa->sentringing)) { > ast_indicate(in, AST_CONTROL_PROGRESS); > } > } > > . > . > > > Asterisk will attempt to bridge the media only for the case of a single > outgoing channel, but at the same time it will happily forward progress > messages for parallel calls: (!single && !pa->sentringing) as long as no 180 > Ringing message was sent out to the caller yet. The questions still remains > if this should be reported as bug or if there is indeed a use case when > sending 183 progress message, without actually bridging the media stream is > desired. > > > > On Mon, Jul 15, 2013 at 4:14 PM, Hristo Trendev wrote: > >> Hi, >> I am using asterisk 1.8.22 and have a problem when calling in parallel >> several SIP endpoints and I am not sure how to resolve it. In this case >> Asterisk will not bridge any audio to the caller before the 200 OK. Which >> means any progress announcements, including remotely generated ringback, >> are not passed back to the caller. >> >> This behavior is completely correct, because there is no way to know >> which early media audio stream to pass back to the caller in a parallel >> call scenario (as in this case several endpoint may indicate session >> progress all at the same time). >> >> The question is why is asterisk still sending 183 session progress back >> to the caller if no audio is to be bridged before the 200 OK anyway? If 183 >> are not passed back to the caller, then at least a 180 Ringing that may >> come from another endpoint will cause the calling endpoint to generate >> local ringback. This won't happen if the caller has received a 183 already. >> >> So it's a bit of a race condition as well - if the first endpoint to >> reply sends a "183 session progress" this means the caller will not hear >> any ringback even if some of the other endpoints are sending back 180 >> Ringing. >> >> The question is can I somehow block 183 messages from being passed back >> to the calling endpoint when dialing several destinations in parallel? I >> don't see a point (please correct me if I'm wrong) to pass only the 183 SIP >> message back to the caller without the corresponding RTP stream, so it may >> be much better to actually ignore it when dealing with parallel call >> scenarios (bug?). >> >> BR, >> Hristo >> > > > -- > _ > -- 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 > multiple_early_media Description: Binary data -- _ -- 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] FW: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6
Xavier, DoNotDisturb generates a "Busy" indication. Insert that into my earlier response, and you have an explanation of why the call tries to go from RING to BUSY, and confirms my theory. No you cannot replace the Zaptel card driver on its own (and the problem was bigger than that anyway), as Asterisk is compiled and linked to a specific Zaptel (Dahdi) version. As mentioned, you need to call IPCortex. Regards, Steve On 11 July 2013 16:23, Xavier Singer - EcuTek wrote: > Update: > I can reproduce the error by putting the reception phone (call queue 0) in > Do Not Disturb mode, then call in from outside using a mobile, then pick up > the call from the 2nd phone in the queue. It will cause the error only if I > hang up _before_ the mobile hangs up. The error doesn't seem to happen if > the external call hangs up, or if the call is answered by the reception > phone (first call in the queue). > > Thanks again, > Xavier > > > -Original Message- > From: Xavier Singer - EcuTek > Sent: 11 July 2013 12:02 > To: 'asterisk-users@lists.digium.com' > Subject: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number > when in state 6 > > We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. > We have recently implemented Call Queuing for our main incoming line with > hold music. The call queue type is: Ring all - One call at a time (no > position announcement). > > Since implementing this feature we've been receiving the below error daily > in the mornings and lunchtime when the queue will jump to the next > available phone, as the main reception phone is in Do Not Disturb mode: > > Jul 11 08:30:54 WARNING[23444] chan_zap.c: 1 Cause code 17 not > allowed when disconnecting an active call. Changing to cause 16. > Jul 11 08:30:54 ERROR[23444] chan_zap.c: You cannot use cause 17 > number when in state 6! Corrected. > Jul 11 08:30:54 WARNING[7133] chan_zap.c: Call specified, but not > found? > Jul 11 08:30:54 NOTICE[7133] chan_zap.c: Hangup, did not find cref > 1, tei 127 > Jul 11 08:30:54 WARNING[7133] chan_zap.c: Hangup on bad channel > 0/1 on span 1 > Jul 11 08:30:58 WARNING[7133] chan_zap.c: Call specified, but not > found? > Jul 11 08:30:58 NOTICE[7133] chan_zap.c: Hangup, did not find cref > 1, tei 127 > Jul 11 08:30:58 WARNING[7133] chan_zap.c: Hangup on bad channel > 0/1 on span 1 > Jul 11 08:47:04 WARNING[7133] chan_zap.c: 1 received SETUP message > for call that is not a new call (retransmission), peercallstate 19 > ourcallstate 0 cr 1, > Jul 11 08:47:08 WARNING[7133] chan_zap.c: 1 received SETUP message > for call that is not a new call (retransmission), peercallstate 19 > ourcallstate 0 cr 1, > Jul 11 08:47:19 WARNING[7133] chan_zap.c: 1 received SETUP message > for call that is not a new call (retransmission), peercallstate 19 > ourcallstate 0 cr 1, > Jul 11 08:47:23 WARNING[7133] chan_zap.c: 1 received SETUP message > for call that is not a new call (retransmission), peercallstate 19 > ourcallstate 0 cr 1, > > The ERROR happens when the call is ended. I can't replicate the error > either... > > I suspect that the chan_zap driver has a bug and is possibly trying to > hang up the call on the first phone in the queue, rather than the phone > that answered the call. > > I have investigated the different state and causes listed in the above log > file, and this is what I think they mean (please correct me if I got it > wrong): > ISDN State 6 = not initialised > Cause 16 = normal call clearing > Cause 17 = user busy > TEI 127 = reserved as the broadcast TEI > > > So my questions are: > 1. What could be causing the error and any suggestions on how to > troubleshoot this issue? > 2. Can I upgrade the chan_zap driver for the ISDN card without breaking > the IPcortex frontend (we have root access)? > 3. Should I supply any config files? > > > Thanks! > Xavier > > > > -- > _ > -- 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] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6
Hi Xavier, The issue you are seeing is an old Asterisk/Bristuff bug that was fixed years ago. Basically ISDN is unable to understand a call going from RING state to BUSY state, so Asterisk converts the BUSY into a HANGUP/Normal Clearing, and warns that this is happening. Sadly, in that old version there was a resource leak of the call object when this happened. I would suggest calling IPCortex directly to see what can be done about this. Regards, Steve On 11 July 2013 12:04, Mitul Limbani wrote: > Chan_zap has been deprecated more then 2-3 yrs back. You might have to > ping ipcortex helpdesk to get fix. > > Mitul > On Jul 11, 2013 4:32 PM, "Xavier Singer - EcuTek" > wrote: > >> We use an IPcortex PABX running Asterisk >> 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. We have recently implemented Call Queuing >> for our main incoming line with hold music. The call queue type is: Ring >> all - One call at a time (no position announcement). >> >> Since implementing this feature we've been receiving the below error >> daily in the mornings and lunchtime when the queue will jump to the next >> available phone, as the main reception phone is in Do Not Disturb mode: >> >> Jul 11 08:30:54 WARNING[23444] chan_zap.c: 1 Cause code 17 not >> allowed when disconnecting an active call. Changing to cause 16. >> Jul 11 08:30:54 ERROR[23444] chan_zap.c: You cannot use cause 17 >> number when in state 6! Corrected. >> Jul 11 08:30:54 WARNING[7133] chan_zap.c: Call specified, but not >> found? >> Jul 11 08:30:54 NOTICE[7133] chan_zap.c: Hangup, did not find >> cref 1, tei 127 >> Jul 11 08:30:54 WARNING[7133] chan_zap.c: Hangup on bad channel >> 0/1 on span 1 >> Jul 11 08:30:58 WARNING[7133] chan_zap.c: Call specified, but not >> found? >> Jul 11 08:30:58 NOTICE[7133] chan_zap.c: Hangup, did not find >> cref 1, tei 127 >> Jul 11 08:30:58 WARNING[7133] chan_zap.c: Hangup on bad channel >> 0/1 on span 1 >> Jul 11 08:47:04 WARNING[7133] chan_zap.c: 1 received SETUP >> message for call that is not a new call (retransmission), peercallstate 19 >> ourcallstate 0 cr 1, >> Jul 11 08:47:08 WARNING[7133] chan_zap.c: 1 received SETUP >> message for call that is not a new call (retransmission), peercallstate 19 >> ourcallstate 0 cr 1, >> Jul 11 08:47:19 WARNING[7133] chan_zap.c: 1 received SETUP >> message for call that is not a new call (retransmission), peercallstate 19 >> ourcallstate 0 cr 1, >> Jul 11 08:47:23 WARNING[7133] chan_zap.c: 1 received SETUP >> message for call that is not a new call (retransmission), peercallstate 19 >> ourcallstate 0 cr 1, >> >> The ERROR happens when the call is ended. I can't replicate the error >> either... >> >> I suspect that the chan_zap driver has a bug and is possibly trying to >> hang up the call on the first phone in the queue, rather than the phone >> that answered the call. >> >> I have investigated the different state and causes listed in the above >> log file, and this is what I think they mean (please correct me if I got it >> wrong): >> ISDN State 6 = not initialised >> Cause 16 = normal call clearing >> Cause 17 = user busy >> TEI 127 = reserved as the broadcast TEI >> >> >> So my questions are: >> 1. What could be causing the error and any suggestions on how to >> troubleshoot this issue? >> 2. Can I upgrade the chan_zap driver for the ISDN card without breaking >> the IPcortex frontend (we have root access)? >> 3. Should I supply any config files? >> >> >> Thanks! >> Xavier >> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fundemental changes to CDR within single asterisk family
On 4 April 2013 09:05, Ishfaq Malik wrote: > On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote: > > On 03/26/2013 05:22 AM, Ishfaq Malik wrote: > > > Hi > > > > > > In asterisk 1.8.7.0, an inbound call that was transferred to another > > > peer would have 2 cdr entries. > > > > > > In asterisk 1.8.18.0 this same activity has a single cdr entry. > > > > > > This is a rather large and fundamental change to be enacting halfway > > > through a single family branch, was there any reason why this happened? > > > It means we can't upgrade without doing significant extra development > > > and testing. > > > > > > > This was most likely an unintended consequence of some other change > > (most likely dealing with masquerades). Is 1.8.18.0 the exact version > > when the behaviour changed? > > > > Just so I'm clear on the scenario, what are the channel technologies > > involved? Is the transfer initiated via a protocol message or via a DTMF > > feature? > > > > Thanks, > > > > Matt > > > > Hi Matt > > Did you ever spot/recreate the change I was referring to? > > > "me too" - I can confirm a behaviour change and will go try and pin down at what point it happened. This may take me a while :( Steve -- _ -- 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] Congestion() forcing PRI channels to be not available
On 19 December 2012 21:54, Christopher Harrington wrote: > You probably already know this, but 1.4x is very old (released in 2006) > and is officially end-of-life. > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions > > You might get more help or better behavior by updating to a newer more > current version of Asterisk, such as 1.8 which will be receiving bug fixes > into October 2014. > > > On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna wrote: > >> Hi, >> I have a PSTN Asterisk box that's connected to other dialplan PBXes >> through IAX2. >> >> Recently this box was upgraded to 1.4.44 with the latest DAHDI version. >> I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI >> will return ISDN code 34 (as its supposed to do). >> However, the issue is that subsequent calls into that PRI channel are >> immediately responded by a Code 44 (channel not available) even though >> there is no live call on the channel. >> >> Has anyone else experienced this behavior? Its a pretty crippling >> behavior since all of our channels eventually become unresponsive until a >> 'dahdi restart' is issued. >> >> Thanks. >> >> -- James >> > I believe that what you are describing is a very old bug, which is fixed somewhere in the 1.8 timeline when the interface between DAHDI and Asterisk is changed slightly. I encountered the same issue some time ago. I do not recall the exact conditions under which the issue happens, but I believe it is the attempt to cancel an unanswered inbound call with a specific subset of cause codes. If you are using an older Asterisk version, the only workaround is to use Playtones + Hangup() instead of sending the Congestion() or Busy() cause codes. Regards, Steve -- _ -- 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] SIP Question - Early audio one-way or 2-way?
