| 00:36.49 | *** join/#asterisk HyP3r (~HyP3r@unaffiliated/hyp3r) |
| 00:38.24 | HyP3r | Hello, I try to create a sip-trunk where the sip-trunk is just a extension of a differnet pbx. My aim is that when I dial e.g. 200 in my pbx the sip-trunk is dialing a specific number in its pbx. So for example the sip-trunk has the number 90 and it dials 91. |
| 00:38.44 | HyP3r | My question is: do you have a example manual how I should create that? |
| 00:39.13 | HyP3r | And the second question: are sip-trunks and outgoing routes the correct way to realize that? |
| 00:39.31 | Samot | Asterisk doesn't have outgoing routes. |
| 00:40.30 | Samot | HyP3r: Is this an install of pure Asterisk or something that is using Asterisk? |
| 00:40.32 | HyP3r | I'm using FreePBX. And under Connectivvity there are Outbound Routes |
| 00:40.40 | HyP3r | :) |
| 00:41.00 | HyP3r | I hope its ok to ask here questions about FreePBX too...? |
| 00:41.15 | Samot | It's better to ask in FreePBX. |
| 00:41.17 | Samot | Now you're asking about specific things to it. |
| 00:41.26 | *** part/#asterisk MLC (~MLC@63.249.40.11) |
| 00:41.54 | HyP3r | Ok. Thanks. I always thought that freepbx is just a webinterfaace for asterisk |
| 00:42.29 | Samot | It basically is, however, it does things in a certina way. |
| 00:42.39 | Samot | It generates dialplan, it does the configs. |
| 00:42.45 | Samot | So it's not 1:1 |
| 00:43.05 | HyP3r | Ok |
| 00:46.09 | *** join/#asterisk tmoore (~tmoore@2601:40e:8180:db80:cd09:7d0f:4650:996c) |
| 01:33.16 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 01:36.53 | *** join/#asterisk dakudos (~dakudos@c-73-229-175-50.hsd1.co.comcast.net) |
| 01:45.29 | *** join/#asterisk deavmi (~quassel@165.255.253.162) |
| 02:03.08 | *** join/#asterisk chandoo (~chandoo@pool-108-50-150-199.nwrknj.fios.verizon.net) |
| 02:53.41 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 04:33.00 | *** join/#asterisk jkroon (~jkroon@165.16.203.107) |
| 04:52.41 | *** join/#asterisk gerhard7 (~gerhard7@ip5657ee30.direct-adsl.nl) |
| 05:00.35 | *** join/#asterisk scampbell (~scampbell@mail.scampbell.net) |
| 06:25.13 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 06:38.44 | *** join/#asterisk tomaluca95 (~quassel@kde/developer/tomaluca) |
| 06:49.23 | *** join/#asterisk ganbold (~ganbold@66.85.186.234) |
| 07:11.46 | *** join/#asterisk zapata_ (~zapata@2a02:1748:fad4:7260:f4ae:5fa:7291:d26a) |
| 07:19.48 | *** join/#asterisk pchero_work (~pchero@87.213.247.82) |
| 08:07.59 | *** join/#asterisk tsal (~tsal@i59F52D6A.versanet.de) |
| 08:20.41 | *** join/#asterisk jkroon (~jkroon@165.16.203.61) |
| 09:08.50 | *** join/#asterisk ganbold_ (~ganbold@66.85.186.234) |
| 09:35.55 | *** join/#asterisk Chotaire (chotaire@unaffiliated/chotaire) |
| 11:31.13 | *** join/#asterisk zapata_ (~zapata@2a02:1748:fad4:7260:e96e:766f:6060:33fb) |
| 12:21.04 | *** join/#asterisk pabe (~pabe@81.24.66.208) |
| 12:21.04 | *** join/#asterisk forgotmynick (uid24625@gateway/web/irccloud.com/x-zgpaydhpddymeiic) |
| 12:42.48 | *** join/#asterisk aoeui (~aoeui@unaffiliated/aoeui) |
| 13:34.41 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:53.57 | *** join/#asterisk sahmed (~sahmed@cpe-70-114-236-63.austin.res.rr.com) |
| 14:28.43 | *** join/#asterisk [TK]D-Fender (~joe@216.191.106.165) |
| 14:56.28 | *** join/#asterisk tozhu (~tozhu@171.221.44.75) |
