| 00:09.19 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 00:20.10 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 00:30.12 | ChkDigit | If I have a call coming in with: Context: incoming-btt Extension: 200, should Pickup(200@incoming-btt) not answer that call? |
| 00:33.39 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 00:34.30 | [TK]D-Fender | "core show application pickup" |
| 00:34.43 | ChkDigit | Thanks, did that. |
| 00:36.57 | ChkDigit | Console tells me pickup_exec: No target channel found for 200@incoming-btt. |
| 00:37.39 | ChkDigit | Note 3 on core show application pickup tells me I should expect it to pickup. |
| 00:40.01 | [TK]D-Fender | pastebin the instructions.... |
| 00:41.56 | ChkDigit | Looks like the very last little bit where the context is defined by Dial() got me. |
| 00:44.18 | ChkDigit | And... it isn't. It is using the default context for the SIP peer... |
| 00:48.13 | ChkDigit | So, I have my problem solved, but this does not feel right. |
| 00:48.26 | ChkDigit | I have a receptionist phone at SIP/221 |
| 00:48.41 | ChkDigit | It has a default context of from-internal |
| 00:49.29 | ChkDigit | A call comes in from the PSTN over a DAHDI board, which goes to 200@incoming-btt |
| 00:49.51 | ChkDigit | after some stuff that context decides to Dial(SIP/221) |
| 00:50.18 | ChkDigit | At which point context needs to match from-internal, but extension needs to match 200. Is this expected? |
| 00:51.08 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 00:51.34 | [TK]D-Fender | Show the call and the instructions |
| 00:55.34 | ChkDigit | I'm seeing how they match up, I'm just complaining that it doesn't seem intuitive. Mostly because I don't understand why Dial() would change context. |
| 00:56.12 | [TK]D-Fender | it doesn't |
| 00:56.51 | ChkDigit | [incoming-btt] |
| 00:57.03 | ChkDigit | exten 200,1,Dial(SIP/221) |
| 00:58.07 | ChkDigit | core show channel SIP/221... |
| 00:58.11 | ChkDigit | <PROTECTED> |
| 00:58.11 | ChkDigit | <PROTECTED> |
| 00:58.11 | ChkDigit | <PROTECTED> |
| 00:58.11 | ChkDigit | <PROTECTED> |
| 00:58.27 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:59.25 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 01:00.27 | [TK]D-Fender | Lets try this again ... |
| 01:00.32 | [TK]D-Fender | Show the CALL. |
| 01:01.13 | [TK]D-Fender | Not "show a manual random chunk of the raw config". Show what is actually happening. |
| 01:01.46 | ChkDigit | So, you'd like to see the console output, yes? |
| 01:02.48 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 01:02.53 | [TK]D-Fender | yes. |
| 01:03.01 | [TK]D-Fender | And the instructions as dumped earlier... |
| 01:05.20 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 01:09.36 | ChkDigit | http://pastebin.com/Ga4Btkhq |
| 01:09.40 | ChkDigit | That is the call. |
| 01:09.55 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-hbckbyflkbwksgax) |
| 01:10.22 | ChkDigit | http://pastebin.com/ZFyR65GM |
| 01:10.30 | ChkDigit | Those are the instructions. |
| 01:29.07 | *** join/#asterisk sgriepentrog (~sgriepent@2602:306:bdc4:84e0:20c:29ff:fe01:6302) |
| 01:29.13 | [TK]D-Fender | Hrm |
| 01:34.43 | *** join/#asterisk shido6 (~shido6@98.234.178.27) |
| 01:38.10 | [TK]D-Fender | ChkDigit: is that due to an INCLUDED context? |
| 01:42.53 | *** part/#asterisk Russ (foobar@pool-108-0-19-96.lsanca.fios.verizon.net) |
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| 02:59.23 | BeeBuu | > 0x7f3574022ee0 -- Probation passed - setting RTP source address to XXX.XXX.XXX.XXX:8444 |
| 02:59.36 | BeeBuu | what's above text mean? |
| 02:59.52 | BeeBuu | could anyone tell me ?please |
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| 06:17.48 | ayrjola | Hi, any news on svn.asterisk.org? It still seems to be down... |
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| 07:54.54 | riess82 | good morning |
| 08:05.52 | riess82 | I am back, better prepared then yesterday. I would like to run an asterisk inside the following system: ISDN form Telephone COmpany --- AVAYA Integral 5 D with 1 ISDN station on S0, 3 analog phones, 1 free T0, 1 free S0 and 2 free analog lines |
| 08:07.49 | riess82 | is the s0 the right place to plug in the asterisk with a generic ISDN card? i found a post online that mentions the disadvantage of blocking 2 ISDN lines if i call from the isdn station to the asterisk on the same bus |
