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| 00:01.19 | WIMPy | You want the simple "proceed with dialplan execution". |
| 00:08.23 | mjordan | (g) |
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| 00:52.13 | Dick-Tracy | Well Thanks for all the help Guys, it appears as though i'm going to be making .call files anways, as the subscript fails on dial |
| 00:53.55 | Dick-Tracy | still am going to have to find out how to keep track of the redials, and when finished send an email - but thats a job for another day |
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| 01:07.13 | bitkill3r | hi, i'm looking for help on fully uninstalling dahdi from a CentOS server. I found a few guides online, but it doesnt seem to have fully uninstalled it... Any help appreciated |
| 01:13.19 | ChannelZ | make uninstall |
| 01:13.19 | ChannelZ | ? |
| 01:14.51 | bitkill3r | tried it, and it didnt work |
| 01:14.59 | bitkill3r | for some reaso |
| 01:15.42 | bitkill3r | <PROTECTED> |
| 01:17.42 | ChannelZ | Looks like there is only an uninstall target in the 'linux' directory (the drivers) and not one for 'tools' (or for the parent directory which is both.) |
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| 02:34.03 | esaym153 | so my new sip phone supports 722, but my outgoing provider is 711, should I use 722 on the phone or will it cause trans coding? (and thus more cpu and lower quality) |
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| 03:23.12 | ChannelZ | If your provider is only ulaw there's no point in running 722 on your phone |
| 03:25.20 | esaym153 | ChannelZ: yea figured |
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| 03:34.11 | esaym153 | in voicemail.conf, if I have a plan like: '111 => 3111,111,xxx' can I remove the password 3111 and have non-pas protected voicemail? |
| 03:35.44 | Cyford33_p_l_y | easier to have no password protection for devices that dial from same exten |
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| 03:56.07 | esaym153 | Cyford33_p_l_y: yea, that is what I want |
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| 04:13.30 | esaym153 | I see, VoiceMailMain(#,s) :) |
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| 04:32.55 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 07:33.47 | mzf | hi. how can i enable music on hold without restarting asterisk? |
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| 08:40.27 | Spartapatate | Hey guys! |
| 08:40.59 | Spartapatate | Just a weird question, i'm trying to use Asterisk Realtime with PGSQL but can't find the Schemas anywhere! Has the whole thing been depracated, or am I just not looking in the right place? |
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| 09:21.53 | Nouuu | hello i have a problem with some agent on queue who seems to not hangup et be busy is theire sommething to do to check if they hangup correctly ? |
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| 10:04.42 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 11:43.40 | WIMPy | Oh. Are the SIP scaners back again? |
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| 12:10.14 | Rico | hello |
| 12:10.29 | Rico | using asterisk 13, I've removed chan_sip, and my asterisk server does not bind to port 5060 |
| 12:10.34 | Rico | how do I do that with pjsip ? |
| 12:10.54 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 12:10.55 | WIMPy | Configure it. |
| 12:11.10 | Rico | nice |
| 12:11.21 | WIMPy | IIRC there's a page on the wiki on how to migrate. |
| 12:11.46 | Rico | https://wiki.asterisk.org/wiki/display/AST/PJSIP+Transport+Selection |
| 12:11.52 | Rico | I('ve seen that |
| 12:14.30 | file | https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples |
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| 13:39.41 | Rico | strange behavior : asterisk 13.4 is registering to my SBC. SIP capture shows that registering is OK, but I still have errors about asterisk unable to register in CLI |
| 13:40.01 | Rico | I have REGISTER => 401 => register with auth > OK |
| 13:40.19 | Rico | is it the first 401 which causes this error ? |
| 13:40.45 | [TK]D-Fender | it is an automatic challenge, not an error |