On 1 September 2012 09:08, Olle E. Johansson wrote: > > 31 aug 2012 kl. 13:13 skrev Steve Davies : > >> On 31 August 2012 07:49, Olle E. Johansson wrote: >>> >>> 24 aug 2012 kl. 16:18 skrev Steve Davies : >>> >>>> Hi SIP Gurus, >>>> >>>> I've tried to find the relevant RFCs, but am struggling. I can find >>>> the odd opinion online, but was wondering if anyone could give a >>>> definitive answer. >>>> >>>> If a SIP call is initiated (INVITE) and receives either a "180 with >>>> SDP", or a "183 with SDP", then the remote party will start to send >>>> audio for the inband-ringing. Asterisk then passes this audio, and it >>>> is correctly heard by the caller. >>>> >>>> At present, Asterisk will also start to pass back any handset audio in >>>> return, in theory allowing a conversation to occur on an unanswered >>>> channel if an endpoint were designed to allow this (free phonecalls >>>> here we come!). >>>> >>>> My question: >>>> >>>> Should: >>>> 1) Asterisk block outbound audio between the 183 Progress and the 200 >>>> OK packets? >>>> 2) Replace any outbound audio with silence? >>>> 3) Replace outbound audio with a special NULL RTP of some sort? Does that >>>> exist? >>>> 4) Allow any audio to be sent regardless? >>>> >>>> I have implemented 1) at present on our test rig, but the lack of >>>> outbound RTP causes issues with firewall state not being set-up to >>>> allow the inbound audio. I am not sure how hard/easy it would be to do >>>> 2) as you'd need to create silence of the correct duration to replace >>>> each audio frame. >>>> >>>> Thoughts please? >>> >>> First, because of Asterisk's RTP implementation we have to send some RTP >>> packets at this point. You could mute the calling channel before calling >>> and unmute the channel at answer if needed, but normally sending audio >>> won't hurt. A normal user should not be able to send early media on a >>> pstn-like installation where you bill the calls, so there should be little >>> risc of two-way conversations before an answer. >>> >>> In some cases you have to let the caller send DTMF (the famous fed ex >>> example) in >>> early media, so we can't block any media by default in Asterisk. >>> >>> Using the "r" option in dial causes a lot of issues, since you can still >>> get busy or congestion when you have early media, so that is not a good >>> solution. >>> >>> /Olle >>> >> >> Excellent information as always Olle. Many thanks. >> >> My intention is to make the early-audio prevention in SIP a little >> more harsh, such that if SIP receives audio before a 183 or 200 is >> received, it is dropped. >> >> This fixes the case where "useless" early-audio is received from a >> non-SIP (eg ISDN) technology, and can cause an onward node to >> auto-enable early audio mode, causing silent ringing and other broken >> behaviours. > > This is one of my pet issues. THe problem today is that many gateway vendors > ALWAYS send 183 with sdp, > regardless if it's a ring tone or a service provider message. If you kill the > 183, service provider messages > will disappear. My recommendation (which I've mentioned in tons of mails and > blog entries) is to send > 180 ringing with SDP for ring tones and 183 for other messages. That way I > could kill the 180 SDP in a Kamailio > proxy before it hits the Asterisk server. In reality today, by killing 183 > you will also block important information > for the caller, like "this subscriber has a new number". > I have 2 similar scenarios. One like yours is a SIP provider who interchangeably uses 180 with sdp, 183 with sdp or 180 no sdp to indicate ringing. There seems to be no logic, so we have to honour all of those cases when they happen. Asterisk handles this already. The case I am now killing can be caused by a SIP to ISDN call where the ISDN provides a "RINGING" response which becomes 180 no sdp, but the ISDN then proceeds to send early audio regardless. Asterisk will currently pass that audio even though the SDP exchange is incomplete, by using the SDP in the initial INVITE to send one way audio. This is then propagated along the call chain and is most inconvenient. I think I have a solution, currently in 1.6.2 and 1.8, and if you think it is legitimate as a change to block that unexpected early audio, I can put the patch on jira. Regards, Steve -- _ -- 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] SIP Question - Early audio one-way or 2-way?
On 31 August 2012 07:49, Olle E. Johansson wrote: > > 24 aug 2012 kl. 16:18 skrev Steve Davies : > >> Hi SIP Gurus, >> >> I've tried to find the relevant RFCs, but am struggling. I can find >> the odd opinion online, but was wondering if anyone could give a >> definitive answer. >> >> If a SIP call is initiated (INVITE) and receives either a "180 with >> SDP", or a "183 with SDP", then the remote party will start to send >> audio for the inband-ringing. Asterisk then passes this audio, and it >> is correctly heard by the caller. >> >> At present, Asterisk will also start to pass back any handset audio in >> return, in theory allowing a conversation to occur on an unanswered >> channel if an endpoint were designed to allow this (free phonecalls >> here we come!). >> >> My question: >> >> Should: >> 1) Asterisk block outbound audio between the 183 Progress and the 200 >> OK packets? >> 2) Replace any outbound audio with silence? >> 3) Replace outbound audio with a special NULL RTP of some sort? Does that >> exist? >> 4) Allow any audio to be sent regardless? >> >> I have implemented 1) at present on our test rig, but the lack of >> outbound RTP causes issues with firewall state not being set-up to >> allow the inbound audio. I am not sure how hard/easy it would be to do >> 2) as you'd need to create silence of the correct duration to replace >> each audio frame. >> >> Thoughts please? > > First, because of Asterisk's RTP implementation we have to send some RTP > packets at this point. You could mute the calling channel before calling and > unmute the channel at answer if needed, but normally sending audio won't > hurt. A normal user should not be able to send early media on a pstn-like > installation where you bill the calls, so there should be little risc of > two-way conversations before an answer. > > In some cases you have to let the caller send DTMF (the famous fed ex > example) in > early media, so we can't block any media by default in Asterisk. > > Using the "r" option in dial causes a lot of issues, since you can still get > busy or congestion when you have early media, so that is not a good solution. > > /Olle > Excellent information as always Olle. Many thanks. My intention is to make the early-audio prevention in SIP a little more harsh, such that if SIP receives audio before a 183 or 200 is received, it is dropped. This fixes the case where "useless" early-audio is received from a non-SIP (eg ISDN) technology, and can cause an onward node to auto-enable early audio mode, causing silent ringing and other broken behaviours. Cheers, Steve -- _ -- 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] SIP Question - Early audio one-way or 2-way?
On 24 August 2012 15:34, Faisal Hanif wrote: > Steve Davies wrote: >>Hi SIP Gurus, >> >>I've tried to find the relevant RFCs, but am struggling. I can find >>the odd opinion online, but was wondering if anyone could give a >>definitive answer. >> >>If a SIP call is initiated (INVITE) and receives either a "180 with >>SDP", or a "183 with SDP", then the remote party will start to send >>audio for the inband-ringing. Asterisk then passes this audio, and it >>is correctly heard by the caller. >> >>At present, Asterisk will also start to pass back any handset audio in >>return, in theory allowing a conversation to occur on an unanswered >>channel if an endpoint were designed to allow this (free phonecalls >>here we come!). >> >>My question: >> >>Should: >>1) Asterisk block outbound audio between the 183 Progress and the 200 >>OK packets? >>2) Replace any outbound audio with silence? >>3) Replace outbound audio with a special NULL RTP of some sort? Does that >>exist? >>4) Allow any audio to be sent regardless? >> >>I have implemented 1) at present on our test rig, but the lack of >>outbound RTP causes issues with firewall state not being set-up to >>allow the inbound audio. I am not sure how hard/easy it would be to do >>2) as you'd need to create silence of the correct duration to replace >>each audio frame. >> >>Thoughts please? >> >>Many thanks, >>Steve >> > hi, > > you can simply avoid this by using local ring r option in dial command. > azterisk pass local ring voice to caller and will not bridge b leg audio > until b leg is answered.iin > Regards, > > Faisal Hanif > (sent from phone) Nice thought, but what if there is a useful reason for the progress audio? If it is sent we want to hono[u]r it, and if it is not, we expect a "180 ringing", and let the SIP device generate the tone, rather than send an unwanted audio stream to use up bandwidth :) For example, some UK ISDN services will give a useful call failure message by sending a progress-frame, followed by some audio. DAHDI and SIP handle this nicely with a 183 progress message, and pass on the message on the un-answered SIP channel. Regards, Steve -- _ -- 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] SIP Question - Early audio one-way or 2-way?
Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer. If a SIP call is initiated (INVITE) and receives either a "180 with SDP", or a "183 with SDP", then the remote party will start to send audio for the inband-ringing. Asterisk then passes this audio, and it is correctly heard by the caller. At present, Asterisk will also start to pass back any handset audio in return, in theory allowing a conversation to occur on an unanswered channel if an endpoint were designed to allow this (free phonecalls here we come!). My question: Should: 1) Asterisk block outbound audio between the 183 Progress and the 200 OK packets? 2) Replace any outbound audio with silence? 3) Replace outbound audio with a special NULL RTP of some sort? Does that exist? 4) Allow any audio to be sent regardless? I have implemented 1) at present on our test rig, but the lack of outbound RTP causes issues with firewall state not being set-up to allow the inbound audio. I am not sure how hard/easy it would be to do 2) as you'd need to create silence of the correct duration to replace each audio frame. Thoughts please? Many thanks, Steve -- _ -- 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] CONNECTEDLINE() updated during SIP events?
On 25 April 2012 18:05, Kevin P. Fleming wrote: > On 04/25/2012 11:54 AM, Steve Davies wrote: > >> A further question... It appears that for SIP endpoints, this facility >> only updates RPID and PAI headers? I have found that there appear to >> be 4 different SIP CID-update mechanisms "out there" as follows: >> >> - Update RPID and PAI (ITSP and trunks often understand this) >> - Update Contact: header (Aastra handsets use this) >> - A SIP INFO packet if "Supported: callerid" is specified (Older snom >> firmware uses this) >> - Update From: header if "Supported: from-change" is specified (RFC >> 4916, snom, Yealink) >> >> Are there existing plans to support any of these other methods? If >> not, I will almost certainly add them for my own use, and submit the >> code. > > > No, we have no plans at this time to go beyond RPID and PAI support. Those > two appear to cover all the current endpoints that we have been able to test > with, and many community members have also used with other endpoints and had > success. Thanks for that, I'll have to test further and see whether all the devices we use support RPID/PAI. It would certainly be easier than messing about with headers that should not really be changed! > Changing the Contact header seems quite wrong; the display-name in a URI in > the Contact header is pretty much irrelevant. Changing the From header also > seems wrong; that should indicate who sent the initial INVITE, not who > redirected it. I don't think we'd want to merge patches that added support > for either of those mechanisms. The From: header change is a relatively recent RFC, but I've seen several handsets supporting it, and several non-Asterisk SIP stacks using this to achieve COLP updates. I completely agree that changing the Contact: header is daft, and I have no idea why Aastra use this method. Regards, Steve -- _ -- 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] CONNECTEDLINE() updated during SIP events?
On 25 April 2012 16:55, Richard Mudgett wrote: [snip] > >> - Is it possible to have the COLP/COLR information updated when a SIP >> attended transfer is completed? If so how? > > Transfers generate connected line update events automatically. The connected > line interception macros give you a chance to alter the connected line > information as it is passed between the connected endpoints of the bridge. > >> In both of the above cases, there is no obvious dialplan execution >> when the calls are redirected, diverted or masqueraded, so we cannot >> update the CONNECTEDLINE() information trivially. Or am I missing an >> obvious trick? > > This is the purpose of the interception macros. Ah, thank you. I was looking at it back-to-front. The key bit is "Transfers generate connected line update events automatically." - I can now see this in the source code in ast_do_masquerade() and elsewhere. This then lets you use the macros as you describe. A further question... It appears that for SIP endpoints, this facility only updates RPID and PAI headers? I have found that there appear to be 4 different SIP CID-update mechanisms "out there" as follows: - Update RPID and PAI (ITSP and trunks often understand this) - Update Contact: header (Aastra handsets use this) - A SIP INFO packet if "Supported: callerid" is specified (Older snom firmware uses this) - Update From: header if "Supported: from-change" is specified (RFC 4916, snom, Yealink) Are there existing plans to support any of these other methods? If not, I will almost certainly add them for my own use, and submit the code. Regards, Steve -- _ -- 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] CONNECTEDLINE() updated during SIP events?
Hi, I have read the excellent information here: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information and believe I have an understanding of what is offered. I have a couple of questions: - Is it possible to update COLP/COLR when a SIP redirect occurs, or when a SIP divert is in place? If so, how? - Is it possible to have the COLP/COLR information updated when a SIP attended transfer is completed? If so how? In both of the above cases, there is no obvious dialplan execution when the calls are redirected, diverted or masqueraded, so we cannot update the CONNECTEDLINE() information trivially. Or am I missing an obvious trick? Thanks, Steve -- _ -- 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 ACL
On 2 April 2012 14:06, Mark Farmer wrote: > > > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro > Dardini > Sent: 02 April 2012 13:53 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk ACL > > > > Your understanding of the problem seems incorrect. The problem seems due to > the extension not available in your dialplan. Please check carefully in > which context the call is placed and if the extension is defined in that > context. > > > > Maybe it can be useful to define a _X. extension to catch all not defined > extensions. > > > > Leandro > > [Mark Farmer] > > The problem is that the inbound call is not being matched by the correct > peer and as such falls through to the default context which is not supposed > to match. > > The problem is around the matching of a range of IP addresses to one peer. > > Thanks > > Mark. Mark, This is a problem I have encountered regularly. Your mistake is thinking that setting deny/permit will cause a peer to be matched if it falls in the permitted range. It will not. The peer will only match if the source IP address matches the host= value, and in the case of "dynamic" it must match the IP address of the party that registered. deny/permit will also restrict a 'type=user' or 'type=friend' so that the username can only be attempted from specified IP ranges. IAX does what you expect, and I have thought regularly of implementing in SIP what you expected to be the normal behaviour, but in fact, the deny/permit will limit where the original registration can come from, but AFAIK does not get used for subsequent (INVITE) matches until after the host IP match is completed. At present, the best solution is to change type to 'friend' and use username/password based authentication. "Buyer beware" - I believe the above to be true, and I hope it makes sense! Regards, Steve -- _ -- 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] Fwd: Re: Asterisk CLI unresponsive
On 6 February 2012 10:45, Jonas Kellens wrote: > ** > Hello, > > is there anyone that can give me some more information on these > "deadlocks" ?! > > How can these deadlocks occur and what is "good practise" to avoid these > problems ?? > > > Jonas. > > The only way to avoid deadlocks is to report them when they happen, provide the requested debug, and hope for a fix. Deadlocks are just a type of bug. Nothing you can do to avoid it unless you can work out the events that lead up to it, and avoid doing that. I am still using 1.6.2.22 in many places, and if I see a deadlock "show locks" trace on the mailing list - I will certainly be looking into it. Regards, Steve. -- _ -- 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 CLI unresponsive
On 3 February 2012 12:12, Jonas Kellens wrote: > On 02/03/2012 01:05 PM, Mikhail Lischuk wrote: > > Jonas Kellens писал 03.02.2012 12:09: > > > using asterisk 1.6.2.22 > > What is wrong with Asterisk when the CLI becomes unresponsive ?! > > > Greetings. I am using the same version, Asterisk 1.6.2.22 > > IDK is your problem is same as mine. My CLI also becomes unresponsive > sometimes, however it does not affect calls at all. And all commands I type > get processed and output given after some time - 3 or 5 minutes. > > As I determined, this happens when Asterisk is rotating logs. > > You can see it by issuing `ps aux | grep asterisk` at that time. > > IDK if this is normal (due to logfile locks or whatever) or not. > > > Hello, > > my logs get rotated at 8h15, 12h15 and 17h15. The deadlock occurred at > 10h34. Don't think this is the relation between problem and deadlock. > > There is no more output on the CLI, so I don't see if the calls which are > going on are affected and are hang up. > > I think you are best to rebuild your asterisk with lock tracking enabled (it is one of the debug options) and then when it locks, do a "core show locks" and post it here. Normally I would suggest posting it to the bugtracker, but 1.6.2 is no-longer supported, so that would be a waste of time. Regards, Steve -- _ -- 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] SIP and NAT best practices since recent changes?
