| 00:01.56 | jeffspeff | brut-, you're right. i'm still writing this thing. haven't got all the parts together yet. |
| 00:02.35 | jeffspeff | [TK]D-Fender, I know that the ARG won't work there, I just put it there so you can see what value i'm trying to get back from the macro |
| 00:06.25 | [TK]D-Fender | jeffspeff: that isn't getting "back" from anything.. that is SETTING to a value |
| 00:06.32 | [TK]D-Fender | jeffspeff: and you aren't showing the actua problem |
| 00:07.34 | jeffspeff | [TK]D-Fender, i think this might work. do you see any problems? I'm concerned with calling out ${CONFusers} like that on line 32. |
| 00:07.41 | jeffspeff | http://pastebin.com/m0uhvMT4 |
| 00:07.54 | [TK]D-Fender | jeffspeff: You are asking us to guess your code an\d values aren't being set where we can see... |
| 00:08.24 | jeffspeff | [TK]D-Fender, what can't you see? |
| 00:08.33 | [TK]D-Fender | jeffspeff: Go run your dialplan and see if there is a problem. Don't use us as a parsing engine in place of actual teting |
| 00:08.46 | [TK]D-Fender | testing* |
| 00:09.18 | jeffspeff | [TK]D-Fender, this is the first time i've ever messed with macros, and was just looking for a general review of best practices, etc. |
| 00:09.47 | [TK]D-Fender | jeffspeff: the only best practices is to avoid nesting and being very careful when you have to |
| 00:09.52 | [TK]D-Fender | aside from that it's all just dialplan |
| 00:10.15 | jeffspeff | what do you mean by nesting? |
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| 00:10.50 | [TK]D-Fender | calling macros within macros |
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| 00:12.25 | jeffspeff | ok, thanks |
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| 04:40.49 | nicola_pav | hello. anyone familiar with an open source application to be integrated with asterisk that does speaker verification? |
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| 09:19.26 | kenshin | hi, I think I found a bug ??? (in asterisk 10.3.1 - on gentoo ) |
| 09:21.55 | kenshin | mwi => use:pass@host/mbox in sip.conf makes asterisk crash if [ host is IP ] or [host is name && dnsmgr is disabled] |
| 09:22.41 | kenshin | meaning if one wants to use that one has to put host as a hostname and enable dnsmgr |
| 09:25.06 | kenshin | for some reason AST_UNREFF is killing mwi obj (inspire of ref count of 3 ) --lien 12787 of cha_sip.c) which leads to a set fault on line 12792 (mwi is 0x0) |
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| 10:11.48 | joobie | hey guys.. is it possible to call a script (via agi() for example) when a call is answered by a queue member? |
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| 11:08.39 | skirmisha | hi guys |
| 11:08.45 | skirmisha | i have little issue |
| 11:09.08 | skirmisha | chan_sip.c: Unsupported SDP media type in offer: image 5008 udptl t38 |
| 11:09.16 | skirmisha | why do i get this error? |
| 11:13.06 | skirmisha | ??? |
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| 11:14.27 | WIMPy | Looks like someone wants to send a fax and you don't allow that. |
| 11:18.47 | skirmisha | the fax is workng |
| 11:18.55 | skirmisha | but i got 5 faxes instead of 1 |
| 11:19.18 | skirmisha | and i see diff causes like image 5000, image 5002 and so on untill image 5008 |
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| 11:22.45 | coppice | not having T.38 support enabled won't prevent FAXing. it will force the call to use audio for FAX |
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| 12:15.51 | weinerk | Please help - I have an agi for Incoming call to exec Dial(outside) and based on get_variable('DIALSTATUS') - I know if billable + count seconds. |
| 12:15.51 | weinerk | Problem if the caller hangs up first - the callflow is interrupted - and I dont get to check DIALSTATUS - cant do billing - even if I have pcntl_signal(SIGHUP,"agi_hangup_handler"); |
| 12:15.51 | weinerk | <PROTECTED> |
| 12:18.23 | leifmadsen | well there is no call at that point if there is no Dial(). Also you should probably check the CDRs, not just the dialstatus for billing |
| 12:24.36 | weinerk | leifmadsen: thanks. But if I was in the middle of a successfull dial - I need to know if to bill seconds or not. But caller-hangup interrupts script flow. |
| 12:25.03 | leifmadsen | then that's the issue with your script since as soon as Dial() is called, then a DIALSTATUS would be created |
| 12:25.55 | weinerk | but control does not return until the called party hangs up or timeout or similar |
| 12:26.20 | weinerk | i can be waiting inside Dial for minutes while parties talk, |
| 12:26.27 | leifmadsen | sure |
| 12:26.48 | weinerk | then at the end see if it was a success and decide to bill seconds or not |
| 12:27.09 | leifmadsen | why aren't you just looking at a CDR or setting up CEL to track that? |
| 12:27.11 | weinerk | but if caller hangs up - I dont return from Dial |
| 12:28.08 | weinerk | honestly I did not do it with CDR - I just do a timestamp before Dial and after Dial and if billable - I do diff |
| 12:28.23 | weinerk | also what is CEL? |
| 12:28.33 | leifmadsen | and you're only tracking a single channel |
| 12:28.38 | leifmadsen | a call is made up of 2 separate channels |
| 12:28.43 | leifmadsen | ~cel |
| 12:28.43 | infobot | CEL is Channel Event Logging, or http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html#Monitoring_id246970 |
| 12:29.05 | leifmadsen | anyways, it appears your approach is incorrect for what you're trying to do |
| 12:31.11 | weinerk | ~CDR |
| 12:31.11 | infobot | hmm... cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
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| 12:50.37 | mjordan | ~cdrw |
| 12:50.37 | infobot | The Unix CD-Writer compatibility list is at http://www.shop.de/cgi-bin/winni/lsc.pl, or http://newbiedoc.sourceforge.net/tutorials/cdrw/index-debian-cdrw.html |
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| 13:21.55 | shamelessn00b | http://pastebin.com/J03m7is4 |
| 13:21.58 | shamelessn00b | [TK]D-Fender: |
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| 13:22.57 | shamelessn00b | dtmf doesn't work, also, I don't get any audio on the spied on channel |
| 13:27.45 | shamelessn00b | unable to get chanspy working on local channel :| |
| 13:28.45 | [TK]D-Fender | shamelessn00b, You did not show 1111 and we have no idea what gets called when... |
| 13:28.58 | shamelessn00b | can I paste here |
| 13:29.24 | shamelessn00b | exten => 1111,1,Answer() |
| 13:29.25 | shamelessn00b | exten => 1111,n,Goto(abc,testuser1,1) |
| 13:29.46 | shamelessn00b | lemme paste the message log on asterisk console |
| 13:30.40 | [TK]D-Fender | shamelessn00b, that channel itself was never answered <- |
| 13:30.41 | *** join/#asterisk yetanotheruser (~butterfly@p4FDC1EAE.dip.t-dialin.net) |
| 13:31.17 | yetanotheruser | hi |
| 13:31.19 | shamelessn00b | oh ok, lemme add that line |