| 13:40.49 | Rico | pjsip show registrations tells me that trunk is registered |
| 13:40.57 | [TK]D-Fender | it does this to obfuscate valid devices |
| 13:41.07 | Rico | yes |
| 13:41.14 | [TK]D-Fender | It's fine |
| 13:41.15 | Rico | but why do I have an arror ? |
| 13:41.18 | Rico | error |
| 13:41.33 | [TK]D-Fender | show us the error |
| 13:41.43 | [TK]D-Fender | because "REGISTER => 401 => register with auth > OK" says OK |
| 13:41.51 | Rico | let me pastebin |
| 13:43.41 | Rico | [TK]D-Fender: http://pastebin.com/0Hw6t3YP |
| 13:43.52 | Rico | I'm looking about xmit error |
| 13:44.02 | [TK]D-Fender | taht's says timeout... |
| 13:44.05 | [TK]D-Fender | not an auth issue |
| 13:44.22 | Rico | I did not say it was an auth issue |
| 13:44.29 | [TK]D-Fender | clearly not getting answers on time |
| 13:44.31 | Rico | <PROTECTED> |
| 13:44.52 | file | er, well |
| 13:44.53 | [TK]D-Fender | check your networking, firewalls, etc. |
| 13:45.00 | file | that message is from chan_sip |
| 13:45.02 | *** join/#asterisk Fira_ (2edae5ac@gateway/web/freenode/ip.46.218.229.172) |
| 13:45.06 | [TK]D-Fender | you'll need to trace the packet flow to confirm where they are getting caught up |
| 13:45.08 | file | and yet you are showing an outbound registration from PJSIP |
| 13:45.20 | Rico | file: ... ok thank youy |
| 13:45.21 | Fira_ | Uh, hey guys, anyone managed to get rtptimeout to work ? I can't |
| 13:45.25 | Rico | that's it |
| 13:45.30 | [TK]D-Fender | Um.... |
| 13:45.32 | Rico | forgot to unload chan_sip after reboot |
| 13:45.33 | [TK]D-Fender | and the BIG PRINT |
| 13:45.45 | [TK]D-Fender | [Jul 16 15:40:44] ERROR[704]: chan_sip.c:4167 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data |
| 13:45.47 | [TK]D-Fender | [Jul 16 15:40:44] NOTICE[704]: chan_sip.c:15307 sip_reg_timeout: -- Registration for '1004@sipproxy.xx.x' timed out, trying again (Attempt #159) |
| 13:45.48 | [TK]D-Fender | CHAN_SIP <------------------------ |
| 13:45.52 | WIMPy | Fira_: Yes, but it take a second more than expected. |
| 13:45.53 | [TK]D-Fender | this is NOT pjsip at all |
| 13:45.59 | [TK]D-Fender | You left old crap in there |
| 13:46.06 | Rico | [TK]D-Fender: yes, that is what file said |
| 13:46.08 | Fira_ | WIMPy: what do you mean by that ? |
| 13:46.11 | Rico | and this is the problem |
| 13:46.15 | Rico | thanks |
| 13:46.27 | [TK]D-Fender | And that... would be ceause of changing the PORT the way I'm betting you just did and that error is because it's bound to PJSIP now |
| 13:46.47 | WIMPy | Fira_: If you set it to 10 it will kill the call at the end of the 11th second without rtp. |
| 13:46.52 | Fira_ | I set my rtptimeout both globally and on peers to 15 seconds, but upon cutting the RTP feed, nothing happens at all |
| 13:46.57 | [TK]D-Fender | port__overlapp++ |
| 13:47.26 | WIMPy | Fira_: Aure you using directmedia? |
| 13:47.53 | Fira_ | WIMPy: um, nope |
| 13:50.04 | Fira_ | I'm making an ARI App, trying to catch when channels go unresponsive (as this /will/ happen in our usecase), so i'm trying to catch ChannelDestroyed events on it. But it seems even with rtptimeout, I don't get any kind of notification, nothing shows up in RTP debug, and the channel is still listed in sip show channels |
| 13:50.40 | Fira_ | tried with and without rtpkeepalive |
| 13:50.45 | Fira_ | am I doing something wrong here ? |
| 13:50.52 | Fira_ | i saw there's been a number of issues with rtptimeout in the past |
| 13:52.06 | Fira_ | it's not a config problem because the timeout appears in show sip settings |
| 13:52.07 | Fira_ | and yet |
| 13:52.09 | Fira_ | nothing happens :/ |
| 13:52.38 | Fira_ | i do see noise being sent out from rtpkeepalive when it is enabled, in the debug output |
| 13:52.43 | Fira_ | but absolutely nothing regarding timeouts |
| 13:54.06 | Fira_ | WIMPy: i haven't touched the directmedia setting and can see RTP packets reaching Asterisk in debug, so i doubt that's the problem |
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| 14:01.48 | Fira_ | anyone else can give me a hand ;( ? |
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| 14:03.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.58 | Rico | Fira_: sorry, no |