On 11 January 2012 15:43, Kevin P. Fleming wrote: > On 01/11/2012 05:29 AM, Steve Davies wrote: >> >> Hi, >> >> Since the recent update to the NAT configuration options and defaults >> in chan_sip.so, I am interested in any SIP/NAT best practices advice. >> >> What I've always done in the past is: >> >> Global: nat=no >> SIP handsets that are local: nat=no >> SIP handsets that are remote: nat=yes >> ITSP SIP trunks: nat=yes >> >> I will then set externip and localnet to reflect the local setup, >> UNLESS there is a functional SIP ALG doing the work in the gateway >> device. I make this statement because I've found one or two firewalls >> where it is best to disable the SIP ALG, and one or two where it is >> best to leave it enabled. >> >> The above always worked very well, but I now find my asterisk logs >> being spammed with warnings containing lots of "!!" and I'd like to >> know the best way to operate to achieve what I've always had while >> following the new rules in order to be as secure as possible with >> "clean" logs. I should add that we do not accept unsolicited >> connections, and 99% of attempts to connect will be stopped at the >> firewall. > > > The simplest answer is to always use 'nat=yes' (or at least > 'nat=force_rport' in recent versions of Asterisk that support it), until you > come across a SIP endpoint that fails to work properly with that setting. If > you do come across such an endpoint, try hard to get it to work with that > setting; if you can't, then set 'nat=no' for that endpoint, and understand > that the endpoint's name could be discoverable using the attack methods > previously disclosed. If the endpoint's configuration is suitably locked > down (permit/deny, for example) this may not be a concern for you. If it's > not locked down (for example, if it has to register to your Asterisk server > from random locations), then the next step would be to seriously consider > requesting that the user of that endpoint consider switching to some other > SIP endpoint. > > To date, the only endpoints that have been identified that do *not* work > with Asterisk's 'rport' handling forced upon them are Cisco phones. > Excellent. Thanks as always Kevin. (Why am I not surprised about Cisco!) Regards, Steve -- _ -- 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] SIP and NAT best practices since recent changes?
Hi, Since the recent update to the NAT configuration options and defaults in chan_sip.so, I am interested in any SIP/NAT best practices advice. What I've always done in the past is: Global: nat=no SIP handsets that are local: nat=no SIP handsets that are remote: nat=yes ITSP SIP trunks: nat=yes I will then set externip and localnet to reflect the local setup, UNLESS there is a functional SIP ALG doing the work in the gateway device. I make this statement because I've found one or two firewalls where it is best to disable the SIP ALG, and one or two where it is best to leave it enabled. The above always worked very well, but I now find my asterisk logs being spammed with warnings containing lots of "!!" and I'd like to know the best way to operate to achieve what I've always had while following the new rules in order to be as secure as possible with "clean" logs. I should add that we do not accept unsolicited connections, and 99% of attempts to connect will be stopped at the firewall. Thanks, Steve -- _ -- 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 1.4 compatible with 1.8.7 ?
On 28 December 2011 03:02, Joseph wrote: > No, it makes no difference, on the other end is asterisk 1.4.39 > > and 1.8.8 is still giving me: > > Executing [4@internal:1] Dial("SIP/11-0003", > "IAX2/home_server:@192.168.141.1/4,30,rw") in new stack > -- Called IAX2/home_server:@192.168.141.1/4 > [Dec 27 20:00:16] WARNING[16398]: chan_iax2.c:10672 socket_process: Call > rejected by 192.168.141.1: Unable to negotiate codec > -- Hungup 'IAX2/192.168.141.1:4569-5678' > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing [4@internal:2] Hangup("SIP/11-0003", "") in new stack > == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0003' > > -- > Joseph > [snip] Have you tried enabling IAX2 debug at both ends to see if the packet decode provides any more clues? Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.x segfaulting daily
On 14 December 2011 12:56, Paulo Santos wrote: > Hello list, > > An Asterisk installation that was doing fine suddenly stared segfaulting a > couple of times per day. I enabled all the logging and debugging to try to > find a pattern but there was too much information to see exactly where it > broke. So I enabled core dump and did backtraces and all of them seem to > break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty > much the only thing I can make of it, don't even know if that's correct. > > Does anyone have any ideas on why this is happening? The backtrace is > attached. > > P.S.: I've switched the whole hardware already, including the BRI card > (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and > mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. > If I was guessing, I'd say that the channel structure that is being modified by the ast_setstate() call is incomplete, and contains some garbage pointers. If I was guessing further, I'd say that the callerID pointers are the most likely candidate - Does the issue happen when a caller-id withheld call is hung-up? hung-up before being answered perhaps? You'd need to add some debug reporting into ast_setstate() to know for sure. Just my 2p - 1.4.42 is an old version, so the chance of a solid answer is fairly low. Steve -- _ -- 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] one way voice with IVR
On 17 October 2011 11:01, gincantalupo wrote: > Hi, > > found where the problem is.I tried with a Grandstream phone and it > works!!! > > The problem is my (crappy) Snom phone. > > I'm investigating the probhope to find the cause asap. > FYI: snom firmware 7.3.30 is very old. I remember it being quite reliable, but it is still worth looking at the 8.4.32 release or newer. Regards, Steve -- _ -- 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] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
On 5 October 2011 10:21, Nasir Iqbal wrote: > You can do this by an AMI based transfer (Redirect) to Local channel, and > then in local channel's dialplan you need to add your desired custom sip > header followed by a dial command. > Nasir Iqbal > > ICT Innovations > http://www.ictinnovations.com/ > Broadcom invented some SIP NOTIFY extensions to cover this case - Several "open" SIP handsets support the Broadcom extensions, which revolve around sending a NOTIFY "Event: hold" or "Event: talk" Asterisk does not support these NOTIFY messages at present, though I expect they could be added reasonably simply. Regards, Steve -- _ -- 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] Possibly odd sip.conf security requirements. Possible?
Hi, Is the following possible in some way? I want to have several SIP providers able to send me calls, each provider may send calls into many possible DDIs. Each provider has a cluster of servers, but is unable to authenticate with me, so the following would be a sort of pseudo-code sip.conf example. [general] context = barred ; Unknown/other source of calls [provider 1] type = peer context = provider1-context ; deal with provider's calls 1 deny = 0.0.0.0/0.0.0.0 permit = 12.13.14.0/24 ; This provider has servers in this range [provider 2] type = peer context = provider2-context ; deal with provider's calls 2 deny = 0.0.0.0/0.0.0.0 permit = 22.23.24.0/24 ; This provider has servers in this range [provider 3] type = peer context = provider3-context ; deal with provider's calls 3 deny = 0.0.0.0/0.0.0.0 permit = 32.33.34.0/24 ; This provider has servers in this range Normally a call into SIP has one of 3 paths: 1) Unauthenticated, so use the default 2) Identifiable username 3) Identifiable IP address In the above example, we have a BLOCK of IP addresses instead of a single address. Can this be made to work? Thanks for any pointers. Regards, Steve -- _ -- 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] 1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)
On 14 August 2011 08:36, Eric Wieling wrote: > > I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. > Below is a dialplan snippet and the resulting CLI output. This is running in > an 'h' extension. > > Noop(DIALSTATUS=${DIALSTATUS}) > Noop(CDR(disposition)=${CDR(disposition)}) > > -- Executing [h@pbxmax-dial-simple:1] NoOp("SIP/msx_01-005b", > "DIALSTATUS=ANSWER") in new stack > -- Executing [h@pbxmax-dial-simple:2] NoOp("SIP/msx_01-005b", > "CDR(disposition)=NO ANSWER") in new stack > > Unless I seriously misunderstand the CDR(disposition) function, this looks > like a bug to me. Does any else have this issue? > What call events lead up to that? There is a case I encountered some time ago where a blind-transfer can mess-up the CDR disposition. I started to re-factor all of the CDR code to fix all of these issues, but with 1.6 EOL, 1.8 etc etc, it needs revisiting. As far as I know, 1.8/1.10 has the same issues. Regards, Steve -- _ -- 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] Seg Faults with 1.6.2.19
On 18 July 2011 14:05, Lee Archer wrote: > Seems to be an already reported problem but since no more fixes for 1.6 > it's back to 1.6.2.18.2 for me. > > https://issues.asterisk.org/jira/browse/ASTERISK-18103 > > Regards > > Lee > If it is a regression introduced in 1.6.2.19, then it should still be fixed. At least I believe that's the rules. Regards, Steve -- _ -- 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] Seg Faults with 1.6.2.19
On 18 July 2011 13:00, Lee Archer wrote: > Hi Steve, I think it's related to my ODBC connection. I started with a > random hang where it looked ODBC related which led me to try a few things. > Reloading the config a few times is causing core dumps which 1.6.2.18.2 just > doesn't have, however my main reason for using 1.6.2.19 is a fix to ODBC so I > don't really want to downgrade. I will try and get some traces from one of > my test boxes. > > Thanks > > Lee > I can confirm that we are NOT using ODBC, and that our box does NOT crash, so your theory is still holding up. :) Steve -- _ -- 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] FAX with SIP
On 18 July 2011 12:20, Eduardo Carpes wrote: > Hello guys > I need some help to do works FAX using SIP, anybody know the secret to > this? Have asterisk 1.6. > Thanks!! > > -- > Enviado do meu celular > > Eduardo Carpes > E-mail: car...@bsd.com.br > www.freebsd.org The magic sauce that you need is "T.38" - Asterisk 1.6 supports this to a limited degree, and your ITSP will need to support it. The sip.conf.sample file and the voip-info wiki has all the information you need to try it out. Regards, Steve -- _ -- 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] Seg Faults with 1.6.2.19
On 18 July 2011 12:03, Lee Archer wrote: > Hi, is anyone else having problems with the reload command crashing Asterisk > 1.6.2.19? I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but > 1.6.2.19 after a few reloads is just dumping and restarting. > > Thanks > > Lee > I've not had a problem here with 1.6.2.19. What are you reloading that causes the issue, and can you post the usual gdb backtrace somewhere? Perhaps on the bug tracker. Regards, Steve -- _ -- 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] OT: Google Plus
On Saturday, 9 July 2011, Gordon Henderson wrote: > On Sat, 9 Jul 2011, Steve Davies wrote: > > > On 9 July 2011 12:34, randulo wrote: > > Go ahead and lambast me for this post, it isn't specific to Asterisk, but: > > G+ has only been open at all for a week and I already am chatting with > over 200 people who are into VoIP, Asterisk and all the rest of the > stuff we here care about. If you don't care or are anti-social, fine. > But you owe it to yourself to check it, because a lot of cool VoIP > people are there and after all, Google themselves are doing some > great stuff with VoIP, XMPP and video, and steadily moving towards > open source. Come drink the Kool-Aid! > > > > Can you suggest a good way of finding/following appropriate > VoIP/Asterisk people once on Google+? How do you then group them? Just > in a Circle, or some other mechanism? > > > I've just created a "VoIPy" circle - So I can then invite people I know into > the circle by email address, and/or looking at someone else's circles and > seing if they have something relevant in their summary tag and adding them > into your own circle... (Or using their people search - e.g. for 'randulo' :) > > You can have people in more than one circle. Right now, it's a bit like a > media-rich version of twitter with excellent filtering (the circles). I don't > have camera/microphone/speakers on my PC, (got real desk SIP phones!) so > haven't tried the audio/video chat yet, but the typing "instant messaging" > type chat works just fine. > > I think Google are still slowly gating people into + though. I did have some > invites, but seem to have used them all up now (google didn't tell me how > many, the "invite" button just went away after a while!) > > I'd love to see SIP integration into it, so I can use my existing SIP toys > with it. > > Gordon Thanks for that Gordon. What appears to be missing at the moment is the ability to interface or collaborate with a group of 'strangers'. It would be good if there were a way to broadcast a 'we're here, come join us' to bring a group of VoIP people together, a bit like an IRC channel name can do, or a Facebook fan page. I thought that sparks might cover that, but I'm not entirely sure how sparks work yet. I agree that SIP integration would be great. I think it'll be a while yet but if anyone will allow it, it'll be Google. Cheers, Steve -- _ -- 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] OT: Google Plus