| 13:31.38 | [TK]D-Fender | shamelessn00b, And who said Chanspy passed on DTMF? |
| 13:32.01 | [TK]D-Fender | shamelessn00b, Its for LISTENING IN |
| 13:32.14 | shamelessn00b | even if I set DTMF to inband? |
| 13:32.36 | [TK]D-Fender | exten => testuser1,1,Set(SPYGROUP="spyonthis") <- quotes are BAD |
| 13:32.47 | [TK]D-Fender | Chanspy does not parse DTMF |
| 13:33.25 | shamelessn00b | wouldn't inband dtmf be equivalent to playing a dtmf tone on the channel |
| 13:33.58 | shamelessn00b | hence it would reach to all bridged channels as well |
| 13:34.10 | shamelessn00b | I could always use the manager interface, just giving it a try |
| 13:34.19 | [TK]D-Fender | God fix your first errors |
| 13:34.29 | [TK]D-Fender | including "core show application record" <- |
| 13:34.33 | [TK]D-Fender | Go* |
| 13:36.07 | shamelessn00b | no description for record available |
| 13:36.15 | shamelessn00b | I checked on voip-info.org |
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| 13:37.54 | [TK]D-Fender | your syntax is wrong and I told you this already |
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| 13:43.06 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:44.38 | shamelessn00b | http://pastebin.com/1hNUpz1h |
| 13:44.55 | shamelessn00b | that's the message log on asterisk console, autofailthrough this time |
| 13:45.29 | shamelessn00b | Record(filename:format[|silence][|maxduration][|option]) |
| 13:46.56 | shamelessn00b | http://pastebin.com/embbQua3 (updated dialplan) |
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| 13:49.16 | *** join/#asterisk yetanotheruser (~butterfly@p4FDC1EAE.dip.t-dialin.net) |
| 13:53.18 | yetanotheruser | has anyone ever used asterisk win 32 with sipgate? |
| 13:53.28 | yetanotheruser | is it possible or should i rather go for voip buster? |
| 13:56.15 | yetanotheruser | just to have said it: there is no way around windows, i wish there was but the decision about the server isn't up to me |
| 13:57.19 | Qwell | yetanotheruser: Not in here, no. Everybody that has used astwin32 has been taken out back and shot. |
| 13:57.31 | [TK]D-Fender | shamelessn00b, -- Auto fallthrough, channel 'Local/start@abc-fb69;1' status is 'UNKNOWN' |
| 13:57.34 | [TK]D-Fender | shamelessn00b, Ran out of dialplan |
| 13:57.51 | Qwell | yetanotheruser: 12 years later, and people still fall for the april fools joke... |
| 13:58.01 | shamelessn00b | [TK]D-Fender: shouldn't it block on ChanSpy? |
| 13:58.08 | Hive | lol |
| 13:58.51 | [TK]D-Fender | Qwell, Apparently we missed one... lemme get the blunderbuss.... |
| 13:59.17 | shamelessn00b | exten => start,n,chanspy(,swg(spyonthis)) |
| 13:59.20 | shamelessn00b | :/ |
| 13:59.40 | [TK]D-Fender | crams in a volley worth of the few metal bits from a series of demolished GrandStream phones.... |
| 13:59.54 | yetanotheruser | Qwell: what you mean is it is disallowed to talk about asterisk win 32 on this channel? |
| 14:02.39 | shamelessn00b | [TK]D-Fender: and I'm still not getting those played back digits on the spied on channel |
| 14:09.28 | [TK]D-Fender | shamelessn00b, what pat of "won't do DTMF" are you getting stuck on? |
| 14:10.01 | shamelessn00b | [TK]D-Fender: right now, even chanspy isn't working properly |
| 14:10.05 | shamelessn00b | forget DTMF |
| 14:10.38 | shamelessn00b | that channel shouldn't be running out of dialplan |
| 14:10.52 | shamelessn00b | as it is supposed to block while it executes chanspy |
| 14:11.10 | p3nguin | If I got forced into a Windows server, I'd install virtualbox and set up a headless Linux server in a virtual machine. |
| 14:11.32 | [TK]D-Fender | shamelessn00b, You should take a VERY close look at line #17 of your dialplan pastebin.......... |
| 14:12.06 | [TK]D-Fender | yetanotheruser, Asterisk doesn't care what provider you use it with. |
| 14:12.16 | shamelessn00b | its commented out |
| 14:12.34 | [TK]D-Fender | http://pastebin.com/J03m7is4 <- #17 |
| 14:13.03 | shamelessn00b | lol wait |
| 14:15.44 | shamelessn00b | ok, dialplan is working fine now, but still not getting any audio on the spied on channel |
| 14:15.56 | p3nguin | The good old "asteriskspeaker(loop)" extension. |
| 14:16.55 | p3nguin | I'm also curious how dialing 1111 on the phone runs any of what I see pasted. |
| 14:17.03 | *** join/#asterisk wonderworld (~ww@dsdf-4db53493.pool.mediaWays.net) |
| 14:17.34 | p3nguin | Ah, that part is described in here as opposed to in the pastebin. |
| 14:18.07 | shamelessn00b | p3nguin: I fixed that |
| 14:18.24 | yetanotheruser | i heard about connecting issues with asterisk and sipgate, this is why i asked |
| 14:18.34 | shamelessn00b | lemme re-paste my dialplan |
| 14:19.03 | yetanotheruser | asterisk is supposed to solve a nat-problem |
| 14:19.43 | [TK]D-Fender | yetanotheruser, there are tons of sipgate users. It has always worked. People however are often broken. We do not warranty them. |
| 14:20.14 | [TK]D-Fender | yetanotheruser, And * doesn't "solve" problems... it may replace something you may have been using that had one of it's own however |
| 14:20.18 | p3nguin | its own |
| 14:20.39 | *** join/#asterisk Defraz (~Defraz@69.20.176.131) |
| 14:20.50 | [TK]D-Fender | That two ;) |
| 14:20.56 | [TK]D-Fender | Am I rite? |
| 14:21.11 | shamelessn00b | http://pastebin.com/mM6qTZZ6 |
| 14:21.11 | [TK]D-Fender | Bet her? |
| 14:21.43 | yetanotheruser | well, our sip-clients have a problem with connecting to sipgate, and we want to use asterisk to connect and then use our softphones |
| 14:22.09 | [TK]D-Fender | yetanotheruser, Go right ahead.... |
| 14:23.14 | yetanotheruser | think i'll do so. fine if i heard it worked. then people who told me "impossible" didn't configure it the right way |
| 14:23.17 | yetanotheruser | thanks |
| 14:24.25 | shamelessn00b | [TK]D-Fender: do you see me doing anything wrong in the dial plan now? |
| 14:24.56 | [TK]D-Fender | I know I don't see a new call.... |
| 14:25.01 | shamelessn00b | chanspy shows it to be attaching itself to the channel I want it to be spying on, and despite the whisper flag being set, I'm unable to hear anything that is played on the local channel |
| 14:25.30 | [TK]D-Fender | yetanotheruser, You should reconsider how much you trust these things you "heard" and who you "heard" them from.... |
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| 14:27.19 | shamelessn00b | http://pastebin.com/fkD6zg8J |
| 14:27.35 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:28.06 | shamelessn00b | one more thing, chanspy keeps running when I hangup the call to 1111, would I have to kill that manually? |
| 14:29.42 | yetanotheruser | well, if you hear something and don't know if it is true, collecting information is probably not too bad |
| 14:30.54 | Hive | does anyone know why this dial command would cuase 4-105 to ring 3 times? |
| 14:30.55 | Hive | http://pastebin.com/5KMwCrJw |
| 14:31.17 | Hive | doesnt seem like a problem, but it's kind of curious behavior |