| 15:00.25 | tozhu | hello , I have a question, when build the asterisk with pjproject, I found a lot of option is disabled, such as g711 and g722, even gsm is disabled too, may I know why? and how to enable them? https://pastebin.com/gw8MeaYB big thanks |
| 15:02.36 | file | we don't use the media stack in pjproject, so those aren't built |
| 15:02.56 | file | there's no reason to enable them for bundled as they wouldn't be used or matter |
| 15:03.11 | *** join/#asterisk eharris (~eharris@unaffiliated/eharris) |
| 15:03.30 | tozhu | okay, got it, thank you |
| 15:06.45 | *** join/#asterisk spatel (~spatel@static-71-174-102-211.bstnma.fios.verizon.net) |
| 15:11.30 | *** join/#asterisk pabe (~pabe@81.24.66.208) |
| 15:24.35 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-itbpdfipldtacszm) |
| 15:24.35 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:27.00 | *** join/#asterisk cp- (~cp-@b157153.ppp.asahi-net.or.jp) |
| 15:30.24 | *** join/#asterisk Janos (~Janos@201.204.94.76) |
| 15:32.57 | *** join/#asterisk chandoo (~chandoo@pool-108-50-150-199.nwrknj.fios.verizon.net) |
| 15:49.15 | igcewieling | gads I hate cellphones. |
| 15:56.00 | *** join/#asterisk deavmiQuassel (~quassel@165.255.253.162) |
| 15:56.52 | *** join/#asterisk spatel (~spatel@static-71-174-102-210.bstnma.fios.verizon.net) |
| 16:04.08 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-bicvkayyyrpavdps) |
| 16:04.08 | *** mode/#asterisk [+o bford] by ChanServ |
| 16:28.16 | *** join/#asterisk dw5304 (~dw5304@xigmanas/dw5304) |
| 16:30.50 | dw5304 | good morning guys, Im having a bit of an issue with a pjsip trunk. asteirsk 15.7.3, I have a aor that asterisk can see with out an issue but when i ask it to reload the endpoint it states "unable to retrieve endpoint {endpointname} |
| 16:31.10 | dw5304 | commend 'pjsip reload qualify endpoint {endpointname} failed" |
| 16:31.33 | dw5304 | looking for thoughts as to what i should check according to the pjsip.conf the endpoint exists |
| 16:31.34 | igcewieling | try: module reload res_pjsip.so |
| 16:32.25 | dw5304 | Module 'res_pjsip.so' reloaded successfully. |
| 16:32.26 | dw5304 | [2020-02-12 10:31:58] ERROR[34640]: res_pjsip_config_wizard.c:1087 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf' |
| 16:32.26 | dw5304 | [2020-02-12 10:31:58] NOTICE[34640]: sorcery.c:1285 sorcery_object_load: Type 'system' is not reloadable, maintaining previous values |
| 16:32.26 | dw5304 | [2020-02-12 10:31:58] ERROR[34640]: res_pjsip_config_wizard.c:1087 object_type_loaded_observer: Unable to load config file 'pjsip_wizard.conf' |
| 16:32.38 | dw5304 | any issues with that? |
| 16:32.59 | igcewieling | that depends on if you are using the pjsip wizard. |
| 16:33.31 | dw5304 | nope |
| 16:33.42 | dw5304 | endpoint didnt show up |
| 16:33.49 | igcewieling | you can put this in /etc/asterisk/modules.conf: noload => res_pjsip_config_wizard.so |
| 16:34.02 | dw5304 | kk |
| 16:34.06 | igcewieling | but you'll have to restart asterisk for that to apply |
| 16:34.10 | dw5304 | got it |
| 16:34.18 | dw5304 | save that for later then |
| 16:34.57 | dw5304 | so yeah module reload didnt seem to do anything |
| 16:34.58 | igcewieling | try this, then reload pjsip: module unload res_pjsip_config_wizard.so |
| 16:35.14 | igcewieling | I assume some other issue than a reload issue. |
| 16:35.26 | file | what's the actual config? |
| 16:35.33 | dw5304 | appernelty its in use lol.. |