On 9 July 2011 12:34, randulo wrote: > Go ahead and lambast me for this post, it isn't specific to Asterisk, but: > > G+ has only been open at all for a week and I already am chatting with > over 200 people who are into VoIP, Asterisk and all the rest of the > stuff we here care about. If you don't care or are anti-social, fine. > But you owe it to yourself to check it, because a lot of cool VoIP > people are there and after all, Google themselves are doing some > great stuff with VoIP, XMPP and video, and steadily moving towards > open source. Come drink the Kool-Aid! > Can you suggest a good way of finding/following appropriate VoIP/Asterisk people once on Google+? How do you then group them? Just in a Circle, or some other mechanism? Thanks, Steve -- _ -- 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] Aastra phone # key in dialplan
On 22 June 2011 17:09, marvin horst wrote: > I want to use extension numbers that begin with the # key in my dialplan, > but I can't get my Aastra phone (6731i) to transmit the # key to asterisk. > It works fine for the * key. > > I've tried numerous Local Dial Plan patterns in the aastra web configuration > but none of them worked. My current Local dial Plan pattern is > "x+#|xx+*|#x+". Any help would be appreciated. > The following works here: sip dial plan terminator: 1 sip digit timeout: 99 sip dial plan: "X+^" I believe that '#' acts as a send button by default. Regards, Steve -- _ -- 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] iLBC re-licence
On 22 June 2011 17:14, Patrick Lists wrote: > On 06/22/2011 03:32 PM, Steve Davies wrote: >> >> Does anybody know if the updated licence on iLBC makes it safe to >> include in Asterisk when used in a commercial environment again? >> >> https://sites.google.com/site/webrtc/ilbc-freeware >> >> It seems to require that the Google iLBC licence document is on the >> box, but that otherwise it is free-to use by all in any way (BSD >> licence style). I believe that prior to that there was a requirement >> to register every commercial use of the codec with the licence holder, >> or some-such thing? > > IANAL but that's what I also understand from reading the software license. > Seems you just need to ship the license with the binaries and all should be > well. Please also note that the IP Rights Grant seems to note some > limitations that you may want to study before shipping to your customers. > > Regards, > Patrick Yes, it seems to grant the rights to use the iLBC implementation that comes with WebRTC. I'll have to take a look and see how that compares (if at-all) with the code that can be plugged into Asterisk. Cheers, Steve -- _ -- 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] iLBC re-licence
Does anybody know if the updated licence on iLBC makes it safe to include in Asterisk when used in a commercial environment again? https://sites.google.com/site/webrtc/ilbc-freeware It seems to require that the Google iLBC licence document is on the box, but that otherwise it is free-to use by all in any way (BSD licence style). I believe that prior to that there was a requirement to register every commercial use of the codec with the licence holder, or some-such thing? Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
On 9 June 2011 15:49, satish patel wrote: >> Date: Wed, 8 Jun 2011 18:15:14 +0100 >> From: davies...@gmail.com >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI >> >> On 8 June 2011 17:20, satish patel wrote: >> > Interesting thing is when i reload sip.conf i got MWI lamp working on >> > polycom 501 >> > >> > But its not working when anyone leave voicemail. Do you know its some >> > timeout or polling setting in sip.conf ? >> > >> > Still my question is my my asterisk not sending NOTIFY message ? Do i >> > need >> > to subscribe my phone to asterisk ? >> > >> >> Does this help? >> >> https://issues.asterisk.org/jira/browse/ASTERISK-17866 >> >> Regards, >> Steve >> > Thanks steve, > > But you know if i connect X-lite softphone my asterisk sending NOTIFY . > > But its not sending NOTIFY to polycom 501 phone ? Do you think i need to > subscribe my phone to asterisk ? > > -S > X-Lite automatically SUBSCRIBEs for MWI indication. Polycom and snom do not do this by default, instead they assume that the REGISTER will automatically cause MWI notifications. chan_sip changed behaviour (by accident I suspect) somewhere between version 1.2 and 1.6, and the patch basically puts back what went missing. It is crude, but has not caused me any problems so far. Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 broken MWI
On 8 June 2011 17:20, satish patel wrote: > Interesting thing is when i reload sip.conf i got MWI lamp working on > polycom 501 > > But its not working when anyone leave voicemail. Do you know its some > timeout or polling setting in sip.conf ? > > Still my question is my my asterisk not sending NOTIFY message ? Do i need > to subscribe my phone to asterisk ? > Does this help? https://issues.asterisk.org/jira/browse/ASTERISK-17866 Regards, Steve -- _ -- 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] Refactor of CDR - Comments please.
Hi, Since raising this ticket about broken CDR data: https://issues.asterisk.org/jira/browse/ASTERISK-17826 I have been researching how CDR records work in various circumstances. CEL will do most things that people want, but that does not change that CDR records are likely to persist into future versions, and should be as correct as possible. Is this something people want? Is it "worth the risk" to change CDRs again? I am working on a patch within 1.6.2, and if it is considered worthwhile I will clean-up and migrate the patch to 1.8 and put it on the review-board It struck me that the fundamental issue was the "bridge_cdr" construct that appears to have been added in version 1.4. While this solved the problems it was aimed at solving, it appears to have caused other problems, and perhaps even worsened support of CDR records when channels are transferred/masqueraded. The "special cdr reset" feature seems to be able to clear out valuable CDRs that are no-longer related to the bridge when it ends, and copying bridge_cdr back into chan->cdr often overwrites other valuable CDR data! A classic example of the current failing is: - A UK based receptionist calls Australia. - Then a 2nd call leg is placed UK to America. - The receptionist bridges the 2 calls. The desired outcome would be 2 CDR records detailing the full duration of both calls for billing purposes. Sadly, at present, one of the 2 CDRs stops when the transfer happens because Asterisk sees only one bridge in progress. Here is the outline of my solution: 1) There are 2 basic types of CDR a) A CDR that tracks a running PBX/dialplan. b) A CDR that tracks an outbound dialled channel. 2) Channels can be bridged and re-bridged. When bridging, the existing code provides a way to merge caller (PBX) and callee (Dialed) CDRs, and for consistency this will be maintained as closely as possible. 3) A CDR on a dialled channel should track that channel for its lifetime. 4) A CDR that tracks a running PBX should track that PBX for its lifetime or until merged into a dialled call. Side-note: 3) and 4) Affect the way cdr data is masqueraded. 5) Aim: Valid CDR output should be changed minimally over the current system. 6) Aim: NoCDR, ResetCDR and ForkCDR etc will remain untouched and function the same. 7) A bridge CDR will be un-needed, instead, the bridged CDR will be stored the dialed channel. 8) It is intended that all call legs will have a CDR, which will accurately reflect the duration of that call leg, but no attempt will be made to record transfers/masquerades beyond current mechanisms. CEL data should provide that additional information if needed. The changes to achieve the above are surprisingly minor (relatively speaking). I am testing all of the cases that I can think of: - Simple call in. - Simple call out. - Call-out, then blind transferred by caller and callee. - Call-out, then att. transferred by caller and callee. - AMP or call-file originated call. - Masq-away of Local channel. - Bridge() channels in the dialplan. - Feature park of calls. - Local/ channel calls. - Feature transfer of calls. - Transfer call to IVR/Playback - Transfer IVR/Playback to a handset - Feature Pickup and Pickup() - SIP blind xfer - SIP att xfer - SIP xfer to ringing channel Thoughts? Many thanks, Steve -- _ -- 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] busy hangup HDLC Bad FCS (8) on Primary D-channel
On 1 June 2011 15:10, randall wrote: > On 06/01/2011 03:55 PM, randall wrote: >> On 06/01/2011 03:41 PM, Tzafrir Cohen wrote: >>> On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote: Hi all, After running fine for a few months now asterisk seems to hang frequently , still functioning but the DAHDI channels seem busy (users report a busy signal when calling or being called) A reboot will allow it to run for another day or maybe 2 or 3 till the problem occurs again. running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel 2.6.32-5-686 i get the following errors: pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2 (happens on all 4 spans) and the following in dmesg: [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252 [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX: current packet[0..2]: 55 55 FC [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71 Channel 0/1, span 1 got hangup, cause 18 >>> >>> Is this happening in the middle of a call? Or only a while after the >>> call ended? >>> >> >> the "bad fcs" messages seem to happen random > there seems to be a relation indeed, have seen them happen randomly > quite spurious, but they indeed tend to happen a while after the call is > made. >> >> the hangup happens when a call through DAHDI is attempted, >> (usually after it has been working fine for a while a day or 2) In my experience, FCS errors are caused by line quality issues, and usually (not always) are in the telco's equipment. If they are only happening occasionally, it may be a marginal, but mostly-OK signal on the wire. Do you also get occasional poor-quality audio on calls? The issue will happen more when a call is being setup, or is progressing because there are more frames being exchanged when a call is in progress. I have also seen a bad component or dry solder on a voice card cause this, and even a badly made ISDN cable can be part of the problem. If none of that helps, I would ask the telco to put a trace on the line. Hope that helps, Steve -- _ -- 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] AstManProxy
On 24 May 2011 10:43, Patrick Lists wrote: > On 05/24/2011 11:02 AM, Steve Davies wrote: > [snip] >> >> I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most >> recent version is on Github, and is not that old. In fact that reminds >> me that I really must upload my latest changes to my github fork of >> the project! > > The last update to Dave Troy's version was in December 2008 and the last > update to your repo was more than a year ago. Not very old but not crispy > fresh either :) > > For the archives, Dave Troy's astmanproxy repo can be found here on github: > https://github.com/davetroy/astmanproxy > > And Steve Davies' fork: https://github.com/davies147/astmanproxy > > I look forward to seeing your repo updated with your latest changes. > Repo updated. I have tried to merge all of the other changes that are "out there" also. Regards, Steve -- _ -- 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] AstManProxy
On 20 May 2011 16:16, Ishfaq Malik wrote: > On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: >> On 11-05-20 09:37 AM, Ishfaq Malik wrote: >> > Do many people use this? >> > Is it reliable and safe? >> >> It may still work, but that code is quite old, and I'm not even sure it's >> necessary any more. >> >> Leif. >> I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most recent version is on Github, and is not that old. In fact that reminds me that I really must upload my latest changes to my github fork of the project! > The reasons I'm considering it are as follows: > Building a web page which uses AJAX to get information from the AMI > every 10-30 seconds or so and not wanting to log on and off via AMI that > many times. > > We'll soon be using multiple asterisk servers so having a single point > of access would be very useful. Astmanproxy does seem to offer that, but it is not a feature I've used. > I'd love for you to elaborate on why it's not necessary any more, is > there something simple I've overlooked? - I run asterisk in high-priority mode, meaning that every AMI connection gets a "high priority socket" - Astmanproxy runs at a normal priority - That just feels like a nicer thing to be doing :) - If a client fails to read its net socket fast enough, it affects astmanproxy, and not asterisk with the backlog.. Again, that just feels like a better way to do it. - Astmanproxy offers filtering, which can be used to either reduce load on the client, to present information about a single device/channel only, or as a sanity layer to block or fix-up invalid requests by a client. Hope that helps. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering
On 6 May 2011 16:30, Eric Wieling wrote: >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >> Cassius Smith >> Sent: Friday, May 06, 2011 11:23 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not >> registering >> >> Hi all, >> I have a production server running with about 90 Cisco >> 79[46]1's and SIP release 8.5(2)SR1 from last year. I was >> running Asterisk 1.6.2.9 and upgraded last night after hours. >> (Seemed low risk to me!) >> >> Much to my surprise, not a single one of the Cisco 79XX >> phones would register. Since it's a production server, I >> rolled back to 1.6.2.9 and everything was fine. All my >> Linksys SPA phones and Polycom speaker phones registered just fine. >> >> I am now setting up test servers with both 1.6.2.18 and >> 1.8.3.3 to collect some debug. >> >> I am just curious - has anyone else had SIP issues with these >> phones and updating Asterisk broke them? >> >> I will post results of my findings after I have time to collect them. >> >> Cassius Smitha >> > > I seem to recall this issue mentioned on asterisk-dev. Check > issues.digium.com and see if there is anything similar to your issue. > I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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] IAX2 codec selection and video
Hi, Can anyone let me know how I can enable video (h.263) on SIP, but if a video call is passed over IAX, it will remove the video and pass the audio only. What I tried was: SIP - videosupport=yes - disallow=all - allow=alaw - allow=h263 IAX - disallow=all - allow=alaw What appears to occur is that the SIP call negotiates h263 video, and when passed over IAX, the h263 frames are passed, and are also accepted at the far end which also does not have a video codec allowed. Should that be happening? This is with 1.6.2.18-rc1. Am I missing a setting somewhere? Thanks, Steve -- _ -- 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] Possible bug in Hangup() (Asterisk 1.4.x)
On 15 April 2011 13:02, Vlasis Hatzistavrou wrote: > Hello, > > On an Asterisk 1.4.33.1 in a simple scenario: > > [test] > exten => _X.,1,Dial(SIP/12345@peer01,,,) > > exten => i,1,Hangup(${HANGUPCAUSE}) > exten => t,1,Hangup(${HANGUPCAUSE}) > exten => h,1,Hangup(${HANGUPCAUSE}) > > > I have noticed that no matter what value we set in the Hangup() > commands, if the call is not answered by peer01 for any reason, the actual > cause code returned to the calling party is a 503, no matter what the > ${HANGUPCAUSE} is. > > Even if we set a fixed value like Hangup(1) (which should give a 404) or > Hangup(17) (which should give a 486), the cause code returned is always a > 503. > > Has anyone else noticed this? I went through the issue tracker but I > couldn't find any relevant bug posted in the past. I am certain that in > previous versions I could set the reply message to the desired value, so I > wonder if this is a bug in this particular version (1.4.33.1). > Strictly speaking you can only Hangup (BYE) an answered and fully established call. In SIP terms, a hangup that occurs before an answer is a CANCEL, and I believe a CANCEL is always represented by a 503 code in chan_sip. Regards, Steve -- _ -- 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] No ringback even though progressinband=yes is set
On 7 April 2011 23:04, Douglas Mortensen wrote: > Steve. Thanks for the insight. I won't pretend to know what "early-audio" is, > but I guess I'm about to find out :-). > > Also, I believe that I have a nearly identical setup like this with the exact > same SIP provider w/o any trouble. However, I think that system must be > running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to > confirm). Is there a significant difference between 1.2/1.4 & 1.6 in this > scenario? > > Thanks a million!! :-) > > - > Doug Mortensen > Network Consultant > Impala Networks > P: 505.327.7300 > . > > > -Original Message- > From: Steve Davies [mailto:davies...@gmail.com] > Sent: Thursday, April 07, 2011 10:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] No ringback even though progressinband=yes is > set > > On 7 April 2011 17:02, Douglas Mortensen wrote: >> Any ideas on why callers who call into my customer's SIP trunk are not >> hearing a ringback tone? I had this on one other asterisk system, and wound >> up needing to set progressinband=yes in the SIP trunk config. >> >> I have set this on the current system & restarted asterisk, but to no avail. >> >> I am using: >> >> AsteriskNOW distro >> Asterisk build is 1.6 from AsteriskNOW repository: >> asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9 >> >> Any help would be greatly appreciated! :-) >> >> - >> Doug Mortensen > > > In my personal experience with SIP and 1.6.x, that mostly depends on where > you are sending the call to. It depends on whether the next or subsequent leg > tries to use early-audio for the ring tone, or uses a Ringing event to signal > that is what is happening. It then depends on whether the originating > caller's equipment can understand early-audio ringing. > > We have a setup here where all our trunks support early-audio ringing except > one (an ISDN30 circuit) and we have to juggle things a bit sometimes to > ensure ringing occurs. > > Perhaps provide more details? Or you may find that tracing the SIP gives you > the clue that you need. > > Hope that helps, > Steve > > Early audio is audio that is sent before the call is "answered", usually in the form of a custom ring-tone or perhaps a "cannot connect, try later" message. Some systems do not support it as it can be abused to communicate at least basic information for free. We had a problem with this when connecting Asterisk 1.2 to Asterisk 1.6 via IAX. A 1.2 SIP system will automatically switch into early audio if it sees an early audio frame. 1.6 defaults to not doing this, but there is a parameter to re-enable it. In this case we solved the problem by upgrading to 1.6 everywhere :) Regards, Steve -- _ -- 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] No ringback even though progressinband=yes is set
On 7 April 2011 17:02, Douglas Mortensen wrote: > Any ideas on why callers who call into my customer's SIP trunk are not > hearing a ringback tone? I had this on one other asterisk system, and wound > up needing to set progressinband=yes in the SIP trunk config. > > I have set this on the current system & restarted asterisk, but to no avail. > > I am using: > > AsteriskNOW distro > Asterisk build is 1.6 from AsteriskNOW repository: > asterisk16-1.6.2.17.2-1_centos5 > FreePBX 2.9 > > Any help would be greatly appreciated! :-) > > - > Doug Mortensen In my personal experience with SIP and 1.6.x, that mostly depends on where you are sending the call to. It depends on whether the next or subsequent leg tries to use early-audio for the ring tone, or uses a Ringing event to signal that is what is happening. It then depends on whether the originating caller's equipment can understand early-audio ringing. We have a setup here where all our trunks support early-audio ringing except one (an ISDN30 circuit) and we have to juggle things a bit sometimes to ensure ringing occurs. Perhaps provide more details? Or you may find that tracing the SIP gives you the clue that you need. Hope that helps, Steve -- _ -- 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] SIP channel able to add codecs once up and running?
>From my observations, if a video capable device starts the call in non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with exceptions for T.38 fax) as any SDP update is masked using the existing 'jointcapability' and knocks out the newly requested codec. Is that right? Thanks, Steve -- _ -- 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 from '"000000" x 1000
On 2 April 2011 09:46, Jonas Kellens wrote: > Hello list, > > I often see the following in my message log : > > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > [Apr 2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '"00" > ' failed for '184.106.109.168' - No matching peer found > > And there are hundreds of them... > > > Is there a setting so I can make Asterisk not respond to SIP PEER > registrations which are not in my sip.conf or my realtime MySQL DB ?? Yes, you add a rule to your firewall! Even better, get it filtered further out so that it does not waste your inbound Internet bandwidth, because in my experience, once those SIP spammers start, they continue for weeks at the very least. IIRC, the way SIP registrations works basically requires than an failed/un-authorised attempt is responded to, so that the other party knows to authenticate. If you stop sending that response, no-one can authenticate. Hope that helps. Steve -- _ -- 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] DAHDI, IAX2 and SIP considerations for Early-Media / Alerting
Hi, Short version: Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA indication into a DAHDI/q.931 ALERTING signal when your ISDN provider does not pass early media on receipt of an PROGRESS(8) indication? Long version: I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1 line), also, the system has IAX2 trunks, and several SIP handsets. All 3 protocols (q.931/IAX2/SIP) have a mechanism to indicate either ALERTING/RINGING, or to specify PROGRESS/EARLY-MEDIA. Based on this you'd think call setup would all work happily all of the time :) What happens based on the call direction is as follows: SIP -> DAHDIISDN returns ALERTING, SIP uses 180 Ringing, all OK SIP -> IAX2 IAX2 returns PROGRESS, SIP uses 183 Progress, early audio works OK IAX2 -> DAHDI ISDN returns ALERTING, IAX2 uses RINGING, all OK IAX2 -> SIP SIP returns 180 ringing, IAX2 uses RINGING, all OK DAHDI -> SIPSIP returns 180 ringing, ISDN uses ALERTING, all OK DAHDI -> IAX2 IAX2 returns PROGRESS, ISDN uses PROGRESS(8), but the caller hears no ringing. I believe that my issue is that my UK ISDN provider does not accept early media, and will simply send silence instead of using the provided early audio stream. DAHDI is configured with: priindication=outofband The IAX2 trunk provider is using early-media to send the ringing tone, and as above, this mostly seems to work okay. The exception is when the call is bridged to ISDN, where I believe the ISDN provider does not pass on early media. I checked the IAX2 RFCs 5456/5457, but cannot find a definition of how RINGING/PROGRESS is meant to work. Is my IAX2 trunk provider doing something wring by not also sending RINGING? Is there a workaround that converts either IAX2 PROGRESS into RINGING, or allows DAHDI to send ALERTING if it receives an early media indication? I suspect the code to do the latter would be reasonably simple, but would appreciate pointers for any badness that it may cause. Thanks in advance for any suggestions. Regards, Steve -- _ -- 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] One Way Audio
On 10 March 2011 11:17, Ishfaq Malik wrote: > Just fixed our problem with > > directmedia=no > > but this only applies if your extensions are behind a nat > > Ish > There are several reasons why "directmedia=no" might be the correct configuration. 1) NAT - probably the most common reason 2) Routing - Sometimes devices cannot route to each other directly 3) ITSP calls. Many SIP providers will not accept a redirect and I am sure there are many more... Cheers, Steve -- _ -- 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] IAX channel name incorrect - Found in 1.2 still happens in 1.6
*Bump* No takers? Perhaps no-one else thinks this is a bug? Regards, Steve On 7 February 2011 16:45, Steve Davies wrote: > Hi, > > The following IAX config (slightly edited) causes an issue for me in > version 1.6.2.16.1, where my CDR data is unreliable. > > [user1] > type=friend > auth=md5 > accountcode=user1 > notransfer=yes > context=context1 > host=10.0.0.250 > username=user1 > secret=secret1 > disallow=all > allow=alaw > > [user2] > type=friend > auth=md5 > accountcode=user2 > notransfer=yes > context=context2 > host=dynamic > deny=0.0.0.0/0.0.0.0 > permit=10.0.0.0/24 > username=user2 > secret= > disallow=all > allow=alaw > > If a call comes in from 10.0.0.250, using username "user2" and with no > password, then it is correctly authenticated against the [user2] > section. > Accountcode is set to user2 > Context is set to context2 > and the call mostly proceeds correctly, BUT the source channel name is > set to IAX2/user1-, which is then seen both in the dialplan debug > output, and in the CDR. I would expect the channel name to reflect the > section name that was used to authenticate the call ie. > IAX2/user2-; I specifically put a password onto [user1] so there > is no possibility that the call is authenticating there. > > Am I missing something? Or is this a bug? > > Thanks, > Steve > -- _ -- 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] Barge in.
On 16 February 2011 10:13, Peter den Hartog wrote: > I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge > in" solution running. > I was thinking of using chanspy, but i would like that the original call > would be dropped, and the new call would be the only one there. What you are describing looks to me like a third party controlled transfer, and not a barge-in at all. I suspect that the Asterisk Manager API action "Redirect" will be your friend. Regards, Steve -- _ -- 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] Aastra phones cannot transfer calls?