| 14:04.35 | WIMPy | has only tried with chan_sip. I'm stuck on 11. |
| 14:05.07 | Rico | WIMPy: I'm trying to understand how pjsip works |
| 14:05.27 | Rico | hard after many years with chan_sip |
| 14:08.24 | Rico | for now, pjsip does not seem to be able to load registrations from my db using odbc |
| 14:10.12 | Rico | what do I have to set in extconfig.conf for using table ps_registratins ? |
| 14:10.17 | Rico | registrations |
| 14:11.05 | Rico | https://issues.asterisk.org/jira/browse/ASTERISK-23807 <- shit |
| 14:11.17 | file | what do you want to gain from doing that? |
| 14:11.38 | *** join/#asterisk superscrat (asanders@nat/digium/x-fmvptcgaohzyxuma) |
| 14:11.41 | Rico | file: just put maximum stuff in DB |
| 14:11.58 | file | you would still need to tell Asterisk to reload if it changed |
| 14:12.29 | Rico | ok, but is it supported in 13.4.0 ? |
| 14:13.11 | Rico | <PROTECTED> |
| 14:13.11 | Rico | <PROTECTED> |
| 14:13.23 | file | I do not know off the top of my head, but probably not |
| 14:13.26 | file | not via that mechanism |
| 14:13.30 | Rico | ok |
| 14:13.35 | file | static realtime would work |
| 14:14.01 | Fira_ | k |
| 14:14.04 | Fira_ | got it |
| 14:14.14 | Fira_ | looks like rtptimeout is broken when using adaptative jitterbuffer |
| 14:14.42 | Fira_ | found an old post saying it broke in Asterisk 12, and this seems to still be the case in 13.0.4 |
| 14:17.48 | *** join/#asterisk kharwell (kharwell@nat/digium/x-cnlshzghmryhfxmu) |
| 14:19.06 | Fira_ | pretty uncool, i absolutely need the JBs :/, running VoIP on Wireless... gonna have to hack around with voice detection and absolute timeout in my ARI App i guess :( |
| 14:20.08 | WIMPy | Where does the call go? You don't want a JB in the middle. |
| 14:23.38 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-dljwkkhuhyouivzr) |
| 14:23.57 | Fira_ | WIMPy: mixing between wireless clients. where else do i put JBs ? |
| 14:25.03 | WIMPy | On the endpoint. |
| 14:25.06 | Fira_ | hmm |
| 14:25.13 | WIMPy | That's the only place they make sense. |
| 14:25.43 | Fira_ | don't you also need to resync the audio with jbs before mixing it ? |
| 14:25.44 | WIMPy | You only want Asterisk to have one when converting to TDM. |
| 14:26.13 | Fira_ | i mean, i get that it'd correct any downstream issue if it's on the endpoint, |
| 14:26.13 | WIMPy | If you do conferencing, ConfBridge will also have to do it, yes. |
| 14:26.39 | Fira_ | but what about upstream ? you'd need a jb to prevent asterisk from mixing garbage audio, no ? |
| 14:26.49 | Fira_ | well |
| 14:26.51 | WIMPy | BUT IIRC COnfBridge has its own JB. |
| 14:27.05 | Fira_ | that's sad since i use normal bridges and not confbridges |
| 14:27.07 | Fira_ | :p |
| 14:27.25 | WIMPy | Then what do you mix? |
| 14:27.36 | Fira_ | what do you mean ? |
| 14:28.02 | Rico | there's no doc about pjsip and realtime registrations.... |
| 14:28.09 | Fira_ | i'm using ARI mixing_bridges WIMPy |
| 14:28.13 | WIMPy | Is it a multi-party conference or what is it? |
| 14:28.16 | Fira_ | Yeah |
| 14:28.32 | Fira_ | multi-party over jitterful wireless (also named "Wi-Fi") |
| 14:28.35 | WIMPy | has no idea about that. |
| 14:29.18 | WIMPy | Bad idea to use wifi for realtime applications. |
| 14:29.33 | Fira_ | it is. that's why i need the JB |
| 14:30.03 | WIMPy | It probably won;t become usefull anyway. |
| 14:30.05 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-ljrgnouadisrzhyh) |
| 14:30.10 | Fira_ | i actually tested with up to 350+-100ms latency and the adaptive JB does its job very well at correcting it |
| 14:30.27 | Fira_ | which is well within the margin of what you can expect from a proper wireless network with QOS |
| 14:30.30 | WIMPy | In multi-party conferences delays are even more annoying than on normal calls. |
| 14:31.30 | Fira_ | well what do you suggest then ? |
| 14:31.54 | WIMPy | Get rid of the wifi. |
| 14:31.59 | Fira_ | and use ? |
| 14:32.23 | WIMPy | How DO you use it? |
| 14:34.13 | *** join/#asterisk cstk421 (~cstk421@50.153.109.45) |