| 16:35.34 | dw5304 | count of 1 |
| 16:35.45 | dw5304 | sure sec |
| 16:36.46 | dw5304 | https://pastebin.com/EMCdS0jf |
| 16:37.21 | file | and does it show up in "pjsip show endpoints"? |
| 16:37.40 | dw5304 | no |
| 16:38.08 | dw5304 | but it does show up in identiry |
| 16:38.13 | file | is the file included in pjsip.conf? |
| 16:38.37 | dw5304 | yes |
| 16:39.00 | dw5304 | #include pjsip.endpoint.conf |
| 16:39.29 | file | wait, 15 something? |
| 16:39.35 | dw5304 | yes 15.7 |
| 16:39.53 | file | send_connected_line doesn't exist there |
| 16:40.00 | dw5304 | lol |
| 16:40.07 | dw5304 | ok ill will report this bug to freepbx then |
| 16:40.18 | dw5304 | let me change this and see if it shows up |
| 16:41.00 | file | https://issues.freepbx.org/browse/FREEPBX-21062?jql=text ~ "send_connected_line" |
| 16:41.35 | dw5304 | thank you |
| 16:41.41 | *** join/#asterisk salviadud (~rgonzalez@189.207.212.148) |
| 16:42.21 | dw5304 | looks good now |
| 17:00.06 | *** join/#asterisk lankanmon (~LKNnet@CPEb4fbe4e331bd-CM64777d632380.cpe.net.cable.rogers.com) |
| 17:19.28 | *** join/#asterisk Helenah (~s98259@unaffiliated/iveeee) |
| 17:33.36 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:101a:2472:a0e3:dfa9:786a:4fd4) |
| 17:41.18 | *** join/#asterisk sarlalian (~sarlalian@107.170.239.102) |
| 18:14.55 | *** join/#asterisk defsdoor (~Andrew@cpc120600-sutt6-2-0-cust232.19-1.cable.virginm.net) |
| 18:25.05 | *** join/#asterisk khronos (~khronos@072-238-042-188.res.spectrum.com) |
| 18:25.22 | khronos | Hi guys. |
| 18:27.46 | khronos | Have an issue I am experiencing with Asterisk 13.11.2 and even Asterisk 17.1.0. |
| 18:28.41 | khronos | When I receive calls from Inteliquant they are not processed on these two versions but are processed on a much older 1.4 install. |
| 18:29.27 | khronos | I can receive calls from all my other providers using the same extensions.conf on all versions except for this provider. |
| 18:32.42 | seanbright | sucks |
| 18:36.20 | seanbright | are you getting any kinda of error message? |
| 18:36.43 | *** join/#asterisk jkroon (~jkroon@165.16.203.107) |
| 18:37.39 | khronos | Not anything really useful. extension 2345678900 not found in context inbound as if I don't have the DID programmed in the system even though I do. |
| 18:38.21 | seanbright | well, good luck |
| 18:39.49 | khronos | All DIDs from other providers are in this context as well and they work just fine. |
| 18:40.37 | seanbright | paste the exact error message |
| 18:42.46 | igcewieling | does "dialplan show 2345678900@inbound" give you anything helpful? |
| 18:45.01 | *** join/#asterisk matrix1233 (~matrix123@2a04:cec0:101a:2472:a0e3:dfa9:786a:4fd4) |
| 18:47.14 | seanbright | well this has been fun |
| 18:50.02 | *** join/#asterisk Ai9zO5AP (~BQcdf9eiZ@105.158.149.212) |
| 18:53.06 | *** join/#asterisk dougquaid (dougquaid@gateway/vpn/privateinternetaccess/dougquaid) |
| 18:56.16 | dougquaid | I'm having a weird issue with an Android device running the Mizudroid SIP client. I had setup a test FreePBX VM and made two extensions, 101 and 102. A few days ago I completed my testing and shut down the VM but forgot to uninstall Mizudroid from the device with extension 101. However, in the past few minutes, extension 101 received two incoming calls from extension 102, despite my FreePBX VM being |
