00:06.48 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
00:09.34 | igcewieling1 | pjensen00: see sip.conf option promiscredir if using chan_sip, no idea what to check if running pjsip. |
00:10.20 | snadge | so apparently some polycom piece of junk has an intercom feature which adds an alert info header to an invite packet.. and asterisk strips it |
00:10.37 | snadge | i probably should have just googled it now i think about it :D |
00:11.17 | igcewieling1 | snadge: correct in that Asterisk doesn't forward headers from the incoming call to the outgoing call. |
00:11.33 | snadge | is this something i can do relatively easily? |
00:11.39 | igcewieling1 | you would read the header in the dialplan, then sipaddheader() to add it to the outbound leg |
00:11.45 | pjensen00 | uhh..... hrm. |
00:12.11 | pjensen00 | promiscredir I haven't heard of before |
00:12.29 | igcewieling1 | see sip.conf.sample in the asterisk source code |
00:12.44 | pjensen00 | Ok, I wonder if that'll translate to PJSIP. :( |
00:13.06 | igcewieling1 | pjensen00: I have no idea if has anything to do with your specific problem, but good to check. |
00:15.03 | igcewieling1 | snadge: see "core show function SIP_HEADER" and "core show application sipaddheader" |
00:20.55 | pjensen00 | ;redirect_method=user ; How redirects received from an endpoint are handled |
00:20.56 | pjensen00 | <PROTECTED> |
00:21.07 | pjensen00 | What other options are there? I can't seem to find any other mentions in the wik |
00:21.55 | pjensen00 | Oh wait, nevermind. It's in some pdf |
00:27.53 | *** part/#asterisk kharwell (kharwell@nat/digium/x-tozoffxywkhdxlzh) |
00:35.04 | snadge | hmmm.. 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 |
00:41.44 | WIMPy | gruetzkopf: BTW: What were the limits you found on the mobile carriers? |
00:42.09 | WIMPy | And does it make a difference if you use GSM, UMTS or LTE? |
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01:21.43 | pjensen00 | *sigh* I cannot for the life of me figure out why asterisk is choking so hard on this. http://pastebin.com/ZUCh7ynD |
01:22.11 | pjensen00 | Unless I'm missing someting obvious, A13 can't handle redirects. |
01:22.47 | [TK]D-Fender | Where do we see the failure associated with this? |
01:22.55 | [TK]D-Fender | You jsut showed a packet |
01:23.00 | [TK]D-Fender | Where is it a PROBLEM? |
01:23.23 | pjensen00 | The problem is that the local channels it spawns has no reference to the process that spawned it. |
01:23.44 | pjensen00 | Sorry I'm on hour 11 strait of working on this and I'm just at wits end. |
01:23.58 | pjensen00 | Well, hour 11 including this issue. |
01:25.09 | pjensen00 | All 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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01:35.41 | pjensen00 | 302 being processed - http://pastebin.com/BwyPWuCD |
01:37.22 | pjensen00 | Part 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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02:19.48 | jpsharp | Is there a dialplan function to get a list of channels in a group? |
02:20.09 | jpsharp | Or a dialplan way, not neccesarily a function, to get a list of channels in a GROUP()? |
02:46.54 | pjensen00 | Oh man... I know one brute force method. |
02:47.33 | jpsharp | AGI exec asterisk -rx "group show channels" and then parse it to death. |
02:48.04 | jpsharp | Which is what I'm having to do. |
02:48.06 | jpsharp | Which sucks. |
02:55.38 | pjensen00 | I had to do that for blind transfers |
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03:07.38 | jpsharp | Now 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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03:48.50 | jpsharp | Well, that's just downright completely fugly. |
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04:45.47 | igcewieling1 | jpsharp: have you considered While and/or AEL? |
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08:55.15 | whizzi | Good day all |
08:58.17 | whizzi | Did 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' wonât fix it, only a hard restart of asterisk or unloading chan_sip and loading it again (and all registries are back). Itâs not an internet related issue. Running Asterisk 11.17.0. |
08:59.07 | WIMPy | Are you sure? Maybe a transient DNS failure? |