On 16 February 2011 00:22, Ernie Dunbar wrote: >> At 12:12 PM 2/15/2011, you wrote: >>>I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk >>>v1.6.2.1. They can call each other's extensions (and make and receive >>>calls otherwise), but they cannot transfer calls, not even to outside >> >> I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had >> no problem at all with transfers. Have you considered trying a newer >> version? >> > > Nope. Upgraded to 1.6.2.16.1, and I still see the same effect. > > It may be a setting on the phone or a SIP setting. I'll investigate this > elsewhere but report back about the solution. > > I also tried this with a 6757i and a 6753i with no problems (blind and attended) on Asterisk 1.6.2.16.1. Have you updated the handset firmware to 2.6.0.2010? Cheers, Steve -- _ -- 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] IAX channel name incorrect - Found in 1.2 still happens in 1.6
Hi, The following IAX config (slightly edited) causes an issue for me in version 1.6.2.16.1, where my CDR data is unreliable. [user1] type=friend auth=md5 accountcode=user1 notransfer=yes context=context1 host=10.0.0.250 username=user1 secret=secret1 disallow=all allow=alaw [user2] type=friend auth=md5 accountcode=user2 notransfer=yes context=context2 host=dynamic deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/24 username=user2 secret= disallow=all allow=alaw If a call comes in from 10.0.0.250, using username "user2" and with no password, then it is correctly authenticated against the [user2] section. Accountcode is set to user2 Context is set to context2 and the call mostly proceeds correctly, BUT the source channel name is set to IAX2/user1-, which is then seen both in the dialplan debug output, and in the CDR. I would expect the channel name to reflect the section name that was used to authenticate the call ie. IAX2/user2-; I specifically put a password onto [user1] so there is no possibility that the call is authenticating there. Am I missing something? Or is this a bug? Thanks, Steve -- _ -- 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] queue_log in MySQL database
On 13 January 2011 16:28, Jonas Kellens wrote: > > > I actually found this : > http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL > > But a second question : > > how can I know how long a caller stayed inside the queue untill it was > answered by a member ?? > > The queue_log table contains exactly that information - Along with a few other events, it indicates when a caller joined a queue, and when an agent gets given the call. Take the difference between the 2 times and you have the number that you need. Cheers, Steve -- _ -- 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] No MOH with parked call
On 24 December 2010 15:44, Steve Davies wrote: > On 23 December 2010 18:01, Steve Davies wrote: >> Hi Again, >> >> I thought I had this sorted, but it appears that in a clean >> environment I did not in fact fix it. There appears to be a bit of a >> contradiction. >> >> 1) In 1.6.2.x, musiconhold requires DAHDI (which we have) >> 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded... >> >> I am confused. MOH in general terms works just fine, but if I park a >> call with res_timing_dahdi loaded, I get silence after the orbit >> announcement. If I unload res_timing_dahdi, all works fine, but my >> timing sources are less reliable. >> >> I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this >> does not seem to fix things - is the problem elsewhere? Is there a fix >> that I can try, or perhaps backport? > > Further to this, I have been slowly tracing through the codepath for a > parked call - ast_settimer is called correctly for the MOH generator, > and seems to set up the DAHDI timer exactly the same way as it does > for the alternative timing modules, and all of the setup calls return > success. I don't have a verbose output trace to hand, but the sequence > seems to be as follows: > > -- Start moh_files_generator successfully using ast_settimer() > > -- Stop moh_files_generator successfully > -- Play the parking position numbers > -- Start moh_files_generator using ast_settimer() > > > Any thoughts where to look next? It is odd that it appears to be > working immediately before the position is read out, but then fails to > restart afterwards... > Thanks to Lee pointing me in the right direction, I discovered https://issues.asterisk.org/view.php?id=18262 Which seems to fix this issue. Many thanks, Steve -- _ -- 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] One way crappy audio in iax call - Asterisk 1.6.2.15
On 24 December 2010 14:40, Administrator TOOTAI wrote: > Hi, > > We had 2 asterisk 1.4 connected together in iax, all was fine. One of them > was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 > > When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling > from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you > can't understand the words). On callee party it's still good. > > We replace 1.6.2 version with 1.4.38 and everything is going back to normal, > good audio on both side does'nt matter who call. > > I already opened another thread about problem with iax and Asterisk 1.6.2 > (rsa auth not working anymore). Are there some known problems with iax and > 1.6 version of Asterisk? > > Thanks for any hint > Not 100% sure, but I think there was a fix for IAX audio in 1.6.2.16-rc1 - Perhaps try that? Regards, Steve -- _ -- 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] No MOH with parked call
On 23 December 2010 18:01, Steve Davies wrote: > Hi Again, > > I thought I had this sorted, but it appears that in a clean > environment I did not in fact fix it. There appears to be a bit of a > contradiction. > > 1) In 1.6.2.x, musiconhold requires DAHDI (which we have) > 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded... > > I am confused. MOH in general terms works just fine, but if I park a > call with res_timing_dahdi loaded, I get silence after the orbit > announcement. If I unload res_timing_dahdi, all works fine, but my > timing sources are less reliable. > > I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this > does not seem to fix things - is the problem elsewhere? Is there a fix > that I can try, or perhaps backport? Further to this, I have been slowly tracing through the codepath for a parked call - ast_settimer is called correctly for the MOH generator, and seems to set up the DAHDI timer exactly the same way as it does for the alternative timing modules, and all of the setup calls return success. I don't have a verbose output trace to hand, but the sequence seems to be as follows: -- Start moh_files_generator successfully using ast_settimer() -- Stop moh_files_generator successfully -- Play the parking position numbers -- Start moh_files_generator using ast_settimer() Any thoughts where to look next? It is odd that it appears to be working immediately before the position is read out, but then fails to restart afterwards... Thanks, Steve -- _ -- 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] No MOH with parked call
On 7 December 2010 17:47, Steve Davies wrote: > On 7 December 2010 15:00, Steve Davies wrote: >> On 7 December 2010 14:17, Lee Archer wrote: >>> Hi, try unloading res_timing_dahdi.so then trying again. >>> >>> Lee >>> >>> -Original Message- >>> From: asterisk-users-boun...@lists.digium.com >>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve >>> Davies >>> Sent: 07 December 2010 12:54 >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [asterisk-users] No MOH with parked call >>> >>> Hi, >>> >>> Has anybody else noticed that MOH does not play on parked calls in >>> 1.6.2.14? Or is it just my setup? MOH seems to work in every other >>> respect (Call Held or in-Queue), but when a call is parked, the logs >>> show MOH being started, but the parked party hears nothing. >>> >>> The verbose logs show the following. Any thoughts on whet to check next? >>> >>> Thanks, >>> Steve >>> >>> >>> ### Call comes in here and is answered >>> -- SIP/snom360-0d6f answered DAHDI/2-1 >>> -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f", >>> "0?done") in new stack >>> -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f", >>> "CHANNEL(musicclass)=m-default") in new stack >>> -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f", >>> "") in new stack >>> >>> ### Here the call is being blind transferred to the Park number >>> -- Started music on hold, class 'default', on DAHDI/2-1 >>> -- Stopped music on hold on DAHDI/2-1 >>> == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1' >>> -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack >>> -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in >>> new stack >>> >>> ### Not sure why I send "Ringing" here, but I tried NoOP() and >>> Answer() too just in case >>> -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack >>> -- Executing [...@local:4] Set("DAHDI/2-1", >>> "CHANNEL(musicclass)=default") in new stack >>> -- Executing [...@local:5] Set("DAHDI/2-1", >>> "CHANNEL(parkinglot)=default") in new stack >>> -- Executing [...@local:6] Goto("DAHDI/2-1", >>> "parkedcalls_default,park,1") in new stack >>> -- Goto (parkedcalls_default,park,1) >>> -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in >>> new stack >>> == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to >>> extension [parkedcalls_default] s, 1 in 90 seconds >>> -- Added extension '211' priority 1 to parkedcalls_default >>> (0xbe2e528) >>> >>> # The "211" announcement is heard perfectly >>> -- Playing 'digits/2.alaw' (language 'en') >>> == Extension Changed 211[extensions] new state InUse for Notify User >>> steve >>> -- Playing 'digits/1.alaw' (language 'en') >>> -- Playing 'digits/1.alaw' (language 'en') >>> >>> # The system claims to start MOH "default" which works elsewhere, but >>> the caller gets silence >>> -- Started music on hold, class 'default', on DAHDI/2-1 >>> == Spawn extension (parkedcalls_default, s, 1) exited non-zero on >>> 'Parked/DAHDI/2-1' >>> >> >> Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but >> it has killed call quality on ISDN calls - I think it interferes with >> the software echo canceller somehow. >> >> Is there a ticket open on this? A patch to try? >> >> Thanks, >> Steve > > For anyone searching/finding this issue, the patch here: > > https://issues.asterisk.org/view.php?id=17726 > > Applies to 1.6.2 with only a trivial tweak, and with minimal testing > is working here. We now get music on hold when a call is parked, even > when we are using res_timing_dahdi.so - Call quality remains high > under these circumstances too. > Hi Again, I thought I had this sorted, but it appears that in a clean environment I did not in fact fix it. There appears to be a bit of a contradiction. 1) In 1.6.2.x, musiconhold requires DAHDI (which we have) 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded... I am confused. MOH in general terms works just fine, but if I park a call with res_timing_dahdi loaded, I get silence after the orbit announcement. If I unload res_timing_dahdi, all works fine, but my timing sources are less reliable. I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this does not seem to fix things - is the problem elsewhere? Is there a fix that I can try, or perhaps backport? Thanks for any pointers. Regards, Steve -- _ -- 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 hangs up call after 20s
On 22 December 2010 12:44, Gilles wrote: > Hello > > I have an Asterisk 1.4 server and two XLite softphones, where > Asterisk and the local XLite phone are located in a LAN behind a NAT > router, and the remote XLite phone is located elsewhere on the Net > behind its own NAT router: > > http://img252.imageshack.us/img252/3749/asterisknat.png > > I'm having the following issue: When the _local_ XLite calls out the > remote XLite, everything works fine; However, when the _remote_ XLite > calls the local XLite, things work OK until precisely 20s, where > Asterisk decides to hang up, and displays the following error message > in the console: > > == > WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on > transmission > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno > 2 (Critical Response) > > WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no > reply to our critical packet. > == Spawn extension (my-phones, local-xlite-extension, 1) exited > non-zero on 'SIP/unused-008008e4' > == > > I'm no SIP expert, but based on the debug session, before deciding to > hang up, Asterisk tries 6 times to send an OK message to the remote > XLite, and doesn't seem to get an answer. FWIW, after Asterisk has > hung up, the remote XLite remains off-hook, oblivious to this error > and keeps displaying "Call established": > > www.pastebin.com/x6MgnrpG > > There's also this oddity on line 50: "Scheduling destruction of SIP > dialog". > > FWIW, in sip.conf, for the remote XLite user, I tried "nat=no" and > "nat=yes", with no difference. I'm actually not sure how to configure > a remote user which happens to be listed in sip.conf (it's behind a > NAT router but it registers with Asterisk, so... is it NATed or not?), > and am surprised it actually rings and sends/receives voice with no > problem, regardless of this parameter. > > I found discussions about using "t1min=500" in sip.conf, but it made > no difference either. > > Has someone already experienced this and knows what can be done? > > Any hint much appreciated. Look in the XLite advanced network settings and disable the 2 timeout settings (RTP and RTCP?). This is not always necessary, but there are sufficient cases where the packets XLite expects appear early on, but do not persist, thus causing a hangup. I think the default timeout is 20 seconds. Cheers, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.
On 21 December 2010 22:06, Tilghman Lesher wrote: > On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote: >> We have an issue with our Asterisk install where Asterisk produces many >> Zombie processes (on the order of several hundred per minute) until >> either the Asterisk server is restarted (and the zombies die a natural >> death), or the kernel runs out of PID space (happens within hours) and >> brings the system to a halt. >> >> This problem only happens when the server is under some non-trivial >> load. We were testing this server with 8 SCCP phones, making up to five >> simultaneous calls through the DAHDI interface (a Digium Wildcard >> TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients) >> start logging on and we get around 7 or 8 simultaneous DAHDI calls, >> Asterisk starts producing zombie processes at a high rate. > > I know what the issue is. Please open a report on > https://issues.asterisk.org and I'll get a patch uploaded pronto. > Please let us know the issue number once raised - I'd like to follow this one. Regards, Steve -- _ -- 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] UDP buffer overflows?
On 10 December 2010 17:33, Steve Davies wrote: > On 10 December 2010 17:21, Shaun Ruffell wrote: >> On 12/10/2010 11:02 AM, Steve Davies wrote: >>> On 10 December 2010 16:45, Steve Davies wrote: >>>> Hi, >>>> >>>> On one of our asterisk systems that is quite busy, we are seeing the >>>> following from 'netstat -s': >>>> >>>> Udp: >>>> 17725210 packets received >>>> 36547 packets to unknown port received. >>>> 44017 packet receive errors >>>> 17101174 packets sent >>>> RcvbufErrors: 44017 <--- this >>>> >>> [snip] >>>> >>>> net.core.rmem_max = 1048575 >>>> net.core.wmem_max = 1048575 >>>> net.core.rmem_default = 1048575 >>>> net.core.wmem_default = 1048575 >>>> net.core.optmem_max = 1048575 >>>> net.core.netdev_max_backlog = 1 >>>> >>> >>> Additional question - Do I need to restart Asterisk for these settings >>> to apply to SIP? I have not done so yet, and further reading suggests >>> that this is a per-process buffer and may be assigned when the >>> listener is created. >>> >> >> I had to double check myself in the kernel sources (in >> net/core/sock.c:sock_init_data), but yes you will need to recreate the >> socket for the rmem_default option to take effect. >> >> I was just about to ask if changing the sysctl changed the frequency >> that you see the error counter increment when I saw your follow on. >> >> Cheers, >> Shaun > > Thanks for making the extra effort Shaun :) I will restart Asterisk > when I get a chance, and re-check. > > Regards, > Steve > False alarm? FYI, it seems that the issue is not at-all what we thought. It transpires that the unit is an old Asterisk 1.2 system, and that the DNS server that the unit had been pointed to had been taken out of service. This caused lots of threads to back-up while attempting to resolve things, and this dragged down the whole of asterisk, seemingly to the point where it was not reading UDP data out of its buffers fast enough! Thanks, Steve -- _ -- 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] UDP buffer overflows?
On 10 December 2010 17:21, Shaun Ruffell wrote: > On 12/10/2010 11:02 AM, Steve Davies wrote: >> On 10 December 2010 16:45, Steve Davies wrote: >>> Hi, >>> >>> On one of our asterisk systems that is quite busy, we are seeing the >>> following from 'netstat -s': >>> >>> Udp: >>> 17725210 packets received >>> 36547 packets to unknown port received. >>> 44017 packet receive errors >>> 17101174 packets sent >>> RcvbufErrors: 44017 <--- this >>> >> [snip] >>> >>> net.core.rmem_max = 1048575 >>> net.core.wmem_max = 1048575 >>> net.core.rmem_default = 1048575 >>> net.core.wmem_default = 1048575 >>> net.core.optmem_max = 1048575 >>> net.core.netdev_max_backlog = 1 >>> >> >> Additional question - Do I need to restart Asterisk for these settings >> to apply to SIP? I have not done so yet, and further reading suggests >> that this is a per-process buffer and may be assigned when the >> listener is created. >> > > I had to double check myself in the kernel sources (in > net/core/sock.c:sock_init_data), but yes you will need to recreate the > socket for the rmem_default option to take effect. > > I was just about to ask if changing the sysctl changed the frequency > that you see the error counter increment when I saw your follow on. > > Cheers, > Shaun Thanks for making the extra effort Shaun :) I will restart Asterisk when I get a chance, and re-check. Regards, Steve -- _ -- 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] UDP buffer overflows?
