IRC log for #asterisk on 20151202

00:06.48*** join/#asterisk snadge (~snadge@unaffiliated/snadge)
00:09.34igcewieling1pjensen00: see sip.conf option promiscredir if using chan_sip, no idea what to check if running pjsip.
00:10.20snadgeso apparently some polycom piece of junk has an intercom feature which adds an alert info header to an invite packet.. and asterisk strips it
00:10.37snadgei probably should have just googled it now i think about it :D
00:11.17igcewieling1snadge: correct in that Asterisk doesn't forward headers from the incoming call to the outgoing call.
00:11.33snadgeis this something i can do relatively easily?
00:11.39igcewieling1you would read the header in the dialplan, then sipaddheader() to add it to the outbound leg
00:11.45pjensen00uhh..... hrm.
00:12.11pjensen00promiscredir I haven't heard of before
00:12.29igcewieling1see sip.conf.sample in the asterisk source code
00:12.44pjensen00Ok, I wonder if that'll translate to PJSIP.  :(
00:13.06igcewieling1pjensen00: I have no idea if has anything to do with your specific problem, but good to check.
00:15.03igcewieling1snadge: see "core show function SIP_HEADER" and "core show application sipaddheader"
00:20.55pjensen00;redirect_method=user   ; How redirects received from an endpoint are handled
00:20.56pjensen00<PROTECTED>
00:21.07pjensen00What other options are there?  I can't seem to find any other mentions in the wik
00:21.55pjensen00Oh wait, nevermind.  It's in some pdf
00:27.53*** part/#asterisk kharwell (kharwell@nat/digium/x-tozoffxywkhdxlzh)
00:35.04snadgehmmm.. it looks like thirdlane has macro-tl-page which seems to fiddle with the alert-info header.. but thirdlane is an undocumented pile of crap
00:41.44WIMPygruetzkopf: BTW: What were the limits you found on the mobile carriers?
00:42.09WIMPyAnd does it make a difference if you use GSM, UMTS or LTE?
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01:21.43pjensen00*sigh*  I cannot for the life of me figure out why asterisk is choking so hard on this.  http://pastebin.com/ZUCh7ynD
01:22.11pjensen00Unless I'm missing someting obvious, A13 can't handle redirects.
01:22.47[TK]D-FenderWhere do we see the failure associated with this?
01:22.55[TK]D-FenderYou jsut showed a packet
01:23.00[TK]D-FenderWhere is it a PROBLEM?
01:23.23pjensen00The problem is that the local channels it spawns has no reference to the process that spawned it.
01:23.44pjensen00Sorry I'm on hour 11 strait of working on this and I'm just at wits end.
01:23.58pjensen00Well, hour 11 including this issue.
01:25.09pjensen00All I get is the EXTEN from the contact header, and the local channel ID/name.  Every other channel variable is blank.  Ack, I didn't post the debug output from asterisk-ari chan
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01:35.41pjensen00302 being processed - http://pastebin.com/BwyPWuCD
01:37.22pjensen00Part I can't figure out is that I don't SEE any problem, but there's clearly some information missing in the channel asterisk is throwing back into the dialplan.
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02:19.48jpsharpIs there a dialplan function to get a list of channels in a group?
02:20.09jpsharpOr a dialplan way, not neccesarily a function, to get a list of channels in a GROUP()?
02:46.54pjensen00Oh man... I know one brute force method.
02:47.33jpsharpAGI exec asterisk -rx "group show channels" and then parse it to death.
02:48.04jpsharpWhich is what I'm having to do.
02:48.06jpsharpWhich sucks.
02:55.38pjensen00I had to do that for blind transfers
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03:07.38jpsharpNow I'm jus trying to figure out how to loop through it in the dialplan.  Which I probably can't do.
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03:48.50jpsharpWell, that's just downright completely fugly.
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04:45.47igcewieling1jpsharp: have you considered While and/or AEL?