| 14:34.31 | Fira_ | i don't even see your point, it works very well with JBs on Wi-Fi. The issue here is that rtptimeout doesn't work with them. Trust me, there aren't any real alternatives to 2.4/5 Ghz bands for this that don't involve either custom radio hardware or non-IP radio. |
| 14:35.11 | robmal | How long is the wifi link? |
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| 14:36.08 | WIMPy | Well, for telephony there's CAT-iq that actually works. But we still don't know what you're doing there. Are you using wifi phone? Laptops? Just a wifi link somewhere in the loop?/ |
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| 14:36.14 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:37.38 | Rico | mjordan: hi, do you know if outbound registrations are supported in realtime pjsip ? |
| 14:37.48 | Rico | https://issues.asterisk.org/jira/browse/ASTERISK-23807 |
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| 14:39.20 | Fira_ | hmm |
| 14:40.29 | Fira_ | CAT-iq could actually be a good variation indeed for just VoIP, sadly we're using other IP services |
| 14:40.33 | *** join/#asterisk cstk421_ (~cstk421@50.153.110.180) |
| 14:41.12 | Fira_ | in a nutshell we need to run a bunch of random services on smartphones deployed in the middle of nowhere, including a VoIP (or cellular) link to Asterisk |
| 14:41.13 | WIMPy | CAT-iq has nothing to do with VOIP per se, but it can be used with VOIP. |
| 14:42.05 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:42.39 | WIMPy | In the middle of nowhere, you could set up your own GSM or UMTS base. |
| 14:43.44 | Fira_ | we actually already have something like that, but can't use it in many places due to authorization problems, hence the need to have an unlicensed and rapid deployment alternative, hence 2.4 and 5 ghz |
| 14:45.05 | robmal | But 5GHz links on a few km are pretty stable and nowhere near 300ms |
| 14:45.21 | mjordan | Rico: it depends on your expectation. |
| 14:45.35 | mjordan | Rico: You can store outbound registrations in a realtime backend, but Asterisk will only read them on startup |
| 14:46.03 | mjordan | Rico: Think about it this way: nothing ever tells Asterisk that "hey, you should go do an outbound registration". So if you update the database, there's no way Asterisk will ever know about it. |
| 14:46.15 | mjordan | Now, that being said, there's some interesting use cases for storing outbound registrations in a database still. |
| 14:46.32 | Fira_ | robmal: yeah, and as i said, even 300ms is fine with adaptive jb |
| 14:46.33 | mjordan | For one, you can just issue a reload to res_pjsip_outbound_registration, as opposed to the entire stack, and it should re-read the database |
| 14:46.51 | Rico | mjordan: yes I understood that. I think my problem is more global, as realtime endpoints does not appear in "pjsip show endpoints"... |
| 14:46.59 | mjordan | er |
| 14:47.03 | mjordan | they do for me. |
| 14:47.14 | Rico | mjordan: even if they're hnot registered ? |
| 14:47.19 | robmal | Fira_: But if you have 300ms latency your wifi link is fucked up, stop tweaking asterisk to compensate that and focus on wifi. |
| 14:47.21 | mjordan | sure |
| 14:47.22 | Rico | ok |
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| 14:47.29 | Fira_ | robmal: we don't, that's testing. |
| 14:47.32 | Rico | so I'll try to make that working first... |
| 14:47.32 | mjordan | The endpoint object itself has nothing to do with its contact information stored on its AoR |
| 14:47.33 | Rico | thanks |
| 14:48.27 | Fira_ | robmal: fact is you're gonna get random jitter due to CSMA/CA and retransmits either way, which is why it's almost imperative to use JB to avoid random chopping |
| 14:48.38 | Fira_ | even if it's not 300ms consistently |
| 14:48.39 | Fira_ | it happens |
| 14:48.43 | Fira_ | wireless is just fucked up like that |
| 14:48.50 | robmal | Lies. |
| 14:49.17 | Fira_ | you ever tried to run dot11 with even minimum interference ? |
| 14:49.26 | Fira_ | and even do a simple ping ? |
| 14:49.30 | robmal | I can make a hd voice voip call on 5km 5ghz link without any problems. In fact, 15 calls at the moment. |
| 14:49.40 | Fira_ | 0.5ms. 0.5ms. 0.5ms. 0.5ms. 80ms. 0.5ms. 0.5ms. |
| 14:49.44 | Fira_ | i'm not talking ptp |