| 14:31.18 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:31.21 | shamelessn00b | dial plan seems to execute the way I want it to, just that the bridging part isn't working how I want it to be working |
| 14:31.50 | [TK]D-Fender | Hive, that is where you see the other side REPORT that it is ringing... |
| 14:32.02 | [TK]D-Fender | Hive, so you've have to look at what it is that you are calling... |
| 14:32.07 | *** join/#asterisk DavidNGonzalez (~dgonzalez@99-158-162-221.uvs.cicril.sbcglobal.net) |
| 14:32.30 | [TK]D-Fender | Hive, and it doesn't actually necessarily mean that it physically made a sound with a given frequency, etc |
| 14:34.28 | Hive | [TK]D-Fender: ok so it could be the behavior of the device that is ringing, to send multiple ringing notifications like that? |
| 14:34.35 | [TK]D-Fender | yes |
| 14:34.41 | Hive | cool, thanks :) |
| 14:34.44 | [TK]D-Fender | it IS the notification you're seeing. |
| 14:34.59 | DavidNGonzalez | Having an issue with 911 calling, wondering if any of you can help me diagnose thise. Any time someone calls 911, we can hear the operator but they cannot hear us. The emergency calls are set to go out the same sip trunk as the rest of our voice calls, and yet that is the only number that misbehaves that way. Thoughts? |
| 14:35.17 | Hive | strange that it's sending multiple, but as long as it's not some hidden bug in my system, im ok with that :P |
| 14:35.34 | [TK]D-Fender | Hive, it's actually common enough for one/multiple |
| 14:35.56 | shamelessn00b | [TK]D-Fender: I've pasted a new call |
| 14:43.07 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 14:43.07 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:43.51 | *** join/#asterisk bobb_WU (~bobb_WU@206.74.211.13) |
| 14:54.42 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 14:54.52 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 14:56.57 | bobb_WU | how can i troubleshoot dtmf issues? i'm having no trouble from my asterisk phones, but our mitel side can't dial their passwords to the asterisk VM server. it is confirmed that mitel is sending rfc2833 tones to the voice mail. |
| 15:03.45 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 15:03.47 | bobb_WU | from having sip debugging on, i see Non-codec capabilities (dtmf) : us - 0x1 (telephone-event) and the same for the peer / combined |
| 15:04.32 | jeffspeff | when i enter a confbridge conference, it doesn't ask to record your name anymore. |
| 15:05.09 | bobb_WU | there is a central relay between the voicemail box and the mitel side. i made the SIP peer table dtmfmode 'rfc2833' and 'auto', both did nothing new |
| 15:16.53 | *** part/#asterisk yetanotheruser (~butterfly@p4FDC1EAE.dip.t-dialin.net) |
| 15:24.17 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-202.mobile.uci.edu) |
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| 15:27.26 | *** join/#asterisk d0uglas (~d0uglas@batteryboss.org) |
| 15:27.46 | d0uglas | Hi folks. Can Blackberry MVS be used with Asterisk or Cisco only? |
| 15:39.21 | p3nguin | Can someone tell me about ANI when using SIP? |
| 15:41.38 | shamelessn00b | what about it? |
| 15:43.20 | p3nguin | Is it important for that channel tech? |
| 15:44.15 | p3nguin | If I send a specific caller id number on a call which goes via ITSP to the PSTN, where does the ANI come into the equation? |
| 15:45.31 | p3nguin | Tell me all that you know about ANI with regard to SIP calling. |
| 15:45.52 | shamelessn00b | don't think so, I think it would only read the CLI information |
| 15:46.07 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.228) |
| 15:46.08 | shamelessn00b | the number basically |
| 15:46.34 | shamelessn00b | ANI would be for the outbound trunk connecting to the PSTN |
| 15:48.50 | WIMPy | Is it supported at all? |
| 15:49.03 | shamelessn00b | I've been using SS7 over asterisk for quite some time now, don't really need to set ANI for calls coming over SIP and being routed to SS7 trunks |
| 15:49.40 | WIMPy | only knows that chan_sip doesn't set ANI. |
| 15:49.45 | shamelessn00b | still stuck on that stupid dialplan issue, can't seem to get chanspy work properly :( |
| 15:51.35 | _Corey_ | p3nguin: It depends a lot on your provider... some will accept what you have in the From header, others require a different field be added to the SIP header |
| 15:52.02 | _Corey_ | p3nguin: (the custom sip header guys are usually running on broadsoft platforms that can't get billing right any other way) |
| 15:52.07 | WIMPy | The whole caller ID thing is a little messed up. |
| 15:53.16 | WIMPy | On one side we differ between (num) and (ANI), but on the other side they both have the -pres property that includes the screening indicator. |
| 15:53.35 | _Corey_ | yup |
| 15:54.07 | WIMPy | That makes the ehole thing rather unpredictable :-( |
| 15:54.16 | WIMPy | s/eh/wh/ |
| 15:54.30 | shamelessn00b | lol |
| 15:54.32 | jeffspeff | when i enter a confbridge conference, it doesn't ask to record your name anymore. what did i accidentally change or not set? |
| 15:54.36 | shamelessn00b | wimpy can you guide me on this |
| 15:55.18 | shamelessn00b | http://pastebin.com/fkD6zg8J |
| 15:55.53 | shamelessn00b | here's the dialplan http://pastebin.com/mM6qTZZ6 |
| 15:56.02 | p3nguin | I'm wanting to test how my carrier responds to setting ANI and pres. Is it enough to set CALLERID(ANI-num) to a valid arbitrary number to test it? |
| 15:56.12 | p3nguin | calling with SIP |
| 15:56.57 | WIMPy | p3nguin: I'd look at sip debug to be sure anything happens at all. |
| 15:56.59 | p3nguin | I don't fully understand the various data types for the CALLERID function. |
| 15:57.16 | shamelessn00b | anything that the local channel 'says' isn't audible on the SIP channel |
| 15:57.31 | shamelessn00b | but that shouldn't be the case... |
| 15:57.35 | _Corey_ | p3nguin: Just do CALLERID(number) |
| 15:57.45 | _Corey_ | p3nguin: That SHOULD be what they need in most cases |
| 15:57.57 | WIMPy | I found out that using the REDIRECTING function doesn't seem to do anythign for chan_sip. |
| 15:58.07 | p3nguin | I already do that, but I'm interested in ANI. |
| 15:58.44 | p3nguin | I personally do not ever try to set ANI for calls to the ITSP. Someone has brought up a situation regarding ANI and SIP, and I need to get to the bottom of it. |
| 15:59.04 | WIMPy | Ich which direction? |
| 16:00.31 | WIMPy | I've got one ITSP that can send two caller ids to me. |
| 16:00.41 | _Corey_ | p3nguin: You'd have to check with your ITSP to see if they've implemented something |
| 16:00.43 | WIMPy | They're using Huawei. |
| 16:00.59 | WIMPy | But Asterisk doesn't understand that. |
| 16:01.43 | *** join/#asterisk jdnwest (~Adium@c-98-201-128-17.hsd1.tx.comcast.net) |
| 16:02.01 | jdnwest | Anyone know how long it takes for CNAM's (caller ID database entries) to get updated once you change them? |