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08:55.15whizziGood day all
08:58.17whizziDid anyone here ever had the issue that SIP trunks lost all registration all of a sudden (sip show registry show unregistered) while the peers are still happily online (sip show peers = all OK)? A 'sip reload' won’t fix it, only a hard restart of asterisk or unloading chan_sip and loading it again (and all registries are back). It’s not an internet related issue. Running Asterisk 11.17.0.
08:59.07WIMPyAre you sure? Maybe a transient DNS failure?
09:00.21whizzithat would be very weird. Another system, located on the same network using the same DNS doesn’t have the issue
09:00.55whizzias a matter of fact, both machines are 'virtual' (OpenVZ) and located on the same host
09:01.36whizzieven more strange, there are 5 machines on the host, on the same network, using the same DNS and only one has issues.
09:02.21whizziwhile it only has like 30 registers outgoing and other (who don’t have issues) are way over 400
09:04.18whizziI even killed the machine, created a new container on OpenVZ and changed IP-address back to original.
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12:02.14Phil-Workpbx_config.c:1649 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 76 of extensions.conf
12:02.45Phil-Workso I need to add _1. _2. _3. etc.?
12:03.18WIMPyDid you read that message?
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12:03.37Phil-WorkI did
12:03.46Phil-Workthough, now I re-read it, I realise it's literal _X.
12:03.53Phil-Workrather than X representing a number
12:03.53WIMPyThen I don't understand your question.
12:04.03WIMPyIndeed.
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12:40.42SunnyB_Guys, would you suggest a nice, cheap, voip phone similar to the cisco 7940 ? Call quality and BLF are a must.
12:44.16Phil-WorkSIP?
12:44.27carrar7960
12:44.34carraroh no BLF sorry
12:44.36carrar;)
12:45.03Phil-Workthe Snom 320 isn't terrible
12:45.05carrarI would recommend Polycom
12:47.58carrarLot of people like the Yealink T4 series
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12:54.38SunnyB_Yes, I use them but blf seems to be important
12:58.53whizziSunnyB_: how many BLF we’re talking about?
12:59.14SunnyB_Well, 2 or 4 would be enough I guess.
13:00.01SunnyB_I'm looking at the low-end side of the market. Like 60-90$. Is it too low?
13:00.21whizzilook for the Mitel 6865i then, it has 8 programmable BLF keys
13:00.29whizzinot sure how expensive that one is
13:01.12SunnyB_thank you
13:01.31whizziperhaps you can find an Aastra 6731 somewhere, that one has 8 BLF keys too
13:01.42SunnyB_looks quite expensive.
13:02.37whizziAastra 6731i  or 6730i
13:03.08whizziit’s an older model, but still working quite good.
13:03.48whizziI’ll check Yealink for you, hold on
13:04.20whizziYealink is basically a copy of Aastra/Mitel, in the few latest firmwares you can actually start seeing difference between the brands ;)
13:05.21SunnyB_the t48g looks nice. probably on the higher end of the price range.
13:07.40whizzithe cheapest yealink is the T26P, around 114 euro in NL
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13:08.10whizziso, the cheapest option is Aastra 6730, followed by 6731
13:08.48SunnyB_Maybe a digium phone will be better at that price point.
13:08.55whizziI have to say, these 2 phones have issues with speaker and not-so-good audio. So if you are using them to call hands free without headset, don’t go for these. Spend a little more money and get the Mitel 6865i
13:09.32SunnyB_ok this is a deal breaker then. I love the cisco handsfree and would like something similar.
13:11.00WIMPyThe Snom handsfree is really good.
13:11.06RZeroHi everyone, I've got an odd issue with asterisk 13.4,  I have a peer that is sending the invites and I see the invites using ngrep and asterisk sip debug. But Asterisk seems to be ignoring them.
13:14.55whizziSunnyB_: Snom 320 it is then, that’s cheaper than Mitel’s 6865. I don’t have experience with these phones myself, but they’re probably good and have BLF :)
13:15.33SunnyB_Yes, they look good. Not too expensive either.
13:15.45WIMPyDoesn't even the 300 have some keys with LED?