| 14:49.54 | WIMPy | Use CAT-iq phones with Android. Should do the same as any other smartphone , just that it has DECT instead of GSM/UMTS/LTE. |
| 14:50.10 | Fira_ | of course ptp is going to be fine, it's directional |
| 14:50.27 | Fira_ | i'm talking about the link to endpoints.. |
| 14:50.48 | robmal | So, p2mp, how far from the base station? |
| 14:50.55 | Fira_ | -sigh |
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| 14:50.57 | Fira_ | i mean |
| 14:50.59 | Fira_ | mobile stations |
| 14:51.13 | Fira_ | not a site-to-site link |
| 14:51.35 | robmal | Ok, how far are they from the access point? |
| 14:51.51 | Fira_ | since when do you measure radio in distance ? |
| 14:51.56 | WIMPy | Does that matter? |
| 14:52.18 | robmal | Could you at least answer my question? |
| 14:52.39 | Fira_ | WIMPy: so there are CAT-iq Android Phones around for abordable prices? got some tips regarding manufacturers? |
| 14:52.47 | WIMPy | Well, you definitely do when using TDMA. But on wifi doesn't care AFAIK. |
| 14:53.13 | Fira_ | robmal: "as far as possible" ? how far do you expect to get 11 mbps datarate on .11n ? |
| 14:53.34 | Fira_ | i have no idea since it depends on environment, so if you do, i'm glad for you |
| 14:53.39 | WIMPy | Fira_: Sorry. I just found them rather senseless, but they do exist. |
| 14:54.03 | WIMPy | And for your application they seem to be ideal. |
| 14:54.30 | Fira_ | yeah, i'll have a look, thanks for the tip ! |
| 14:55.49 | WIMPy | DECT for voice and wifi for other applications. That should work as good as it can. |
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| 15:04.05 | esaym153 | nat=yes in iax.conf is no longer valid right? |
| 15:06.29 | [TK]D-Fender | I don't recall that ever actually being a valid concept |
| 15:07.20 | WIMPy | Doesn't make sense, no. |
| 15:09.03 | Rico | if somebody can take a look : http://pastebin.com/kc98nEkP |
| 15:09.13 | Rico | don't understand why I don't see my pjsip endpoints |
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| 15:20.26 | pjensen00 | what does your pjsip.conf look like? |
| 15:21.35 | Rico | pjensen00: let me pastebin |
| 15:22.38 | Rico | pjensen00: http://pastebin.com/X6i7mud1 |
| 15:24.57 | pjensen00 | I think it's because you don't actually have an entry for an actual "endpoint". You have a registration, transport, and auth type I see. |
| 15:25.06 | pjensen00 | I'll post an example of one of my working ones |
| 15:27.54 | pjensen00 | Rico: http://pastebin.com/VU8Q18LX |
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| 15:31.40 | pjensen00 | It might be as simple as just adding the "[1004]\r type=endpoint" entry. But I'm not entirely sure. |
| 15:36.06 | Rico | pjensen00: I'm talking about realtime endpoints |
| 15:36.20 | Rico | mi 1004 is an outbound registration, it works fine |
| 15:36.44 | Rico | that's just the realtime part which does not work |
| 15:36.51 | pjensen00 | awww... I see. :( |
| 15:41.06 | bipolar | I'm trying to get native bridging to work on an outgoing call. I'm using Digium analog FXS ports, and a Digium PRI for PSTN connection. Is there anything that would make this impossible? I'm trying to track down what's going on, and I have it natively bridging on *incoming* calls from the PRI. I'm using Asterisk 13. |
| 15:41.53 | bipolar | I know asterisk needs to listen for the DTMF tones when the phone dials so it knows what number to use for the outgoing call, but after it bridges I need it to stop emulating DTMF. :P |
| 15:42.23 | WIMPy | Depends on your Dial() options. |
| 15:42.52 | WIMPy | If you have anythong enablesd that needs DTMF, there will be no native bridging. |
| 15:43.47 | WIMPy | Code exists that would make that possible, but it doesn't seem to work any mor, which is rather annoying. |
| 15:44.40 | WIMPy | And if anyone knows, BTW, since when that doesn't work any more, I'd like to know. |
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| 15:47.02 | bipolar | WIMPy: Here's my config and context: http://pastebin.com/XAx9WV8J |
| 15:47.05 | bipolar | It's very simple. |
| 15:49.03 | bipolar | This is for alarm panels sending contact ID, btw, since it sends signals via dtmf tones. |
| 15:49.34 | bipolar | I was able to get incoming to receivers working great! Now this has me banging my head on the wall. :P |