On 10 December 2010 16:45, Steve Davies wrote: > Hi, > > On one of our asterisk systems that is quite busy, we are seeing the > following from 'netstat -s': > > Udp: > 17725210 packets received > 36547 packets to unknown port received. > 44017 packet receive errors > 17101174 packets sent > RcvbufErrors: 44017 <--- this > [snip] > > net.core.rmem_max = 1048575 > net.core.wmem_max = 1048575 > net.core.rmem_default = 1048575 > net.core.wmem_default = 1048575 > net.core.optmem_max = 1048575 > net.core.netdev_max_backlog = 1 > Additional question - Do I need to restart Asterisk for these settings to apply to SIP? I have not done so yet, and further reading suggests that this is a per-process buffer and may be assigned when the listener is created. All pointers gratefully received :) Steve -- _ -- 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] UDP buffer overflows?
Hi, On one of our asterisk systems that is quite busy, we are seeing the following from 'netstat -s': Udp: 17725210 packets received 36547 packets to unknown port received. 44017 packet receive errors 17101174 packets sent RcvbufErrors: 44017 <--- this When this number increases, we see SIP errors, and in particular Qualify packets are lost, and temporarily disable handsets, causing all sorts of minor chaos. I have already tuned from the defaults of: net.core.rmem_max = 131071 net.core.wmem_max = 131071 net.core.rmem_default = 111616 net.core.wmem_default = 111616 net.core.optmem_max = 10240 net.core.netdev_max_backlog = 1000 up to: net.core.rmem_max = 1048575 net.core.wmem_max = 1048575 net.core.rmem_default = 1048575 net.core.wmem_default = 1048575 net.core.optmem_max = 1048575 net.core.netdev_max_backlog = 1 with no luck. Any more suggestions? Many thanks, Steve -- _ -- 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] No MOH with parked call
On 7 December 2010 15:00, Steve Davies wrote: > On 7 December 2010 14:17, Lee Archer wrote: >> Hi, try unloading res_timing_dahdi.so then trying again. >> >> Lee >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve >> Davies >> Sent: 07 December 2010 12:54 >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: [asterisk-users] No MOH with parked call >> >> Hi, >> >> Has anybody else noticed that MOH does not play on parked calls in >> 1.6.2.14? Or is it just my setup? MOH seems to work in every other >> respect (Call Held or in-Queue), but when a call is parked, the logs >> show MOH being started, but the parked party hears nothing. >> >> The verbose logs show the following. Any thoughts on whet to check next? >> >> Thanks, >> Steve >> >> >> ### Call comes in here and is answered >> -- SIP/snom360-0d6f answered DAHDI/2-1 >> -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f", >> "0?done") in new stack >> -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f", >> "CHANNEL(musicclass)=m-default") in new stack >> -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f", >> "") in new stack >> >> ### Here the call is being blind transferred to the Park number >> -- Started music on hold, class 'default', on DAHDI/2-1 >> -- Stopped music on hold on DAHDI/2-1 >> == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1' >> -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack >> -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in >> new stack >> >> ### Not sure why I send "Ringing" here, but I tried NoOP() and >> Answer() too just in case >> -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack >> -- Executing [...@local:4] Set("DAHDI/2-1", >> "CHANNEL(musicclass)=default") in new stack >> -- Executing [...@local:5] Set("DAHDI/2-1", >> "CHANNEL(parkinglot)=default") in new stack >> -- Executing [...@local:6] Goto("DAHDI/2-1", >> "parkedcalls_default,park,1") in new stack >> -- Goto (parkedcalls_default,park,1) >> -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in >> new stack >> == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to >> extension [parkedcalls_default] s, 1 in 90 seconds >> -- Added extension '211' priority 1 to parkedcalls_default >> (0xbe2e528) >> >> # The "211" announcement is heard perfectly >> -- Playing 'digits/2.alaw' (language 'en') >> == Extension Changed 211[extensions] new state InUse for Notify User >> steve >> -- Playing 'digits/1.alaw' (language 'en') >> -- Playing 'digits/1.alaw' (language 'en') >> >> # The system claims to start MOH "default" which works elsewhere, but >> the caller gets silence >> -- Started music on hold, class 'default', on DAHDI/2-1 >> == Spawn extension (parkedcalls_default, s, 1) exited non-zero on >> 'Parked/DAHDI/2-1' >> > > Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but > it has killed call quality on ISDN calls - I think it interferes with > the software echo canceller somehow. > > Is there a ticket open on this? A patch to try? > > Thanks, > Steve For anyone searching/finding this issue, the patch here: https://issues.asterisk.org/view.php?id=17726 Applies to 1.6.2 with only a trivial tweak, and with minimal testing is working here. We now get music on hold when a call is parked, even when we are using res_timing_dahdi.so - Call quality remains high under these circumstances too. Thanks for the pointers. Regards, Steve -- _ -- 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] No MOH with parked call
On 7 December 2010 14:17, Lee Archer wrote: > Hi, try unloading res_timing_dahdi.so then trying again. > > Lee > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Davies > Sent: 07 December 2010 12:54 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] No MOH with parked call > > Hi, > > Has anybody else noticed that MOH does not play on parked calls in > 1.6.2.14? Or is it just my setup? MOH seems to work in every other > respect (Call Held or in-Queue), but when a call is parked, the logs > show MOH being started, but the parked party hears nothing. > > The verbose logs show the following. Any thoughts on whet to check next? > > Thanks, > Steve > > > ### Call comes in here and is answered > -- SIP/snom360-0d6f answered DAHDI/2-1 > -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f", > "0?done") in new stack > -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f", > "CHANNEL(musicclass)=m-default") in new stack > -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f", > "") in new stack > > ### Here the call is being blind transferred to the Park number > -- Started music on hold, class 'default', on DAHDI/2-1 > -- Stopped music on hold on DAHDI/2-1 > == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1' > -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack > -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in > new stack > > ### Not sure why I send "Ringing" here, but I tried NoOP() and > Answer() too just in case > -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack > -- Executing [...@local:4] Set("DAHDI/2-1", > "CHANNEL(musicclass)=default") in new stack > -- Executing [...@local:5] Set("DAHDI/2-1", > "CHANNEL(parkinglot)=default") in new stack > -- Executing [...@local:6] Goto("DAHDI/2-1", > "parkedcalls_default,park,1") in new stack > -- Goto (parkedcalls_default,park,1) > -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in > new stack > == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to > extension [parkedcalls_default] s, 1 in 90 seconds > -- Added extension '211' priority 1 to parkedcalls_default > (0xbe2e528) > > # The "211" announcement is heard perfectly > -- Playing 'digits/2.alaw' (language 'en') > == Extension Changed 211[extensions] new state InUse for Notify User > steve > -- Playing 'digits/1.alaw' (language 'en') > -- Playing 'digits/1.alaw' (language 'en') > > # The system claims to start MOH "default" which works elsewhere, but > the caller gets silence > -- Started music on hold, class 'default', on DAHDI/2-1 > == Spawn extension (parkedcalls_default, s, 1) exited non-zero on > 'Parked/DAHDI/2-1' > Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but it has killed call quality on ISDN calls - I think it interferes with the software echo canceller somehow. Is there a ticket open on this? A patch to try? Thanks, Steve -- _ -- 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] No MOH with parked call
Hi, Has anybody else noticed that MOH does not play on parked calls in 1.6.2.14? Or is it just my setup? MOH seems to work in every other respect (Call Held or in-Queue), but when a call is parked, the logs show MOH being started, but the parked party hears nothing. The verbose logs show the following. Any thoughts on whet to check next? Thanks, Steve ### Call comes in here and is answered -- SIP/snom360-0d6f answered DAHDI/2-1 -- Executing [...@macro-set-moh-call:1] GotoIf("SIP/snom360-0d6f", "0?done") in new stack -- Executing [...@macro-set-moh-call:2] Set("SIP/snom360-0d6f", "CHANNEL(musicclass)=m-default") in new stack -- Executing [...@macro-set-moh-call:3] NoOp("SIP/snom360-0d6f", "") in new stack ### Here the call is being blind transferred to the Park number -- Started music on hold, class 'default', on DAHDI/2-1 -- Stopped music on hold on DAHDI/2-1 == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1' -- Executing [...@local:1] ForkCDR("DAHDI/2-1", "") in new stack -- Executing [...@local:2] Set("DAHDI/2-1", "CDR(userfield)=") in new stack ### Not sure why I send "Ringing" here, but I tried NoOP() and Answer() too just in case -- Executing [...@local:3] Ringing("DAHDI/2-1", "") in new stack -- Executing [...@local:4] Set("DAHDI/2-1", "CHANNEL(musicclass)=default") in new stack -- Executing [...@local:5] Set("DAHDI/2-1", "CHANNEL(parkinglot)=default") in new stack -- Executing [...@local:6] Goto("DAHDI/2-1", "parkedcalls_default,park,1") in new stack -- Goto (parkedcalls_default,park,1) -- Executing [p...@parkedcalls_default:1] Park("DAHDI/2-1", "") in new stack == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to extension [parkedcalls_default] s, 1 in 90 seconds -- Added extension '211' priority 1 to parkedcalls_default (0xbe2e528) # The "211" announcement is heard perfectly -- Playing 'digits/2.alaw' (language 'en') == Extension Changed 211[extensions] new state InUse for Notify User steve -- Playing 'digits/1.alaw' (language 'en') -- Playing 'digits/1.alaw' (language 'en') # The system claims to start MOH "default" which works elsewhere, but the caller gets silence -- Started music on hold, class 'default', on DAHDI/2-1 == Spawn extension (parkedcalls_default, s, 1) exited non-zero on 'Parked/DAHDI/2-1' -- _ -- 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] Avoided deadlock Error(solved)
On 25 November 2010 13:02, wrote: > The proble is dialplan I configure fine > -- > Sent from my BlackBerry® > VoIP, Windows/Linux Administration and Network Management > US Numbers: 561-886-0664 > Nicaragua Mobile: +505.8488.6876 > > -Original Message- > From: Stefan Schmidt > Sender: asterisk-users-boun...@lists.digium.com > Date: Wed, 24 Nov 2010 22:59:56 > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [asterisk-users] Avoided deadlock Error > > Am 24.11.2010 13:48, schrieb Bayardo Sanchez: >> My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem >> is : >> >> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided >> deadlock for '0x861f6d8', 9 retries! >> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided >> deadlock for '0x85a6420', 9 retries! >> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided >> deadlock for '0x85bc510', 9 retries! >> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided >> deadlock for '0x85f9e68', 9 retries! >> Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided >> deadlock for '0x85e1db0', 9 retries! >> >> this error comes only when I call spain saturated my CLI with the message >> error >> >> > hello, > > as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch > of this messages on different servers and they dont cause any problem at > all. > > if you have some problems with this (except the warning message) you > should upgrade. > AFAIK, this message is still there in 1.6, but has been downgraded to a DEBUG message, so it is no longer visible most of the time. Regards, Steve -- _ -- 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] Contradiction in GROUP() function
On 24 November 2010 10:12, Steve Davies wrote: > I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: > "Setting a group requires an argument (group name)" > > But the syntax is shown as: "Syntax: GROUP([category])" > > The [category] square brackets indicate to me an "optional" parameter, > which contradicts the error. > > Verison 1.6 is non-committal in its definition, but I always assumed > that an empty string was still a valid category-name, so GROUP()=123 > is as valid as GROUP(X)=123. > > Could this be clarified? I suspect from further reading the code that this might just be a misleading error message. Regards Steve -- _ -- 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] Contradiction in GROUP() function
I am confused. In Asterisk 1.2 and 1.4, in the code there is an error: "Setting a group requires an argument (group name)" But the syntax is shown as: "Syntax: GROUP([category])" The [category] square brackets indicate to me an "optional" parameter, which contradicts the error. Verison 1.6 is non-committal in its definition, but I always assumed that an empty string was still a valid category-name, so GROUP()=123 is as valid as GROUP(X)=123. Could this be clarified? Many thanks, Steve -- _ -- 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 parking question
On 18 November 2010 17:43, Mike wrote: > I tried thator I think I did something similar, but that may or may not > apply (depending on my understanding of parking lots) > > > Here is my relevant contexts. The SIP phones are registered under this > context: > > > [some_context] > include => parkinglotA > include => outboundcalls > > exten => t,1,Verbose(1|parking timeout!!!) > > Here, in features.conf, here is parkinglotA's definition > > [parkinglotA] > context => parkinglotA > parkpos => 101-120 > findslot => next > parkingtime=>60 > > > The thing is, I never hit the "Verbose" command. > > So my questions: > 1) Why won't this work? > And more importantly: > 2) what's this park-dial context you speak of ? Is this a hardcoded context > calls go back to? Can I set one per parkilots (remember: I use multiple > parking lots) > Your call is in the [parkinglotA] context, but you are adding the 't' to your [some_context] context, perhaps the following will work. I have not tried it: [some_context] include => parkinglotA include => outboundcalls [parkinglotA] exten => t,1,Verbose(1|parking timeout!!!) Regards, Steve -- _ -- 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] Licensing of Default MOH
On 22 October 2010 14:24, Miguel Molina wrote: > I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that > came with asterisk. I wonder why the change from the fpm sounds to the > opsound ones, it was a licensing issue? > I think the original 'fpm' files were not as freely licenced as originally thought, so there was a need to change. The details and discussion will be in the mailing list archives, so I won't bother repeating it here. The licences mentioned earlier in this thread all relate to the newer MoH files. Regards, Steve -- _ -- 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] SIP authentication - Thoughts please
On 7 October 2010 10:10, Stefan Schmidt wrote: > Am 07.10.10 10:52, schrieb Steve Davies: >> Hi, >> > > > Hello, > > i just want to say something about point 4 which comes to my mind about > security. > >> >> 4) I am not sure whether it is worth dropping through and testing auth >> against other peers if there is no username match. Can auth ever >> succeed under those circumstances (password matches, but not >> username?) > > If you use UDP its very easy to fake the source ip of a call so do you > really want to open a door to an attacker by authenticate only by ip and > passwort which can match to any peer with the same ip adress? To > bruteforce this would be much easier than to bruteforce against sending > IP, right username and right password. I was not clear. By option 4) I intended that you test the password against other peers with a matching IP address. I am not sure whether the username is included in the SIP password hash, so do not know whether there is even any point in doing so. As far as I can tell, in the EXISTING sip stack, digest username is not used to determine which peer to authenticate with, it just uses the first peer with a matching IP. > Have you tried to use different ports to register? i think this could help. AFAIK, Asterisk will only operate on one port, and the remote end is a major ITSP who will not be wanting to listen to me making odd requests :) Thanks for the feedback! Regards, Steve -- _ -- 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] SIP authentication - Thoughts please
Hi, We have a scenario where we need multiple discrete SIP trunks (peers) from/to a single endpoint. Because the authentication system starts by matching IP address, it only ever matches on one of the SIP peer entries, and ignores the others. This is documented behaviour and as such is "correct". I would like to propose an extension to how SIP peers are authenticated in this case: 1) Initial INVITE arrives, scan the list of all matching peer IP addresses. If a peer with no password is found, proceed with that peer immediately. 2) ...otherwise, a password is required, so send 407 challenge 3) INVITE arrives with Basic-Auth. Scan for /all/ matching peers based on IP address. If one of the matching peers has a section name matching the Basic-Auth username, use it to proceed. 4) I am not sure whether it is worth dropping through and testing auth against other peers if there is no username match. Can auth ever succeed under those circumstances (password matches, but not username?) Thanks for any feedback. Regards, Steve -- _ -- 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] High volume BLF - Suggestions?