09:00.21 | whizzi | that would be very weird. Another system, located on the same network using the same DNS doesnât have the issue |
09:00.55 | whizzi | as a matter of fact, both machines are 'virtual' (OpenVZ) and located on the same host |
09:01.36 | whizzi | even more strange, there are 5 machines on the host, on the same network, using the same DNS and only one has issues. |
09:02.21 | whizzi | while it only has like 30 registers outgoing and other (who donât have issues) are way over 400 |
09:04.18 | whizzi | I even killed the machine, created a new container on OpenVZ and changed IP-address back to original. |
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12:02.14 | Phil-Work | pbx_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 |
12:02.45 | Phil-Work | so I need to add _1. _2. _3. etc.? |
12:03.18 | WIMPy | Did you read that message? |
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12:03.37 | Phil-Work | I did |
12:03.46 | Phil-Work | though, now I re-read it, I realise it's literal _X. |
12:03.53 | Phil-Work | rather than X representing a number |
12:03.53 | WIMPy | Then I don't understand your question. |
12:04.03 | WIMPy | Indeed. |
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12:40.42 | SunnyB_ | Guys, would you suggest a nice, cheap, voip phone similar to the cisco 7940 ? Call quality and BLF are a must. |
12:44.16 | Phil-Work | SIP? |
12:44.27 | carrar | 7960 |
12:44.34 | carrar | oh no BLF sorry |
12:44.36 | carrar | ;) |
12:45.03 | Phil-Work | the Snom 320 isn't terrible |
12:45.05 | carrar | I would recommend Polycom |
12:47.58 | carrar | Lot of people like the Yealink T4 series |
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12:54.38 | SunnyB_ | Yes, I use them but blf seems to be important |
12:58.53 | whizzi | SunnyB_: how many BLF weâre talking about? |
12:59.14 | SunnyB_ | Well, 2 or 4 would be enough I guess. |
13:00.01 | SunnyB_ | I'm looking at the low-end side of the market. Like 60-90$. Is it too low? |
13:00.21 | whizzi | look for the Mitel 6865i then, it has 8 programmable BLF keys |
13:00.29 | whizzi | not sure how expensive that one is |
13:01.12 | SunnyB_ | thank you |
13:01.31 | whizzi | perhaps you can find an Aastra 6731 somewhere, that one has 8 BLF keys too |
13:01.42 | SunnyB_ | looks quite expensive. |
13:02.37 | whizzi | Aastra 6731i or 6730i |
13:03.08 | whizzi | itâs an older model, but still working quite good. |
13:03.48 | whizzi | Iâll check Yealink for you, hold on |
13:04.20 | whizzi | Yealink is basically a copy of Aastra/Mitel, in the few latest firmwares you can actually start seeing difference between the brands ;) |
13:05.21 | SunnyB_ | the t48g looks nice. probably on the higher end of the price range. |
13:07.40 | whizzi | the cheapest yealink is the T26P, around 114 euro in NL |
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13:08.10 | whizzi | so, the cheapest option is Aastra 6730, followed by 6731 |
13:08.48 | SunnyB_ | Maybe a digium phone will be better at that price point. |
13:08.55 | whizzi | I 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, donât go for these. Spend a little more money and get the Mitel 6865i |
13:09.32 | SunnyB_ | ok this is a deal breaker then. I love the cisco handsfree and would like something similar. |
13:11.00 | WIMPy | The Snom handsfree is really good. |
13:11.06 | RZero | Hi 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. |
13:14.55 | whizzi | SunnyB_: Snom 320 it is then, thatâs cheaper than Mitelâs 6865. I donât have experience with these phones myself, but theyâre probably good and have BLF :) |
13:15.33 | SunnyB_ | Yes, they look good. Not too expensive either. |
13:15.45 | WIMPy | Doesn't even the 300 have some keys with LED? |
13:16.56 | RZero | I 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. |
13:17.12 | RZero | I 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-Fender | RZero: Show us the full debug, and the config, peer dump, etc |
14:05.29 | RZero | I figured it out [TK]D-Fender the server had dropped some routing information for the nic (sigh) |
14:09.05 | Emilis- | 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-Fender | Firewall or other networking issue |
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14:12.07 | RZero | No, 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. |