| 16:03.02 | p3nguin | If I set my CALLERID(num) on the SIP call which is going to the ITSP, and I call a telco on their toll-free number, and their system tells me that I am calling from the number I have set on the CALLERID(num), I feel like that number is also my ANI number. Does that sound accurate? |
| 16:03.16 | _Corey_ | jdnwest: It depends on your provider... :) I've had some update same day, others taking many days or weeks |
| 16:03.38 | jdnwest | I did it through their pretty web portal (Paetec) |
| 16:04.14 | _Corey_ | jdnwest: Hmm, I think it took 24-48hrs the last time I did one w/Paetec |
| 16:04.28 | p3nguin | That, to me, indicates that the ITSP is probably setting the ANI and CID numbers when they terminate the call on their trunk. |
| 16:06.41 | jdnwest | Corey, Not bad. I like Paetec's web portal thingy, but their organization is so random. I terminated a T1 with them using Frame Relay.... |
| 16:07.21 | *** join/#asterisk tomwish (~Tom@97-117-164-105.phnx.qwest.net) |
| 16:08.16 | _Corey_ | jdnwest: Don't get me started about Paetec... :-) |
| 16:09.26 | *** join/#asterisk twanny796 (~twanny@46.11.81.69) |
| 16:12.55 | p3nguin | I just tested a call via flowroute, setting the CID num to one number and the ANI num to another number, and called MCI... it reported that I was calling from the number I set in CID. I watched the SIP debug, and the number I set in ANI was not used anywhere at all. |
| 16:13.19 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:14.06 | *** join/#asterisk pdtpatr1ck (~pdtpatric@firewall2.dc1.tech-corps.com) |
| 16:14.25 | p3nguin | Same thing when testing via VoIP.ms. |
| 16:14.27 | _Corey_ | p3nguin: I doubt any ITSPs will support it |
| 16:14.54 | _Corey_ | You will have more luck with a more traditional ILEC/CLEC |
| 16:15.15 | p3nguin | I'm not even sure why the whole ANI thing was brought up, but I wanted to put a stop to the concerns that were raised. |
| 16:15.45 | _Corey_ | what concerns? |
| 16:16.21 | p3nguin | The subject that brought up the ANI in the first place was placing anonymous calls. |
| 16:17.55 | p3nguin | The dial plan apparently sets the CID num to unknown (which is also invalid in my experience), but the ANI was not affected. Subsequently, the dial plan was going to be changed to set the ANI to 0000000000. |
| 16:18.26 | p3nguin | If I try to set ANY INVALID caller identification, it results in my ITSP's default caller ID number being sent on calls. |
| 16:18.49 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:22.02 | *** join/#asterisk AaronCharpentier (~IceChat77@pool-173-48-243-195.bstnma.fios.verizon.net) |
| 16:23.30 | AaronCharpentier | We currently have a distro of Asterisk on our server from about a year ago (Never updated) This was the ISO, Ready-To-Go solution (OS+Asterisk), the GUI never worked - lots of problems, but we've been using it and manually editing users/etc (Gui works, just doesnt write...I guess) We just purchased a new machine to take the place of our old phone server, I'm curious - Anyone have any input as to 32bit vs 64bit? Hav |
| 16:24.15 | p3nguin | You got truncated at "input as to 32bit vs 64bit? Hav" |
| 16:24.52 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 16:25.35 | AaronCharpentier | <PROTECTED> |
| 16:25.50 | jeffspeff | hey, if anybody wants to create dynamic conference rooms using confbridge in 10.x here's a nice chunk that i wrote to do it.... http://pastebin.com/gHJgv7qz |
| 16:26.04 | AaronCharpentier | I just know from personal use, when I was using Linux as my main os, a few years back - 64bit support was lacking. |
| 16:26.24 | AaronCharpentier | <PROTECTED> |
| 16:26.55 | p3nguin | Will your system have more than 4GB system RAM? |
| 16:27.11 | AaronCharpentier | Negative, 4 GB Exactly. Going to suggest just sticking with 32B Then? |
| 16:27.16 | p3nguin | Yes. |
| 16:27.25 | AaronCharpentier | Convinced me, thanks ;) |
| 16:27.30 | p3nguin | You will gain nothing by switching to 64-bit OS. |
| 16:27.49 | AaronCharpentier | Figured as much, I know that any performance gains would be so negligable that we would never notice them. |
| 16:27.50 | [TK]D-Fender | AaronCharpentier, Ther are many different all-in-one-distros. We have no idea which release of which distro you used or the origin of your problems or why you decided to try to walk all over it as your solution. They generally all work. |
| 16:28.28 | AaronCharpentier | Very sorry TK, I only saw two available as current all-in-ones, 32 and 64. Off of Asterisks site, I wasn't so much hoping for troubleshooting of our old distro, we've just given up on that machine. |
| 16:28.44 | p3nguin | I'm quite satisfied with AsteriskNOW 32-bit. |
| 16:28.51 | [TK]D-Fender | AaronCharpentier, Also... no mention of what GUI... |
| 16:29.31 | AaronCharpentier | No worries TK, like I said - was more mentioning of why we are changing, rather than hoping for support of our existing install. P3nguin answered my question, I'm all set. |
| 16:30.10 | p3nguin | I don't use any GUI with AsteriskNOW, but I'm sure that the FreePBX packaged with AsteriskNOW will be as good as, or better than, any other distribution of FreePBX available. |
| 16:31.02 | AaronCharpentier | Gotcha, does AsteriskNOW come equipped with the ability to have a gui, though? Straight from the ISO? |
| 16:31.31 | AaronCharpentier | We edit all of our users/settings through ssh and manual edits, gui isn't important to me per say, but the other techs here might get some use out of it. |
| 16:32.07 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:32.12 | p3nguin | AsteriskNOW provides your choice of no GUI, the not-recommended Asterisk GUI, and FreePBX. |
| 16:32.38 | p3nguin | and it is "per se." |
| 16:32.46 | AaronCharpentier | Gotcha, I'll go with FreePBX, we currently use the Asterisk Gui |
| 16:33.15 | AaronCharpentier | It's also "Penguin" Mr.1337 ;) |
| 16:33.30 | p3nguin | Forget about most manual configuration when you use FreePBX. |
| 16:33.44 | AaronCharpentier | Alright, that about takes care of it! Thanks everyone, I appreciate the help - especially p3nguin! Have a great day everyone. |
| 16:34.23 | p3nguin | My nick is p3nguin because p1nguin and p2nguin were already taken. :) |
| 16:36.48 | [TK]D-Fender | See in #asterisk you're not just a number .... you're ALPHAnumeric! |
| 16:39.41 | *** join/#asterisk shamelessn00b (~bwahahaha@182.177.114.73) |
| 16:40.07 | shamelessn00b | quick question, how would I add a local channel to a conference bride, my initial attempt failed |
| 16:40.15 | shamelessn00b | _Corey_: |
| 16:41.39 | _Corey_ | how are you trying to add it? originate? call spool file? |
| 16:41.45 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 16:41.59 | *** join/#asterisk retentiveboy (~retentive@72.54.144.26) |
| 16:44.22 | shamelessn00b | dial, _Corey_ using the G(context^exten^pri) option in dial command |
| 16:45.02 | shamelessn00b | A lands at extension n, B at n+1, I call confbridge for A |