| 08:09.06 | sekil | yes s0 is the only input to use |
| 08:09.08 | michael_work | ayrjola, so it's not only me :( |
| 08:09.25 | michael_work | wanted to prepair patch for review and no go as can't update |
| 08:10.01 | ayrjola | michael_work I have been trying for about 3 hour now |
| 08:10.03 | riess82 | sekil: would i get rid of this problem if i find another slot module for the integral with another S0? |
| 08:10.25 | michael_work | i wonder who we need to ping for that |
| 08:10.33 | sekil | I would get rid of isdn phone |
| 08:10.39 | michael_work | let's try asterisk-dev |
| 08:11.26 | riess82 | sekil: if money wouldn't be an issue, i would agree |
| 08:12.00 | ayrjola | michael_work that could work, they had maintanance break that strated 9CST |
| 08:12.23 | ayrjola | michael_work that sould have been only about one hour |
| 08:12.51 | michael_work | ok. i assume the best option to wait for highlight here or to chck in few hours :P |
| 08:15.02 | sekil | riess82: if you call from isdn phone to asterisk you're assuming both channels |
| 08:15.46 | riess82 | ok |
| 08:16.45 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:16.47 | ayrjola | michael_work its 3am at huntsville, so I would bet that 4-5 hours wait before anybody is at the office |
| 08:17.02 | sekil | riess82: as I said get rid of that and use IP phone on asterisk |
| 08:17.35 | riess82 | no option, unfortunately |
| 08:17.41 | sekil | riess82: oh avaya integral is a pbx? |
| 08:17.55 | sekil | riess82: I thought it's an isdn phone... |
| 08:18.06 | riess82 | integral is an pbx |
| 08:18.13 | sekil | riess82: that you can interconnect two pbxes differently |
| 08:18.18 | sekil | s/that/than/ |
| 08:18.21 | sekil | err then |
| 08:18.59 | riess82 | wow, nice service infobot |
| 08:19.46 | sekil | riess82: i read h323 is supported on that avaya...asterisk has ooh323..and there you go |
| 08:19.51 | riess82 | sekil: i would have one free plugin slot on the integral pbx that could take another module with 2xt0 and 2xs0 |
| 08:21.38 | sekil | riess82: why not use voip to connect * and avaya |
| 08:22.09 | riess82 | you mean the asterisk between telco and current avaya? |
| 08:22.36 | sekil | no |
| 08:23.04 | riess82 | sekil: i can't find anything on google with "integral 5 d h.323" |
| 08:23.11 | *** join/#asterisk bulkorok (~Benjamin@85.183.61.47) |
| 08:24.04 | sekil | riess82: bah..it seems like 5E does voip |
| 08:24.40 | riess82 | so back to the extra slot module for another s0... |
| 08:25.33 | sekil | if * can do nt mode you connect via isdn |
| 08:26.13 | sekil | you can program digium/openvox cards for nt mode I think |
| 08:27.25 | *** join/#asterisk CeBe (~CeBe@91-65-183-227-dynip.superkabel.de) |
| 08:27.43 | riess82 | here is a thread putting asterisk infront of integral 5, so should work (somehow) http://www.ip-phone-forum.de/showthread.php?t=220476 |
| 08:30.19 | sekil | yeah... |
| 08:30.50 | sekil | you can use q.sig |
| 08:31.23 | riess82 | *scared* another thing i don't know anything about |
| 09:01.08 | riess82 | sekil: do i understand this correctly that with this card i could put * between telco and integral or on s0 of integral? http://www.junghanns.net/en/duobri_express_produkt.html |
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| 09:34.55 | sekil | riess82: you buy an isdn card....put it in the box ...make it work in nt mode...install asterisk.. and then connect the card to an isdn card of the avaya |
| 09:35.09 | sekil | riess82: so the connection is just between pbxes..no telco |
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| 10:08.57 | riess82 | ok. i think i understand |
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| 11:05.58 | *** join/#asterisk fabio1591 (c56037ee@gateway/web/freenode/ip.197.96.55.238) |
| 11:06.48 | fabio1591 | Hey Guys, maybe a silly question but I'm sort of new to Asterisk and SIP in general... |
| 11:07.45 | fabio1591 | Long story short, I hav an incoming SIP call to my asterisk server which works fine, however I need to specifically use data in a later SIP invite (in-dialog) which I need to be able grab |
| 11:08.39 | fabio1591 | The updated SIP Invite contains a different value inside P-Asserted-intentity, so how to I grab that data specifically and not the data in the first SIP invite |