14:12.49 | Emilis- | well if I do nmap -v -sU -p 5060 <ip> -- it shows opened |
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14:15.11 | [TK]D-Fender | Emilis-: that's in inbound text. |
14:15.15 | [TK]D-Fender | check OUTBOUND rules |
14:15.40 | [TK]D-Fender | RZero: Wasn't emant for you. |
14:15.44 | [TK]D-Fender | meant* |
14:15.52 | RZero | Yes, just saw sorry :D |
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14:20.52 | Emilis- | 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-Fender | that message should be unrelated |
14:29.01 | [TK]D-Fender | Show us the full debug |
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14:35.32 | Emilis- | e.g. http://pastebin.com/8SgpJkPi |
14:41.45 | [TK]D-Fender | iptables --list |
14:41.49 | [TK]D-Fender | route -an |
14:41.50 | [TK]D-Fender | ^ |
14:42.37 | [TK]D-Fender | Also 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-Fender | IE chan_sip & chan_pjsip both being set to the same. |
14:43.00 | [TK]D-Fender | or another SIP software on that same machine, etc |
14:44.17 | Emilis- | iptables is empty - I turned it off already, as part of my troubleshooting. |
14:45.18 | Emilis- | route -a 0.0.0.0 10.0.0.1 0.0.0.0 UG 0 0 0 eth0 |
14:45.41 | Emilis- | how do I see if chan_sip and pjsip are being set to the same? |
14:48.50 | Emilis- | res_pjsip* is set as noload in modules.conf |
14:49.30 | [TK]D-Fender | chan not just res |
14:49.39 | [TK]D-Fender | And check the configs themselves to be sure |
14:51.14 | Emilis- | grep chan modules.conf - only preload = chan_local.so - rest show noload |
14:52.29 | Emilis- | what other configs should I look at? |
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14:57.31 | igcewieling1 | sip.conf might be a good place to start. |
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15:03.41 | Emilis- | well yea, sip.conf has a few includes, but I don't see anything of note |
15:04.17 | Phil-Work | what's the syntax to originate to a local channel in a given context? :S |
15:04.29 | igcewieling1 | Local/extension@context |
15:05.22 | Phil-Work | thanks |
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15:22.46 | Phil-Work | should BackGroundDetect() work on an outbound call? Dialplan is Dial() then BackGroundDetect() |
15:23.29 | [TK]D-Fender | Show us |
15:23.52 | igcewieling1 | Phil-Work: dialplan stops when Dial is run. This is the way it works. |
15:24.52 | Phil-Work | [TK]D-Fender: https://gist.github.com/phil-lavin/4a66b8df327f0dfc5c68 |
15:25.14 | [TK]D-Fender | You don't seem to understand how Dial works... |
15:25.30 | [TK]D-Fender | you STAY in that app until the call ends. |
15:25.43 | [TK]D-Fender | You don't continue running dialplan in the background |
15:26.06 | Phil-Work | but the message is played, so the BackgroundDetect is running, no? |
15:26.26 | igcewieling1 | Phil-Work: that would be obvious in the Asterisk CLI output |
15:26.52 | Phil-Work | <PROTECTED> |
15:26.52 | Phil-Work | <PROTECTED> |
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15:27.23 | [TK]D-Fender | That isn't the leg of the call with a backgrounddetect AFTER the dial <- |
15:27.37 | [TK]D-Fender | Line 10 is NOT going to happen during that Dial |
15:27.41 | Phil-Work | ok, that makes sense |
15:27.45 | Phil-Work | what should I use rather than Dial()? |
15:28.00 | igcewieling1 | [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-Fender | Gosub <- |
15:28.12 | igcewieling1 | it would look something like THAT. |
15:28.22 | [TK]D-Fender | If you want to detect after the answer but before bridging |
15:29.10 | igcewieling1 | When we ask for CLI output we are looking for the sort of stuff I pasted. |
15:29.27 | Phil-Work | after the answer is right, yes |
15:30.39 | Phil-Work | if I'm understanding the Gosub manual correctly, the logic should be... Gosub(dial), BackgroundDetect(), (dial)Dial()? |
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17:06.42 | tenspeed705 | Morning guys |
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17:07.07 | tenspeed705 | anyone have some code to a wake up system I can steal. Im lazy :) |
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17:34.57 | igcewieling1 | too lazy to google? |