| 16:45.14 | *** join/#asterisk elanz (~elanz@69.169.150.15.provo.static.broadweavenetworks.net) |
| 16:45.31 | *** join/#asterisk hehol (~Adium@ip-178-202-243-147.unitymediagroup.de) |
| 16:45.46 | shamelessn00b | call originate(local/someexten@somecontext,app,confbridge,parameters) at n+1 |
| 16:45.54 | shamelessn00b | and then confbridge at n+2 |
| 16:46.11 | shamelessn00b | someexten@somecontext would contain the EAGI script |
| 16:48.43 | *** join/#asterisk twanny796 (~twanny@46.11.81.69) |
| 16:49.32 | WIMPy | Ok, so just as I suspectd: No ANI in chan_sip at all. |
| 16:50.13 | _Corey_ | shamelessn00b: Well, without really thinking about this... it looks like the first thing you're doing wrong is the originate |
| 16:50.46 | _Corey_ | shamelessn00b: You want to call the Local/x@y exten that will run the ConfBridge command and originate an application (EAGI) into it |
| 16:51.04 | elanz | question: when I leave a vm message at extension 214 for example, a wav file is created under: /var/spool/asterisk/monitor/exten-214-801-xxx-xxxx-20120501-104316-1335890582.6267.wav then under: /var/spool/asterisk/voicemail/default/214/INBOX the same .wav file is labeled msg0000.wav is this normal * behavior? |
| 16:51.37 | shamelessn00b | _Corey_: I can use the G option and do it on the second leg of the call then? |
| 16:51.45 | shamelessn00b | at n+1 |
| 16:51.50 | _Corey_ | shamelessn00b: I don't use originate() in the dialplan ever really, so I'd have to try it out |
| 16:52.27 | shamelessn00b | I still don't have a clue why didn't the chanspy thing work |
| 16:52.31 | _Corey_ | It sounds like you're close |
| 16:53.05 | p3nguin | This is interesting. Using voipms, I can set my pres to any of the prohibited values, and the caller id number always appears on the called phone, but calling MCI makes it say I am calling from 714. |
| 16:53.24 | shamelessn00b | basically what I'm trying to design is an IVR that 2 people will be able to use at the same time |
| 16:53.28 | shamelessn00b | while talking to each other |
| 16:53.33 | shamelessn00b | :) |
| 16:53.41 | shamelessn00b | 2 (or more) |
| 16:53.47 | p3nguin | Using flowroute, setting the pres to any of the prohitied values, the caller id always shows unavailable on the called phone, but calling MCI always reports the actual caller id number. |
| 16:54.45 | _Corey_ | shamelessn00b: I remember you describing this before. I need to run out for lunch now, but you should get there if you keep experimenting. It sounds like you're nearly done |
| 16:55.26 | shamelessn00b | lol, I feel like I'm nearly done for the last 2-3 days, and then I get stuck on some really PITA issue |
| 16:55.39 | shamelessn00b | and have to start over again |
| 16:58.15 | *** join/#asterisk wonderworld (~ww@dsdf-4db53493.pool.mediaWays.net) |
| 16:58.51 | WIMPy | p3nguin: Isn;t it great to do some basic research? And the great thing is that it won't really help. |
| 16:59.25 | WIMPy | You still don;t know if it warks the way you found out intentionally or if it will change on the next release of whatever. |
| 16:59.55 | p3nguin | What I'm finding out doing the tests is that the best way to restrict a phone number is to set a fully random (valid) caller id number. |
| 17:00.24 | *** join/#asterisk shamelessn00b (~bwahahaha@39.47.154.253) |
| 17:00.40 | p3nguin | When I got unavailable to show up on my cell phone display, calling my GV number still showed the number. |
| 17:00.44 | tomwish | I am having a wierd clid issue. A call comes in with a good clid/ani. I can see the info in the ngrep. I answer the call, play a msg and then forward the call to a tfn. The clid does not carry through. It ends up being sent with asterisk as the clid. I tried to put a def clid in but it ignores it. I have put the pertinant parts of sip.conf and extensions.conf here http://pastebin.com/U87ekgYL. I really just want the orig clid to pass. |
| 17:02.16 | *** join/#asterisk gusto (~gusto@ip-109-43-0-101.web.vodafone.de) |
| 17:02.27 | *** join/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 17:02.45 | talntid | is there a way to bump the overall recieve volume for my reps, without using those amplifier box things? |
| 17:02.50 | talntid | I'm on pure SIP |
| 17:02.52 | *** join/#asterisk serafie (~erin@nat/digium/x-igchgweufiwqpwoq) |
| 17:03.01 | p3nguin | There's the VOLUME() function. |
| 17:04.01 | WIMPy | And the devices surely have volume settings as well. Might be the better option. |
| 17:04.11 | talntid | they are all the way up |
| 17:04.16 | talntid | for the people who are complaining |
| 17:05.06 | WIMPy | Are you using POTS? |
| 17:05.17 | *** join/#asterisk shamelessn00b (~bwahahaha@39.47.90.186) |
| 17:05.29 | talntid | negative |
| 17:05.46 | talntid | polycom ip550's |
| 17:06.25 | WIMPy | And the other end? |
| 17:07.02 | talntid | various customers |
| 17:07.25 | WIMPy | Connecting to you how? |
| 17:08.02 | talntid | polycom ip550 -> switch -> asterisk server -> switch -> internet -> flowroute/vitelity -> |
| 17:09.54 | talntid | is that what you are asking? |
| 17:09.55 | p3nguin | I would use VOLUME() or AGC() to make things louder. |
| 17:09.59 | WIMPy | Yes |
| 17:10.20 | WIMPy | Yes, VOLUME is the last option then. |
| 17:10.45 | *** join/#asterisk AndyHarris2 (~AndyHarri@dsl-217-155-202-52.zen.co.uk) |
| 17:10.51 | WIMPy | And I get a feeling again that I was lucky that I didn't get any of the Polycoms. |
| 17:11.38 | talntid | most of my people are happy |
| 17:11.43 | talntid | just a few of the older crowd.. |
| 17:12.23 | [TK]D-Fender | Phone is fine. ITSP is not |
| 17:13.42 | talntid | other suggestion? |
| 17:14.21 | [TK]D-Fender | Adjust * or your phones to compensate... or change providers |
| 17:14.39 | talntid | phone volumes are all the way up |
| 17:15.08 | Qwell | get employees that don't have defective ears |
| 17:15.53 | [TK]D-Fender | talntid, you adjusted the master gains in the configs? |
| 17:16.02 | [TK]D-Fender | (of course not) |
| 17:16.28 | talntid | negative. |
| 17:16.46 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:17.03 | talntid | I thought those were for zap/dahdi stuff? |
| 17:17.58 | [TK]D-Fender | .. pardon? |
| 17:18.18 | talntid | "master gain settings" |
| 17:18.41 | talntid | I thought those are in the settings for dahdi, zap? |
| 17:18.43 | [TK]D-Fender | Who said anything about Zap/DAHDI? |
| 17:18.44 | WIMPy | Hint: Asterisk doesn't have such a thing. |
| 17:18.52 | talntid | gotcha. |
| 17:18.54 | [TK]D-Fender | talntid, POLYCOM CONFIGS. |
| 17:19.02 | talntid | ohhh |
| 17:19.03 | [TK]D-Fender | Imaginations are in the bin on your left... |
| 17:19.04 | WIMPy | Ony per channel, just like you said. |
| 17:19.08 | Qwell | [TK]D-Fender: what? the polycom configs don't apply to dahdi? |
| 17:19.11 | Qwell | since when?! |
| 17:19.17 | [TK]D-Fender | TOMRROW! |