| 11:08.56 | fabio1591 | any tips, tricks and or advise would be highly appreciated :) |
| 11:11.14 | fabio1591 | been struggling with this one for a while googling has just led me in circle with nothing conclusive so sort of hitting a wall here, everytime I try to make use of the P-Asserted-Identify header, it always displays the value from the first invite... am I doign something horribly wrong or is there a reason behind this sort of behaviour |
| 11:11.30 | fabio1591 | Asterisk 1.8 btw |
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| 11:54.19 | fabio1591 | Anyone home? |
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| 12:18.41 | P-NuT | Hi all, I'm looking at a ps axjf on my asterisk server and I see 40 child processes of "/usr/sbin/asterisk -p -U asterisk". |
| 12:18.53 | P-NuT | What is this and how do I bring this number down? |
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| 12:23.27 | file | your system might be showing threads as child processes, in which case there's nothing to worry about |
| 12:28.13 | workingcats | checked my man ps, showing threads as processes has other option letters |
| 12:28.16 | workingcats | H specifically |
| 12:28.32 | workingcats | unless they figured it's a good idea to develop multiple sets of options for ps |
| 12:33.57 | file | depends on what implements ps |
| 12:36.44 | workingcats | yep |
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| 13:01.09 | ayrjola | anybody using pjsip and getting wierd lookig contact Contact: <sip:da001af3-cfea-45c2-a3af-c9231f936b6b@... |
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| 13:20.46 | Katty | GOSH DARNIT QWELL |
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| 13:28.31 | ayrjola | Hi, any news on svn.asterisk.org? It still seems to be down... |
| 13:28.55 | file | no update as of yet, it's morning in Huntsville |
| 13:29.34 | file | if you want to know when it's back up you can keep checking or follow the AsteriskDev twitter account |
| 13:31.29 | ayrjola | @file oh, didn't notice that tweet. Thanks. |
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| 14:56.20 | bmurt | anyone using mediaproxy or rtpproxy? |
| 14:56.29 | bmurt | quick question for helping me understand it |
| 14:56.33 | bmurt | and its handling of rtp |
| 14:58.24 | pabelanger | bmurt, we use rtpproxy, but migrating to rtpengine soon |
| 14:58.31 | puzzled | hi |
| 14:58.38 | pabelanger | that said, #kamailio might also be a better place |
| 15:00.45 | puzzled | is libss7 2.0.0 supposed to work with Asterisk 11? During Asterisk ./configure I see "checking for ss7_set_adjpc in -lss7... no" and at the end of configure I get an error "The SS7 installation appears to be missing or broken" |
| 15:04.40 | puzzled | never mind, found that it's only for upcoming Asterisk 13 |
| 15:09.55 | Naikrovek | 13? last I played with asterisk it was 1.8. busy and/or version numbering change... |
| 15:11.15 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:2f70:21b:63ff:fe31:8426) |
| 15:12.33 | Katty | hugs Naikrovek |
| 15:18.13 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:18.34 | Chainsaw | Naikrovek: 1.8, 10, 11, 12, 13... |
| 15:18.52 | Naikrovek | ah katty :) |
| 15:19.01 | Naikrovek | reciprocates the hug |
| 15:19.07 | Naikrovek | Chainsaw: ah ok |
| 15:19.32 | Katty | Naikrovek: how're you dear? |
| 15:19.49 | Naikrovek | fine. also, dandy. Fine & dandy. You? |
| 15:20.24 | michael_work | if you have penalty set and queue ringall it would call only people with same (minimal penalty) but not people in range. Why is that decision was made or is it a bug? |
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| 15:21.36 | Katty | drmessano: you around sweety? |
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| 15:41.08 | bmurt | pabelanger: thanks, im researching moving rtp out of asterisk, but was reading about both mediaproxy & rtpproxy, but i was trying to clarify whether once both endpoints are aware of the proper addressing for each other, do they communicate directly or does the media pass through mediaproxy or rtpproxy |
| 15:46.20 | pabelanger | bmurt, depends on your network layout. Usually media goes through the proxy |
| 15:46.34 | pabelanger | that's what we did |
| 15:46.42 | pabelanger | we wanted to move RTP out of the core, and to the edge |