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17:41.26 | gevans | I 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 |
17:42.12 | gevans | could someone lend some insight, please? |
17:42.39 | [TK]D-Fender | Where dop we see a call being terminated? |
17:42.57 | igcewieling1 | gevans: SIP OPTIONS packets are used for keepalive packets. They have nothing to do with a dropped call |
17:45.02 | gevans | igcewieling1: 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. |
17:46.28 | gevans | igcewieling1: 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? |
17:46.38 | igcewieling1 | check whatever the pjsip versions of chan_sip'externip and localnet |
17:46.55 | igcewieling1 | check whatever the pjsip versions of chan_sip's externip and localnet settings are |
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17:48.05 | igcewieling1 | no. |
17:48.31 | gevans | igcewieling1: ha ha, well that a good start. Care to englighten me |
17:49.15 | igcewieling1 | What is the IP address of your provider in that paste? |
17:49.43 | igcewieling1 | I doubt I can help much. You are using pjsip and I use chan_sip. But we'll see. |
17:51.31 | gevans | 10.185.38.35 is the provider gateway |
17:51.51 | gevans | any and all help is welcomed |
17:52.44 | igcewieling1 | you have a private connection to your provider? |
17:52.49 | gevans | vpls |
17:52.53 | Kobaz | <PROTECTED> |
17:52.58 | Kobaz | how do i tell who is watching? |
17:53.00 | gevans | so yes |
17:53.02 | Kobaz | without sip debug |
17:54.01 | igcewieling1 | Kobaz: core show hints |
17:54.14 | igcewieling1 | gevans: is vpls anything like mpls? |
17:54.15 | Kobaz | that doesn't tell you the who |
17:54.19 | *** join/#asterisk saratogga (~saratogga@wsip-98-174-232-77.ph.ph.cox.net) |
17:54.23 | Kobaz | igcewieling1: that's who is *being* watched |
17:54.29 | Kobaz | not who *is* watching |
17:54.33 | igcewieling1 | Kobaz: ah. |
17:55.33 | gevans | igcewieling1: somewhat |
18:00.28 | tenspeed705 | igcewieling1, I did google. I found several apps, but all of them come back with a 404. |
18:02.47 | gevans | igcewieling1: 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 |
18:03.31 | gevans | igcewieling1: and for the most part yes vpls is a lot like mpls |
18:10.08 | gevans | igcewielng1: what is your inteprentation of the log by the way |
18:11.52 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
18:11.54 | igcewieling1 | my I do have other jobs to work on. |
18:14.51 | igcewieling1 | the sip resolve message is a pjsip message. |
18:42.27 | *** part/#asterisk jpsharp (~jsharp@c-76-101-138-247.hsd1.fl.comcast.net) |
18:44.27 | todwetsprock | warum sind die virtual-machines auf srv07 & srv08 xfs gemountet und nicht btrfs ? |
18:45.06 | robmal | Da. |
18:45.23 | robmal | Sorry, Ja. |
18:45.34 | robmal | Entschuldigung, Ja. |
18:45.47 | robmal | can has into duolingo |
18:45.51 | todwetsprock | sry, wrong channel. my mistake |
18:45.53 | robmal | ;-) |
18:45.56 | robmal | No problem. |
18:47.08 | *** join/#asterisk pankid (~quassel@104.236.22.159) |
18:49.05 | pankid | I 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 |
18:49.37 | gevans | igcewieling1: 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-Fender | What are you asking about? |
18:49.57 | [TK]D-Fender | Is there a MODEL # from a vendor for the AHRDWARE? |
18:52.31 | pankid | [TK]D-Fender: I don't have physical access. Trying to figure out the model from the shell. |
18:53.17 | robmal | dmidecode |
18:53.59 | *** join/#asterisk tm1000 (sid6728@gateway/web/irccloud.com/x-sdiupkkgfwtmomtm) |
18:54.58 | [TK]D-Fender | Nodel of what? |
18:55.02 | [TK]D-Fender | Thte overall server? |
18:55.19 | [TK]D-Fender | We don't even have a brand |
18:55.55 | *** join/#asterisk ryang (sid10904@gateway/web/irccloud.com/x-dtvcomipdbpnerck) |
18:56.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
18:56.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
18:57.27 | igcewieling1 | gevans: I do not know how to enable options support onm pjsip |
18:58.54 | gevans | igcewieling1: how would you do in sip? |
18:59.04 | pankid | robmal: thanks. Might be able to find some clues in here |
18:59.14 | gevans | igcewieling1: is it a configuration file? |
19:00.57 | *** join/#asterisk qloogkm (sid20721@gateway/web/irccloud.com/x-obawvunqmxvqysef) |