| 17:19.43 | talntid | I didn't know he was talking about polycom configs, obviously. and the only other "gain" settings I had seen were in the pri card's configs |
| 17:19.59 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 17:20.37 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-130.telkomadsl.co.za) |
| 17:25.35 | talntid | http://pastebin.com/nLy8Ggra |
| 17:25.52 | talntid | the 6000 works, but * and # don't appear to do anything, and the volume sounds the same. Is this a bad test case? |
| 17:28.25 | *** join/#asterisk Elv1313 (~lepagee@2607:fad8:4:0:4261:86ff:fe62:9560) |
| 17:29.30 | Elv1313 | Hi, is there any client side (without server side module) way to download individual voicemail as audio file using sip credentials? |
| 17:29.30 | [TK]D-Fender | Considered the possibility that MoH isn't a "call", and that you picked the wrong one to bump? |
| 17:30.03 | [TK]D-Fender | Elv1313, SIP can't magically download files |
| 17:30.20 | [TK]D-Fender | Elv1313, And has nothing to do with "files" so yes, clearly you will have to have something on the server side |
| 17:31.02 | [TK]D-Fender | talntid, also, 2db isn't that much. 6 should stand out. |
| 17:31.59 | talntid | roger that |
| 17:32.27 | *** join/#asterisk qakhan (~qakhan@180.178.150.247) |
| 17:32.41 | Elv1313 | [TK]D-Fender: I was thinking more about Asterisk specific protocols, but I could not find anything. So, if we are developping a native phone client, we can't have an "hack" to download voicemail from any unmodified asterisk server (just to be clear)? |
| 17:34.14 | qakhan | hi all, i have Digium TDM 800p card. when i hangup call from my phone it doesnt hangup call. when caller hangup from his phone then its hangup |
| 17:34.25 | qakhan | please help |
| 17:35.11 | shamelessn00b | [TK]D-Fender: did you look at the second call yet? |
| 17:35.24 | [TK]D-Fender | Elv1313, * doesn't transmit "files". The closest "on demand" concept is a serious security risk |
| 17:35.41 | [TK]D-Fender | Elv1313, And not really "sane" |
| 17:36.15 | Elv1313 | Not a serurity risk over TLS, but anyway, if there is none, that answer my question, thanks |
| 17:37.21 | [TK]D-Fender | Elv1313, there is no SIP way for this period |
| 17:37.50 | [TK]D-Fender | Elv1313, I was saying there is a vague ASTERISK means (other) in some odd sense for doing something like this |
| 17:43.14 | talntid | volume, when on call, seems to have helped :) |
| 17:46.00 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:48.17 | *** join/#asterisk F|shie (~chatzilla@182.177.125.72) |
| 17:51.27 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-202.mobile.uci.edu) |
| 17:55.58 | shamelessn00b | [TK]D-Fender: |
| 17:56.22 | shamelessn00b | surprisingly it works fine if I replace playback/wait with Dial to a SIp account |
| 18:03.42 | qakhan | hi all, i have Digium TDM 800p card. when i hangup call from my phone it doesnt hangup call. when caller hangup from his phone then its hangup |
| 18:03.43 | qakhan | please help |
| 18:10.48 | carrar | ~sipnat |
| 18:10.48 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 18:10.52 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 18:11.12 | cj | thanks, carrar |
| 18:11.20 | carrar | np |
| 18:11.35 | tomwish | I am having a strange caller id issue. I can see the correct callerid being sent to asterisk from an ngrep. I take the call, answer it, play a message then forward it to a toll free. for some reason the caller id doesn't follow. I even tried putting a def caller id in the dial plan but it doesn't pass either. here is the pertinant parts of the conf files http://pastebin.com/U87ekgYL. I have read the docs and it looks like it should work. |
| 18:12.32 | rgsteele | Is it necessary to have TCP ports 5004-5082 open for SIP signalling traffic, or will that range over UDP suffice? |
| 18:14.52 | Qwell | range? why? |
| 18:16.30 | cj | carrar: the asterisk server is not behind NAT, just the client |
| 18:16.39 | cj | I've already got nat=yes for the client |
| 18:18.11 | Qwell | tomwish: Your provider is likely blocking it. |
| 18:19.02 | tomwish | I can see it in the ngrep. We are sending asterisk as the from and contact. not the callerid info |
| 18:19.20 | tomwish | I can post an ingress and egress ngrep if that would help |
| 18:19.53 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 18:22.51 | wdoekes2 | tomwish: it might |
| 18:23.09 | *** join/#asterisk Agro (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net) |
| 18:24.14 | p3nguin | rgsteele: SIP is port 5060. |
| 18:24.37 | adyn | My boss wants to put in ASA 5505 devices in front of about 40 phones connected to a remote Asterisk PBX. Anyone have any knowlege of whether this might be an issue? I can't lab it out and I have no experience with ASA's. My google research has been inconclusive. |
| 18:25.28 | p3nguin | tomwish: Quotes in a caller ID name are incorrect. You should consider fixing that. |
| 18:26.14 | tomwish | ok |
| 18:26.19 | p3nguin | Set(CALLERID(all)=Some Name <3145551212>); <---- no quotes |
| 18:27.04 | tomwish | ok, let me try that |
| 18:27.21 | p3nguin | Dial(SIP/out/18888590786); Dial(ChannelTech/peer/extension) |
| 18:27.38 | wdoekes2 | p3nguin: that dial works either way |
| 18:27.50 | p3nguin | I know what it does. Do you? |
| 18:28.32 | p3nguin | Have you confirmed that your ITSP allows you to send your own caller id number? |
| 18:29.14 | tomwish | I am doing the ngreps from my opensips server. before it even hits the carrier |
| 18:29.28 | tomwish | here is the ingress/egress traces http://pastebin.com/pajtgJtm |
| 18:30.13 | Qwell | Remote-Party-ID: "john" <sip:3039549453@216.66.79.69>;party=calling;privacy=full;screen=no. |
| 18:30.19 | p3nguin | Are you saying that the caller id you are setting is not going out of your asterisk? |
| 18:30.38 | p3nguin | privacy=full? |
| 18:30.58 | p3nguin | Someone is trying to block their caller id number. |
| 18:31.19 | tomwish | I am not passing the one that comes in and when I try and inject one it doesn't pass either |
| 18:32.36 | tomwish | when I make the call the cli shows |
| 18:32.38 | tomwish | <PROTECTED> |
| 18:32.38 | tomwish | <PROTECTED> |
| 18:32.56 | tomwish | no errors but it doesn't send that in the sip invite |
| 18:33.10 | Qwell | Yes it does. |
| 18:33.38 | p3nguin | The RPID that qwell pasted indicates that it does. |
| 18:34.00 | wdoekes2 | tomwish: add Set(CALLERPRES(num-pres)=allowed_passed_screen) |
| 18:34.04 | tomwish | that is in the rpid not the from field |
| 18:34.13 | tomwish | or the contact field |
| 18:34.14 | Qwell | tomwish: Which is the correct thing to do. |
| 18:34.27 | wdoekes2 | you want it in the rpid, that's what you state in sendrpid=yes |
| 18:34.33 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 18:35.00 | tomwish | if I set sendrpid to no, then it sends as anon |
| 18:35.09 | Qwell | Don't do that then |
| 18:35.12 | [TK]D-Fender | tomwish, Show us your peer |
| 18:35.32 | tomwish | http://pastebin.com/U87ekgYL |