| 18:56.22 | dougquaid | powered off. How is this possible? Could this be a bug with Mizudroid? |
| 18:58.06 | igcewieling | dougquaid: by default most SIP phones will accept calls from anywhere, including bots scanning for open SIP servers. |
| 18:59.01 | igcewieling | On Polycoms the setting is voIpProt.SIP.strictUserValidation="1" |
| 19:02.17 | dougquaid | I've read about that but I'm not sure that's the issue since I haven't forwarded any ports on my router to the android device and my router doesn't have upnp enabled |
| 19:03.57 | lagzilla | SIP ALG might be your issue too |
| 19:04.04 | igcewieling | if the phone is trying to register to your old server will put a hole in the NAT, allowing incoming packets |
| 19:04.27 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:04.32 | igcewieling | that is called one-to-many NAT or full cone NAT IIRC |
| 19:04.51 | khronos | igcewieling: the dialplan function shows it being routed properly to the client. |
| 19:05.24 | khronos | igcewieling: Weird thing is that if this DID had been with one of my other providers this issue wouldn't even occur. |
| 19:05.44 | dougquaid | Interesting, I didn't know about one-to-many nat. That very well could be the issue |
| 19:05.54 | khronos | igcewieling: I can confirm that the 20 or 30 DIDs I have with this provider are the only ones with problems. |
| 19:06.03 | seanbright | paste the exact error message |
| 19:08.21 | seanbright | khronos: pretty please - paste the exact error message |
| 19:13.58 | seanbright | sigh |
| 19:17.25 | *** join/#asterisk eharris (~eharris@unaffiliated/eharris) |
| 19:17.42 | *** join/#asterisk hfb (~hfb@47.139.23.19) |
| 19:18.38 | *** join/#asterisk _0x5eb_ (~seb@130.104.204.80) |
| 19:48.08 | khronos | [Feb 11 15:46:48] NOTICE[3022][C-00000233] chan_sip.c: Call from 'in-inteliquent2' (1.2.3.4:5060) |
| 19:49.21 | khronos | to extension '2345678900' rejected because extension not found in context 'inbound' |
| 19:50.07 | khronos | If this DID had been with say level3 or other providers I have the DID would pass through just fine. |
| 19:54.55 | *** join/#asterisk Nate54 (26797142@38.121.113.66) |
| 19:55.27 | seanbright | khronos: pastebin the output of 'dialplan show 2345678900@inbound' |
| 19:56.08 | seanbright | i have a feeling that your redaction of the phone number is hiding useful detail |
| 19:56.58 | igcewieling | seanbright: perhaps a leading + ? |
| 19:57.17 | khronos | No, hwat I'm saying is this. |
| 19:57.22 | khronos | what I'm saying is this |
| 19:58.31 | khronos | 10 digit DIDs from all other providers work. |
| 19:58.48 | seanbright | i understand what you are saying |
| 19:58.49 | khronos | Just not this particular one. |
| 19:58.55 | *** join/#asterisk MLC (~MLC@63.249.40.11) |
| 19:59.00 | seanbright | asterisk is telling you that it is not able to match that DID to your dialplan |
| 19:59.10 | khronos | and if my internal cutomers dial the DID the calls pass through. |
| 19:59.13 | khronos | customers |
| 19:59.13 | seanbright | it has nothing to do with Inteliquant |
| 19:59.26 | seanbright | that is my opinion based on the information you have provided |
| 19:59.27 | [TK]D-Fender | That extension does not have a match in that context |
| 19:59.32 | [TK]D-Fender | It means exactly what it says |
| 19:59.54 | MLC | voip.ms gave me a 503 with these headers in response to an INVITE. X-Asterisk-HangupCause: Unallocated (unassigned) number |