13:16.56RZeroI have the host details as [peer] type=peer insecure=port,invite context=from-pstn disallow=all,allow=ulaw,alaw  so its nothing special. I has been working with those settings.
13:17.12RZeroI have the host details as [peer] type=peer host=some ip insecure=port,invite context=from-pstn disallow=all,allow=ulaw,alaw  so its nothing special. I has been working with those settings.
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13:46.02[TK]D-FenderRZero: Show us the full debug, and the config, peer dump, etc
14:05.29RZeroI figured it out [TK]D-Fender the server had dropped some routing information for the nic (sigh)
14:09.05Emilis-hi I upgraded to version 13, and now trunks not registering, and getting error message:  ERROR[8685]: chan_sip.c:4181 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
14:10.48[TK]D-FenderFirewall or other networking issue
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14:12.07RZeroNo, eth configs where never setup correctly looking at, so when the server was rebooted it dropped the routes that had be added by the route command.
14:12.49Emilis-well if I do nmap -v -sU -p 5060  <ip> -- it shows opened
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14:15.11[TK]D-FenderEmilis-: that's in inbound text.
14:15.15[TK]D-Fendercheck OUTBOUND rules
14:15.40[TK]D-FenderRZero: Wasn't emant for you.
14:15.44[TK]D-Fendermeant*
14:15.52RZeroYes, just saw sorry :D
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14:20.52Emilis-also have tshark running, and it shows no traffic while i get this error message: chan_sip.c:15330 sip_reg_timeout:    -- Registration for '##########@sip.flowroute.com' timed out, trying again (Attempt #8)
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14:28.56[TK]D-Fenderthat message should be unrelated
14:29.01[TK]D-FenderShow us the full debug
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14:35.32Emilis-e.g. http://pastebin.com/8SgpJkPi
14:41.45[TK]D-Fenderiptables --list
14:41.49[TK]D-Fenderroute -an
14:41.50[TK]D-Fender^
14:42.37[TK]D-FenderAlso one other thing would be if you are using a port that it couldn't bind to due to a conflict <-
14:42.52[TK]D-FenderIE chan_sip & chan_pjsip both being set to the same.
14:43.00[TK]D-Fenderor another SIP software on that same machine, etc
14:44.17Emilis-iptables is empty - I turned it off already, as part of my troubleshooting.
14:45.18Emilis-route -a 0.0.0.0         10.0.0.1        0.0.0.0         UG    0      0        0 eth0
14:45.41Emilis-how do I see if chan_sip and pjsip are being set to the same?
14:48.50Emilis-res_pjsip* is set as noload in modules.conf
14:49.30[TK]D-Fenderchan not just res
14:49.39[TK]D-FenderAnd check the configs themselves to be sure
14:51.14Emilis-grep chan modules.conf - only preload = chan_local.so - rest show noload
14:52.29Emilis-what other configs should I look at?
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14:57.31igcewieling1sip.conf might be a good place to start.
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15:03.41Emilis-well yea, sip.conf has a few includes, but I don't see anything of note
15:04.17Phil-Workwhat's the syntax to originate to a local channel in a given context? :S
15:04.29igcewieling1Local/extension@context
15:05.22Phil-Workthanks
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15:22.46Phil-Workshould BackGroundDetect() work on an outbound call? Dialplan is Dial() then BackGroundDetect()
15:23.29[TK]D-FenderShow us
15:23.52igcewieling1Phil-Work: dialplan stops when Dial is run.   This is the way it works.
15:24.52Phil-Work[TK]D-Fender: https://gist.github.com/phil-lavin/4a66b8df327f0dfc5c68
15:25.14[TK]D-FenderYou don't seem to understand how Dial works...
15:25.30[TK]D-Fenderyou STAY in that app until the call ends.
15:25.43[TK]D-FenderYou don't continue running dialplan in the background
15:26.06Phil-Workbut the message is played, so the BackgroundDetect is running, no?