| 18:35.36 | wdoekes2 | s/CALLERPRES/CALLERID/ |
| 18:35.57 | p3nguin | wdoekes2: CALLERID(), not CALLERPRES(). |
| 18:36.54 | tomwish | ok. let me try that |
| 18:42.56 | tomwish | that fixed it. Thanks for all the help |
| 18:43.50 | pabelanger | Anybody ever had an issue where dahdi is not passing the clock information correctly between span 1 and span 2? EG: CO -> asterisk -> PBX I have a tech with a monitor on the line and they are telling me clock in to asterisk is good, but out from asterisk is bad |
| 18:48.58 | _Corey_ | pabelanger: Yeah, I've had that... I had some motherboard driver incompatibility thing going on |
| 18:49.35 | pabelanger | _Corey_: how did you solve the issue? Disable IRQs? Or replace the board |
| 18:50.28 | _Corey_ | pabelanger: (I worked with Matt Fredrickson on it if memory serves me right...) zttest/dahdi_test reported really crappy results |
| 18:50.43 | _Corey_ | He ended up sending me a new driver that fixed the compatibility issue |
| 18:51.01 | Qwell | _Corey_: cresl1n is cool like that |
| 18:51.15 | pabelanger | _Corey_: New version of dahdi? Or new dahdi firmware |
| 18:51.56 | _Corey_ | hang on, i'm checking my old e-mail box |
| 18:52.14 | pabelanger | _Corey_: cool, appreciate it |
| 18:53.41 | _Corey_ | pabelanger: If you're running dahdi_test and getting good results with the board loaded then it's probably not the same thing |
| 18:54.02 | cj | pabelanger: o/ |
| 18:54.10 | _Corey_ | mine was pretty striking... 99.999% w/the pseudo driver, 11% with the card loaded |
| 18:55.09 | pabelanger | _Corey_: Ya, dahdi_test works like a champ. And I don't see any errors with that. My issue, is the nortel attached to span 2, the clock is messed up. So, I'm trying to figure out why the clock is messed up between span 1 and 2 |
| 18:55.42 | sruffell | pabelanger: two single spans / dual-span? span 2 set to use clock recovered from other span or internal oscillator? |
| 18:56.10 | _Corey_ | pabelanger: youch... the install I had that crap on was a similar nortel integration. bad mojo on that nortel equipment |
| 18:56.27 | pabelanger | sruffell: dual span. Span 1 to CO, span 2 to nortel. Span 1 is master, span 2 is 0 |
| 18:56.48 | sruffell | span 1 has timing set to 1? |
| 18:57.27 | sruffell | i.e….what is in /etc/dahdi/system.conf |
| 18:58.58 | pabelanger | sruffell: http://pastebin.com/cFLHAqnB |
| 18:59.35 | Qwell | unrelatedly, your echocan lines are b0rked |
| 18:59.53 | pabelanger | Qwell: copy paste error |
| 19:00.46 | sruffell | yeah…that looks correct. Anything in dahdi_maint -s 2? |
| 19:00.48 | pabelanger | The local tech is seeing some 'polarity' issue on the span 2 side. His words 'clock into span 1' is good, but coming out from 'span 2' is bad |
| 19:01.06 | pabelanger | sruffell: nothing specific |
| 19:01.17 | sruffell | using a cross-over from span-2 into the nortel? |
| 19:01.19 | pabelanger | I'll have to check again for output |
| 19:01.25 | sruffell | i.e…greened up? |
| 19:01.37 | pabelanger | sruffell: cross-over... cable? |
| 19:01.40 | sruffell | yes |
| 19:01.51 | pabelanger | Yes, T1 cable good |
| 19:01.56 | pabelanger | changed out twice |
| 19:02.19 | sruffell | if the clock is bad…you should see error in dahdi_maint |
| 19:02.26 | Qwell | I bet you need a J1 cable. |
| 19:02.29 | pabelanger | everything is pointing to an physical issue with the card / computer. We have also changed out the full asterisk box before. We end up have the same problems |
| 19:03.54 | pabelanger | sruffell: they recently recycled the box, but here is the output: http://pastebin.com/sUg2yyck |
| 19:04.36 | pabelanger | Qwell: why do you say J1? |
| 19:04.40 | AndyHarris2 | Would anyone know if the internal asterisk directory can be served to snom phones ? |
| 19:04.41 | Qwell | pabelanger: ignore me |
| 19:04.46 | sruffell | smirks |
| 19:04.51 | Qwell | just a bit of trolling while I compile |
| 19:05.05 | *** join/#asterisk F|shie (~chatzilla@182.177.9.247) |
| 19:06.11 | AndyHarris2 | Sorry --- I mean Asterisk Phonebook served to snom phones. ... |
| 19:06.45 | Qwell | Asterisk has no concept of phonebooks. |
| 19:07.36 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 19:07.51 | pabelanger | sruffell: I'm trying to get some low level debug logs, on the nortel side, to try and better understand what is happening. |
| 19:08.14 | sruffell | pabelanger: if those errored seconds are increasing at a normal rate from the telco side, I would focus on that. If it's recovering the clock from span 1, but that span has problems, it will have trouble providing a stable clock on span 2. You can isolate the provider by unplugging span 1 and seeing if you still have problems reported on span2. |
| 19:08.47 | AndyHarris2 | @Qwell --- apologies it reads 'Asterisk Phonebook' on the PIAF interface -- therefore assumed it was at the asterisk level |
| 19:08.59 | AndyHarris2 | Will look elsewhere |
| 19:09.02 | Qwell | AndyHarris2: You'll have to ask them |
| 19:09.33 | cj | pabelanger: got any spare tuits? I'm using your package and not getting any RTP |
| 19:09.51 | cj | I'm leaning toward it being a misconfiguration on my side |
| 19:09.52 | pabelanger | sruffell: okay, so if I unplug span 1, clock will be generated internal and sent to span 2, right? |
| 19:09.56 | cj | but it could always be you :) |
| 19:10.34 | *** join/#asterisk timahvo1 (~rogue@197.177.176.103) |
| 19:10.36 | sruffell | yes…if all the numbered spans are in red alarm, the framer will only use it's internal oscillator to generate a clock on all the spans set to 0. |
| 19:10.48 | pabelanger | sruffell: okay cool. Let me do that. |
| 19:14.51 | pabelanger | sruffell: when span 1 gets clock, is it simply echo'd (insert technical jargon here) to span 2? Or does dahdi modify it in any way. Trying to understand if there is any way for that clock to get corrupt between span 1 and 2 |
| 19:15.42 | AndyHarris2 | Hmmm --- I found that there is a CLI command database show key --- this shows the phonebook entries |
| 19:16.23 | sruffell | the framer only has one clock source. It will either recover clock from *one* of the spans, or use the internal oscillator. The number in /etc/dahdi/system.conf for timing is the priority of where to recover clock from. That timing source is then used by all the other spans. So…yes…if timing is recovered from span 1, that timing will be used to drive span 2. |
| 19:17.17 | sruffell | and…for spans on a particular framer….dahdi isn't involved at all. |
| 19:18.00 | p3nguin | andyharris2: That is something your web application has created. |
| 19:18.35 | AndyHarris2 | Ahhh IC .. |
| 19:18.43 | AndyHarris2 | seems useful! |
| 19:20.42 | pabelanger | sruffell: okay, cool. Thanks for the information. I'll have to follow up on Thursday with the results of the test. The onsite tech from Bell is gone for the day |
| 19:20.50 | sruffell | cool...np |