| 15:52.55 | WIMPy | IIRC you need a 'transfer=yes'. |
| 15:56.28 | bipolar | I do? I had turned that off because I thought it had to listen for transfer digits, but I'll turn it back on and test. |
| 15:57.55 | Rico | anybody about my realtime pjsip problem ? |
| 16:01.22 | file | Rico, have you followed the wiki page to set it up? |
| 16:01.39 | Rico | yes |
| 16:01.45 | file | did you configure sorcery.conf? |
| 16:02.41 | Rico | ouch |
| 16:02.45 | Rico | slaps himself |
| 16:06.41 | Rico | works better... thanks file |
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| 16:17.22 | bipolar | WIMPy: bah. No dice. |
| 16:17.58 | bipolar | Guess I'll just keep researching this. |
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| 16:38.24 | bipolar | I think I just got it. :D |
| 16:39.17 | bipolar | DYNAMIC_FEATURES was set with something. I set it to null in this context and it's not emulating dtmf any more. |
| 17:10.02 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
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| 17:20.41 | BCS-Satori | Is a hangup request, cause 1 (for a PRI) generated by the carriers equipment or asterisk? We can make inbound/outbound calls no problem over the PRI but when a user call forwards their desk phone to an external number (i.e.: a cell phone) the call makes it to DAHDI but I get an error that says "Span 1: Channel 0/23 got hangup request, cause 1". The call forward phone uses the same outbound dialplan that the the phone norma |
| 17:20.41 | BCS-Satori | lly uses and I still mask the CALLERID(num) to the main number like normal. |
| 17:21.34 | bipolar | BCS-Satori: this may help you: http://pbxbook.com/other/isdncode.html |
| 17:21.34 | bipolar | BCS-Satori: it sounds like the number that it's being forwarded to isn't valid or something. |
| 17:26.40 | BCS-Satori | bipolar: I've seen the definition of the isdn code 1 but still unclear if its the carrier's equipment (i.e.: adtran) or my pri card (sangoma) causing the error message? The number dialing is valid and the calling syntax over DAHDI/G0/<external number> looks identical when the cal is passed. The only difference I see when when the phone does the call forward is the call uses "Local/<external number>" instead of the phone. |
| 17:34.13 | BCS-Satori | Could it be that if a call originates from Local/ that it would fail on DAHDI to the PRI? |
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| 17:43.32 | bipolar | BCS-Satori: I'm sorry I'm not familiar with your setup :( |
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| 18:11.21 | ibercom | Is there any function that returns the peer's register server ? This information is interesting in realtime configurations. |
| 18:11.42 | ibercom | Asterisk11 and chan_sip |
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| 18:53.15 | joker_89 | hi |
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| 19:09.50 | tesaf | exten => dial,1,Dial(SIP/${DB(AGENT/${ARG1}/PHONE)},30,rotwh) ; Ring the interface, 30 seconds maximum |
| 19:09.53 | tesaf | exten => dial,n,Goto(dial-${DIALSTATUS},1) |
| 19:10.09 | tesaf | does that make sense? I'm getting the following warning in my asterisk message logs |
| 19:10.12 | tesaf | app_dial.c:1027 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 19:14.04 | [TK]D-Fender | clearly it's not getting a usable value back from AstDB. |
| 19:14.15 | [TK]D-Fender | Go verify what it's actually looking for, and what's there |
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| 19:38.54 | tesaf | How would I do that? I'm really asterisk noob, just got thrown into debugging these phone systems |
| 19:39.22 | [TK]D-Fender | "database show" |
| 19:39.25 | [TK]D-Fender | ?book |
| 19:39.29 | [TK]D-Fender | ~book |
| 19:39.29 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:39.31 | [TK]D-Fender | ^^^^ |
| 19:40.00 | tesaf | Is there a config file that contains the same information? I'd rather not use the ast cli |
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| 19:45.33 | mjordan | tesaf: The AstDB is a SQLite database. |