| 19:59.55 | MLC | X-Asterisk-HangupCauseCode: 1 |
| 20:00.01 | khronos | I agree, but what do you say if I dial it with in the system itself and the calls passes properly? |
| 20:00.01 | [TK]D-Fender | Either you're sending it to the wrong context, or you don't have an exten there to match. |
| 20:00.02 | MLC | Does that mean that the called number is invalid? |
| 20:00.21 | [TK]D-Fender | <khronos> I agree, but what do you say if I dial it with in the system itself and the calls passes properly? <- obviously it's not the same combination of exten & context |
| 20:00.45 | khronos | If I dial the number with in the same system the call passes though just fine. |
| 20:00.49 | khronos | through |
| 20:00.55 | seanbright | you aren't reading |
| 20:00.55 | [TK]D-Fender | khronos, You're giving a description... and not showing the actual evidence to show why you think it should work |
| 20:00.58 | khronos | I don't get the inbound context error. |
| 20:01.24 | [TK]D-Fender | If I tell you to find an item in the TOP shelp and it['s in the BOTTOM shelf it doesn't matter if the description of the item is eprfect |
| 20:01.25 | khronos | The same config files exist on asterisk 1.4 server and calls are passing though just fine. |
| 20:01.25 | seanbright | and for that reason... |
| 20:01.27 | seanbright | i'm out |
| 20:01.29 | [TK]D-Fender | you're looking in the wrong PLACE |
| 20:01.34 | khronos | through |
| 20:01.41 | [TK]D-Fender | SHO US <---- |
| 20:01.46 | [TK]D-Fender | SHOW |
| 20:02.16 | [TK]D-Fender | <khronos> The same config files exist on asterisk 1.4 server and calls are passing though just fine. <- the error is describing exactly what it's looking for . SHOW US WHY YOU THINK IT'S WRONG. |
| 20:02.51 | khronos | If I use the same config files on both servers one server passes the calls from this provider just fine. |
| 20:02.52 | [TK]D-Fender | Asterisk version has no impact on what that error is saying |
| 20:03.04 | [TK]D-Fender | You're giving excuses still.... |
| 20:03.07 | [TK]D-Fender | it's wrong. |
| 20:03.09 | khronos | I know the error doesn't make sens. |
| 20:03.14 | khronos | doesn't make sense. |
| 20:03.15 | [TK]D-Fender | If you want help in fixing it, show us |
| 20:03.35 | khronos | The error says the DID isn't in the extensions file, but I know damn well it is. |
| 20:03.40 | [TK]D-Fender | "But I did everything right!". No. You didn't. That error is blatant evidence |
| 20:03.50 | [TK]D-Fender | The error isn't lying |
| 20:03.56 | [TK]D-Fender | And you're refusing to show the evidence |
| 20:04.14 | [TK]D-Fender | Every time we see someone say "But I did everything right" ... they didn't |
| 20:04.22 | khronos | Ok, I'll give you an example of completed call with did from other server. |
| 20:04.27 | [TK]D-Fender | no |
| 20:04.34 | seanbright | how would that be useful? |
| 20:04.38 | [TK]D-Fender | Show us the config for THIS server that SHOULD process the call |
| 20:04.42 | khronos | What I'm saying is this. |
| 20:04.48 | [TK]D-Fender | You're trying to pawn off OTHER crap |
| 20:04.49 | seanbright | we fucking know what you are saying |
| 20:04.55 | seanbright | but you're wrong |
| 20:04.56 | [TK]D-Fender | You're insisting you're right |
| 20:04.57 | khronos | I have the same sip.conf and extensions.conf files on both servers. |
| 20:04.59 | [TK]D-Fender | and NOT showing us |
| 20:05.03 | [TK]D-Fender | Which is useless |