15:26.26igcewieling1Phil-Work: that would be obvious in the Asterisk CLI output
15:26.52Phil-Work<PROTECTED>
15:26.52Phil-Work<PROTECTED>
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15:27.23[TK]D-FenderThat isn't the leg of the call with a backgrounddetect AFTER the dial <-
15:27.37[TK]D-FenderLine 10 is NOT going to happen during that Dial
15:27.41Phil-Workok, that makes sense
15:27.45Phil-Workwhat should I use rather than Dial()?
15:28.00igcewieling1[Dec  2 10:27:33]     -- Executing [~~s~~@sm_dial:41] Dial("SIP/nextone-00118794", "SIP/igv0746/16095855506,600,gb(sm_pre_dial^16095855500^1)U(sm_answer^16095855500^1)F(stub_sm_intercept^16095855500^1)") in new stack
15:28.11[TK]D-FenderGosub <-
15:28.12igcewieling1it would look something like THAT.
15:28.22[TK]D-FenderIf you want to detect after the answer but before bridging
15:29.10igcewieling1When we ask for CLI output we are looking for the sort of stuff I pasted.
15:29.27Phil-Workafter the answer is right, yes
15:30.39Phil-Workif I'm understanding the Gosub manual correctly, the logic should be... Gosub(dial), BackgroundDetect(), (dial)Dial()?
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17:06.42tenspeed705Morning guys
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17:07.07tenspeed705anyone have some code to a wake up system I can steal. Im lazy :)
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17:34.57igcewieling1too lazy to google?
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17:41.26gevansI have an options request coming in from my sip provider, that seems to get resolved to the wrong target 10.50.50.110 when it should be 10.50.50.181 consequently the call gets terminated. That is how a read the log anyhow. pastebin.com/n9TWTitt
17:42.12gevanscould someone lend some insight, please?
17:42.39[TK]D-FenderWhere dop we see a call being terminated?
17:42.57igcewieling1gevans: SIP OPTIONS packets are used for keepalive packets.   They have nothing to do with a dropped call
17:45.02gevansigcewieling1: Yes of course, but the state of the options request is trying and after some time the sip provider sends BYE request and call is terminated. If more of the log is requried please let me know.
17:46.28gevansigcewieling1: is my interprenation of the snippet of the log correct though? Sip provider sends options request, asterisk server resolved target to 10.50.50.110 and forwards request?
17:46.38igcewieling1check whatever the pjsip versions of chan_sip'externip and localnet
17:46.55igcewieling1check whatever the pjsip versions of chan_sip's externip and localnet settings are
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17:48.05igcewieling1no.
17:48.31gevansigcewieling1: ha ha, well that a good start. Care to englighten me
17:49.15igcewieling1What is the IP address of your provider in that paste?
17:49.43igcewieling1I doubt I can help much.  You are using pjsip and I use chan_sip.  But we'll see.
17:51.31gevans10.185.38.35 is the provider gateway
17:51.51gevansany and all help is welcomed
17:52.44igcewieling1you have a private connection to your provider?
17:52.49gevansvpls
17:52.53Kobaz<PROTECTED>
17:52.58Kobazhow do i tell who is watching?
17:53.00gevansso yes
17:53.02Kobazwithout sip debug
17:54.01igcewieling1Kobaz: core show hints
17:54.14igcewieling1gevans: is vpls anything like mpls?
17:54.15Kobazthat doesn't tell you the who
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17:54.23Kobazigcewieling1: that's who is *being* watched
17:54.29Kobaznot who *is* watching
17:54.33igcewieling1Kobaz: ah.
17:55.33gevansigcewieling1: somewhat
18:00.28tenspeed705igcewieling1, I did google. I found several apps, but all of them come back with a 404.
18:02.47gevansigcewieling1: pjsip version is 2.4 localnet is 10.50.50.0/24 and 10.50.70.0/23 nad exernip is 10.185.38.38
18:03.31gevansigcewieling1: and for the most part yes vpls is a lot like mpls
18:10.08gevansigcewielng1: what is your inteprentation of the log by the way
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18:11.54igcewieling1my I do have other jobs to work on.
18:14.51igcewieling1the sip resolve message is a pjsip message.