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| 19:53.38 | [TK]D-Fender | No, AstDB isn't a "config files"", it's a database |
| 19:53.57 | [TK]D-Fender | doesn't always read all the answers given to the same question... |
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| 20:13.49 | tesaf | exten => s,1,Set(ci=${DB(PHONE/${CALLERID(num)}/AGENT)}) |
| 20:13.52 | tesaf | exten => s,n,Set(CALLERID(name)=${ci}) |
| 20:14.14 | tesaf | to retrieve that from the DB, would it be database get PHONE CALLERID ? |
| 20:14.31 | tesaf | not sure how to retrieve it from the DB when its doing param expansion |
| 20:15.18 | *** join/#asterisk pppingme (~pppingme@unaffiliated/pppingme) |
| 20:15.42 | [TK]D-Fender | "database show" |
| 20:15.50 | [TK]D-Fender | just dump the whole thing and examine |
| 20:16.13 | tesaf | how can I write it to a file? Can I use std bash redirection inside the ast cli? |
| 20:16.14 | [TK]D-Fender | it'll be PHONE/SOMETHING/AGENT |
| 20:16.19 | [TK]D-Fender | copy/paste |
| 20:16.23 | tesaf | too big |
| 20:16.27 | [TK]D-Fender | that's why God invented scroll-back buffers |
| 20:16.35 | [TK]D-Fender | No such thing as "too big. |
| 20:16.40 | [TK]D-Fender | I have 5000 lines of it |
| 20:16.57 | [TK]D-Fender | That's because I didn't want to seem presumptuous and go for 5-digits |
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| 20:21.36 | tesaf | hmm I guess -x switch worked for dumping |
| 20:22.09 | [TK]D-Fender | often does |
| 20:22.12 | [TK]D-Fender | sometimes you get junk |
| 20:23.47 | tesaf | im seeing a lot of /PHONE/5006/AGENT : <bad value> |
| 20:24.03 | tesaf | im guessing that is the issue? How would I resolve that. |
| 20:25.02 | [TK]D-Fender | Resolve what? |
| 20:25.37 | tesaf | why do a bunch of them say bad value |
| 20:25.46 | [TK]D-Fender | Change them |
| 20:25.47 | tesaf | yet some have extensions |
| 20:27.25 | tesaf | I guess I'm asking is... Where are these families getting their values from? What conf file? |
| 20:28.39 | [TK]D-Fender | WThere are several ways they could have been set |
| 20:28.45 | [TK]D-Fender | And we don't know |
| 20:28.48 | [TK]D-Fender | this is your server |
| 20:28.57 | [TK]D-Fender | They got there however you had it set up to run |
| 20:29.03 | [TK]D-Fender | Could be slowly via dialplan |
| 20:29.10 | [TK]D-Fender | Could be via CLI, or AMI as well |
| 20:29.43 | tesaf | okay, ill poke around. Thanks for the help so far :) |
| 20:29.57 | [TK]D-Fender | You're welcome |
| 20:30.07 | [TK]D-Fender | packs up to head home |
| 20:40.33 | WIMPy | BCS-Satori: If it says "got hangup request" that should mean it comes from the other end. |
| 20:40.40 | WIMPy | If you want the full story, enable pri debug. |
| 20:41.39 | *** join/#asterisk kchehab (~kchehab@77.42.185.236) |
| 20:41.50 | kchehab | hi ppl |
| 20:42.07 | kchehab | i can plenty of warrning WARNING[19545][C-00009360]: channel.c:4860 ast_prod: Prodding channel 'SIP/ ---failed |
| 20:42.15 | kchehab | as i have 250 active call |
| 20:42.34 | kchehab | cpu load 11 % af ram usage 25% |
| 20:42.55 | kchehab | can i do anything to solve this issue ? |
| 20:49.32 | BCS-Satori | WIMPy: yep just got off the phone with the carrier its on their end and the issue is they claim was caused by the configuration file that allows me set my outbound caller id |
| 20:50.04 | WIMPy | Sounds like they need to fix something. |
| 20:50.34 | BCS-Satori | Yep they had to add a wildcard mask to their CID. All set now :) |
| 20:51.09 | WIMPy | No excuse for wrong error messages. |
| 20:53.31 | WIMPy | thought Adtran did less shitty stuff. |
| 20:54.08 | BCS-Satori | I'm always happy when its not on my end :) |
| 20:55.02 | WIMPy | I'm always happy if I'm not mislead by wrong errors. |
| 20:55.57 | kchehab | any idea why asterisk makes alot of delay when calls raech more than 250 calls and how to avoid ot |
| 20:55.58 | kchehab | it |
| 20:56.43 | WIMPy | First find out where it happens. |
| 20:57.21 | WIMPy | Chan_sip easily blocks, most likely dut ot DNS delays/failures. But it could also be your dialplan or whatever. |
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| 22:33.21 | zopsi | So when I run pjsip show aors, pjsip show endpoints, pjsip show channels; it just shows unknown and nan for status and RTT(ms). Shouldn't it at least be showing Disconnected or something? |
