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| 04:01.43 | Kyosh | i have a difficult question (difficult because i'm not sure what happened). asterisk v11.21.0 paging "used to" work. intercom is still fine, but out of the blue (who knows maybe my boss logged in and messed with it, no idea) the paging feature doesn't work. originally paging took place on 299 (all phones), 399 (executive phones), 499 (floor staff phones) and 599 (back office phones). none |
| 04:01.43 | Kyosh | of them are working now. instead you hear "your call cannot be completed as dialed" message for any of the paging numbers dialed. I am completely stumped by this. i set up a test pbx and copied the paging config (working on the test) and pasted it into the production pbx and still nothing. it's almost as the pbx is intentionally ignoring the extension dialed (maybe it doesn't understand |
| 04:01.43 | Kyosh | the context - not sure really). is there any way to figure this out instead of taking down a 150 user pbx and replacing it (I REALLY could not explain that to the boss)? |
| 04:09.50 | Samot | What do the debugs show/ |
| 04:10.11 | Kyosh | <PROTECTED> |
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| 04:20.51 | Kyosh | give me a minute to create a fresh debug on all of the numbers |
| 04:30.29 | Samot | OK |
| 04:50.19 | Kyosh | http://pastebin.com/FCgcsam5 |
| 04:50.30 | Kyosh | thats for 299 |
| 04:53.05 | Samot | No. Not an Asterisk debug. |
| 04:53.29 | Samot | A call debug, set the verbosity to 10 |
| 04:53.36 | Samot | core set verbose 10 |
| 04:54.17 | Samot | Want to see the call going throgh the dialplan without all the extra details like AMI connections. |
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| 04:58.26 | Kyosh | here is 399 http://pastebin.com/dgP7A2hH |
| 04:58.45 | Kyosh | ack |
| 04:58.55 | Kyosh | 10? gotta gimme a min please |
| 04:59.00 | Kyosh | no debug level set? |
| 05:00.30 | Samot | No, none. |
| 05:00.35 | Kyosh | k |
| 05:00.36 | Samot | Don't want those details. |
| 05:07.29 | Kyosh | i created a new one also, page 800 |
| 05:07.35 | Kyosh | same issue |
| 05:07.40 | Kyosh | getting pastebin ready now |
| 05:09.06 | Kyosh | http://pastebin.com/qsvMY1Yh |
| 05:09.10 | Kyosh | thats 299 |
| 05:11.01 | Kyosh | here is 299 again |
| 05:11.03 | Kyosh | http://pastebin.com/RX1czP72 |
| 05:11.16 | Kyosh | sorry 599 |
| 05:12.02 | Kyosh | its as if the dialplan has nothing for it |
| 05:12.12 | Samot | First, this is FreePBX so you will need to continue this in #freepbx |
| 05:12.17 | Samot | Second, -- Executing [599@from-internal-original:1] ResetCDR("SIP/1899-00000d16", "") in new stack |
| 05:12.22 | Samot | You are correct. |
| 05:12.59 | Samot | So move over to #freepbx because you'll need to start providing screenshots of your FreePBX system which is not Asterisk related. |
| 05:14.33 | Kyosh | ack man ok |
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| 14:39.06 | polysics | hello! how do you do RTP analysis for packet loss etc? Wirseshark only does 1 stream at a time |
| 14:39.56 | zhold | perhaps rtpbreak |
| 14:40.22 | zhold | Detects, reconstructs and analyzes any RTP session. |
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| 14:43.10 | polysics | what I am looking for is something that can caggregate stats from a lot of calls, like the volume from a SIPp scenario |
| 14:44.37 | zhold | im looking for a way to remove FAS from my calls |
| 14:44.57 | zhold | i tell you what, ill write you a program to analyse all your calls and the packetloss |
| 14:45.01 | zhold | and you solve my FAS issue :D ? |
| 14:45.11 | zhold | hehe |
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| 15:36.08 | rolledgold | Hello all. |
| 15:40.37 | zhold | hello rolledgold |
| 15:41.01 | rolledgold | having an issue where a carrier is sending a reinvite to change a ptime after the 200 OK and ACK of the initial SDP. For some reason Asterisk does not respond to the new invite and the call dies with a 408. |
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| 15:41.56 | zhold | jesus christ |
| 15:42.02 | zhold | good luck bro i have no idea |
| 15:42.08 | zhold | maybe you can directly msg voipmonk and pay him small fee |
| 15:42.12 | zhold | he has solved my problems many times before |
| 15:42.16 | zhold | truly a voip expert |
| 15:42.36 | zhold | or wait for advice in the channel :) sorry i cant help |
| 15:45.55 | rolledgold | weird thing is that we offer a ptime of 20, they send a 200 with SDP ptime of 30 and then after the call is established they invite with ptime 20. |
| 15:49.55 | igcewieling | *grumble* My "quick hack" is now 4,000 lines of code. |
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| 15:51.06 | mutilator | what's the latest solution for making your own efax server? asterfax/noojee seem to be discontinued for a while.. |
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| 16:00.16 | [TK]D-Fender | Pre-dating even that : Hylafax |
| 16:00.31 | igcewieling | I assume most people roll their won. |
| 16:00.41 | igcewieling | s/won/own/ |
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| 19:24.45 | zhold | [DAHDI_OUT] |
| 19:24.45 | zhold | exten => _0XXXXXXXXX,1,Dial(DAHDI/g0/${EXTEN},23,r) |
| 19:24.45 | zhold | exten => _0XXXXXXXXX,2,Waitforsilence(4250,1) |
| 19:24.45 | zhold | exten => _0XXXXXXXXX,3,Answer() |
| 19:24.52 | drmessano | ~pb |
| 19:24.52 | infobot | [pastebin] a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:24.55 | zhold | <PROTECTED> |
| 19:24.57 | zhold | lucky it was short |