| 20:05.23 | khronos | Common dial plan on both systems. |
| 20:05.33 | [TK]D-Fender | Stop wasting our time |
| 20:05.36 | seanbright | ok, this has to be a troll at this point |
| 20:05.42 | [TK]D-Fender | You are coming up with excuses |
| 20:05.46 | seanbright | i refuse to believe this is a real person with a real problem |
| 20:05.47 | [TK]D-Fender | the error means it |
| 20:06.05 | Nate54 | Well, if he's gonna just waste time I've got a fun one: |
| 20:06.07 | Nate54 | Asterisk: 16.5.1 |
| 20:06.08 | Nate54 | send ~9 RTP packets to the phone and then it just stops. I continue to see bidirectional RTP packets between Vitelity and Asterisk but after those initial few packets Asterisk stops sending RTP to the phone but the phone continues sending RTP packets. Looking at the RTP debug output it doesn't say anything but just stops sending the packets. When |
| 20:06.08 | Nate54 | this happens the user can speak into their phone and the person dialing in can hear them but the user can't hear the person dialing in (because Asterisk stops sending RTP to the phone). I'm a little baffled at what's going on and I'm opening to suggestions on what could be going on. My Asterisk configuration is pretty out of the box with only |
| 20:06.09 | Nate54 | extensions.conf, sip.conf, voicemail.com and manager.conf edited. |
| 20:06.16 | Samot | seanbright: I refuse to believe you don't have real problems. |
| 20:06.44 | seanbright | i don't know what that measn |
| 20:06.46 | seanbright | means* |
| 20:07.42 | Samot | It means, you have problems. |
| 20:07.47 | seanbright | ok? |
| 20:08.00 | seanbright | what does this have to do with anything? |
| 20:08.09 | Samot | Oh I thought we were trolling you. |
| 20:08.23 | Samot | Is that not the thing? |
| 20:08.43 | seanbright | sorry, i'm confused. |
| 20:08.57 | Samot | No, no. It's my bad. Totally read the room wrong. |
| 20:09.15 | Samot | Continue. |
| 20:11.07 | file | Nate54: if you disable strictrtp in rtp.conf does it work? |
| 20:12.16 | Nate54 | I'll give it a try. Unfortunately the issue is pretty sporadic so I don't know if I'll be able to get you an answer real quick but I'll give it a go! Thank you! |
| 20:16.28 | khronos | Ok, here's what I've got. |
| 20:16.58 | khronos | Here's the cdr from a call on the server that works. |
| 20:17.02 | khronos | "","7186282811","5162899019","fba_did","7186282811","SIP/in-inteliquent2-c41e37d0","SIP/fba-cc-05b642d0","Dial","SIP/5162899019@fba-cc","2020-02-12 15:08:00","2020-02-12 15:08:05","2020-02-12 15:10:13",133,128,"ANSWERED","DOCUMENTATION","1581538080.230603","" |
| 20:17.32 | khronos | In extensions.conf I have a line that looks like: |
| 20:17.48 | khronos | exten => 5162899019,1,Goto(fba_did,${EXTEN},1) |
| 20:17.52 | [TK]D-Fender | <PROTECTED> |
| 20:17.57 | khronos | This sends it to the context for the client. |
| 20:18.07 | khronos | Thar routes to their system. |
| 20:18.10 | khronos | that routes |
| 20:18.29 | [TK]D-Fender | <khronos> [Feb 11 15:46:48] NOTICE[3022][C-00000233] chan_sip.c: Call from 'in-inteliquent2' (1.2.3.4:5060) |
| 20:18.29 | [TK]D-Fender | <khronos> to extension '2345678900' rejected because extension not found in context 'inbound' |
| 20:18.34 | [TK]D-Fender | not the same |
| 20:18.37 | [TK]D-Fender | And 1 line means nothing |