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18:44.27todwetsprockwarum sind die virtual-machines auf srv07 & srv08 xfs gemountet und nicht btrfs ?
18:45.06robmalDa.
18:45.23robmalSorry, Ja.
18:45.34robmalEntschuldigung, Ja.
18:45.47robmalcan has into duolingo
18:45.51todwetsprocksry, wrong channel. my mistake
18:45.53robmal;-)
18:45.56robmalNo problem.
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18:49.05pankidI am trying to figure out what prebuilt asterisk box this is. Anyone know a good way to figure out the vendor? The web gui doesnt seem to have any clues and there is nothing obvious in /etc/asterisk. I know it has a  Digium Wildcard TE121
18:49.37gevansigcewieling1: so i talked to provider and they said that our pbx is not repsonding to the options request
18:49.41[TK]D-Fender"what prebuilt asterisk box this is"
18:49.48[TK]D-FenderWhat are you asking about?
18:49.57[TK]D-FenderIs there a MODEL # from a vendor for the AHRDWARE?
18:52.31pankid[TK]D-Fender: I don't have physical access. Trying to figure out the model from the shell.
18:53.17robmaldmidecode
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18:54.58[TK]D-FenderNodel of what?
18:55.02[TK]D-FenderThte overall server?
18:55.19[TK]D-FenderWe don't even have a brand
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18:57.27igcewieling1gevans: I do not know how to enable options support onm pjsip
18:58.54gevansigcewieling1: how would you do in sip?
18:59.04pankidrobmal: thanks. Might be able to find some clues in here
18:59.14gevansigcewieling1: is it a configuration file?
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19:03.24igcewieling1gevans: in chan_sip it is enabled by default
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19:34.26gevansigcewieling1: thank you anyhow
19:44.52lvlinuxgevans: do you mean SIP OPTIONS?
19:45.40lvlinuxgevans: if so, with pjsip you can use "qualify_frequency = 15"
19:46.17gevanslvlinux: what does lowering the qualify_frequency do? it is set to 60 by default
19:46.30igcewieling1lvlinux: respond to, not send
19:47.11lvlinuxigcewieling1: huh?
19:47.29lvlinuxgevans: the time is how many seconds between sending OPTIONS packets
19:47.37gevansyes, thats right it doesn't not respond to sip provider's options request to ensure the call is still alive so provider sends BYE
19:48.08gevanswithin a 15 minute time span it never once sends an option packet to the provider
19:48.14gevansor a repsond to the options
19:48.25WIMPyOPTIONs are not related to calls.
19:48.36igcewieling1lvlinux: his asterisk is not RESPONDING to OPTIONS packets sent to it.
19:48.38gevansfull log is here goo.gl/fcVnRD
19:49.13gevansWIMPy: Is this hte same WIMPy from jbc?
19:49.29WIMPyWhat is jbc?
19:49.39gevansjupiter broadcasting
19:49.52lvlinuxigcewieling1: ah ok I hadn't read back that far, just saw the request about "enabling options support" so yeah he has a more in depth problem. Maybe a SIP ALG perhaps, gevans?
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19:50.44igcewieling1lvlinux: I suspect a fundamental problem somewhere else, but I don't have the time, inclination or knowledge to prove or disprove it.
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19:52.14acidfoo-hello world, does Asterisk 11 support end-to-end codec renogiciation to avoid transcoding ? Or it's still nogociated per-leg ?
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19:53.07gevansigcewieling1: well thanks, God, I see some issues opened to the fact but no solution
19:53.12igcewieling1acidfoo-: asterisk will not transcode if both legs of the call use the same codec.
19:53.44acidfoo-yeah that part I understand
19:54.18igcewieling1However, getting Asterisk to use the codec you want is a bit more complicated.
19:54.51acidfoo-the two legs might have negociated a different codec with Asterisk, even tho both phone support a common one
19:54.53igcewieling1As far ask I know it will not change the codec in the middle of the call.
19:55.16igcewieling1acidfoo-: the most common solution I use it to only allow one codec on all the devices.