19:03.24 | igcewieling1 | gevans: in chan_sip it is enabled by default |
19:03.39 | *** join/#asterisk foamz (sid25727@gateway/web/irccloud.com/x-kictzejdmaujcewp) |
19:09.12 | *** join/#asterisk nunne (sid38499@gateway/web/irccloud.com/x-nzsfzkrenlwlwbbk) |
19:32.23 | *** join/#asterisk marlinc (~marlinc@bouncer.cvo.technology) |
19:34.26 | gevans | igcewieling1: thank you anyhow |
19:44.52 | lvlinux | gevans: do you mean SIP OPTIONS? |
19:45.40 | lvlinux | gevans: if so, with pjsip you can use "qualify_frequency = 15" |
19:46.17 | gevans | lvlinux: what does lowering the qualify_frequency do? it is set to 60 by default |
19:46.30 | igcewieling1 | lvlinux: respond to, not send |
19:47.11 | lvlinux | igcewieling1: huh? |
19:47.29 | lvlinux | gevans: the time is how many seconds between sending OPTIONS packets |
19:47.37 | gevans | yes, thats right it doesn't not respond to sip provider's options request to ensure the call is still alive so provider sends BYE |
19:48.08 | gevans | within a 15 minute time span it never once sends an option packet to the provider |
19:48.14 | gevans | or a repsond to the options |
19:48.25 | WIMPy | OPTIONs are not related to calls. |
19:48.36 | igcewieling1 | lvlinux: his asterisk is not RESPONDING to OPTIONS packets sent to it. |
19:48.38 | gevans | full log is here goo.gl/fcVnRD |
19:49.13 | gevans | WIMPy: Is this hte same WIMPy from jbc? |
19:49.29 | WIMPy | What is jbc? |
19:49.39 | gevans | jupiter broadcasting |
19:49.52 | lvlinux | igcewieling1: 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? |
19:50.13 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
19:50.44 | igcewieling1 | lvlinux: I suspect a fundamental problem somewhere else, but I don't have the time, inclination or knowledge to prove or disprove it. |
19:50.56 | *** join/#asterisk acidfoo- (18303355@gateway/web/cgi-irc/kiwiirc.com/ip.24.48.51.85) |
19:52.14 | acidfoo- | hello world, does Asterisk 11 support end-to-end codec renogiciation to avoid transcoding ? Or it's still nogociated per-leg ? |
19:52.17 | *** join/#asterisk Echo6 (~Echo6@64.136.247.50) |
19:53.07 | gevans | igcewieling1: well thanks, God, I see some issues opened to the fact but no solution |
19:53.12 | igcewieling1 | acidfoo-: asterisk will not transcode if both legs of the call use the same codec. |
19:53.44 | acidfoo- | yeah that part I understand |
19:54.18 | igcewieling1 | However, getting Asterisk to use the codec you want is a bit more complicated. |
19:54.51 | acidfoo- | the two legs might have negociated a different codec with Asterisk, even tho both phone support a common one |
19:54.53 | igcewieling1 | As far ask I know it will not change the codec in the middle of the call. |
19:55.16 | igcewieling1 | acidfoo-: the most common solution I use it to only allow one codec on all the devices. |
19:56.06 | acidfoo- | yeah I can't do that |
19:58.14 | igcewieling1 | acidfoo-: your life will be very interesting |
20:05.13 | *** join/#asterisk DanQuinney (sid18169@gateway/web/irccloud.com/x-gstpetukmfoeyihp) |
20:05.13 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-hzcwkweeuhnqliir) |
20:09.03 | gevans | so 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 |
20:13.08 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-lddbqivpdayzfmwh) |
20:13.24 | *** part/#asterisk gevans (~gevans@air-ses.morganairservices.com) |
20:13.41 | *** join/#asterisk gevans (~gevans@air-ses.morganairservices.com) |
20:13.53 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-genbfmfyaoubsnub) |
20:20.49 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
20:26.09 | *** join/#asterisk dan_j (sid21651@gateway/web/irccloud.com/x-lzylwrhszhqnqzhh) |
20:26.25 | gevans | igcewieling1: 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? |
20:28.07 | gevans | I honestly believe the issue is in these couple of lines: |
20:28.08 | gevans | 40156 [2015-12-01 18:03:22] DEBUG[12560] pjsip: sip_endpoint.c Distributing rdata to modules: Request msg OPTIONS/cseq=12627 (rdata0x2817e48) |
20:28.08 | gevans | <PROTECTED> |
20:28.08 | gevans | <PROTECTED> |
20:28.08 | gevans | <PROTECTED> |
20:28.13 | gevans | <PROTECTED> |
20:28.16 | gevans | <PROTECTED> |
20:28.19 | gevans | <PROTECTED> |
20:28.22 | gevans | <PROTECTED> |