| 20:19.15 | [TK]D-Fender | You're still playing this shitty "secret squirrel" game. Stop wasnting our time and actually show us a meaningful amount of config and actual call failure output. |
| 20:26.35 | MLC | At this pause in the action... |
| 20:26.37 | MLC | voip.ms gave me a 503 with these headers in response to an INVITE. X-Asterisk-HangupCause: Unallocated (unassigned) number. X-Asterisk-HangupCauseCode: 1. Does that mean that the called number is invalid? |
| 20:26.38 | *** join/#asterisk war9407 (war@2600:4040:4001:1e00::f7aa) |
| 20:27.23 | seanbright | it means that the called number is unallocated (unassigned) |
| 20:27.33 | seanbright | that'll be $10 |
| 20:27.43 | MLC | :) check's in the mail |
| 20:28.34 | MLC | unallocated meaning not assigned to a phone or company,etc? |
| 20:29.18 | seanbright | that would be my interpretation, though you would expect that to return a 404 |
| 20:29.43 | seanbright | but response codes are all over the map because SIP sucks and everyone that implements SIP sucks |
| 20:29.55 | seanbright | sorry for the technical jargon |
| 20:30.03 | MLC | LOL |
| 20:33.35 | igcewieling | a 503 indicates a problem. 404 is "number not found" |
| 20:40.09 | sibiria | add to that the pleasure of dealing with shit providers who pull stupid tricks like responding 486/busy when they're congested, just because it looks better |
| 20:47.15 | seanbright | khronos: we don't need stories. we don't need to know what works. we need two things: 1. the exact error message you are getting (don't replace the phone number in the error message - no one cares what phone number you have) and 2. the output of 'dialplan show <the exact phone number from the error message>@inbound' |
| 20:47.35 | [TK]D-Fender | seanbright, not holding my breath on that one... |
| 20:48.28 | [TK]D-Fender | seanbright, If we haven't gotten it in the first 3/4 hour ... |
| 20:48.52 | sibiria | and if nothing within 72 hours, presume "op" is dead |
| 20:49.37 | seanbright | i don't understand why you would come to an IRC channel looking for help and then not provide information to those that are actually _trying_ to help |
| 20:49.49 | seanbright | i'm not cut out for support |
| 20:51.57 | [TK]D-Fender | Bring car to garage. Refuse to take it into the repair bay. Refuse to allow them access under the hood. Demand they fix the problem. WIN. |
| 20:55.06 | khronos | [Feb 11 16:39:58] NOTICE[1738][C-000000ce] chan_sip.c: Call from 'in-inteliquent2' (216.221.155.24:5060) to extension '5162899002' rejected because extension not found in context 'inbound' |
| 20:57.13 | seanbright | ok, so that's #1, now the output of "dialplan show 5162899002@inbound" |
| 20:58.14 | khronos | Context 'inbound' created by 'pbx_config' ] '_516289900[0-9]' => 1. Goto(fba_did,${EXTEN},1) [extensions.conf:4533] |
| 20:58.24 | khronos | -= 1 extension (1 priority) in 1 context. |
| 21:00.51 | *** join/#asterisk Hooloovo0 (Hooloovoo@sorunome.de) |
| 21:01.25 | Hooloovo0 | I've got a sangoma a104 card hooked to an adit600, and I'm having trouble with pulse dialing |
| 21:02.00 | Hooloovo0 | something seems to be interpreting pulses sometimes as hookflashes, sometimes mising pulses, etc |
| 21:02.11 | khronos | If I connect a phone into the same context of inbound and dial the 516 number the call routes fine. |
| 21:02.22 | Hooloovo0 | How do I debug this? I'm not really sure where pulses are actually interpreted |