19:56.06acidfoo-yeah I can't do that
19:58.14igcewieling1acidfoo-: your life will be very interesting
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20:09.03gevansso looking more through the log and it does seem the server does repsond to queries from the server it just does not respond for the sessions it puts those into a state of trying
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20:26.25gevansigcewieling1: you said the sip resolve is just a sip resolve, why is the server resolving to the ip addresss when an options request is coming in for another endpoint?
20:28.07gevansI honestly believe the issue is in these couple of lines:
20:28.08gevans40156 [2015-12-01 18:03:22] DEBUG[12560] pjsip:       sip_endpoint.c Distributing rdata to modules: Request msg OPTIONS/cseq=12627 (rdata0x2817e48)
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20:28.27[TK]D-FenderPASTEBIN.  Do not flood in here
20:28.34gevansSorry!
20:28.52WIMPyDamn. I'm too fast.
20:29.01WIMPyI unignored before it ended.
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20:36.23igcewieling1gevans: I should have said the server resolving is a PJSIP thing.
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20:37.39igcewieling1is it evil to point the MX record for my old domain to localhost? 8-)
20:44.01gevansigcewieling1: oh for sure, i just think our provider does not get an options response for an ongoing call because pjsip is resolving to the wrong endpoint
20:44.39WIMPyOPTIONs are not related to calls.
20:45.21WIMPySo it is impossible that they would get an options response to an ongoing call.
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20:47.08gevansWIMPy: Maybe I'm not explaining my scenario the best, there is a similar issue: goo.gl/fb8TLD
20:49.56filethey are unrelated, what version of Asterisk is in use?
20:50.11K0HAXDoes anyone have a working n-way calling script? :( The ones I find on google drop the originating end after placing the call to the 3rd party
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20:53.41gevansasterisk 13.1 cert2
20:53.59gevanspjsip 2.4
20:54.10file13.1-cert2 does not have support for in-dialog OPTIONS, it was added afterwards
20:55.06fileyou may be able to take the res_pjsip_dlg_options.c file from the certified/13.1 branch, drop it into -cert2, build it, and use it
20:56.25gevansfile: thank you.. god! your awesome
20:58.34gevanswhat release is the issue adddressed in?
20:59.17gevansnoticed version 13.2 was still listed as affected in issue 24818
20:59.28filethere is no current certified release with it, it was added in 13.4.0
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21:00.23gevansawesome, uncertified it's going to have to be
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21:22.12xphereshello, I'm trying to configure a trunk and outbound with a voip service "justvoip" but I'm not able to make it work
21:24.38acidfoo-how can I see what codec are currently used by two bridged call ?
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21:24.48acidfoo-(Asterisk11)
21:25.05anubisjimhiyas all
21:27.06anubisjimanyone able to assist with an install of asterisk im having a few issues installing it onto fedora
21:35.23xpheresanyone able to assist to set up outbound trunk with sip service?
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21:37.40moe`hey kids
21:39.36moe`ooohhhh 13.6.0 is LTS, nice, didn't realize that.   sweet.  that's what I last compiled.
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22:21.32CFX_Anything particular I have to do to give another full access over the asterisk terminal? I updated the sudoers file and I'm still getting permission errors.
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23:22.51edong23[TK]D-Fender: just an update. you were right bout the trunk order. but the reason the notify was being denied was because of some options on the class5 switch. And im sorry you got so disgruntled the other day. i was making lots of changes between responses to see the differences in an attempt to see where the issue was. And it was indeed on the clas5 switch.not an issu, but a misconfiguration
23:43.42pankidAnyone recognize this pbx? http://imgur.com/eZkdUAP I know it is centos/redhat 5 and asterisk 1.8.23
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23:46.50monstercoHi everyone, I have a call file which is working fine except for Application:Dial line. Can someone please tell me what is wrong with this: http://pastebin.com/gyUBa1Yx
23:47.18monstercoerro I am getting is: WARNING[3006]: pbx.c:8964 ast_pbx_run_app: No such application 'Dial(.............
23:47.37monstercothis is the line in my call file:   Application:Dial(SIP/sip-provider-ip/4166356000)
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