20:28.25 | gevans | <PROTECTED> |
20:28.27 | gevans | <PROTECTED> |
20:28.27 | [TK]D-Fender | PASTEBIN. Do not flood in here |
20:28.34 | gevans | Sorry! |
20:28.52 | WIMPy | Damn. I'm too fast. |
20:29.01 | WIMPy | I unignored before it ended. |
20:29.19 | *** join/#asterisk moy (sid47040@gateway/web/irccloud.com/x-ewrjftewissbriyq) |
20:36.23 | igcewieling1 | gevans: I should have said the server resolving is a PJSIP thing. |
20:36.28 | *** join/#asterisk bipolar (~bipolar@offsite.guru) |
20:37.39 | igcewieling1 | is it evil to point the MX record for my old domain to localhost? 8-) |
20:44.01 | gevans | igcewieling1: 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 |
20:44.39 | WIMPy | OPTIONs are not related to calls. |
20:45.21 | WIMPy | So it is impossible that they would get an options response to an ongoing call. |
20:46.47 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
20:47.08 | gevans | WIMPy: Maybe I'm not explaining my scenario the best, there is a similar issue: goo.gl/fb8TLD |
20:49.56 | file | they are unrelated, what version of Asterisk is in use? |
20:50.11 | K0HAX | Does 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 |
20:51.44 | *** join/#asterisk CFX_ (~deserver@unaffiliated/cfx-/x-0311648) |
20:53.41 | gevans | asterisk 13.1 cert2 |
20:53.59 | gevans | pjsip 2.4 |
20:54.10 | file | 13.1-cert2 does not have support for in-dialog OPTIONS, it was added afterwards |
20:55.06 | file | you 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 |
20:56.25 | gevans | file: thank you.. god! your awesome |
20:58.34 | gevans | what release is the issue adddressed in? |
20:59.17 | gevans | noticed version 13.2 was still listed as affected in issue 24818 |
20:59.28 | file | there is no current certified release with it, it was added in 13.4.0 |
21:00.16 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
21:00.23 | gevans | awesome, uncertified it's going to have to be |
21:21.18 | *** join/#asterisk xpheres (~bodhi64@dslb-088-073-028-049.088.073.pools.vodafone-ip.de) |
21:21.51 | *** join/#asterisk theron (~theron@2620:10d:c091:200::b:cbf9) |
21:22.12 | xpheres | hello, I'm trying to configure a trunk and outbound with a voip service "justvoip" but I'm not able to make it work |
21:24.38 | acidfoo- | how can I see what codec are currently used by two bridged call ? |
21:24.46 | *** join/#asterisk anubisjim (5ce8485d@gateway/web/freenode/ip.92.232.72.93) |
21:24.48 | acidfoo- | (Asterisk11) |
21:25.05 | anubisjim | hiyas all |
21:27.06 | anubisjim | anyone able to assist with an install of asterisk im having a few issues installing it onto fedora |
21:35.23 | xpheres | anyone able to assist to set up outbound trunk with sip service? |
21:37.32 | *** join/#asterisk moe` (~moe@node-be028882.scarlet.cw) |
21:37.40 | moe` | hey kids |
21:39.36 | moe` | ooohhhh 13.6.0 is LTS, nice, didn't realize that. sweet. that's what I last compiled. |
21:45.55 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
21:46.18 | *** part/#asterisk xpheres (~bodhi64@dslb-088-073-028-049.088.073.pools.vodafone-ip.de) |
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21:51.21 | *** part/#asterisk DragonAzul (~DragonAzu@187.208.73.113) |
22:05.49 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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22:21.32 | CFX_ | 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. |
22:49.33 | *** join/#asterisk paraxor (~paraxor@unaffiliated/prazial) |
23:00.06 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca) |
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23:22.51 | edong23 | [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 |
23:43.42 | pankid | Anyone recognize this pbx? http://imgur.com/eZkdUAP I know it is centos/redhat 5 and asterisk 1.8.23 |
23:46.24 | *** join/#asterisk monsterco (~monsterco@bas2-toronto47-1242435693.dsl.bell.ca) |
23:46.50 | monsterco | Hi 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 |
23:47.18 | monsterco | erro I am getting is: WARNING[3006]: pbx.c:8964 ast_pbx_run_app: No such application 'Dial(............. |
23:47.37 | monsterco | this is the line in my call file: Application:Dial(SIP/sip-provider-ip/4166356000) |
23:51.28 | *** part/#asterisk kharwell (kharwell@nat/digium/x-fobxygvqtyycyidc) |