| 15:47.13 | bmurt | yeah, we're trying to do the same |
| 15:47.24 | bmurt | trying to get it out of asterisk |
| 15:55.38 | pabelanger | works well |
| 15:55.53 | pabelanger | infact, we're running some webrtc test clients with 1.8 |
| 15:56.06 | pabelanger | because media is processed in the rtpengine |
| 15:56.15 | pabelanger | otherwise, we'd have to upgrade to asterisk 11 / 12 |
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| 17:16.26 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
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| 19:02.00 | ayrjola | Hi, I'm getting wierd looking contact header with pjsip Contact: <sip:2608db84-3e53-4af2-a334-1100fc5793e1@... |
| 19:02.32 | ayrjola | do I have something configured wrong or is this feature of pjsip? |
| 19:09.56 | file | it will place a uuid in the Contact user portion, depending on things it may or may not be replaced |
| 19:10.50 | file | any particular reason you ask? |
| 19:11.09 | rmudgett | file: Reparse your previous statement and correct. :) |
| 19:11.50 | file | no correction needed :P |
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| 19:45.28 | pa | hi |
| 19:45.56 | pa | did someone here had to face this bug due to the sloppiness of Canonical? https://bugs.launchpad.net/ubuntu/+source/dahdi-linux/+bug/1312421 |
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| 20:22.34 | brad[] | Hi all, I'm having an issue where I can call my asterisk server (which is behind NAT) and the echo test works - an outside caller can speak and hear himself - but when I connect a SIP client to the PBX, which is on the local LAN, a call inbound to this extension can be heard in only one direction - the SIP client can hear the outside caller but cannot speak to them |
| 20:22.41 | brad[] | obviously some kind of NAT issue, not sure what though |
| 20:22.51 | brad[] | I'm using nf_conntrack_sip and nf_nat_sip |
| 20:24.25 | brad[] | lol |
| 20:31.08 | brad[] | Just solved it. My mic wasn't configured correctly. :| |
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| 22:12.12 | infinity_ | who are some good pay as you go sip trunk providers? |
| 22:12.32 | infinity_ | i'm pretty sure there is a bot with an answer. ..lol |
| 22:13.22 | WIMPy | doesn't think so. |
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| 23:14.40 | cmendes0101 | One of my vendors is charging a bit for short duration calls. I'm trying to see how I can force a call to be 6 seconds even if the user hangs up. Any ideas? |
| 23:19.25 | [TK]D-Fender | cmendes0101: "core show application dial" |
| 23:25.56 | cmendes0101 | Well I looked over this again in case I missed something but I do the the absolute timeout but was trying to see how to do like a minimum length. Only other thing I can think of is do g with condition checking to do a wait |
| 23:26.09 | cmendes0101 | see the absolute timeout* |
| 23:26.30 | WIMPy | That's the way. |
| 23:27.27 | cmendes0101 | Ok |
| 23:29.10 | WIMPy | Well, that actually depend on the direction. What way do you need the delay? |
| 23:30.58 | cmendes0101 | Traffic is majority outbound. Our main vendor is charging a 1cent fee for calls under 6 seconds so trying to eliminate that fee by just making the call longer if its less than the 6. |
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| 23:31.44 | cmendes0101 | Rather pay 12 seconds @ specific rate than the 1 cent |
| 23:32.25 | WIMPy | g won't help then. |
| 23:32.50 | dan_j | If i'm using dundi, can i set entity id to anything? I've got asterisk in a pacemaker setup, with two nodes, only one running at any one time, but obviously the mac address changes depending on which node is running. Does entity-id have to belong to that specific machine? Or does it simply have to be globally unique? |
| 23:33.27 | WIMPy | It has to be unique and look like a mac. |
| 23:34.05 | dan_j | ok. thanks |
| 23:34.37 | WIMPy | cmendes0101: Take a look at F. |
| 23:36.19 | cmendes0101 | WIMPy: Oh ok, I'll look into that. I guess it matters to see which side is hanging up first since callee first might be F and caller would be g? |
| 23:37.01 | WIMPy | If the callee hangs up, ther's nothing you can do on your providers side to delay it. |
| 23:37.51 | WIMPy | And it's the opposite. |
| 23:38.07 | cmendes0101 | Ah, got it. thanks |
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