| 19:25.03 | zhold | i want to dial my extension |
| 19:25.21 | zhold | and wait for the ringing to stop then connect the call |
| 19:25.42 | zhold | but what i have above is incorrect but shows pretty much what i want in order |
| 19:25.51 | zhold | does anyone know the correct way to write this? |
| 19:28.37 | [TK]D-Fender | that doesn't work at all |
| 19:28.44 | [TK]D-Fender | Dial() is a complete call. |
| 19:28.58 | [TK]D-Fender | when it answers.. you STAY in that call. When it ends, so does your call |
| 19:29.07 | zhold | im feeling ya |
| 19:29.09 | zhold | thats whats happening |
| 19:29.13 | [TK]D-Fender | Things do not continue when the call gets answered |
| 19:29.13 | zhold | but Fender i have a unique problem here |
| 19:29.32 | [TK]D-Fender | You need a better description of what you want to have happen, how it is to start, etc |
| 19:29.57 | zhold | i am using pulse dialing over a chan_dahdi adapter on a PSTN that doesnt give me polarity reversal to know when the call has actually started |
| 19:30.15 | zhold | so im left to remove the FAS on the call through some strange method |
| 19:30.44 | zhold | caller ---> dials..... --> (asterisk ringing) ---> (billing starts) ----> PSTN ringing heard -----> callepicks up |
| 19:30.48 | zhold | i need it like |
| 19:30.57 | [TK]D-Fender | So far dial does that.... |
| 19:31.04 | zhold | caller ---> dials -----> (asterisk ringing) -----> calle picks up ("Hello?") ---> billing starts |
| 19:31.46 | [TK]D-Fender | if you need Dial() to wait before dialing the digits... then add "w"'s before the number |
| 19:32.25 | zhold | Fender do you know what FAS is ? |
| 19:33.05 | zhold | WHY DO WE EXPERIENCE FAS? |
| 19:33.06 | zhold | FAS normally takes place when there is no synchronization between a VoIP leg and PSTN leg of the call on a VoIP-to-PSTN gateway. When the call reaches the gateway from the VoIP network, the gateway tries to establish a connection with the called number, but due to incorrect configuration it cannot detect the states of the call, which are advertised by the PSTN network (the states are: "called party ringing", "called party connected"). And thus the gateway f |
| 19:33.26 | zhold | WHY FAS IS A "BAD THING"? |
| 19:33.26 | zhold | FAS is considered to be a negative phenomenon due to two reasons: |
| 19:33.26 | zhold | 1. The calling enduser may notice that he/she is billed for non-connected calls |
| 19:33.38 | zhold | seems everyone on #asterisk never heard of this term before thought i'd just paste here.... |
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| 19:34.24 | [TK]D-Fender | That isn't the more common term |
| 19:34.30 | [TK]D-Fender | "Call Progress" is |
| 19:34.46 | [TK]D-Fender | And if you're using anaolg this could be an issue |
| 19:34.49 | [TK]D-Fender | not all PSTN is analog |
| 19:35.00 | [TK]D-Fender | So their wording makes assumptions |
| 19:35.15 | zhold | yep im using analog |
| 19:35.20 | zhold | you could say i have incorrect configuration |
| 19:35.27 | [TK]D-Fender | "a VoIP-to-PSTN gateway" <- and that is ALSO vague |
| 19:35.37 | zhold | but in reality right now i have no solid way to detect when the person has actually picked up the call |
| 19:35.39 | [TK]D-Fender | DAHDI supports call progress if you tell it to, and what standard |
| 19:35.40 | zhold | thats my main problem |
| 19:35.54 | [TK]D-Fender | But you risk calls getting disconnected due to false reads, etc |
| 19:35.56 | zhold | my chan_dahdi.conf has all the correct settings for polarity detection |
| 19:36.02 | zhold | but simply fails to recieve one |
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| 19:36.33 | [TK]D-Fender | make sure you have a proper indications.conf for audio detection as well |
| 19:36.41 | [TK]D-Fender | and pastebin your configs |
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| 19:42.29 | igcewieling | looks like zhold is trying the randomlydisconnectmycalls=yes |
| 19:42.36 | igcewieling | config option |
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| 19:44.16 | zhold | i dont even mind if your calls get randomly disconnected |
| 19:44.21 | zhold | as long as im not billing you for mins your not using |
| 19:44.37 | zhold | infact disconnect all calls after 5 mins just for fun |
| 19:44.42 | zhold | just please have normal call :/ |
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| 19:55.10 | igcewieling | My customers would be at my doors with torches and pitchforks I they had their calls randomly disconnected. |
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| 19:57.16 | zhold | yeah wait till you have FAS on your line |
| 19:57.20 | zhold | they wont even use your service at all |
| 19:57.29 | zhold | at least you can have a quick word to ya mum |
| 19:57.49 | zhold | ;) |
| 20:00.25 | igcewieling | facility associated signaling? We use it for several customers to combine multiple PRIs |
| 20:00.36 | zhold | no false answer supervision :( |
| 20:01.02 | zhold | hey [TK]D-Fender |
| 20:01.06 | zhold | i just found the indications.conf of the country |
| 20:01.11 | zhold | someone put them up on the net |
| 20:01.15 | igcewieling | you know all those issues go away if you switch away from analog. |
| 20:02.08 | zhold | i really do man :( i really do |
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| 23:08.03 | *** join/#asterisk BakaKuna (~BakaKuna@145.129.205.133) |
| 23:17.47 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:34b1:6abf:8949:6124) |
| 23:22.58 | *** join/#asterisk celer (~celer@pc1.robertwaltersjp-unet.ocn.ne.jp) |