On 13 September 2010 19:12, Cassius Smith wrote: > Steve > I have 64 channels being monitored with an SPA962 with two SPA932 > sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very > happy with this. Latest firmware is a must. > > HTH > Cassius Smith > Any chance you could send me the output of http://1.2.3.4/admin/spacfg.xml from that phone? Of course, remember to remove any passwords :) I would like to compare it to mine in case I am doing something stupid! Thanks, Steve -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
On 13 September 2010 16:58, Carlos Chavez wrote: > On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote: >> On 13 September 2010 11:07, Antonio Berrios >> wrote: >> > Gotcha. Yeah, I'm looking at implementing that (searching call >> > recordings by agent that took the call) here but since our asterisk call >> > recording is a separate server to the ones dealing with queues I'll be >> > looking at tie-ing the agent to the call recording via a unique call ID >> > in a database rather than in the filename itself. I'll post my >> > findings/solution to the list if you would like? >> > >> >> Sounds like a plan :) >> >> I may try to modify Asterisk to send more parameters to the >> MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and >> CDR(dstchannel) seem like obvious choices. >> >> I'll post results here if I manage it. >> > What we do is get the ${MEMBERINTERFACE} variable from the Queue and > then rename the file in the h extension. To get this variable you need > to set "setinterfacevar=yes" in your queue definition. We then run an > AGI from the h extension to rename and move the file. > Excellent. thank you. :) Steve -- _ -- 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] High volume BLF - Suggestions?
On 13 September 2010 12:16, Stefan Schmidt wrote: > Hello, > > Am 13.09.10 11:56, schrieb Steve Davies: >> Hi, >> >> We have a user who is putting large call volumes through Asterisk, and >> wants to BLF monitor up to 90 extensions. We are struggling to find a >> handset that can keep up with Asterisk :) >> >> 1) Is there a handset that will do this? > we only use the spa962 and spa525 for this, but didnt have any problems > so far. We find that the spa962 works pretty well with one sidecar (32 buttons) we only have one spa932 sidecar to hand though. The spa525 seems to not like our sidecar configuration, and allows only 59 subscriptions, so the last few lights do not work, and after a while it decides to reboot regularly. Not quite sure how to debug that one :( The spa504 works pretty well with a sidecar, but crashed under load over a weekend. I have updated 7.4.3 to 7.4.4 firmware and we're trying again. >> 2) Is there a different (standard) way to send BLF and allow directed >> pickups? > > have a look at www.fop2.com maybe this could be interesting for you? its > a web based solution to see the state of extensions and do a pickup or > transfer calls. > Yes, we have a web-based BLF solution just like FOP, but some end-users have very specific requirements *sigh* :) Thanks for the input. Cheers, Steve -- _ -- 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] High volume BLF - Suggestions?
On 13 September 2010 11:43, Olivier wrote: > > > 2010/9/13 Steve Davies [snip] >> Our test involves about 10 BLF-NOTIFY messages per second to each >> handset with a 5-second pause every 5 seconds. This will either crash >> or render unusable all of the following combinations: >> >> snom360 + 1 x sidecar > > As Snom phones have a parameter to express a time period during which BLF's > SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones > would handle this load more easily. > > [snip] The SUBSCRIBE is handled fine, it is the NOTIFY messages that cause a problem. Thanks, Steve -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
On 13 September 2010 11:07, Antonio Berrios wrote: > Gotcha. Yeah, I'm looking at implementing that (searching call > recordings by agent that took the call) here but since our asterisk call > recording is a separate server to the ones dealing with queues I'll be > looking at tie-ing the agent to the call recording via a unique call ID > in a database rather than in the filename itself. I'll post my > findings/solution to the list if you would like? > Sounds like a plan :) I may try to modify Asterisk to send more parameters to the MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and CDR(dstchannel) seem like obvious choices. I'll post results here if I manage it. Regards, Steve -- _ -- 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] High volume BLF - Suggestions?
Hi, We have a user who is putting large call volumes through Asterisk, and wants to BLF monitor up to 90 extensions. We are struggling to find a handset that can keep up with Asterisk :) 1) Is there a handset that will do this? 2) Is there a different (standard) way to send BLF and allow directed pickups? 2a) Or even a handset specific way? Asterisk handles the BLF volume fine, even on quite low-end hardware, but we cannot find any handsets that can cope with it longer term. Our test involves about 10 BLF-NOTIFY messages per second to each handset with a 5-second pause every 5 seconds. This will either crash or render unusable all of the following combinations: snom360 + 1 x sidecar Yealink T28 + 1 x sidecar Yealink T28 + 2 x sidecar Cisco SPA504g + 1 x sidecar Cisco SPA504g + 2 x sidecar Cisco SPA525g + 1 x sidecar (reboots often) Cisco SPA525g + 2 x sidecar (reboots quickly) Aastra 55i + non-LCD sidecar Did not try Polycom as they do not do directed pickup and only small sidecars. Linksys SPA962 with one sidecar is "OK" but is discontinued hardware. Help? Thanks, Steve -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
On 11 September 2010 20:36, Antonio Berrios wrote: > On 09/09/10 17:59, Steve Davies wrote: >> On 9 September 2010 17:52, Antonio Berrios >> wrote: >> >>> Steve Davies wrote: >>> >>>> Hi, >>>> >>>> I am using 1.6.2.11, and I need to be able to include the name of the >>>> channel that answered a call in the call-recording filename. >>>> >>>> At a guess we need to use the Queue(name,,macro) or >>>> Dial(chan1&chan2,,M(macro)) and use the macro to update the call >>>> recording filename. But, the macro runs on the calling channel, and I >>>> need the called channel - Is this accessible? >>>> >>>> Thanks, >>>> Steve >>>> >>> Where ever the MixMonitor recording is done add in the ${CHANNEL} >>> variable to the filename parameter. Or even add in the line below to the >>> context that contains Dial(QueueName). >>> >>> For example: >>> >>> exten => >>> s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49) >>> >>> [snip] > Why is it you require the answering channel in the recording filename? > There may be an easier way to get you what you need. And a quick copy > pasta of the dial plan you're using could be handy. > I will look at putting together an example of the dialplan. We need the answering channel so the we can identify which handset was recorded (which agent took the call). At present we have caller-id and dialled extension, but an extension might be ringing any one of 10 phones, and the recording needs to be associated with that handset/agent. Regards, Steve -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
On 9 September 2010 17:52, Antonio Berrios wrote: > Steve Davies wrote: >> Hi, >> >> I am using 1.6.2.11, and I need to be able to include the name of the >> channel that answered a call in the call-recording filename. >> >> At a guess we need to use the Queue(name,,macro) or >> Dial(chan1&chan2,,M(macro)) and use the macro to update the call >> recording filename. But, the macro runs on the calling channel, and I >> need the called channel - Is this accessible? >> >> Thanks, >> Steve > > Where ever the MixMonitor recording is done add in the ${CHANNEL} > variable to the filename parameter. Or even add in the line below to the > context that contains Dial(QueueName). > > For example: > > exten => > s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49) > I was under the impression from the documentation that the ${CHANNEL} variable held the _Calling_ channel, I need the _Called_ channel (the channel that eventually picks up the call) Am I mistaken? Perhaps I should give it a try :) Thanks, Steve -- _ -- 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/Dial Recording - Capture answering channel name.
Hi, I am using 1.6.2.11, and I need to be able to include the name of the channel that answered a call in the call-recording filename. At a guess we need to use the Queue(name,,macro) or Dial(chan1&chan2,,M(macro)) and use the macro to update the call recording filename. But, the macro runs on the calling channel, and I need the called channel - Is this accessible? Thanks, Steve -- _ -- 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] How to debug this specific issue?
On 25 August 2010 08:22, Matt Riddell wrote: > On 25/08/10 7:20 PM, Tilghman Lesher wrote: >>> I really thought that the canary should have sounded if Asterisk got in >>> a loop - or maybe that only happens with high priority? >> >> The canary only runs in high priority mode, and it's only able to do anything >> if high priority scheduling is the culprit. If it's something else, like >> memory swapping, there's nothing the canary can do to fix that. > > Aha, explains why I've never seen the canary die :D > ...and yes, I was running in high priority mode - I thought I was turning it off for testing, but looks like I left the setting in asterisk.conf so leaving '-p' off the command line was making no difference *sigh* I believe the issue was actually with the devicestate thread. It was trying to update state on a locked channel, and was trying to grab the lock so regularly that it caused asterisk to grab lots of CPU cycles (because of -p mode) The lock was not released because it was waiting on a database write, which was being done by a lower priority external process that was getting no time scheduled to it. The database write is a local hack to record some extra call data - I changed it to occur after the locks are released as I should have done in the first place. 1.6.2.11 does not seem to have quite the same issue - It recovers after the usual 200 lock attempts and gets on with life much more happily. I cannot see any changes between 1.6.2.10 and 1.6.2.11 that would have improved this behaviour. Cheers, Steve -- _ -- 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] How to debug this specific issue?
On 24 August 2010 14:34, Steve Davies wrote: > On 24 August 2010 08:07, Stefan Schmidt wrote: >> Steve Davies schrieb: >>> On 23 August 2010 18:32, Stefan Schmidt wrote: >>> >>>> hello, >>>> >>>> have you allready tried strace ? >>>> you could just easily start asterisk with this command: >>>> >>>> strace asterisk - >>>> >>> >>> Yes, I tried this. Output just stops along with everything else and >>> there are no clues. >>> >> if you know in which function this happens you could also patch some >> ast_verbose rows into this function to see where it happens. >> >> another thing you could try is using gprof to see which functions waste >> much time. for this you need to compile asterisk like this: >> make ASTCFLAGS="-pg" ASTLDFLAGS="-pg" >> make install >> >> after this you will see a gmon.out file in the directory from where you >> have started asterisk (your home) and then you could use gprof with this >> gmon.out file. >> >> maybe you will find something with this. > > I found that doing a build with debug symbols included and running > under gdb slowed down asterisk enough for me to get debug output. > > Thanks for the pointers. I'll start a separate thread on working out > what the hell is going on :) Turns out that 1.6.2.11 fixes the symptom, and the cause is in my own "hack" to the code. Holding locks too long == verybad. Steve -- _ -- 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] How to debug this specific issue?
On 24 August 2010 08:07, Stefan Schmidt wrote: > Steve Davies schrieb: >> On 23 August 2010 18:32, Stefan Schmidt wrote: >> >>> hello, >>> >>> have you allready tried strace ? >>> you could just easily start asterisk with this command: >>> >>> strace asterisk - >>> >> >> Yes, I tried this. Output just stops along with everything else and >> there are no clues. >> > if you know in which function this happens you could also patch some > ast_verbose rows into this function to see where it happens. > > another thing you could try is using gprof to see which functions waste > much time. for this you need to compile asterisk like this: > make ASTCFLAGS="-pg" ASTLDFLAGS="-pg" > make install > > after this you will see a gmon.out file in the directory from where you > have started asterisk (your home) and then you could use gprof with this > gmon.out file. > > maybe you will find something with this. I found that doing a build with debug symbols included and running under gdb slowed down asterisk enough for me to get debug output. Thanks for the pointers. I'll start a separate thread on working out what the hell is going on :) Cheers, Steve -- _ -- 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