| 19:29.59 | *** part/#asterisk elanz (~elanz@69.169.150.15.provo.static.broadweavenetworks.net) |
| 19:31.12 | cj | okay, it's getting better. inbound SIP calls are connecting and media from sip client to external devices is finding its way through. media from external device to sip client is not coming through, though. wah. |
| 19:31.32 | pabelanger | cj: you might want to ask for help here. Digium packages should be fine. Unfortunately I don't package them anymore but will be helping get mjordan up to speed |
| 19:31.52 | cj | oh? that's too bad. I enjoyed working with you in the past. |
| 19:32.15 | *** join/#asterisk timahvo1 (~rogue@197.177.176.103) |
| 19:46.27 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 19:47.31 | Katty | hello my asterisk does not work at all how tof ix plz |
| 19:49.01 | talntid | kicks Katty in the chin |
| 19:49.02 | talntid | :) |
| 19:49.41 | Katty | :< |
| 19:49.46 | Katty | abusseess!!! |
| 19:54.15 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 19:54.50 | talntid | i'm sorry, Katty |
| 19:55.16 | talntid | don't report me to Qwell. |
| 19:56.02 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 19:57.06 | tomwish | i am using app_swift to read back an account number. The issue is that it reads it as two thousand one hundred forty-five. I would like it to say 2 1 4 5. Accordinging to cepestral I need to add a space between the letters or add \<say-as interpret-as=\"letters\"\>$varible\</say-as\>. I tried the ssml tag but it didn't work. Is there away from the dialplan to run the ssml or to modify the varible to have spaces? |
| 19:58.48 | Katty | Qwell: i'd like to report a crime. |
| 19:58.56 | Katty | Qwell: someone STOLE A PEN RIGHT OFF MY DESK |
| 19:59.07 | Katty | the nerve. |
| 20:00.18 | talntid | kick them in the chin. :) |
| 20:03.39 | *** join/#asterisk shamelessn00b (~bwahahaha@39.47.154.253) |
| 20:03.43 | shamelessn00b | _Corey_: |
| 20:03.47 | shamelessn00b | DID IT!!!!!!!!!! |
| 20:04.09 | shamelessn00b | Thanks, all you guys for your help :) |
| 20:05.35 | _Corey_ | shamelessn00b: Congrats... :) |
| 20:05.51 | shamelessn00b | let me share the dialplan |
| 20:06.12 | shamelessn00b | its still a very crude form, I'll have to handle DTMF by either features, or AMI |
| 20:06.18 | shamelessn00b | but it works |
| 20:06.53 | shamelessn00b | [TK]D-Fender: your approach was simpler, but I was unable to retreive audio via chan spy |
| 20:12.12 | shamelessn00b | http://pastebin.com/YbTyCNjq |
| 20:24.05 | shamelessn00b | _Corey_: what would be the optimal way of handlign DTMF in your opinion? |
| 20:24.56 | shamelessn00b | also, you were saying that we can bridge multiple conferences with each other, can you share how can we do that? |
| 20:25.33 | shamelessn00b | I might switch to asterisk 10, if I have to use confbridge this extensively |
| 20:31.15 | *** join/#asterisk magnet (~magnet@AMontpellier-651-1-178-204.w86-209.abo.wanadoo.fr) |
| 20:31.22 | magnet | evening. |
| 20:31.37 | talntid | or morning, depending on where in the world you are. |
| 20:35.55 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 20:36.06 | magnet | ah right. |
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| 20:41.43 | ChannelZ | Good non-zone-specific time of day to you. |
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| 21:12.22 | kenshin | http://www.asterisk.org/doxygen/asterisk1.4/astobj_8h-source.html <--- there's something fishy on line 234 |
| 21:12.45 | kenshin | (although that seems from 1.4 it's still there in 10.3.1) |
| 21:13.23 | kenshin | if that gets executed obj is NULL at the end regardless of refcount |
| 21:16.02 | *** join/#asterisk zaf (~zaf@76.72.88.14) |
| 21:21.04 | mjordan | kenshin: that's worth looking into, although not much uses ASTOBJ anymore. Please open an issue in the issue tracker |
| 21:21.19 | mjordan | there are still some things that do, most notably MWI in chan_sip |
| 21:21.46 | kenshin | I'm looking at a (old?) patch that does the exact same thing -- after a reqrite |
| 21:22.01 | kenshin | https://issues.asterisk.org/jira/browse/ASTERISK-3106 <--- |
| 21:22.23 | kenshin | obj should be st to null if we really intend to destroy it, no ??? |
| 21:23.12 | kenshin | (that thing kept me up all night :)) (was trying to use MWI => blablabla
asterisk kept segfaulting |
| 21:23.16 | mjordan | I'm not as familiar with ASTOBJ as I am with astobj2, but I would imagine that we would only want to set (object) to NULL if the destructor was called on it |
| 21:23.33 | mjordan | kenshin: have you tried it in on 1.8? |
| 21:23.43 | kenshin | I'm on 10.3.1 |
| 21:23.46 | mjordan | ah |
| 21:23.50 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 21:23.59 | mjordan | yeah. Please file a bug report, attach a backtrace generated from the core, and reference this conversation |
| 21:24.36 | nny | i am trying to send a call to two cell phones with a challenge/response. Currently asterisk treats the channel as answered (probably due to how the carrier handles the event) and hanging up the second call. What is the ideal way to handle this? |
| 21:24.53 | mjordan | kenshin: thanks for hunting this down and finding it :-) |
| 21:25.11 | kenshin | it's not the parsing (I spent a lot oi time on bug reports on that and it wash;t that) .. had to open old laptop compile manually asterisk with debug and check it
the crash happens after UNREFF mwi is null (if the host name is an IP or dnsmgr is disabled) |
| 21:25.19 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 21:25.39 | kenshin | now I wash checking the output of the preprocessor (chan_sip.i) and I noticed that |
| 21:26.19 | kenshin | I have to create an account to fill a bug report don;t I ? :) |
| 21:27.09 | mjordan | yup |
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| 21:40.48 | kenshin | ok almost done : |
| 21:41.15 | kenshin | is there a way to tell it to put the text verbatim (no pretty stuff -- in the description)? |
| 21:41.40 | Qwell | {noformat}stuffgoeshere{noformat} I think |
| 21:42.51 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ptimzzlqovyukksn) |
| 21:45.41 | kenshin | thanks Qwell |
| 21:45.44 | kenshin | https://issues.asterisk.org/jira/browse/ASTERISK-19827 <-- there |
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| 23:41.16 | ddickenson | trying to upgrade from 1.6.2.7 to asterisk 10 (or even 1.8.x) and its as if the newer version of "realtime" is ignoring my contexts. Using the same Mysql database I can get phones to register and even receive calls but when I call out it thinks it should be looking in the "default" context which I don't use. |
| 23:41.42 | ddickenson | Anyone have ideas? I can't seem to find documentation with a google search on anything but 1.6.x setups for realtime |
| 23:45.44 | ddickenson | anyone? |
| 23:46.09 | ddickenson | a little kick in the right direction, or anything? |
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