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| 22:40.38 | file | zopsi, why? |
| 22:44.22 | *** join/#asterisk pavelsc (~Miranda@87.252.227.65) |
| 22:52.32 | pavelsc | hello guys. My question is about snooping channels |
| 22:53.07 | pavelsc | ["id"]=> string(16) "1437079924.43185" |
| 22:53.07 | pavelsc | ["state"]=> string(2) "Up" |
| 22:53.07 | pavelsc | ["name"]=> string(31) "Snoop/1437079391.43134-00000000" |
| 22:54.04 | pavelsc | I got this stuff as a response, so what to do next? I need something like originate |
| 22:54.16 | WIMPy | From what? |
| 22:54.52 | WIMPy | And define "snooping". |
| 22:54.54 | pavelsc | from ari snoopChannel |
| 22:58.50 | pavelsc | so i want to listen or whisper channel, it seems like snooping channel is created, but I m not sure what to do next |
| 23:03.39 | pavelsc | I believe originate command only creates a channel when passed something like this "local/10001@incoming". Is using existing channel names "Snoop/1437079391.43134-00000000" allowed in originate? |
| 23:07.47 | *** join/#asterisk jetlag (~jetlag@pool-71-168-242-50.cmdnnj.east.verizon.net) |
| 23:11.30 | file | it creates a channel and puts it into your ARI application |
| 23:11.30 | file | you treat it like any other channel |
| 23:13.24 | pavelsc | http://lists.digium.com/pipermail/asterisk-app-dev/2013-November/000218.html also i read some info here, but no one pointed how to answer snooping channel |
| 23:13.35 | file | you don't answer it, it's already answered |
| 23:13.59 | file | it is a channel, the media flowing through it just happens to come from another channel or is injected into it |
| 23:15.50 | pavelsc | well, I have no idea how to catch this media. regular channels at least are ringing on the phone |
| 23:16.06 | file | what do you want to do? |
| 23:16.19 | file | it's a channel - anything you can do to a channel in ARI you can do to a snoop channel |
| 23:16.36 | file | so if you want to implement MixMonitor for example then call /record on the Snoop channel |
| 23:16.44 | file | if you want to whisper in an audio file call /play on it |
| 23:17.03 | file | if you want to have someone listen create a bridge, put them in it, and put the snoop channel in it |
| 23:21.05 | *** join/#asterisk MarqueI (~Marquel@social.pc-tiede.de) |
| 23:21.54 | *** join/#asterisk Marquel (~Marquel@fuchsfanclub/allerdings/marquel) |
| 23:24.31 | pavelsc | @file omg, so I need to create channel, wait until it becomes answered, then put it as well as snooping channel to a bridge? And do other stuff like listening app events to maintain regular call functionality? |
| 23:24.50 | file | define "regular call functionality" |
| 23:25.03 | pavelsc | answer, hangup |
| 23:25.16 | file | well, you have to listen for events - yes |
| 23:25.27 | file | ARI gives you the fundamental building blocks to write applications |
| 23:25.57 | file | how you use those are up to you |
| 23:30.30 | pavelsc | yeah right, I wrote similar app to implement queues, just wondering if easier ways exist ) |
| 23:32.59 | *** part/#asterisk kharwell (kharwell@nat/digium/x-wztdfavqahgkzedv) |
| 23:36.35 | pavelsc | it would be nice to implement this through something like originate. just need fast way for short demo, of course app will be done later |
| 23:37.21 | file | originate creates an outgoing call and once answered sends it into the dialplan, or into the Stasis application |
| 23:37.34 | file | I don't see how you could implement it like originate |
| 23:37.52 | file | or what that really means |
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| 23:48.39 | pavelsc | @file you got it right. another idea through `ami originate` call to extension you want to spy using feature code (e.g. ##1001, where 1001 is extension, we use it to spy via hardphone) which linked with internal chanspy app. |
| 23:51.05 | file | that's specific to how you do things and dialplan logic, if you can work it using what ARI provides or some other way then so be it |
| 23:52.09 | pavelsc | Ok, thanks a lot, completely agree with you |