| 21:02.31 | khronos | It just seems to be handled differently from the peer coming from the provider. |
| 21:04.47 | [TK]D-Fender | I'm not trusting this as we see it now.... |
| 21:04.47 | [TK]D-Fender | "dialplan show @inbound" and the full call output... |
| 21:05.26 | [TK]D-Fender | It shouldn't have taken an hour to get proper backup for this... |
| 21:11.21 | khronos | Sort of giving full access to my server I am unsure how to prove to you guys my experience is actually real. |
| 21:11.24 | khronos | short |
| 21:12.12 | khronos | Logically if you use the same configs on two servers your experience should be the same. |
| 21:12.55 | khronos | I have lots of other DIDs in the inbound context that are with other providers and those are evaluated fine. Just the Inteliquant ones are not one one server. |
| 21:13.41 | khronos | are mpt working on one of the servers. |
| 21:13.59 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 21:13.59 | *** mode/#asterisk [+o ChanServ] by wilhelm.freenode.net |
| 21:21.50 | [TK]D-Fender | "same configs" means nothing |
| 21:21.57 | [TK]D-Fender | all we care about is the one with the issue |
| 21:22.16 | [TK]D-Fender | Stories about "other DID's" is equally worthless |
| 21:22.44 | [TK]D-Fender | These are all deflections from the server at hand |
| 21:22.58 | [TK]D-Fender | <[TK]D-Fender> "dialplan show @inbound" and the full call output... |
| 21:23.15 | [TK]D-Fender | Full context dump and full unfiltered call debug |
| 21:23.43 | khronos | Context output is over 2000 lines. |
| 21:24.04 | [TK]D-Fender | Don't care |
| 21:24.14 | [TK]D-Fender | I want something I should be able to trus |
| 21:24.16 | [TK]D-Fender | t |
| 21:27.57 | khronos | Ok. |
| 21:28.00 | khronos | I have two text files. |
| 21:28.13 | khronos | One with output of dialplan show inbound and dialplan show fba_did |
| 21:28.24 | khronos | and the second has show dialplan fba_did |
| 21:29.46 | drmessano | Pastebin |
| 21:30.47 | khronos | Don't know if I can do this from a text terminal but I'll try. |
| 21:35.31 | drmessano | Then access it from a machine where you can paste the terminal output to a web browser |
| 21:35.43 | drmessano | This is a standard ask in any support forum |
| 21:37.16 | *** join/#asterisk Nate54 (26797142@38.121.113.66) |
| 21:39.21 | khronos | I actually probably need to look at this from another point of view as well. I will have to wait until people go home, but it might help do turn on a sip debug for the peer and then pass a call into one of the dids that does not work. |
| 21:40.02 | khronos | and see if the sip debug shows anything different between the working / non working servers besides the context error. |
| 21:41.47 | [TK]D-Fender | taht basic error should show it |
| 21:41.57 | [TK]D-Fender | as long as we're not loking at any other box |
| 21:46.00 | igcewieling | y'all could simply ignore people who don't do what is suggested. |
| 22:06.06 | *** join/#asterisk i9zO5AP (~BQcdf9eiZ@45.12.221.227) |
| 22:25.57 | *** join/#asterisk GoldenBear_ (~gb@152.89.160.120) |
| 22:45.20 | *** join/#asterisk netman (~netman@185.94.249.222) |
| 23:11.47 | *** join/#asterisk spatel (~spatel@pool-96-237-230-175.bstnma.fios.verizon.net) |
| 23:23.09 | *** join/#asterisk scampbell (~scampbell@mail.scampbell.net) |
| 23:29.01 | drmessano | igcewieling: I whole heartedly agree |
| 23:34.39 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |