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| 00:21.10 | drmessano | I thought 1.6.2 didnt need dahdi for conferences? |
| 00:25.40 | De_Mon | http://lists.digium.com/pipermail/asterisk-commits/2006-August/006569.html |
| 00:25.57 | De_Mon | that compensate thing has been around for some time it seems |
| 00:26.26 | denon | drmessano: 1.6.1 has res_timing_dahdi |
| 00:26.31 | denon | drmessano: that's all ya need |
| 00:26.39 | denon | and res_timing_pthread |
| 00:26.45 | file | nyet |
| 00:26.49 | drmessano | Uh.. |
| 00:26.54 | denon | but meetme needs dahdi |
| 00:26.56 | denon | for now |
| 00:26.57 | file | 1.6.3 has a new application, ConfBridge, which does not use dahdi for timing/mixing |
| 00:27.06 | drmessano | ConfBridge didn |
| 00:27.10 | denon | file: oh, that's done? coo |
| 00:27.12 | drmessano | ConfBridge didnt make it to 1.6.2 |
| 00:27.22 | drmessano | ? |
| 00:27.24 | denon | I'm so far behind |
| 00:27.26 | file | it did not. |
| 00:27.30 | drmessano | I see |
| 00:27.36 | drmessano | Thats what I was referring to |
| 00:27.46 | drmessano | Hmm |
| 00:29.07 | drmessano | So its being removed from Beta1 then? |
| 00:29.20 | file | I don't understand the question |
| 00:29.43 | drmessano | I know someone who installed 1.6.2beta1 and has app_confbridge |
| 00:29.53 | file | it wasn't in there. |
| 00:29.58 | drmessano | ok |
| 00:30.16 | file | I just merged it into the tree a short time ago |
| 00:31.32 | drmessano | What is the app_confbridge in 1.6.2 beta 1? Placeholder? |
| 00:32.09 | file | oh gah |
| 00:32.19 | file | I had an off by one error in my head |
| 00:32.26 | file | it is in 1.6.2 |
| 00:32.32 | file | I forgot we started with 1.6.0 |
| 00:33.14 | file | it's there, it should work, feedback is welcome and wanted |
| 00:33.19 | drmessano | From what I can see, meetme and confbridge are included |
| 00:33.33 | file | as it is a user of the new foundation API for bridging channels internally |
| 00:33.34 | drmessano | Im assuming until meetme is phased out? |
| 00:33.43 | file | meetme will stay around, no reason to get rid of it |
| 00:33.53 | file | and the feature set isn't equal yet |
| 00:33.56 | drmessano | I see |
| 00:34.28 | drmessano | I'm assuming confbridge doesnt mix down the audio, correct? |
| 00:35.10 | file | mix down the audio? |
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| 00:44.04 | drmessano | IIRC, meetme resamples the audio to 8khz due to the limitations of Zaptel and perhaps Dahdi as well (unless something has changed) |
| 00:44.42 | drmessano | Will that no longer be the case confbridge? |
| 00:44.46 | drmessano | with* |
| 00:44.53 | file | the current module that mixes the audio is the same, but I have a new one that supports both if the codec is available |
| 00:45.23 | drmessano | I see |
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| 00:46.37 | drmessano | So the mixing process is the same, just done outside of Dahdi now |
| 00:46.50 | file | it can be |
| 00:46.52 | file | it's pluggable |
| 00:47.21 | file | so that we can easily drop in different methods of doing bridging/mixing |
| 00:48.27 | drmessano | I know the 2 current limitation with "conferencing" in Asterisk are (1) Zaptel/Dahdi (which is going away) and (2) Quality due to the resampling.. Was hoping 2 would have died with 1 |
| 00:49.14 | file | it has. |
| 00:49.14 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 00:49.57 | file | but it's a bit complex because if you want to have both 8kHz and 16kHz channels in the same conference bridge you have to translate |
| 00:50.03 | file | and the module responsible for that isn't built by default |
| 00:50.07 | drmessano | ok |
| 00:50.15 | file | because we can't include the libresample source code, it is distributed separately |
| 00:50.57 | drmessano | Drop in and /configure, like ilbc? |
| 00:51.49 | file | not drop in, build and install and ./configure |
| 00:51.49 | drmessano | Easy enough |
| 00:52.09 | drmessano | Well, I guess I will plan to play with it then |
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| 00:59.27 | jaytee | this is all too confusing and getting so complex! I've gotta find a job working the drive-thru before my head 'splodes! |
| 01:00.05 | drmessano | Yeah |
| 01:00.33 | drmessano | rm app_meetme && mv app_meetme_better app_meetme |
| 01:02.31 | file | DHW time. |
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| 01:44.48 | s14ck | anybody knows how can I pass parameters to the kernel module like wcte12xp? |
| 01:48.31 | mnicholson_ | s14ck: i think 'modprobe wcte12xp param1=val param2=val' will work |
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| 01:54.10 | s14ck | mnicholson, yes but, i wanna do it with init script, do you know how can I do? |
| 01:55.06 | s14ck | mnicholson, yes but, i wanna do it with init script, do you know how can I do? |
| 01:56.56 | jaytee | is there an echo in here? |
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| 02:07.35 | s14ck | mnicholson, yes but, i wanna do it with init script, do you know how can I do? |
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| 02:13.55 | mnicholson_ | s14ck: hmm, im not familiar with zaptel's init script |
| 02:14.12 | mnicholson_ | there may be something in /etc/defaults/dahdi or something |
| 02:14.28 | mnicholson_ | or /etc/sysconfig/dahdi on redhat |
| 02:16.00 | s14ck | mnicholson_, I'm try it with dahdi compiled on debian |
| 02:16.48 | mnicholson_ | s14ck: check /etc/default/dahdi |
| 02:17.02 | mnicholson_ | there may be a way to set module options in there |
| 02:18.01 | s14ck | mnicholson, not found... |
| 02:18.08 | s14ck | mnicholson_, not found... |
| 02:18.28 | mnicholson_ | im not sure then |
| 02:18.30 | naxxfish | /etc/modprobe.d/ <-- look in there |
| 02:18.37 | mnicholson_ | good call |
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| 02:28.15 | Micc | what does asterisk look for to enable the odbc support? I've installed unixodbc, not configured but it is installed. Do I need to tell it where it is? make menuselect always shows the odbc options disabled. |
| 02:29.52 | jaytee | did you install unixODBC-devel? |
| 02:33.42 | Micc | yes. |
| 02:33.50 | Micc | with yum anyways. |
| 02:33.56 | Micc | Maybe I should get the source and build/install it. |
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| 02:35.11 | jaytee | if you install unixODBC-devel, that IS the source |
| 02:35.38 | jaytee | what distro and what version of * are you running? |
| 02:36.39 | Micc | centos 5.2 |
| 02:36.58 | Micc | maybe its 5.1 |
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| 02:37.30 | jaytee | use this as a rough guide, even if you're using Asterisk 1.6, just if you ARE running 1.6 replace the section for zaptel and install DAHDI instead. |
| 02:37.32 | jaytee | http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 02:37.44 | mnicholson_ | unixodbc-devel is not the source, it's only the headers and libs, but it should be enough to build with odbc support |
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| 02:38.04 | mnicholson_ | Micc: did you ./configure after you installed unixodbc-devel? |
| 02:38.10 | Micc | I'll just get the source for unixodbc and reinstall. |
| 02:38.11 | Micc | yeah. |
| 02:38.24 | jaytee | make sure you install all the dependencies in the section titled Quick method and you should get unixODBC to be enabled when you run make menuselect |
| 02:38.29 | Micc | But I noticed I can't find the normal dir in /usr/local/unixODBC |
| 02:38.56 | jaytee | and if you've already tried to do it once then you need to run a make clean first before you rerun ./configure |
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| 02:39.07 | mnicholson_ | Micc: if you installed it from a package it won't be in /usr/local |
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| 02:51.25 | Micc | ./configure | grep -i odbc shows checking for SQLConnect in -lodbc... yes |
| 02:51.38 | Micc | but still make menuselect shows cdr_odbc disabled |
| 02:55.18 | jaytee | yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel |
| 02:56.46 | Micc | I didn't have ltdl |
| 02:57.14 | Micc | that was it |
| 02:57.19 | jaytee | then install those packages and then run make clean and then ./configure and then make menuselect |
| 02:57.35 | Micc | thanks, that worked. |
| 02:57.41 | jaytee | you're welcome |
| 02:57.52 | jaytee | are you using mysql? |
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| 03:01.06 | Micc | no, pgsql |
| 03:01.22 | jaytee | the book covers setting that up |
| 03:01.32 | jaytee | postGRE SQL |
| 03:01.42 | jaytee | ~book |
| 03:01.42 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 03:44.57 | cool^tom | Hi |
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| 03:46.46 | s14ck | i have this error: ERROR[8658]: chan_dahdi.c:9991 mkintf: Signalling requested on channel 33 is MFC/R2 but line is in ISDN PRI signalling |
| 03:47.00 | s14ck | anybody knows about how fix it? |
| 03:48.23 | Kyosh | shut it off? |
| 03:49.43 | s14ck | ? |
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| 03:53.53 | jaytee | s14ck, your signalling in chan_dahdi.conf is wrong |
| 04:02.22 | cool^tom | What distro of Linux would be best for a production Asterisk Box? Would it be better to go with a Distro of astrisk like Asterisk NOW!, TrixBox or Set up astrisk from scratch? |
| 04:03.02 | [TK]D-Fender | cool^tom: CleaRLY FROM SCRATCH SO YOU HAVE CONTROL |
| 04:18.03 | s14ck | jaytee, but the signalling=mfcr2. |
| 04:19.38 | jaytee | s14ck, then something else in your configs must be messed up |
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| 04:37.57 | cool^tom | [TK]D-Fender: Thanks. |
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| 06:31.14 | miloux | from main/dns.c: |
| 06:31.30 | miloux | brief Lookup record in DNS |
| 06:31.31 | miloux | note Asterisk DNS is synchronus at this time. This means that if your DNS does |
| 06:31.31 | miloux | not work properly, Asterisk might not start properly or a channel may lock. |
| 06:31.31 | miloux | <PROTECTED> |
| 06:31.47 | miloux | anyone knows if this is going to change soon? |
| 06:32.39 | bobsaccamano | how do i send alert info with a SIP invite in asterisk? |
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| 06:45.45 | bobsaccamano | anybody? |
| 06:45.50 | aiksa[LV] | miloux: looks pretty scary. I have never really looked into that. |
| 06:46.07 | aiksa[LV] | bobsaccamano: alert like a specific alert tone for the incomming call? |
| 06:47.20 | aiksa[LV] | bobsaccamano: I am not sure if this is what you want, but snom wiki has this: http://asterisk.snom.com/index.php/Asterisk_1.4/Ringtone_Configuration |
| 06:50.07 | bobsaccamano | aiksa[LV], yeah..alertinfo is used for distinctive ringing |
| 06:56.03 | aiksa[LV] | bobsaccamano: then i suppose that link has it |
| 06:57.57 | bobsaccamano | aiksa[LV], it isnt working ;( |
| 06:58.15 | bobsaccamano | aiksa[LV], im not getting any Alert-Info in the SIP Header |
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| 07:00.30 | aiksa[LV] | bobsaccamano: then sorry - I havent been testing that. |
| 07:02.18 | aiksa[LV] | bobsaccamano: you mean that SIPAddHeader doesnt add a header to message at all? |
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| 07:05.05 | aiksa[LV] | bobsaccamano: when checking the SIP data flow with some snifing tool you dont see that header in outbound communication? (Asterisk to device?) |
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| 07:08.01 | bobsaccamano | aiksa[LV], no |
| 07:11.40 | aiksa[LV] | strange |
| 07:19.35 | bobsaccamano | aiksa[LV], exten=>4444,n,SIPAddHeader("Alert-Info: R0") |
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| 07:25.41 | aiksa[LV] | can you see that message (setting SIP header action) in verbose output on asterisk CLI |
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| 08:09.32 | trymi1 | hi |
| 08:09.37 | trymi1 | everybody there? |
| 08:12.02 | czindy | Hello! I'm trying to install libpri, and it has many errors during make: http://pastebin.com/dadd6cb9 I tried to search under google but no answer. Could somebody help please? Thank you. |
| 08:14.34 | trymi1 | hi |
| 08:14.38 | trymi1 | if I can |
| 08:14.39 | trymi1 | I will |
| 08:14.40 | trymi1 | wait |
| 08:15.42 | czindy | Hi, thank you. |
| 08:17.45 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:19.41 | trymi1 | where is error? |
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| 08:21.09 | czindy | I pasted it to the link |
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| 08:30.16 | tzafrir_laptop | czindy, that's not an error |
| 08:31.50 | czindy | So do you mean after this I can do make install and it will be good? |
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| 08:56.49 | aiksa[LV] | czindy: as tzafrir_laptop those are no errors just build messages |
| 08:57.19 | aiksa[LV] | you can try to "make install" though noone would gurantee that there wont be any errors. |
| 08:57.56 | aiksa[LV] | and judging from your_problem_ I suspect that you might be better off with binary packages for your distribution |
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| 09:07.17 | czindy | aiksa[LV]: could you check please I pasted the make install output: http://pastebin.com/d60dba1ae, do you think i is ok? |
| 09:07.55 | c0rnoTa | hello guys |
| 09:09.16 | c0rnoTa | i'v got "ERROR[29183] astobj2.c: user_data is NULL" and my chan_sip.so goes down with asterisk service. why it could be? i'm using asterisk 1.6.1.0-rc2 |
| 09:21.19 | tamiel | c0rnoTa: post a bug report with core dump ? |
| 09:23.30 | c0rnoTa | i'll do it later. Can you tell me sortly, how to make coredump? |
| 09:24.34 | tamiel | just run asterisk with -g option and try to reproduce your problem |
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| 09:24.53 | c0rnoTa | thx |
| 09:25.15 | tamiel | and after this run gdb with core dump and backtrace it :) |
| 09:27.02 | c0rnoTa | i'll try :) thx |
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| 09:35.48 | tzafrir_laptop | czindy, yes, it's OK |
| 09:35.59 | tzafrir_laptop | czindy, hint: make is very noisy on error |
| 09:36.16 | tzafrir_laptop | try: make nosuchtarget |
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| 09:46.56 | freh | Can I use asterisk 1.4.24 and dahdi to configure a ISDN BRI card? (Digium B410P) |
| 09:47.06 | c0rnoTa | O_o new menuconfig gui in asterisk 1.6.1!! Wow |
| 09:47.23 | c0rnoTa | freh, yes, you can |
| 09:47.57 | freh | It's weird, I can only get it to work with 1.6.0.6 but it's not stable. |
| 09:48.24 | freh | and on 1.4.24 I get "unknown signalling method: bri_cpe_ptmp |
| 09:48.43 | Frogzoo | asterisk gui sounds like the dark side |
| 09:48.44 | tzafrir_laptop | freh, no. you need 1.6.x |
| 09:50.02 | freh | 1.6.x is not stable I assume? |
| 09:50.03 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb116-14-131-43.singnet.com.sg) |
| 09:50.26 | yidiyuehan | hi, everyone, |
| 09:50.33 | freh | I need a stable release because it's for use in a production environment |
| 09:50.56 | mort_gib | freh: I'm testing 1.6.X with that card shortly |
| 09:51.05 | yidiyuehan | kindly check regarding to BLF, do i still need to reboot ip phones once i rebooted my pbx server for asterisk 1.4? or 1.6? |
| 09:51.43 | freh | mort_gib: I can confirm it works, But it's not running stable yet. |
| 09:51.57 | yidiyuehan | currently i am testing with asterisk 1.2.31, it seems that I still need to reboot phones once server rebooted. |
| 09:52.11 | mort_gib | freh: Yeah I know it works, I had the exact same issue as you "unknown signalling..." |
| 09:52.42 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-ff72bd7a4b7fe825) |
| 09:54.07 | freh | mort_gib: I was happy I could get it to work with 1.6 but it crashed several times. I was forwarding incoming calls to a sip-phone that wasn't connected and when I got an incoming call it just crashed. |
| 09:54.16 | freh | which of course should not happen |
| 09:54.36 | mort_gib | freh: Huh, I can't say I have seen that... -Yet! |
| 09:54.43 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 09:55.46 | freh | mort_gib: I'm going to try configuring it with the mISDN driver |
| 09:55.54 | freh | see what happens |
| 09:56.12 | mort_gib | freh: I would be interested in the result! |
| 09:56.54 | mort_gib | freh: I have an installation that has been causing aggro for a while |
| 09:57.28 | mort_gib | freh: The have infrequent dropped calls, using a Sangoma A500 card, Telco is useless in troubleshooting |
| 09:57.50 | mort_gib | freh: So decided to try another card, nice to know the different options anyway |
| 09:58.29 | mort_gib | freh: Banged my head against DAHDI not supporting the Digium card in 1.4, so went for 1.6 |
| 09:59.24 | freh | mort_gib: Same here. It's actually my first setup. Dahdi documentation is lacking. |
| 10:00.02 | freh | mort_gib: 1.6 doesn't work for me. I really need a stable release |
| 10:00.37 | mort_gib | freh: Oh, I have 200+ users on Asterisk, and TWO installations that drops calls, I can't find out why and it's driving me f*****g nuts |
| 10:01.25 | mort_gib | freh: I only get the dropped calls on BRI (BT style) installations, BRI in Spain and analogue installs are rock solid |
| 10:02.21 | mort_gib | freh: May Telcos BURN IN HELL! |
| 10:02.39 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 10:02.59 | freh | mort_gib: hehe, I second that |
| 10:03.19 | hrmphh | anyone know what provisioning/switch changes would be required on the Level3 side if i wanted to remove the Adtran IAD box between the T1 jack and my PBX? basically, we no longer use the data portion of the IVAD circuit and the Adtran is dead |
| 10:04.43 | hrmphh | these idiots at L3 off-hours support have no clue |
| 10:06.11 | mort_gib | freh: Telco keeps going on about signalling being off on the second incoming call, indicating that they don't understand that we can receive any calls at all |
| 10:06.35 | mort_gib | freh: Fact is this is a 25 user install and they are on the phone constantly... |
| 10:07.21 | mort_gib | freh: I get "normal hangup, requested by remote" as the hangup cause when the calls drop |
| 10:07.53 | mort_gib | freh: Asking Telco for debug info they told me "We are not obliged to provide you with any information what so ever" -Nice |
| 10:08.40 | Dovid | freh: And you are not obligated to be their client ;) |
| 10:09.17 | *** join/#asterisk thelordmortis (n=lordmort@119.11.17.191) |
| 10:09.18 | mort_gib | Dovid: As a matter of fact I am... There is only that same Telco |
| 10:09.28 | Dovid | that sux. |
| 10:09.30 | Dovid | go to sip |
| 10:10.03 | mort_gib | Dovid: Not an option, I still need local numbers, which are not offered.... |
| 10:10.31 | c0rnoTa | another question. I'm always getting 'Avoiding initial deadlock for channel' after sip users dials throu dahdi to PSTN |
| 10:10.59 | mort_gib | No matter WHAT I do I HAVE to contract the local Telco (Gibdisconnect) |
| 10:12.10 | mort_gib | Hmm, maybe that coffee was a bit too strong :-) |
| 10:13.00 | c0rnoTa | so question is : is codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory can be wired with deadlocks? |
| 10:15.04 | mort_gib | c0rnoTa: Check the codecs you are using! |
| 10:16.50 | c0rnoTa | mort_gib, some of them have more then 1000ms to retranslate :( why it could be? |
| 10:17.30 | mort_gib | c0rnoTa: Dunno, but make sure you enable the transcode in /etc/dahdi/modulesXX |
| 10:20.19 | c0rnoTa | mort_gib, i haven't pci transcode card and /etc/dahdi/modules have only my openvox module uncommented. nothing else |
| 10:29.25 | c0rnoTa | i'm still could not understand. why i must load transcode module? i haven't it :) |
| 10:30.24 | c0rnoTa | so after modprobe wctc4xxp, i have /dev/dahdi/transcode, many thx to mort_gib! |
| 10:30.55 | mort_gib | c0rnoTa: Np |
| 10:31.00 | bobsaccamano | guys..does anyone know how to configure distinctive ringing for sip channels |
| 10:31.00 | bobsaccamano | ? |
| 10:31.07 | bobsaccamano | Ive tried a lot of things |
| 10:33.59 | nfi|ermes | i'd like to be notified when a new message is leaved in my voicemail |
| 10:34.42 | nfi|ermes | how can i dot it ? |
| 10:35.31 | nfi|ermes | i tried using externnotify , but it notify me also when someone check the voicemail (i dont want it) |
| 10:36.24 | mort_gib | nfi|ermes: What phones are you using?? |
| 10:55.40 | c0rnoTa | Guys, it could be interesting for you. I had deadlocks of channel.c and some errors of /dev/dahdi/transcode. But after i have loaded wctc4xxp (driver for PCI TRANSCODE CARD) deadlocks has gone. But i haven't PCI TRANSCODE card - it's very important. |
| 10:56.23 | mort_gib | c0rnoTa: Yeah, well I though it odd too, but it worked for me :-) |
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| 11:09.50 | lesouvage | x |
| 11:11.18 | Dovid | wow. that was a nice split |
| 11:12.59 | Stese | looks like I missed it |
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| 11:20.39 | defswork | just got asterisk installation go ahead |
| 11:20.49 | defswork | 20 exts E1 |
| 11:26.08 | Stese | lucky you |
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| 11:47.37 | *** join/#asterisk bilke (n=bilke@93.86.192.73) |
| 11:47.50 | bilke | hi people ;) |
| 11:47.55 | bilke | i'm new in asterisk |
| 11:48.01 | bilke | and I have one quick question, if someone knows |
| 11:48.07 | bilke | i need to create system where voluntears will login to one web page, and as long as they are logged in, asterisk will forward incoming calls to them |
| 11:48.14 | bilke | is it possible to do out of box on asterisk ? |
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| 11:54.41 | beek | bilke: The components to make that happen are there "out of the box." There is no application, per se, "out of the box." |
| 11:56.00 | bilke | I see, thanks! |
| 11:56.39 | bobsaccamano | how do i reset register expiry timer in asterisk |
| 11:56.40 | bobsaccamano | ? |
| 11:56.43 | bobsaccamano | for SIP Channels |
| 11:56.47 | *** join/#asterisk _theHub (n=_theHub@ool-ad02788c.dyn.optonline.net) |
| 11:56.48 | bilke | beek: is it more like programming to make it works or integration job ? |
| 11:57.09 | beek | bilke: Integration. |
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| 11:59.18 | beek | bilke: You didn't give any details about exactly you wanted this to work but I'd assume you'd use agents in a queue and use the AMI to tell Asterisk to log them in via a web page. |
| 11:59.56 | beek | That's one way to do it. There are a multititude of others depending upon your needs and desires. |
| 12:02.59 | jaytee | morning beek |
| 12:04.02 | beek | morning jaytee |
| 12:04.33 | beek | jaytee: You know... I've often told people that "I work in a zoo..." and it just dawned on me... ;-) |
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| 12:06.00 | unixdawg | ok who knows how to hard reset a aastra |
| 12:06.00 | Corydon76-dig | beek: I'd consider configuring Asterisk to be more like programming |
| 12:06.35 | unixdawg | other then the gui |
| 12:07.11 | Corydon76-dig | Even with the GUI... you still have to understand how to assemble a dialplan |
| 12:07.17 | beek | Corydon76-dig: It depends on what he already has available to him. He gave few details. |
| 12:08.36 | beek | Corydon76-dig: I guess after thirty years of programming writing a dialplan doesn't seem much like programming to me -- it's more like scripting. |
| 12:08.50 | jaytee | hehehe |
| 12:09.04 | destructure | scripting is programming, heh |
| 12:09.18 | Corydon76-dig | True enough, but for sysadmin types, anything of the sort is programming |
| 12:09.41 | destructure | I think there is an implication of difficulty of programming over configuration though. |
| 12:10.13 | destructure | I think it comes down to logic though. If it has decisions in it, it's programming, if it's just state, it's configuration |
| 12:10.15 | beek | Corydon76-dig: it's all a matter of your definition of programming. If he does a little research he'll likely find out whether it's "integration" or "programming" to him. |
| 12:10.24 | jaytee | dialplan is "kinda" like a basic program more than a script, a boatload of conditionals sprinkled in here and there and every context acting like a switch or select case |
| 12:10.43 | beek | jaytee: Unless you're using AEL, in which its clearly programming. |
| 12:10.44 | beek | ducks |
| 12:10.59 | Corydon76-dig | When I worked at a reseller, the sysadmin regularly snipped bits of dialplan from other sources and wondered why it didn't work. I had to point out to him where he failed to initialize variables -- he needed to actually read what he snipped and understand it. |
| 12:11.05 | beek | In either case, it's a boatload of fun. |
| 12:11.06 | jaytee | I've managed to avoid having to use AEL for now |
| 12:11.30 | destructure | I'd rather use an external language than ael |
| 12:11.39 | beek | And I'm not being sarcastic.... this is the most fun I've had with computers for quite some time. |
| 12:11.50 | Corydon76-dig | Glad you enjoy it |
| 12:12.51 | bobsaccamano | how do i get asterisk to send a 423 SIP response for a register interval smaller than min expiry in asterisk |
| 12:12.52 | bobsaccamano | ? |
| 12:13.14 | Corydon76-dig | bobsaccamano: dust off your C programming skills |
| 12:13.24 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 12:13.25 | destructure | I guess jumping to extensions is technically logic, but it feels more like configuration, unless you have to put conditionals (e.g. gotoif) in there, in which case, definitely programming, and pushes the dialplan too far |
| 12:13.34 | bobsaccamano | Corydon76-dig, eh? |
| 12:13.59 | bobsaccamano | Corydon76-dig, guess you were too succinct |
| 12:14.07 | Corydon76-dig | bobsaccamano: There is no 423 SIP response in Asterisk currently |
| 12:14.12 | bobsaccamano | ohk |
| 12:14.18 | bobsaccamano | Corydon76-dig, thanks |
| 12:14.30 | jaytee | time to head into the office, bbl |
| 12:15.08 | bobsaccamano | Corydon76-dig, asterisk 1.6? |
| 12:15.19 | Corydon76-dig | bobsaccamano: handles it, but never sends it |
| 12:16.49 | Corydon76-dig | I'm out for the next 3 hours (commute time) |
| 12:16.55 | beek | destructure: You're putting much too fine an edge on it. Call it what you will. You'll either be successful or run await foaming at the mouth. |
| 12:17.00 | beek | Corydon76-dig: That's one helluva commute. |
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| 12:19.20 | destructure | beek: I don't think it's too fine. Configuration vs programming is an interesting dichotomy. My point is, it comes down to logic, if you can avoid that in the dialplan, it's just configuration, and therefore should avoid mouthfoaming. |
| 12:19.50 | bobsaccamano | Corydon76-dig, is there some way i can see the official docs..? Feature set or specs or something like it |
| 12:20.15 | destructure | of course, it's hard to avoid, ergo mouthfoaming eventually, particularly if one doesn't understand the dichotomy. "oh wait I thought I was just configuring". |
| 12:20.48 | beek | destructure: <---- author of "The metaphysics of programming" |
| 12:20.48 | unixdawg | anyone here good with aastra 9133 i .. |
| 12:21.02 | beek | You need to write that book! |
| 12:21.24 | beek | "I branch, therefore I am." |
| 12:21.40 | destructure | sounds good |
| 12:22.16 | beek | Next argument: Did "Pluto" get screwed when it was demoted from full planethood? |
| 12:22.31 | *** join/#asterisk doke (n=me@unaffiliated/emrah) |
| 12:22.42 | doke | hi everybody |
| 12:22.50 | doke | I have a weird question. |
| 12:23.00 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:23.02 | coppice | screwing is about parenthood, not planethood |
| 12:23.05 | doke | I just upgraded my Asterisk to asteirsk 1.6 |
| 12:23.28 | doke | and now the h extension does not work as it used to do in 1.4 |
| 12:23.37 | doke | I'm calling the h extension within a macro |
| 12:23.46 | doke | same config works on a 1.4 |
| 12:23.59 | doke | is there any switch to enable or disable h extension? |
| 12:24.15 | doke | any var to set? |
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| 12:26.22 | [TK]D-Fender | doke: "calling" the "h" exten? Do tell... |
| 12:26.41 | [TK]D-Fender | doke: and no, there is nothing to set |
| 12:27.05 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 12:27.16 | doke | [TK]D-Fender: didnt get what you mean |
| 12:27.24 | doke | what I mean is |
| 12:27.41 | doke | I have a notification script to send me a text msg (sms) and an email when I miss a call |
| 12:27.43 | [TK]D-Fender | doke: Pastebin your dialplan and an example of your problem |
| 12:28.11 | [TK]D-Fender | ~pb |
| 12:28.12 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 12:28.33 | doke | [TK]D-Fender: plz listen, there is nothing to do with pastebin |
| 12:28.41 | doke | I'm not new to asterisk |
| 12:28.47 | doke | maybe new to asterisk 1.6 though :) |
| 12:28.53 | doke | well my problem is |
| 12:30.17 | doke | I call en exten that runs a macro. The macro is a simple macro to dial my sip devices. The macro incorporates an h extension to notify me in case of a missed clal. |
| 12:30.25 | doke | this setup works perfectly fine on asterisk 1.4 |
| 12:30.33 | doke | however it does not on asterisk 1.6 |
| 12:30.43 | doke | do you know if there is any change that can justify this? |
| 12:30.50 | [TK]D-Fender | doke: "h" should not get called on a "missed call" |
| 12:31.02 | russellb | doke: please report it on http://bugs.digium.com |
| 12:31.20 | doke | thx russellb |
| 12:31.28 | doke | thx [TK]D-Fender |
| 12:32.01 | doke | I'll make some testing before reporting the bug so I can get a bit more details on when this occure... |
| 12:32.20 | [TK]D-Fender | doke: Pastebin your code so we can see if there is something else wrong with it. |
| 12:32.53 | Pan3D | morning * people |
| 12:33.11 | bobsaccamano | can anyone tell me where im going wrong here: http://pastebin.com/m3efa21b9 Im trying to get asterisk to send alert-info in an incoming SIP Invite |
| 12:34.09 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:34.27 | [TK]D-Fender | bobsaccamano: You prep the header BEFOre you dial your device |
| 12:35.15 | [TK]D-Fender | bobsaccamano: And generally you should have that in quotes |
| 12:35.24 | [TK]D-Fender | shouldn't* |
| 12:35.47 | hrmphh | these idiots at L3 off-hours support have no clue |
| 12:36.19 | hrmphh | anyone know what provisioning/switch changes would be required on the Level3 side if i wanted to remove the Adtran IAD box between the T1 jack and my PBX? basically, we no longer use the data portion of the IVAD circuit and the Adtran is dead |
| 12:36.27 | bobsaccamano | [TK]D-Fender, thanks..wil set that right |
| 12:37.24 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-203.gibnet.gi) |
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| 12:39.15 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:39.15 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 12:41.26 | doke | [TK]D-Fender: it is a bug |
| 12:41.35 | doke | it works if it is out of the macro |
| 12:41.46 | leif[mobile] | what is a bug |
| 12:41.54 | [TK]D-Fender | doke: Other extens than "s" don't belong in a macro anyway |
| 12:42.04 | leif[mobile] | Macros are finicky creatures and have their own sets of rules |
| 12:42.47 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 12:44.25 | doke | [TK]D-Fender: you should core show application macro |
| 12:45.13 | doke | :) |
| 12:45.15 | [TK]D-Fender | doke: and...? |
| 12:45.19 | leif[mobile] | doke: again... What is it you think is a bug? |
| 12:45.24 | doke | sorry leif[mobile] |
| 12:45.44 | [TK]D-Fender | [08:30]<doke>I call en exten that runs a macro. The macro is a simple macro to dial my sip devices. The macro incorporates an h extension to notify me in case of a missed clal. |
| 12:45.46 | [TK]D-Fender | [08:30]<doke>this setup works perfectly fine on asterisk 1.4 |
| 12:45.47 | [TK]D-Fender | [08:30]<doke>however it does not on asterisk 1.6 |
| 12:45.49 | [TK]D-Fender | leif[mobile]: ^ |
| 12:46.10 | [TK]D-Fender | leif[mobile]: And no... we have not seen th issue or anything related to it |
| 12:46.21 | doke | on asterisk 1.4 I have a piece of dialplan that works. It is basically a macro that incorporates an h extension to call a notification script |
| 12:46.30 | doke | the same script on 1.6 does not work |
| 12:46.38 | doke | I checked everything many times |
| 12:46.49 | [TK]D-Fender | doke: and have nothing to show us at all it seems. |
| 12:46.59 | [TK]D-Fender | doke: best of luck with that |
| 12:47.07 | doke | when I put the h extension outside the macro, the macro exits properly and everyhing works |
| 12:47.31 | leif[mobile] | ok @ the subway now, times up, back later. TK can update me |
| 12:48.10 | [TK]D-Fender | leif[mobile]: Doubt there will be anything to do taht with |
| 12:50.01 | doke | [TK]D-Fender: did you read core show application macro? See? It is possible to call an h extension within a macro. It is even clearly stated that the h extension will be called in the current macro context. which is not the case in my situation |
| 12:50.40 | [TK]D-Fender | doke: You are not showing us anything, I'm not going to playt he "assume things are as described, even vaguely" game today. |
| 12:51.16 | [TK]D-Fender | doke: No runarounds. Go right ahead and make that "bug report". |
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| 13:03.38 | elred | Hi there. Do you know a way to detect how many FXO channels are available on a zaptel-compatible card ? |
| 13:04.05 | elred | so I can generate the /etc/zaptel.conf and /etc/asterisk/zapata.conf |
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| 13:09.01 | naxxfish | does ztcfg/dahdi_config not do it? |
| 13:09.33 | *** join/#asterisk proppy (n=proppy@lri11-115.lri.fr) |
| 13:10.04 | QuidProQuo | Morning everybody (dependin where you are). Upgraded from 1.4.22 to .24 and now get continous "[Mar 23 12:56:14] ERROR[4503] channel.c: ast_read() called with no recorded file descriptor" when I connect to the console. Calls go through okay, CPU load is alright but load is really high. Any help with this one? |
| 13:10.32 | proppy | Hi, is it possible to use SIPAddHeader("Console/dsp", "Alert-Info: <http://192.168.2.254/taranis/ringtone/woo.wav>") on polycom phones to set a ringtone from the dialplan ? |
| 13:11.12 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 13:11.28 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 13:11.32 | russellb | QuidProQuo: can you hop in #asterisk-bugs ? |
| 13:11.32 | elred | naxxfish : what do you mean ? I am trying to detect the number of available channel to generate the zaptel.conf file. I think ztcfg will refuse to start if there is no /etc/zaptel.conf present |
| 13:11.42 | russellb | QuidProQuo: we need to get that fixed ... |
| 13:12.38 | russellb | QuidProQuo: basically, if you could post it to bugs.digium.com with as much detail about what is going on in the system when you see the messages, that would be great |
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| 13:13.07 | QuidProQuo | Sure, no problem. |
| 13:13.12 | *** join/#asterisk s0lid (n=s0lid@121.1.18.241) |
| 13:13.55 | naxxfish | elred: oh, yeah, sorry >_< |
| 13:14.52 | [TK]D-Fender | proppy: No, go read the admin guide for the header format and you configure them via provisioning. |
| 13:15.27 | [TK]D-Fender | naxxfish: look in dmesg |
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| 13:16.25 | naxxfish | unf, still getting dahdi errors with 1.6.0.6 |
| 13:16.47 | naxxfish | specifically, when compiling: http://pastebin.com/m615b811 |
| 13:16.48 | proppy | [TK]D-Fender: reading http://www.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_SIP_3_1.pdf |
| 13:18.05 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 13:18.54 | [TK]D-Fender | proppy: There is a decent page on the WIKi for this under |
| 13:19.04 | [TK]D-Fender | "polycom paging" |
| 13:19.09 | [TK]D-Fender | proppy: Go look for it |
| 13:19.11 | [TK]D-Fender | ~wikis |
| 13:19.12 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:19.43 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 13:19.49 | [TK]D-Fender | proppy: You set the Alter-Info tag in yoru sip.cfg accordingly and refer to it as they show |
| 13:20.12 | jaytee | Alter-Info? is that like Alter-Ego? :-) |
| 13:20.28 | jaytee | hands [TK]D-Fender a fresh cup of coffee |
| 13:20.35 | proppy | [TK]D-Fender: thanks :) |
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| 13:33.46 | viraptor | I've got an asterisk that starts dropping sip packets / being weird in general when I cross 500 simultaneous calls (8*xeon, barely reaching 5% CPU usage) - is there any typical problem I should look for? |
| 13:33.51 | viraptor | (calling echo all the time) |
| 13:35.45 | viraptor | err... correction - ~600 calls, all sip |
| 13:38.43 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:42.55 | russellb | viraptor: running out of bandwidth? |
| 13:44.04 | naxxfish | right: http://pastebin.com/m67620f47 trying to compile 1.6.0.6, with DAHD Version: 2.1.0.4 |
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| 13:45.33 | destructure | viraptor: what codec? |
| 13:45.53 | naxxfish | ooh, i fixed it |
| 13:46.52 | DavidR2008 | I have a problem, I've started getting these messages all of the sudden: |
| 13:46.53 | DavidR2008 | ERROR[29949]: cdr_addon_mysql.c:148 mysql_log: cdr_mysql: cannot connect to database server mysqlserver1. |
| 13:46.55 | DavidR2008 | nothing's changed and the sql server is our main sql server that lots of other things are using without any problem. How can I troubleshoot this? |
| 13:47.54 | hrmphh | anyone know what provisioning/switch changes would be required on the Level3 side if i wanted to remove the Adtran IAD box between the T1 jack and my PBX? basically, we no longer use the data portion of the IVAD circuit and the Adtran is dead |
| 13:47.58 | hrmphh | ? |
| 13:48.12 | *** part/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
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| 13:48.34 | viraptor | destructure: alaw |
| 13:49.15 | viraptor | russellb: not even close - I'm testing it on a gigabit switch, both source and destination are gigabit too, all local network |
| 13:49.29 | russellb | ah. |
| 13:50.50 | doke | I have more inputs regarding my bug |
| 13:51.00 | doke | [TK]D-Fender: |
| 13:51.14 | doke | it does go to h extension only when the channel is answered |
| 13:51.19 | doke | not if it is not |
| 13:51.36 | [TK]D-Fender | hrmphh: Odds are its a basic data T1 w/ SIP and the IAD just spits out PRI from it. |
| 13:51.40 | doke | so you still think it is not a but? even when I tell you that the same code works on 1.4? |
| 13:51.50 | doke | bug* |
| 13:51.52 | [TK]D-Fender | doke: I'm not going to comment on what i don't see. |
| 13:51.53 | doke | ;) |
| 13:52.06 | doke | well well |
| 13:52.18 | doke | I have to comment many times on stuff that I can't see |
| 13:52.21 | doke | <PROTECTED> |
| 13:52.24 | hrmphh | tk |
| 13:52.33 | hrmphh | so they would need to make some switch changes right? |
| 13:52.35 | [TK]D-Fender | doke: convenient |
| 13:52.39 | hrmphh | for me to go direct from jack -> pbx |
| 13:52.44 | doke | 7query [TK]D-Fender |
| 13:52.47 | hrmphh | so only PRI comes over it |
| 13:53.07 | [TK]D-Fender | hrmphh: Never said they need to make any changes. |
| 13:53.16 | hrmphh | so you think i can just plug in? |
| 13:53.19 | hrmphh | directly and itd work? |
| 13:53.28 | hrmphh | ISDN PRI? |
| 13:53.31 | [TK]D-Fender | hrmphh: directly into what? |
| 13:53.43 | [TK]D-Fender | hrmphh: depends what kind of circuit it is |
| 13:54.03 | [TK]D-Fender | hrmphh: Could be a partial PRI w/ data. It could be raw T1 + SIP/H.323/IDK |
| 13:54.49 | [TK]D-Fender | hrmphh: You really do need a clue about what it is you've got :) |
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| 13:58.51 | hrmphh | eh tk |
| 13:59.03 | hrmphh | they call it an "IVAD circuit" but dont give specifics |
| 13:59.06 | hrmphh | its broadwing/level3 |
| 13:59.15 | hrmphh | ive been on the horn w/them for hours and their techs have nfc |
| 13:59.23 | hrmphh | trying to get me over to provisioning |
| 13:59.25 | [TK]D-Fender | hrmphh: Oh well |
| 13:59.44 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 14:00.44 | hrmphh | its this beast: http://www.adtran.com/adtranpx/Rooms/DisplayPages/LayoutInitial_webrQS%20_UHJvZHVjdENhdGVnb3J5PWNvbS53ZWJyaWRnZS5lbnRpdHkuRW50aXR5JTVCT0lEJTVCOTVGNTUxNjVEMUUyQTA0NzhCODMxRTVDMzQ0MURCOUIl/NUQlNUQmUHJvZHVjdD1jb20ud2VicmlkZ2UuZW50aXR5LkVudGl0eSU1Qk9JRCU1QkRENzRFMzlDNTFBNkEwNDk5QzExRDMxODFBRjdDMDFFJTVEJTVEJkNvbnRhaW5lcj1jb20ud2VicmlkZ2UuZW50aXR5LkVudGl0eSU1Qk9JRCU1QkEyM0Q1RDMzQTc5MEVBNDdBQzNDRjlFQTc1RDM2ODc3JTVEJTVE/ |
| 14:00.50 | hrmphh | christ thats a longurl |
| 14:04.32 | [TK]D-Fender | hrmphh: Doesn't answer how its configured. |
| 14:04.39 | [TK]D-Fender | hrmphh: Get them to tell you. |
| 14:05.18 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:08.55 | doke | I'd like to submit a bug to mantis but it is necessary to creat an account and to do so I have to copy a captcha... I am blind and therefore can't copy the picture. Can anybody help me? |
| 14:12.43 | Nasra | hi, I am new to asterisk with the help of friend we configured & compiled asterisk ....now it does not want start.....is giving some type of error : it says not connectd.....can any1 tell me what coudl be wrong ?.....thanks |
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| 14:12.55 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:14.32 | [TK]D-Fender | Nasra: apstebin it |
| 14:14.34 | [TK]D-Fender | pastebin* |
| 14:14.36 | [TK]D-Fender | ~pb |
| 14:14.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 14:14.43 | Nasra | okay |
| 14:14.44 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 14:14.45 | [TK]D-Fender | Nasra: "some sort of error" doesn't tell us anything useful |
| 14:14.55 | Nasra | let me go back in |
| 14:18.03 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 14:18.08 | bilke | beek: Thanks a lot!!! |
| 14:18.18 | WeazelON | anyone knows a good place to download FREE music on hold mp3 or wav ? |
| 14:20.40 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:20.44 | [TK]D-Fender | WeazelON: http://freeplaymusic.com/ |
| 14:20.59 | WeazelON | thanks :D |
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| 14:28.07 | *** join/#asterisk moncky (i=moncky@gateway/shell/blinkenshell.org/x-a8a5611898e7e5df) |
| 14:28.10 | moncky | hey all |
| 14:28.17 | *** join/#asterisk gradgrind (n=Gradgrin@216.68.101.253) |
| 14:28.32 | moncky | is there such a thing as a script that will allow call pickup from within a queue? |
| 14:28.54 | *** join/#asterisk QuidProQuo (n=none@99.254.99.244) |
| 14:28.54 | [TK]D-Fender | moncky: nope. |
| 14:30.24 | WeazelON | where do i change the volume db of the freepbx's music on hold ? its too loud |
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| 14:30.43 | [TK]D-Fender | WeazelON: you don't |
| 14:31.11 | [TK]D-Fender | WeazelON: if its the stock boring ones that come with * then they are already normalized and your endpoints are jacked up too high |
| 14:31.49 | WeazelON | but for some wierd reason i hear the same file on different pbx's on different volume rates |
| 14:32.41 | [TK]D-Fender | WeazelON: taht depends on the endpoint involved. |
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| 14:32.46 | [TK]D-Fender | WeazelON: Yours and theirs |
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| 14:36.41 | WeazelON | what do you mean by endpoints ? |
| 14:36.46 | WeazelON | the phones ? |
| 14:37.04 | proppy | In a multiple extension mapped to a single phone scenario, is it possible to display the extension being called in addition to the caller id to the called phone ? |
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| 14:38.35 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:39.01 | [TK]D-Fender | proppy: Add it to the caller-id |
| 14:39.13 | *** join/#asterisk swathanthran (n=user@unaffiliated/shyam-k/x-8459115) |
| 14:39.14 | [TK]D-Fender | WeazelON: Phones, PSTN interfaces, etc |
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| 14:39.33 | [TK]D-Fender | WeazelON: thats 4 devices involved |
| 14:39.49 | [TK]D-Fender | WeazelON: the PBX is right, someone's setup is not |
| 14:40.01 | WeazelON | aight thanks |
| 14:40.29 | proppy | yep I just read about it in http://www.voip-info.org/wiki/view/Setting+Callerid [TK]D-Fender thanks |
| 14:43.51 | nfi|ermes | [TK]D-Fender, i'd like to be notified by a call on my mobile when a new message is leaved in my voicemail |
| 14:43.58 | nfi|ermes | i tried using externnotify , but it notify me also when someone check the voicemail (i dont want it) |
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| 14:44.25 | unixdawg | ok I need a moh.conf file that shows how to stream a mp3 stream for moh |
| 14:45.46 | jaytee | the sample configs have an example of using streaming audio as a source |
| 14:46.50 | unixdawg | the one we have uses a program called streamplayer |
| 14:46.56 | unixdawg | and I dont find the app |
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| 14:48.06 | russellb | streamplayer doesn't actually play an mp3 stream |
| 14:48.10 | russellb | that's something else ... |
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| 14:50.55 | unixdawg | russellb, you have a good howto somewhere I know you do... |
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| 14:51.45 | russellb | I've got nothin' |
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| 15:01.40 | nfi|ermes | i need to execute a perl script from a dialplan extension ! What's the way ? |
| 15:04.11 | [TK]D-Fender | nfi|ermes: "core show application system" |
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| 15:10.01 | nfi|ermes | i used AGI and it looks working |
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| 15:19.35 | *** join/#asterisk CunningPike (n=CunningP@vpn.dnv.org) |
| 15:19.45 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 15:19.50 | ACiDV | Does exist a way to prevent an "agent" (ex. SIP/1234) to receive multiple calls when member of multiple queues ? Ex. SIP/1234 is member of Queue A and B, both queues have waiting callers, when SIP/1234 is available, line 1 and 2 ring at same time, if I pickup call on Queue A, the other return back to the queue B. Must I add something like 'ringinringing' option ? (like similar to ringinuse) |
| 15:21.13 | *** join/#asterisk apeiron (n=apeiron@c-76-124-252-61.hsd1.pa.comcast.net) |
| 15:22.12 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 15:24.07 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 15:31.13 | ayeso | Does anyone here use g729? |
| 15:32.05 | UQlev | ayeso: do you know any free clients supporting it? |
| 15:32.56 | ayeso | My question is has anyone here actually listened to the difference between 729 and 711? I realize they have different MOS scores but I'm curious is anyone here can tell the difference. |
| 15:33.15 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 15:33.25 | [TK]D-Fender | ayeso: Of course, G.729 is a highly compressed codec by comparison. |
| 15:33.43 | [TK]D-Fender | ayeso: at under 1/6th the size do you think it'll be unnoticable? |
| 15:34.12 | ayeso | [TK]D-Fender: right, but have you actually listened to voice over 729? |
| 15:34.21 | UQlev | ayeso: there must be difference because 711 is 64Kbps and 729 is 9 kbps |
| 15:34.21 | [TK]D-Fender | ayeso: Yes |
| 15:34.44 | ayeso | UQlev: I know there is a difference, Im looking for opinions on how much worse it is. |
| 15:35.08 | UQlev | ayeso: 729 is about the same as gsm |
| 15:35.25 | ayeso | [TK]D-Fender: What is your take on 729? Would you use it in a call center? |
| 15:35.30 | [TK]D-Fender | ayeso: Sure |
| 15:35.54 | jasonwoot | ayeso, how much worse? |
| 15:36.47 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 15:37.41 | ayeso | jasonwoot: Yes, 729 has a worse MOS score than 711, but that is just a number as far as i'm concerned I want to know what people think about 729.. do you like it, is it noticeably worse than 711? by worse I am refering to voice quality, |
| 15:38.05 | [TK]D-Fender | ayeso: Its FINE. |
| 15:38.24 | jasonwoot | ayeso, I bit the bullet and licensed it, tried it with zoiper and for the life of me, couldn't tell a difference |
| 15:38.54 | jasonwoot | I'm in a call center, 50% polycom, 50% zoiper, but all >100mbit |
| 15:39.12 | ayeso | jasonwoot: [TK]D-Fender: That's what I wanted to know, thanks. |
| 15:39.17 | *** join/#asterisk QuidProQuo (n=none@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:39.38 | jasonwoot | wanna buy my license? Fender, can I sublet a .729 license? |
| 15:40.40 | Stese | jasonwoot > I very much doubt it |
| 15:40.56 | ayeso | Who licenses it? |
| 15:41.12 | jasonwoot | you would think Microsoft |
| 15:41.27 | [TK]D-Fender | ... |
| 15:41.37 | [TK]D-Fender | has clearly entered CrackWorld |
| 15:41.52 | Stese | ok, now you've lost me... |
| 15:44.50 | ayeso | I guess I don't understand the licensing with g729. Asterisk sites show you have to buy a license through Digium... what if I was running call manager? Would I still have to get a license from digium? |
| 15:45.39 | DavidR2008 | I'm playing around with voicemail. Can I get it to continue in the dialplan after a user hangs up during recording of the vm? |
| 15:46.08 | Stese | DavidR2008 > What would be the purpose? |
| 15:47.32 | [TK]D-Fender | ayeso: Yes |
| 15:47.39 | DavidR2008 | I'm taking baby steps to my goal, but ultimately I'd like to generate an outbound call that notifies another number upon the arrival of a new VM |
| 15:48.22 | *** join/#asterisk Sanjoy (n=sanjoy_r@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 15:48.31 | [TK]D-Fender | DavidR2008: "externnotify" in voicemail.conf is what you want. Read the sameple config |
| 15:48.42 | DavidR2008 | great, thx! |
| 15:48.51 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 15:49.49 | Sanjoy | Hello Everyone - I am Sanjoy Rath - I have attended Asterisk BootCamp at Huntsville |
| 15:50.08 | Sanjoy | If anyone needs any help with Asterisk config - Please let me know |
| 15:50.12 | *** join/#asterisk califus (n=chatzill@210.212.160.101) |
| 15:50.18 | Sanjoy | I wil be glad to help :) |
| 15:51.21 | UQlev | never thought that asterisk is supported from bootcamps, thought that Salvage Army was resposible for it ;) |
| 15:51.50 | Sanjoy | Sorry Asterisk Bootcamp training at Digium |
| 15:52.06 | *** join/#asterisk QuidProQuo (n=none@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:55.38 | proppy | Is there way to extract what matched an extension ? |
| 15:55.47 | proppy | for example if I have _5XX, |
| 15:55.51 | proppy | and I dial 501 |
| 15:55.58 | proppy | can I extract 01 in the dial plan |
| 15:56.51 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:56.55 | *** join/#asterisk digitalirony (i=digitali@207.192.71.175) |
| 15:57.06 | *** join/#asterisk neurosys (n=neurosys@173.9.159.182) |
| 15:57.16 | mort_gib | How do you enable precense in IP 650 phones, when you have hints in place, like what setting is reqiored on the phone?? |
| 15:57.25 | jameswf | So I was thinking anout writing a deal to use google spreadsheets as a CDR.... |
| 15:57.38 | jameswf | *about |
| 15:57.40 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:58.31 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-17edf8adec9acf8a) |
| 16:00.35 | [TK]D-Fender | mort_gib: look for "presence" in sip.cfg |
| 16:01.20 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 16:01.24 | [TK]D-Fender | proppy: ${EXTEN:1} <- Go read CHANNElVARIABLES in your DOCS folder |
| 16:02.01 | mort_gib | [TK]D-Fender: -Not <bw> in 00000000-directory.xml |
| 16:02.27 | proppy | [TK]D-Fender: ah I didn't have the doc package installed |
| 16:02.29 | proppy | thanks |
| 16:02.54 | [TK]D-Fender | mort_gib: that only applies to the first time you boot your phone, because then it laods of the MAX directory, and that set the BW for your contact, but does not enable the overall function. |
| 16:03.04 | [TK]D-Fender | mort_tha has to be enabled in sip.cfg |
| 16:03.13 | mort_gib | [TK]D-Fender: THanks :-) |
| 16:03.37 | *** join/#asterisk UD (n=UD@unaffiliated/underdawg) |
| 16:03.41 | UD | hi |
| 16:05.27 | DavidR2008 | I'm not able to understand how to use externnotify in voicemail.conf here's the line I tried: |
| 16:05.28 | DavidR2008 | 3000 => 1234,Test Users,,,externnotify=/var/lib/asterisk/script/vmnotify |
| 16:05.30 | DavidR2008 | I don't get any errors from asterisk but my program doesn't execute |
| 16:05.30 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 16:06.07 | *** join/#asterisk freh (n=freh@198.0-66-87.adsl-static.isp.belgacom.be) |
| 16:06.15 | [TK]D-Fender | DavidR2008: is not PER BOX |
| 16:06.27 | [TK]D-Fender | DavidR2008: Go read the samples again |
| 16:06.34 | *** join/#asterisk Sanjoy (n=sanjoy_r@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 16:06.34 | DavidR2008 | ok, sorry to bother you |
| 16:07.52 | UD | i am calling broadvoice to cancel my account after 1 day of service |
| 16:08.20 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 16:08.25 | UD | they are the most fraud paranoid people ever |
| 16:08.36 | Stese | ? |
| 16:08.37 | *** join/#asterisk Sanjoy (n=sanjoy_r@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 16:08.40 | hrmphh | UD; yup |
| 16:08.43 | hrmphh | they are a pain in the ass |
| 16:08.44 | UD | i got suspended on day #1 |
| 16:08.55 | hrmphh | altho thankfully we are using them right now for backup |
| 16:08.56 | hrmphh | as our T1 is down |
| 16:08.57 | UD | sent them copies of my credit card twice and they say it got lost |
| 16:08.59 | [TK]D-Fender | UD you are clearly evil |
| 16:09.00 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 16:09.11 | hrmphh | UD; one thing that is pretty sweet tho is that while we only have 1 number through them |
| 16:09.17 | hrmphh | i can get seemingly unlimited calls on it |
| 16:09.20 | hrmphh | same line |
| 16:09.22 | UD | ya |
| 16:09.25 | hrmphh | unlimited simultaneous rather |
| 16:09.29 | UD | thats what i like too |
| 16:09.36 | hrmphh | its pretty sweet for failover |
| 16:09.50 | UD | just now i have to rely on sending my credit card to people |
| 16:09.56 | hrmphh | T1 inbound redirect -> hunt group of (1 x broadvoice, 2 x POTS) |
| 16:09.59 | UD | that just toss it around the office LOL |
| 16:10.05 | hrmphh | yup |
| 16:10.26 | hrmphh | anyone here used a redfone foneBRIDGE2? |
| 16:10.40 | hrmphh | was thinking of getting that for failover to hot standby asterisk box |
| 16:13.08 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 16:13.20 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 16:13.44 | *** join/#asterisk neverblue (n=jezus@unaffiliated/neverblue) |
| 16:13.55 | [TK]D-Fender | hrmphh: craptastic |
| 16:14.00 | neverblue | how do I make a call, in teh asterisk console? |
| 16:14.11 | hrmphh | tk; the redfones suck? |
| 16:14.12 | [TK]D-Fender | hrmphh: Don't aim for cheap proprietary junk. Noone gives a shit about TDMoE |
| 16:14.12 | neverblue | the* |
| 16:14.34 | hrmphh | how would you recommend failing over single T1 in case of asterisk server failure? |
| 16:14.46 | hrmphh | s/recommend failing/recommend automatically failing/ |
| 16:14.56 | hrmphh | hah nice bot |
| 16:15.23 | [TK]D-Fender | hrmphh: jsut get something like an AudioCodes Gateway instead of a T1 card |
| 16:16.50 | hrmphh | which specific model? http://www.audiocodes.com/product-family/cpe-gateways |
| 16:18.03 | freh | Do I need zaptel if I want to use a Digium B410P card? I would use it only in TE mode and with the mISDN driver. It's an ISDN BRI card |
| 16:18.17 | mort_gib | [TK]D-Fender: Worked like a charm, I can sort of see why you like those phones! |
| 16:18.21 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:18.27 | [TK]D-Fender | hrmphh: Mediant 1000/2000 |
| 16:18.42 | [TK]D-Fender | mort_gib: yup, there's a curve, but its quality stuff |
| 16:18.52 | hrmphh | mort; what phones? |
| 16:19.21 | mort_gib | hrmphh: Polycoms, I'm NOT going to say they are THE ONLY PHONES, but nice |
| 16:19.26 | mort_gib | IP 650's |
| 16:20.39 | DavidR2008 | [TK]D-Fender: it's working now. Sorry for the stupid, query. Should anything be displayed in the CLI when the external script executes? |
| 16:21.07 | [TK]D-Fender | DavidR2008: not AFAIK |
| 16:21.14 | DavidR2008 | thx |
| 16:26.49 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 16:26.49 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:27.23 | *** join/#asterisk califus (n=chatzill@210.212.160.101) |
| 16:35.32 | hrmphh | mort; thats all we use |
| 16:35.33 | hrmphh | (polycom) |
| 16:35.40 | hrmphh | ip330s for users, and cant remember the confs |
| 16:35.44 | hrmphh | solid. |
| 16:35.56 | *** join/#asterisk Thiago_Lima (n=tslima@201.13.199.17) |
| 16:37.07 | *** join/#asterisk proppy (n=proppy@lri11-115.lri.fr) |
| 16:37.40 | proppy | Hi, is there a way to query an xml file from a dialplan ? |
| 16:37.55 | russellb | AGI? :-) |
| 16:38.33 | proppy | or a json :) |
| 16:38.40 | proppy | I need a key/value association somewhere |
| 16:38.55 | proppy | to display callerid depending of the extension |
| 16:39.11 | apeiron | How do I determine which context an incoming call is going to as it comes in? |
| 16:39.17 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 16:39.38 | proppy | russellb: ah maybe xmlstartlet you're right |
| 16:41.59 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 16:43.48 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 16:44.04 | BlargMaN00 | has anyone had any experience with BLF and polycoms??? I am trying to get ringing BLF to be different from inuse BLF on the phone itself... anybody?? |
| 16:44.38 | mort_gib | I could do with a bit of help on DAHDI.. I can't make any calls but channels show up... |
| 16:46.45 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:47.20 | mort_gib | I don't even see the incoming call in cli |
| 16:50.29 | *** join/#asterisk PDani (n=pekdanie@adslc3e4e583.fixip.t-online.hu) |
| 16:50.33 | PDani | hi |
| 16:52.54 | PDani | is it a way to livestream out the conversations somehow in asterisk? |
| 16:52.54 | apeiron | Is there a way to log the contexts of incoming calls? Or some other way to find that information? |
| 16:52.54 | PDani | i know, that mixmonitor can save them into i.e. wav |
| 16:52.58 | PDani | but i could'nt find any solution for the livestream problem |
| 16:55.50 | *** join/#asterisk s14ck (i=c82302dc@gateway/web/ajax/mibbit.com/x-fd5f36c108e2f5e7) |
| 16:56.14 | s14ck | anybody knows about this error: "dahdi_r2_on_protocol_error: MFC/R2 protocol error on chan 1: Seize Timeout" |
| 16:58.01 | viraptor | if a user sends a "302 redirect" back to asterisk - can it be somehow connected to the original leg (in the dialplan)? I'd like to be able to bill the redirecting user for that call |
| 16:58.03 | *** join/#asterisk jackson__ (n=jackson@66-188-207-74.dhcp.roch.mn.charter.com) |
| 16:59.26 | moy | s14ck: that means the other end did not gave any response to our call request |
| 16:59.41 | moy | *give |
| 17:00.06 | moy | most of the cases means you should call your telco |
| 17:01.30 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:01.54 | s14ck | moy my telco said: something is wrong in physical level |
| 17:02.22 | s14ck | moy: but the card its ok |
| 17:02.26 | moy | then fix your physical level :) |
| 17:03.01 | moy | what does mfcr2 show channels says? |
| 17:03.02 | apeiron | Is there any way to have asterisk tell me the context an incoming call is on? Or some way to see the context? Anything? |
| 17:03.12 | deadpigeon | can I just issue a dahdi reload when testing echocancel and echotraining settings or do I -have- to reboot? |
| 17:03.21 | apeiron | I'm trying to debug my dialplan. It's saying that an extension isn't found and I want to find out what context the call is coming in on. |
| 17:03.26 | s14ck | moy the latency in the module wcte12xp can do something? |
| 17:04.33 | apeiron | Am I invisible or something? |
| 17:04.41 | deadpigeon | we all feel that way sometimes. |
| 17:04.55 | apeiron | really doesn't think this should be such a difficult thing to do |
| 17:05.00 | deadpigeon | im really not sure myself, apeiron |
| 17:07.13 | [TK]D-Fender | apeiron: depends what the call is coming in on |
| 17:07.29 | apeiron | [TK]D-Fender, It's coming in over SIP. |
| 17:07.44 | [TK]D-Fender | apeiron: enable SIP debug and you'll see |
| 17:07.50 | apeiron | er, no, I don't. |
| 17:07.51 | apeiron | :) |
| 17:07.59 | [TK]D-Fender | apeiron: You will |
| 17:08.00 | apeiron | Either that or 'sip set debug' doesn't enable SIP debug. |
| 17:08.14 | [TK]D-Fender | apeiron: Or at least WE will when you pastebin the complete debug of a call attempt |
| 17:08.27 | apeiron | [TK]D-Fender, Is it not actually specified in plaintext? |
| 17:08.41 | apeiron | [TK]D-Fender, I tried searching for all of the contexts in a log of that and didn't find anything. |
| 17:08.59 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 17:09.50 | [TK]D-Fender | apeiron: You aren't showing us anything, and I know that it will show you so I typically correlate that to an optical deficiency on your part... |
| 17:09.54 | [TK]D-Fender | apeiron: PAStebiN |
| 17:10.36 | apeiron | [TK]D-Fender, le sigh, it shouldn't be this difficult. =\ |
| 17:11.15 | [TK]D-Fender | apeiron: La merde, revielle-toi donc pis nous-montre ton crisse d'affaire :p |
| 17:13.32 | apeiron | hmm. Apparently I turned it off earlier before it wrote the relevant stuff. |
| 17:15.29 | apeiron | Lesson learned today: * does not write all its debug info immediately. |
| 17:16.01 | [TK]D-Fender | apeiron: It'll write it as soon as the invite is auth'd |
| 17:16.21 | [TK]D-Fender | apeiron: which is tpically within a secod or two of the invite |
| 17:16.40 | Stese | Just a thought... isn't the context it starts in part of the SIP trunk config? |
| 17:16.41 | apeiron | [TK]D-Fender, Yes. I was expecting it to be immediate. Took about five seconds here. |
| 17:17.08 | s14ck | moy: see the error http://pastebin.com/m1cea73af |
| 17:17.55 | [TK]D-Fender | Stese: Makes assumptions as to how it comes in... |
| 17:18.33 | moy | s14ck: I know the error, as I said, that means we asked to start a call and we never got a response back from the telco |
| 17:18.40 | Stese | and we know what assumptions makes... just a thought... :P |
| 17:18.41 | moy | CAS Tx >> [SEIZE] 0x00 we transmitted the request |
| 17:18.43 | proppy | is it possible to set the calleridname from an AGI ? |
| 17:19.22 | moy | you have to ask que telco if they see our request, if they don't then is a problem at the driver or hardware level, not the signaling level |
| 17:19.25 | s14ck | moy: the configuration in dahdi/system.conf may be help? |
| 17:19.26 | proppy | I guess I should just use SET VARIABLE |
| 17:22.19 | moy | if you see OK in dahdi_tool then its ok, you should see the channels as IDLE as well, but openr2 would have detected if they are not idle, so I don't think that's your problem |
| 17:22.28 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 17:27.53 | *** join/#asterisk propellerhead (n=yogurt2u@190.246.45.140) |
| 17:28.46 | s14ck | moy http://pastebin.com/m206ff86b |
| 17:29.37 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 17:33.26 | moy | s14ck: you have wrong configuration, read the guide |
| 17:35.44 | *** join/#asterisk BBHoss (n=bbhoss@72.146.26.249) |
| 17:36.44 | s14ck | moy about the x,y:abcd bits? the telco said me 1001 |
| 17:37.28 | [TK]D-Fender | non-PRI E1? Extremely rare |
| 17:40.41 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:40.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:40.46 | leifmadsen | [TK]D-Fender: was it a bug? |
| 17:42.32 | *** join/#asterisk Sanjoy (n=Sanjoy@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 17:43.04 | [TK]D-Fender | leifmadsen: Can't say, like usual another person with nothing to SHOW for their problem |
| 17:44.41 | jaytee | "HI! I have an urgent serious problem! I can't call out! Don't ask me for a pastebin of anything and give me immediate help RIGHT NOW!!!!" |
| 17:44.52 | leifmadsen | [TK]D-Fender: cool :) |
| 17:45.14 | [TK]D-Fender | jaytee: AND DON'T GIVE ME ANY SHIT!!! |
| 17:45.18 | jaytee | hehehe |
| 17:46.05 | jaytee | that was funny evening, I'll never forget that one. |
| 17:46.19 | jaytee | and drmessano was on a roll that night |
| 17:47.06 | [TK]D-Fender | jaytee: that was actually a "how do install * and expect it to just work really fast" one more than a debugging |
| 17:47.38 | jaytee | yeah, he wanted 2 phones and * in like less then 30 minutes and got really hostile |
| 17:48.17 | jaytee | which led to the infamous asterisk highway traffic sign :-) |
| 17:49.29 | *** part/#asterisk swathanthran (n=user@unaffiliated/shyam-k/x-8459115) |
| 17:51.24 | [TK]D-Fender | Oh EMACS is they anything you can't do?! |
| 17:54.05 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-3921680f87ce08e5) |
| 17:54.40 | *** join/#asterisk brunner (n=chris@75-143-70-242.dhcp.aubn.al.charter.com) |
| 17:57.26 | apeiron | Occupy less than 32 MB of core? |
| 18:00.16 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 18:00.18 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 18:01.11 | naxxfish | how do channel modules get loaded? for some reason DAHDI isn't getting loaded |
| 18:03.34 | [TK]D-Fender | naxxfish: and when you try to load it manually? |
| 18:04.02 | mosty | naxxfish: did you compile asterisk yourself? and do you have a valid chan_dahdi.conf |
| 18:04.47 | naxxfish | [TK]D-Fender: doing module load chan_dahdi.so tells me that the module isn't even in /usr/lib/asterisk/modules |
| 18:05.04 | tzafrir_laptop | apeiron, I would expect gedit to take more memory than Emacs nowadays |
| 18:05.17 | apeiron | tzafrir_laptop, yuck |
| 18:05.24 | tzafrir_laptop | (untested) |
| 18:05.51 | [TK]D-Fender | naxxfish: then it isn't |
| 18:06.10 | naxxfish | that's kinda strange - it definitely got compiled |
| 18:06.11 | [TK]D-Fender | naxxfish: Was DAHDI installed and configured BEFORE Asterisk was compiled? |
| 18:06.37 | naxxfish | yup |
| 18:07.12 | mosty | naxxfish: does the chan_dahdi.so exist in the directory where you compiled asterisk? |
| 18:07.20 | naxxfish | http://pastebin.com/d2b84325f <-- shows dahdi modules loaded in dmesg |
| 18:07.27 | naxxfish | mosty: just looking now |
| 18:07.58 | [TK]D-Fender | naxxfish: Perhaps having recompiled * you forgot to re-install it |
| 18:08.13 | [TK]D-Fender | naxxfish: do "make install" again from your source folder and confirm the .so is moved |
| 18:08.24 | naxxfish | make install for asterisk? |
| 18:08.47 | naxxfish | or dahdi |
| 18:09.16 | [TK]D-Fender | naxxfish: asterisk. But also prove DAHDI is insalled & configured correctly in its own right |
| 18:09.35 | naxxfish | dmesg shows that the modules are being loaded |
| 18:09.39 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:09.48 | naxxfish | also dahdi_monitor and so on work fine |
| 18:11.16 | mosty | naxxfish: so was chan_dahdi.so compiled or not? |
| 18:11.21 | naxxfish | oh, hmm, that's weird - there's no chan_dahdi.so in channels |
| 18:12.15 | naxxfish | XXX chan_dahdi <-- says menuselect |
| 18:12.20 | *** part/#asterisk AndyCrawford (n=andy@dynamic-65-161-142-80.tvscable.com) |
| 18:13.47 | naxxfish | i guess maybe it didn't get compiled :p |
| 18:16.17 | *** join/#asterisk s0lid (n=s0lid@122.53.98.50) |
| 18:18.33 | naxxfish | Depends on: res_smdi(M), dahdi(E), tonezone(E) ... so it's missing one of these then i guess ? |
| 18:19.50 | mosty | yes |
| 18:20.02 | naxxfish | well, there all installed |
| 18:20.38 | naxxfish | hmm, i'll try ./configuring without --with-dahdi and see if that helps at all |
| 18:21.16 | mosty | perhaps you can look in config.log to see output related to dahdi |
| 18:21.34 | naxxfish | yup i'll do that |
| 18:31.15 | *** join/#asterisk clintc (n=clintc@n128-227-61-198.xlate.ufl.edu) |
| 18:35.03 | tzafrir_laptop | naxxfish, egrep -i 'dahdi|tonezone' build_tools/menuselect-deps |
| 18:35.15 | tzafrir_laptop | if any of the two is 0, it is missing |
| 18:35.16 | *** join/#asterisk djMax (n=chatzill@66.92.91.133) |
| 18:38.54 | djMax | is there a way to get an extension to go straight to voicemail, with no greeting? |
| 18:39.04 | djMax | i.e. to "beep" |
| 18:42.06 | DavidR2008 | djMax: there's an option on the VoiceMail command to not play a message, I don't remember what it is offhand |
| 18:42.32 | djMax | s, though at least via FreePBX it didn't seem to take. I'll try direct config |
| 18:42.35 | djMax | thx |
| 18:44.02 | *** join/#asterisk foolano (n=magnetic@14.Red-88-26-177.staticIP.rima-tde.net) |
| 18:46.12 | *** join/#asterisk algorithmn (n=root@s15263727.onlinehome-server.com) |
| 18:48.16 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:51.31 | naxxfish | tzafrir_laptop: |
| 18:51.31 | naxxfish | DAHDI=1 |
| 18:51.32 | naxxfish | TONEZONE=1 |
| 18:52.14 | *** join/#asterisk Sanjoy (n=Sanjoy@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 18:52.14 | tzafrir_laptop | naxxfish, do you have: channels/chan_dahdi.so #? |
| 18:52.58 | *** join/#asterisk columbo (n=columbo@pool-98-119-227-123.lsanca.dsl-w.verizon.net) |
| 18:53.26 | naxxfish | tzafrir_laptop: not yet |
| 18:53.33 | naxxfish | i'll try running make again |
| 18:54.09 | *** join/#asterisk bencer (n=bencer@heal.cauterized.net) |
| 18:56.03 | *** join/#asterisk lehel (n=lehel@79.116.195.23) |
| 18:56.10 | lehel | hello |
| 18:57.08 | algorithmn | well POKE back at you |
| 18:58.07 | lehel | anyone has any experience setting up asterisk on a VPS? it's all fine until i get: "Starting Asterisk PBX: Unable to install capabilities." - does has anything to do with cap_net_admin ? |
| 18:58.43 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 18:58.48 | algorithmn | ehh jer! |
| 18:59.00 | naxxfish | <PROTECTED> |
| 18:59.03 | algorithmn | lol |
| 18:59.04 | naxxfish | a hah |
| 18:59.22 | naxxfish | <PROTECTED> |
| 18:59.53 | naxxfish | right, it definitely compiled into channels/ |
| 19:00.55 | lehel | `k |
| 19:01.14 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 19:03.24 | Qwell | NotJerJer: ! |
| 19:03.33 | algorithmn | so !jer, hackin up any ish lately? |
| 19:04.20 | NotJerJer | just tryin to make money |
| 19:04.21 | *** join/#asterisk Magnus_H (n=magnus@217-209-2-180-no145.tbcn.telia.com) |
| 19:04.31 | algorithmn | hear ya |
| 19:05.03 | foolano | hi guys I have already asked this in #asterisk-bugs. I'm gonna ask it here again, just in case it rings a bell to some of you |
| 19:05.12 | foolano | <PROTECTED> |
| 19:08.36 | [TK]D-Fender | foolano: It is accepted? "sip show registry" confirms it? |
| 19:09.36 | foolano | [TK]D-Fender: lemme check now but i guess so, cuz I enabled the sip debug mode and I clearly see how the the user is registered successfully |
| 19:10.12 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 19:10.15 | [TK]D-Fender | foolano: So far is just sounds like the other end is being insecure, not *'s fault. See if ti WORKS. |
| 19:10.43 | foolano | just a sec to check it |
| 19:11.21 | nny_1 | hmm.. need some advice. Had a client move a box over the weekend to a new IP. I changed the IP in iax.conf and now asterisk says "offline" from both sides. Can't find any glaring errors in the logs, any suggestions? |
| 19:11.39 | nny_1 | i can ssh into both boxes from one to the other |
| 19:12.01 | nny_1 | and don't see any firewall rules that would prevent the connection |
| 19:12.36 | nny_1 | actually iptables blow open on both boxes |
| 19:13.26 | mosty | nny_l: does one of the servers register with the other? |
| 19:14.40 | nny_1 | both show as offline UNREACHABLE |
| 19:14.57 | mosty | ok, but can you answer my question? |
| 19:15.02 | naxxfish | way, DAHDI loads :) |
| 19:15.34 | nny_1 | mosty iax2 show registrations ? |
| 19:15.42 | nny_1 | registry* |
| 19:16.06 | algorithmn | nny_1: you have IAX open through firewall? |
| 19:16.25 | nny_1 | algorithmn let me confirm the port is proper |
| 19:16.26 | *** join/#asterisk _theHub (n=_theHub@ool-ad02788c.dyn.optonline.net) |
| 19:17.11 | mosty | nny_l: do you have a register line in one of your iax.conf files? ie is one server trying to register to the other? |
| 19:17.39 | algorithmn | nny_1: make sure you ACL 4569 when your done |
| 19:18.23 | nny_1 | mosty: yes |
| 19:18.40 | *** join/#asterisk isaac (n=isaac@debian/developer/isaac) |
| 19:18.53 | isaac | hi there |
| 19:18.58 | nny_1 | algorithmn: one box has wan interface sitting on the internet connection, and no iptables rules, so no firewall. the other has a linksys router in front of it, but the port (4569) is forwaded |
| 19:19.05 | *** join/#asterisk docid (n=eris@whthyt253-26.northwestel.net) |
| 19:19.34 | algorithmn | k, don't forget to ACL them to the peering hosts |
| 19:19.57 | algorithmn | otherwise things could go wrong |
| 19:19.58 | docid | ok, so im trying to debug my t1 connection here....been advised to use pri instense debug span, but we have no pri, what is equlivant command for an E&M wink trunk? |
| 19:20.10 | naxxfish | hmm, i've got a X100P and here's my chan_dahdi.conf file, http://pastebin.com/m1e071d7e |
| 19:20.44 | naxxfish | and i get WARNING[12135]: app_dial.c:1470 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 19:20.50 | [TK]D-Fender | naxxfish: More-or-less OK |
| 19:20.56 | nny_1 | algorithmn: everything from a network standpoint seems sane |
| 19:20.56 | mosty | nny_1: so what does iax2 show registry show on that box? |
| 19:21.07 | [TK]D-Fender | naxxfish: and it'd be nice to see what GENERATED that error |
| 19:21.17 | naxxfish | set verbose 10? |
| 19:21.24 | [TK]D-Fender | naxxfish: yes |
| 19:21.30 | *** join/#asterisk desdesdesdes (n=kg@196.211.34.2) |
| 19:21.50 | desdesdesdes | anyone worked with xorcom fxs? |
| 19:21.51 | algorithmn | nny_1: sorry to not have been following your question.. I just noticed that you have no firewall w/iax enabled |
| 19:21.54 | [TK]D-Fender | naxxfish: and do "dahdi show status" , "dahdi show channels" as well and PB both |
| 19:21.55 | algorithmn | kinda scary dude |
| 19:22.48 | naxxfish | http://pastebin.com/m6ed9043f hmm, looks like there's no channels |
| 19:23.09 | nny_1 | algorithmn: this was a trixbox install we inherited and removed trixbox garbage from, it's out of the country, so haven't set up that yet |
| 19:23.31 | algorithmn | no doubt |
| 19:23.34 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 19:24.02 | [TK]D-Fender | naxxfish: "module reload chan_dahdi.so" , and repeat |
| 19:24.08 | nny_1 | mosty http://pastebin.com/m530c82b4 |
| 19:24.15 | algorithmn | eeeks is like swiss cheese nny_1 |
| 19:24.27 | algorithmn | follow proper administrative practices you should be fine |
| 19:24.30 | nny_1 | <PROTECTED> |
| 19:24.53 | [TK]D-Fender | naxxfish: ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended <-- you said this file was chan_dahdi.conf |
| 19:24.56 | docid | so, anybody, 'pri intense debug span' equlivant commands for a non-pri E&M / MF span |
| 19:25.02 | mosty | nny_l: there is no registration config on that server |
| 19:25.02 | nny_1 | algorithmn: heh yeah waiting for the client to fly me out to reinstall a proper distro on the box :D |
| 19:25.06 | [TK]D-Fender | naxxfish: if so, it is broken |
| 19:25.26 | docid | ? |
| 19:25.28 | algorithmn | is the box under strain? cpu ramped? |
| 19:25.51 | nny_1 | <PROTECTED> |
| 19:25.57 | algorithmn | k |
| 19:26.18 | naxxfish | [TK]D-Fender: ah - i see |
| 19:26.45 | algorithmn | so your iax insatnce isn't getting tossed around.. |
| 19:27.14 | nny_1 | mosty hmm yes, there is no register line |
| 19:27.24 | *** part/#asterisk isaac (n=isaac@debian/developer/isaac) |
| 19:27.48 | algorithmn | iax2 show channels |
| 19:27.54 | algorithmn | would tell you if your getting tossed around |
| 19:28.17 | mosty | nny_l: so you're using static hosts, there is no registration required. in this case i recommend running a packet trace on one of the servers to see if it's receiving the iax packets from the other |
| 19:28.37 | *** join/#asterisk harryv (n=harry@67.207.147.205) |
| 19:28.53 | nny_1 | http://pastebin.com/m744110aa |
| 19:28.56 | nny_1 | for channels |
| 19:29.18 | nny_1 | looking over the error logs more |
| 19:29.26 | nny_1 | problem may be different than described to me |
| 19:30.09 | nny_1 | [Mar 23 12:01:50] WARNING[3286] app_dial.c: Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 19:30.22 | nny_1 | this is from one host |
| 19:30.26 | nny_1 | (panama) |
| 19:30.36 | mosty | nny_l: can you ping each server from the other? |
| 19:30.39 | algorithmn | is the ip listed one of your ips (wipe if you can) |
| 19:30.50 | nny_1 | yes |
| 19:31.05 | algorithmn | k |
| 19:31.09 | nny_1 | mosty i can ssh between the two, the local side linksys router doesn't respond to pings |
| 19:31.10 | algorithmn | so its trying to establish channels |
| 19:31.39 | nny_1 | just a note, we changed the IP last week (as stated) before then didn't have an issue, the client moved offices |
| 19:31.41 | mosty | nny_l: run a packet trace on one server listening to the iax port, then try to make a call to that server |
| 19:31.48 | nny_1 | ok one sec |
| 19:33.09 | nny_1 | running tcpdump on port 4569 on remote host |
| 19:34.39 | harryv | apparently BACKGROUNDSTATUS was added to asterisk back in 06: http://bugs.digium.com/svnstats/asterisk/trunk/2006-09.html#298 - but yet, GET VARIABLE BACKGROUNDSTATUS returns 0 no matter wether the file was played thorugh or not |
| 19:34.43 | harryv | *through |
| 19:35.32 | harryv | any other way to know wether a file played w/ Background() was finished or not |
| 19:35.35 | naxxfish | right |
| 19:36.20 | naxxfish | i've got a full chan_dahdi.conf file http://pastebin.com/m32dd8d2f |
| 19:36.56 | [TK]D-Fender | naxxfish: looks like progress. |
| 19:37.02 | nny_1 | mosty: algorithmn http://pastebin.com/m70c3f4d0 |
| 19:37.11 | nny_1 | tried to call from local to panama |
| 19:37.19 | nny_1 | remote side had no dump\ |
| 19:37.22 | nny_1 | (no packets) |
| 19:37.24 | naxxfish | the same errors though :/ |
| 19:37.29 | nny_1 | the tcpdump there is from local |
| 19:39.01 | nny_1 | gonna nmap remote from local |
| 19:41.01 | *** join/#asterisk Sazenchan (n=sazencha@85-18-219-234.ip.fastwebnet.it) |
| 19:41.06 | nny_1 | 4569/tcp filtered unknown |
| 19:41.08 | nny_1 | hmm |
| 19:41.19 | nny_1 | ISP blocking? |
| 19:41.45 | NotJerJer | iax is udp |
| 19:41.51 | nny_1 | ha yeah |
| 19:41.57 | nny_1 | er um let me retry that scan |
| 19:42.01 | algorithmn | -sU |
| 19:42.46 | nny_1 | 4569/udp open|filtered unknown |
| 19:42.56 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:42.58 | mosty | nny_l: something is firewalled somewhere, or you have the wrong ip address somewhere |
| 19:43.36 | algorithmn | firewall... |
| 19:43.53 | algorithmn | http://www.securityscraper.com/pingpoke/iaxPingPoker.txt |
| 19:44.06 | algorithmn | use that to do applicaiton level tests |
| 19:44.12 | algorithmn | nmap will only tell you if the port is open |
| 19:44.19 | algorithmn | that will tell you if * is running behind it |
| 19:46.57 | nny_1 | k let me grab the perl deps and try it |
| 19:47.11 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:47.55 | algorithmn | nny_1: brb 10 mins... !jj could help u w/the script im sure |
| 19:48.36 | nny_1 | k trying to figure out what package has Subnets.pm atm |
| 19:48.49 | nny_1 | nm ill read the script |
| 19:53.53 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 19:54.12 | Dovid | TK: Please don't bop me on the head. forgot which one is which for a 180 and a 183 message |
| 19:54.31 | Dovid | 180 is just ringing and 183 means play ur own ringing sound ? |
| 19:54.39 | Maliuta | a 180 indicates ringing |
| 19:54.41 | Sanjoy | where id your asterisk server algorithm |
| 19:54.58 | Dovid | Maliuta: inidcates ringing and they will pass the early media ? |
| 19:55.24 | Maliuta | Dovid: pretty certain on that |
| 19:56.22 | *** part/#asterisk Magnus_H (n=magnus@217-209-2-180-no145.tbcn.telia.com) |
| 19:57.30 | Dovid | Maliuta: http://pastebin.com/mf0bd1ef |
| 19:57.34 | Dovid | trying to figure that one out |
| 19:57.49 | Dovid | if we are being told to play ringing or we get a 183 play our own ? |
| 19:58.56 | nny_1 | hmm expecting more output but got Response from 190.34.201.242 : 4569 |
| 20:01.16 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:01.35 | nny_1 | didnt get any perl dep errors but installing all packages via CPAN |
| 20:02.01 | Maliuta | Dovid: the 180 should indicate incoming media |
| 20:02.13 | [TK]D-Fender | naxxfish: And I don't see your new configs, nor your failed call attempt |
| 20:02.28 | Maliuta | Dovid: the ringing should be from the other end |
| 20:03.08 | Maliuta | Dovid: of course if you have passed it on with Dial() you might not have told it to pass on the ringing |
| 20:03.15 | NotJerJer | Dovid: one of the two messages will include an SDP |
| 20:03.55 | NotJerJer | can't remember which - the one with the SDP includes the media stream. early-media |
| 20:03.56 | *** join/#asterisk AlexGC (n=admin@201.127.34.112) |
| 20:04.06 | AlexGC | good day gentlemen |
| 20:04.13 | Maliuta | NotJerJer: 183 |
| 20:04.21 | Maliuta | NotJerJer: that has the SDP |
| 20:04.28 | NotJerJer | there ya go |
| 20:04.51 | Dovid | so if i hear ringing and not the early media its something by me thats doing it. ok |
| 20:05.09 | nny_1 | algorithmn: when ya get abck, i assume Response from 190.34.201.242 : 4569 is the proper response, tried it against another box not running iax and got nothing.. used perl pingpoker.perl -h 190.34.201.242 \n\t\t |
| 20:05.17 | Maliuta | Dovid: I would expect that it's something in the Dial() |
| 20:05.41 | NotJerJer | ditto |
| 20:05.47 | algorithmn | nny_1: the non responding box is your problem |
| 20:06.14 | Maliuta | Dovid: but I haven't seen your dialplan ... |
| 20:06.46 | *** join/#asterisk Sanjoy (n=Sanjoy@CPE001839a90e41-CM0011e6c3e9a7.cpe.net.cable.rogers.com) |
| 20:06.49 | Maliuta | the Dial application is well documented too |
| 20:06.57 | nny_1 | algorithmn: sorry miscommunication, the non responding box was a test |
| 20:06.59 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:07.14 | s14ck | how can I make call the E1 trunk to sip extensions? |
| 20:07.22 | Dovid | Maliuta: Thanks |
| 20:07.26 | nny_1 | algorithmn: wasn't sure what pingpoker output should look like so i ran it against another non asterisk box |
| 20:07.28 | Dovid | its actually a clinet connecting to me |
| 20:07.38 | Dovid | just wanted to make sure. |
| 20:07.49 | Dovid | gona get a full tcpdump where i can play it for them |
| 20:08.00 | Maliuta | Dovid: connecting to * ... then you have * using Dial() to push it to a sip phone or a hardline? |
| 20:08.09 | algorithmn | ok |
| 20:08.24 | algorithmn | nny_1: you should see the response from x.x.x.x |
| 20:08.29 | algorithmn | if its up |
| 20:08.57 | NovceGuru | awesome, people are calling me and saying they pressed 1 on some security system and its ringing my number |
| 20:09.26 | nny_1 | algorithmn: got Response from 190.34.201.242 : 4569 |
| 20:09.49 | algorithmn | thats good then |
| 20:09.57 | algorithmn | the app is up and you can connect to it |
| 20:10.25 | nny_1 | ok i checked iax.conf and the IP address on both boxes match the remote IP of the other box from each one |
| 20:10.48 | nny_1 | is there some other crazy place I can look i am unaware of? I grepped through all of /etc |
| 20:10.51 | nny_1 | and then / heh |
| 20:11.52 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-22.hsd1.fl.comcast.net) |
| 20:12.06 | algorithmn | hmm.. hard to say |
| 20:12.30 | algorithmn | if both hosts are up and are configured.. considering you followed a proper example.. it should be fine |
| 20:13.18 | nny_1 | algorithmn: heh yeah. the stupid part is, this worked prior to them moving office |
| 20:13.27 | nny_1 | been working for over a year now |
| 20:13.45 | algorithmn | did you try running the script on both hosts? |
| 20:13.49 | nny_1 | I went into the panama box, set the new wan IP, went into the local box, set the IAX.conf |
| 20:13.50 | algorithmn | to try to poke eachother? |
| 20:14.14 | nny_1 | algorithmn: let me try on the remote one, it's running some absurdly crappy distro |
| 20:14.20 | mosty | nny_l: run a packet trace on the server you're calling from- and verify that the packets are really being sent out |
| 20:14.23 | nny_1 | if i have CPAN should be good i hope |
| 20:14.34 | nny_1 | mosty: yeah that was that pastebin earlier |
| 20:14.45 | nny_1 | http://pastebin.com/m70c3f4d0 |
| 20:16.18 | mosty | nny_l: try running iax on a different port? |
| 20:16.39 | nny_1 | mosty: yeah considered that, will do next |
| 20:17.14 | nny_1 | let me satisfy the perl gods real quick and try the remote end |
| 20:18.16 | *** part/#asterisk harryv (n=harry@67.207.147.205) |
| 20:19.08 | algorithmn | lol@perl gods |
| 20:20.06 | desdesdesdes | hi there here is one for the experts |
| 20:21.35 | desdesdesdes | need to know if it is possible to activate a ringback when free(meaning if i dial a fxs extension and it is busy i want to activate a ringback to my extension as soon as that user hangs up the phone ) is this possible? |
| 20:22.27 | [TK]D-Fender | desdesdesdes: Yes, with you doinga whole pile of scripting to poll for the target freeing up and generating the outbound call-file (AMI originate, etc) |
| 20:22.34 | [TK]D-Fender | desPlenty of work, but very possible |
| 20:22.44 | [TK]D-Fender | desdesdesdes: Plenty of work, but very possible |
| 20:23.48 | desdesdesdes | who do i pay to get this working? |
| 20:24.16 | algorithmn | desdesdesdes: i bet notjerjer has a price |
| 20:24.32 | desdesdesdes | was a standard feature on customers old legasy and they want to have the feature? |
| 20:24.57 | desdesdesdes | how do i contact him? |
| 20:26.00 | nny_1 | algorithmn: got a reponse from the other end |
| 20:26.03 | algorithmn | i thought he was in the room |
| 20:26.11 | nny_1 | algorithmn: i can try a new port |
| 20:26.14 | mosty | wouldn't it just be a matter for dialing, and looping with a pause if the channel is busy? |
| 20:26.15 | algorithmn | nny_1: so there is connectivity between the two? |
| 20:26.20 | algorithmn | if so, no need to change ports |
| 20:26.24 | nny_1 | yeah |
| 20:26.29 | algorithmn | don't bother |
| 20:26.33 | nny_1 | k |
| 20:26.39 | algorithmn | fire up wireshark and watch the 4569 packet |
| 20:26.40 | algorithmn | s |
| 20:26.59 | *** join/#asterisk PDani (n=pekdanie@catv-80-99-36-57.catv.broadband.hu) |
| 20:27.26 | nny_1 | both sides i assume? |
| 20:27.48 | algorithmn | try it from both hosts |
| 20:27.58 | algorithmn | not sniff both hosts |
| 20:28.03 | algorithmn | poke from both hosts |
| 20:28.12 | algorithmn | sniff from the authorization server (host, not client) |
| 20:28.37 | mosty | [tk]d-fender: the only annoying part of what desdesdesdes wants to do would be initiating the 'polling call', wouldn't it? |
| 20:28.42 | mosty | bah |
| 20:28.58 | algorithmn | mosty: not sure |
| 20:30.25 | mosty | desdesdesdes: you can do it with some simple dialplan code and maybe a tiny AGI or System call |
| 20:31.24 | nny_1 | haven't used tshark before, any advice on app switches? |
| 20:32.01 | mosty | nny_1: just use tcpdump instead if you are familiar with that |
| 20:33.14 | algorithmn | tcpdump works 2 |
| 20:35.09 | NotJerJer | you mean campon ! |
| 20:35.30 | NotJerJer | class 5 feature - Camp On |
| 20:36.00 | nny_1 | hmm |
| 20:36.01 | NotJerJer | desdesdesdes: ^^ |
| 20:36.11 | desdesdesdes | yes camp on busy |
| 20:36.13 | nny_1 | cannot get any 4569 packets to dump on remote end |
| 20:36.20 | NotJerJer | good luck on that one |
| 20:36.31 | desdesdesdes | why? |
| 20:36.33 | nny_1 | ran poke and it said Response, but no packets spat out on remote |
| 20:36.35 | desdesdesdes | have u tried? |
| 20:36.54 | algorithmn | nny_1: what filter string are you using? |
| 20:37.09 | nny_1 | tcpdump -A -s 0 HOST |
| 20:37.10 | NotJerJer | would have to create a channel agnostic way of requesting a camp on |
| 20:37.14 | nny_1 | er port 4569 at end |
| 20:37.19 | nny_1 | it shows up on localk |
| 20:37.21 | nny_1 | local |
| 20:37.24 | nny_1 | but nada on remote |
| 20:37.35 | nny_1 | tcpdump -A -s 0 HOST port 4569 |
| 20:38.20 | nny_1 | gah nm |
| 20:38.21 | nny_1 | eth1 |
| 20:38.29 | NotJerJer | like res_campon |
| 20:38.42 | nny_1 | ok have packets on remote end |
| 20:38.49 | nny_1 | from local |
| 20:39.29 | nny_1 | odd |
| 20:40.41 | nny_1 | ok to recap. UDP 4569 packets show up on both ends |
| 20:40.49 | nny_1 | iax.conf looks* right |
| 20:41.24 | nny_1 | let me poke around more.. i think this may be something else |
| 20:41.41 | algorithmn | nny_1: let me know bud |
| 20:42.35 | nny_1 | algorithmn: np i suspect this is sip client issue manifesting itself |
| 20:42.58 | nny_1 | as in it's trying to reach a sip client accross IAX and failing, even though the client shows up in sip show peers |
| 20:43.45 | *** part/#asterisk Sazenchan (n=sazencha@85-18-219-234.ip.fastwebnet.it) |
| 20:45.08 | nny_1 | is HHI/panama 64.203.239.92 (S) 255.255.255.255 4569 (T) UNREACHABLE normal if i have the IAX channels set up as trunk=yes? |
| 20:45.16 | nny_1 | (from iax2 show peers) |
| 20:47.31 | mosty | how can a sip client be causing your iax peer to be unreachable? |
| 20:49.02 | *** join/#asterisk [hC] (n=voxter@76.77.95.2) |
| 20:49.13 | nny_1 | mosty: i am reading the console output, it tries a client across the IAX channel (SIP 108) |
| 20:49.22 | nny_1 | so asterisk -IAX - asterisk -SIP - 108 |
| 20:49.29 | nny_1 | i think the last leg is what's broken |
| 20:49.42 | nny_1 | and it says unavail |
| 20:50.28 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-55.dyn.centurytel.net) |
| 20:50.56 | nny_1 | setting up a channel on my phone here to act as a client on the remote |
| 20:51.20 | nny_1 | since calling them is proving difficult, and apparently "Be available" when diagnosing phone issues is something they don't grasp |
| 20:51.34 | docid | ok, so im trying to debug my t1 connection here....need to use command simular to 'pri intense debug span', but we have no pri, what is equlivant command for an E&M wink trunk? |
| 20:52.58 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:54.25 | *** join/#asterisk docidu (n=eris@whthyt253-26.northwestel.net) |
| 20:54.42 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-d0d3c27c490397f6) |
| 20:55.14 | *** part/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-d0d3c27c490397f6) |
| 20:55.32 | AlexGC | Q: Ideas on how to play a msg/recording before a call is connected/dialed? ie "Your call is being connected, please pay your bills!!" then connect. ? ideas, pointers, all appreciated |
| 20:56.03 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:56.24 | docidu | ok, so im trying to debug my t1 connection here....need to use command simular to 'pri intense debug span', but we have no pri, what is equlivant command for an E&M wink trunk? |
| 20:57.00 | jsmith | docidu: There is no equivalent command... |
| 20:57.08 | jsmith | docidu: (but I wish there were!) |
| 20:57.12 | nny_1 | mosty: algorithmn hmm nm back to square one |
| 20:57.15 | docidu | ohh, ok, |
| 20:57.23 | nny_1 | <PROTECTED> |
| 20:57.26 | docidu | umm, any suggestions then? |
| 20:58.07 | algorithmn | nny_1: are you using FQDN's for your IAX2 trunk? |
| 20:58.07 | [TK]D-Fender | nny_1: Stop showing jsut the error and instead the dial & peer that generated it. |
| 20:58.16 | nny_1 | algorithmn: no |
| 20:58.18 | nny_1 | [TK]D-Fender: ok |
| 20:58.25 | jsmith | docidu: Sorry... try sharing your exact problem, and maybe someone can help you |
| 20:58.38 | *** join/#asterisk farkus (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:59.12 | *** join/#asterisk af_ (n=getsmart@88-149-240-30.dynamic.ngi.it) |
| 20:59.45 | nny_1 | full error and entry from extensions.conf |
| 20:59.46 | nny_1 | http://pastebin.com/m68de828a |
| 21:00.09 | *** join/#asterisk drixter (i=drixter@fido.e-utp.net) |
| 21:00.52 | nny_1 | [TK]D-Fender: i can post iax.conf and show that peer 202 is working on other end if need be' |
| 21:00.53 | drixter | hi, may i ask someone to call me using sip drixter@sip.e-utp.net and tell whatever, even 'shit' or 'test' |
| 21:01.02 | docidu | umm, ok....got a multitennant dual t1 setup going on, basically t1 span 1 is telco t1 voice, span 2 is link to cell switch, asterisk is catching certian ranges of numbers for voicemail routing, seemingly randomly, asterisk reacts as though it has recieved no digits....this happens most often cooming from the cell switch but also happens from the telco t1, |
| 21:01.05 | [TK]D-Fender | nny_1: 202 isn't a PEER |
| 21:01.10 | drixter | i just want to test only |
| 21:01.21 | [TK]D-Fender | nny_1: it is an EXTEN. Show us your HHI peer status |
| 21:01.21 | nny_1 | [TK]D-Fender: sorry SIP user |
| 21:01.24 | docidu | ok, when we take the asterisk box out of the loop, the errors stop... |
| 21:01.29 | docidu | mostly |
| 21:01.36 | [TK]D-Fender | nny_1: "202" has nothing to do with this |
| 21:01.39 | [TK]D-Fender | nny_1: at all |
| 21:02.06 | Nugget | there's a new http://www.thewebsiteisdown.com/ |
| 21:02.13 | docidu | in my logs i occationally see 'ignoring wink' why would it ignore wink? im thinking the ignored wink could be where those missing digits are |
| 21:03.10 | nny_1 | [TK]D-Fender: yes |
| 21:03.16 | docidu | umm, if its relevant, supposedly our t1 is coming off the oldest operational telco switch in north america |
| 21:03.18 | nny_1 | [TK]D-Fender: that came up earlier, HHI is offline |
| 21:03.31 | nny_1 | [TK]D-Fender: hence my orignal attempt at diagnostic |
| 21:04.24 | *** part/#asterisk drixter (i=drixter@fido.e-utp.net) |
| 21:05.40 | nny_1 | HHI/panama 64.203.239.92 (S) 255.255.255.255 4569 (T) UNREACHABLE |
| 21:09.01 | nny_1 | is there some way i can get more detail as to why that peer is UNREACHABLE? |
| 21:09.02 | [TK]D-Fender | nny_1: qualify=no |
| 21:09.23 | [TK]D-Fender | nny_1: And go prove the port is reachable |
| 21:09.42 | nny_1 | [TK]D-Fender: i have used pingpoke from both ends and can see the packets |
| 21:10.25 | *** join/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 21:10.40 | nny_1 | qualify may be the issue though |
| 21:11.17 | nny_1 | nm |
| 21:11.39 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 21:11.46 | NMR_1122 | Hi everyone, We're thinking of using Teliax. Is there anything we should be aware of or concerned with? |
| 21:13.06 | *** join/#asterisk brunner (n=chris@75-143-70-242.dhcp.aubn.al.charter.com) |
| 21:13.42 | nny_1 | [TK]D-Fender: still getting chan unavailable, but have port 4569 UDP packets on both ends when doing so. one of the remote packets has "No such context/extension*.." in the text dump though |
| 21:13.44 | [TK]D-Fender | NMR_1122: Generally pretty decent |
| 21:14.20 | nny_1 | [TK]D-Fender: nm its up now |
| 21:15.14 | nny_1 | [TK]D-Fender: it was the qualify? |
| 21:15.33 | [TK]D-Fender | nny_1: 1 failure = * gives up trying usually |
| 21:16.04 | nny_1 | [TK]D-Fender: so the qualify statement was the issue.. |
| 21:16.08 | nny_1 | [TK]D-Fender: so odd |
| 21:16.13 | [TK]D-Fender | nny_1: Could be it |
| 21:16.21 | nny_1 | will qualify fail if it lags out? |
| 21:17.00 | NMR_1122 | Thanks, reviews say they're good, but a bit more expensive than competitors. Just wanted to make sure there were no major quality / support / anything else that may not be apparent at first.... |
| 21:21.33 | nny_1 | need to read up on why qualify would cause it to break. I can understand the remote side not being able to hit the local side as the router doesn't respond to ping, but this all worked prior.. |
| 21:22.10 | nny_1 | and the remote does* respond to ping yet i couldn't dial to it over IAX from local |
| 21:27.36 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:29.22 | nny_1 | If qualify is set to "yes" then, by the looks of it, Asterisk will use the information about round trip time to decide whether or not to bother registering |
| 21:31.41 | *** join/#asterisk BBHoss (n=bbhoss@72.146.26.249) |
| 21:33.15 | mosty | nny_1: try qualify=10000 - which would set the max time to 10s, which is unreasonable but will at least show you if the packets are arriving |
| 21:35.31 | desdesdesdes | join #freepbx |
| 21:36.49 | nny_1 | mosty: well maybe 2000 |
| 21:37.06 | nny_1 | mosty: basically, if the host goes offline, we have the calls route locallt |
| 21:37.08 | nny_1 | locally |
| 21:37.50 | nny_1 | I didn't know you could tell it to qualify in X amount of time |
| 21:38.03 | mosty | i believe 2000 is the default when you have qualify=yes |
| 21:38.08 | nny_1 | hmm |
| 21:38.13 | nny_1 | wow thats some crap latency then |
| 21:38.20 | nny_1 | ok 10k it is |
| 21:39.28 | pfn | you should be able to determine if packets are arriving through sip debug |
| 21:39.36 | pfn | setting qualify to 10 seconds doesn't really solve the problem |
| 21:40.04 | nny_1 | pfn: part of a different issue, iax2 channels not working with qualify=yes, peer is 100 ms away |
| 21:40.47 | pfn | oh, iax, not sip |
| 21:40.53 | pfn | should be same deal, set iax2 debug |
| 21:41.12 | nny_1 | pfn: asterisk thinks the IAX channels are offline if I qualify them |
| 21:42.30 | nny_1 | i honestly don't know why though, the latency isn't bad.. lots of hops though |
| 21:42.58 | *** join/#asterisk talcite_lappy (n=matt@134.117.143.233) |
| 21:42.59 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 21:43.27 | talcite_lappy | hi guys. I have a question about setting up asterisk. How does asterisk connect to the regular phone network? |
| 21:43.37 | talcite_lappy | I'm lacking a bit of fundamental knowledge about VOIP I think |
| 21:43.51 | pfn | talcite_lappy, depends--either through regular phone network cards, or an internet telephony provider |
| 21:44.12 | talcite_lappy | pfn, but in either case, I would need to pay money for a service? |
| 21:44.31 | nny_1 | pfn: basically: moved a remote location to a new office/ IP. IAX2 channels showed "UNAVAILABLE" after, witrh qualify=yes. Sniffed ports, etc and determined the qualify decided the IAX2 was unavail, even though in theory it was |
| 21:44.32 | pfn | talcite_lappy, depends on the service, yes and no |
| 21:44.44 | nny_1 | qualify=10000 in iax2.conf seems to work |
| 21:44.49 | talcite_lappy | pfn, I'm confused. could you elaborate on that please? |
| 21:44.55 | pfn | talcite_lappy, depends on the service you use |
| 21:45.00 | pfn | some services are free |
| 21:45.08 | pfn | some offer small trial periods |
| 21:45.09 | pfn | etc. |
| 21:45.22 | pfn | for the most part, you'll have to pay if you want to interact with the PSTN |
| 21:45.23 | talcite_lappy | really? that's nice. Is there somewhere I can do a bit more reading about this stuff? |
| 21:45.27 | talcite_lappy | PSTN? |
| 21:45.33 | pfn | talcite_lappy, google |
| 21:45.38 | outtolunc | ~book |
| 21:45.39 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:46.51 | talcite_lappy | thanks guys |
| 21:47.05 | nny_1 | gah i always feel like a dumbass after tracking down an issue |
| 21:47.23 | nny_1 | so all that port checking etc, and qualify=yes was causing the issue.. should have guessed as much |
| 21:48.01 | pfn | nny_1, huh |
| 21:48.31 | nny_1 | pfn: nothing, being introspective |
| 21:48.49 | nny_1 | looking over my logs for the obvious PEER LAGGED errors |
| 21:49.20 | nny_1 | chan_iax2.c: Peer 'HHI' is now UNREACHABLE! Time: 0 |
| 21:50.00 | nny_1 | i think the issue is their network connection is flakey TBH, and one net drop and the system never brought the peer back up even after it was stable |
| 21:50.55 | nny_1 | gonna mtr google from the remote site overnight and see the magic |
| 21:51.08 | *** join/#asterisk harryv (n=harry@67.207.147.205) |
| 21:51.09 | nny_1 | anyways thanks algorithmn mosty [TK]D-Fender and pfn for the help |
| 21:52.15 | harryv | every 5 minutes I get this: http://pastie.org/424789 |
| 21:52.21 | harryv | that *looks* nasty |
| 21:52.46 | nny_1 | PRI dropping? |
| 21:52.55 | pfn | sounds like it |
| 21:53.27 | harryv | what would that mean? |
| 21:53.32 | nny_1 | later all thanks again |
| 21:53.34 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 21:53.37 | pfn | it means your T1 is dropping out |
| 21:53.45 | pfn | or E1 |
| 21:53.59 | lesouvage | harryv: do you use a sangoma card |
| 21:54.08 | harryv | Digium |
| 21:55.06 | harryv | can't remember which model, 2sec |
| 21:56.27 | harryv | te-122 |
| 21:56.32 | lesouvage | harryv: if nobody on the channel can help you out, Digium will if you ask them. |
| 21:57.58 | lesouvage | harryv: and every 5 minutes all connection end? |
| 21:58.43 | harryv | every 3-10 minutes |
| 21:59.07 | harryv | so the log says |
| 21:59.32 | harryv | te-121 even |
| 22:00.01 | deadpigeon | hm |
| 22:01.40 | deadpigeon | harryv: is that supposed to be an e1 trunk? |
| 22:03.37 | harryv | yeah |
| 22:03.52 | deadpigeon | harryv: where is this server located? |
| 22:04.15 | deadpigeon | are you facilities based or purchasing trunks from a clec? |
| 22:04.38 | harryv | I've got an ISDN-30 connection right here |
| 22:05.06 | deadpigeon | from whom? i see yer in st. louis |
| 22:05.14 | harryv | I'm in Denmark |
| 22:05.21 | deadpigeon | oh. wierd lookup than. |
| 22:05.23 | harryv | so, from the national telco |
| 22:05.32 | harryv | deadpigeon: I'm IRCing from a VPS :) |
| 22:05.37 | deadpigeon | i realize this now. |
| 22:05.56 | deadpigeon | i was just saying, if yer in st. louis trying to configure an e1 trunk yer gonna have a lot of trouble :) |
| 22:05.59 | *** join/#asterisk PDani (n=pekdanie@catv-89-133-156-227.catv.broadband.hu) |
| 22:06.12 | deadpigeon | what were you using for EC? |
| 22:06.17 | harryv | EC? |
| 22:06.21 | deadpigeon | echo cancellation |
| 22:06.30 | harryv | (and, the connection is kinda-fine, I can call in etc, only, it drops) |
| 22:06.46 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:06.54 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 22:06.56 | harryv | with a digium ec module |
| 22:07.03 | deadpigeon | okay. |
| 22:07.04 | harryv | VPMADT032 |
| 22:07.08 | deadpigeon | whats yer lbo? |
| 22:07.16 | joobie | hey guys.. i have a sip connection to pennytel (ITSP).. Is there anyway i can "reconnect" the pennytel connection via asterisk console? |
| 22:07.35 | [TK]D-Fender | joobie: what "connection"? |
| 22:08.37 | deadpigeon | harryv: definitely check your line build out, not sure the distance of your provider but that'd be where I would first start. also check your gain settings, but lbo is more important. |
| 22:08.47 | joobie | [TK]D-Fender, my asterisk box establishes a sip connection to pennytel .. it maintains that connection and then local phones which are connected to the asterisk box can route out the pennytel sip connection |
| 22:09.00 | harryv | deadpigeon: did a quick google, no luck - how do I check that? |
| 22:09.01 | joobie | i've made a change on the pennytel side and they want me to "disconnect" and "reconnect" the connection |
| 22:09.08 | joobie | i tried restarting asterisk, no joy |
| 22:09.10 | deadpigeon | harryv: dahdi or zaptel? |
| 22:09.43 | harryv | deadpigeon: zaptel |
| 22:10.02 | lesouvage | harryv: Are you sure that you have a proper isdn cable from the endpoint of the telco to your Asterisk server. |
| 22:10.23 | deadpigeon | harryv: okay, that'll be in /etc/asterisk/zaptel.conf |
| 22:11.54 | harryv | deadpigeon: any chance that might be in my zapata.conf ? which ahs nothing but this: http://pastie.org/424816 |
| 22:12.01 | harryv | lesouvage: honestly, no. |
| 22:12.15 | harryv | *has |
| 22:12.27 | *** join/#asterisk DarkRift (n=dark@65.92.167.11) |
| 22:12.38 | deadpigeon | http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 22:12.42 | [TK]D-Fender | joobie: delete the SIP registry entry |
| 22:13.25 | lesouvage | gpuk2: If you mess around with the Asterisk.conf files that are generated by freepbx (the once that say <don't change me>) you can get everything back the way it was by doing a reload from the freepbx gui. |
| 22:13.58 | lesouvage | sorry wrong channel |
| 22:14.03 | deadpigeon | zaptel.conf is for configuring the hardware and spans. signalling, B channels, and D channels, line build out, ecetera. |
| 22:14.14 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 22:14.20 | deadpigeon | framing and whatnot. |
| 22:14.36 | deadpigeon | it could even exist in /etc/zaptel.conf depending on yer installation. |
| 22:15.00 | joobie | ahh serious TK |
| 22:15.07 | harryv | deadpigeon: it's been ages since I touched this setup, so, forgot - it's in /etc/zaptel.conf: http://sprunge.us/BUDc |
| 22:15.08 | joobie | delete it, reload sip conf, then readd? |
| 22:15.12 | joobie | that sucks :) |
| 22:15.25 | deadpigeon | yer zapata.conf doesnt look right to me either. why do you have 2 groups? |
| 22:15.31 | harryv | I have two cards |
| 22:15.38 | harryv | so, this is only concerning span1 |
| 22:15.40 | deadpigeon | oh, yer D channel must be on channel 16? |
| 22:16.14 | deadpigeon | harry if you had a t-berd readaly availible that'd be cool. :) |
| 22:16.23 | harryv | yeah. the second card is connected to another box with a third card :) |
| 22:16.54 | deadpigeon | harry how far away is your provider? |
| 22:17.05 | *** join/#asterisk brunner (n=chris@75-143-70-242.dhcp.aubn.al.charter.com) |
| 22:17.15 | deadpigeon | is the e1 coming in on a loop or is it fiber? |
| 22:17.25 | harryv | I'm in the countryside. so most likely quite far away. |
| 22:17.34 | brunner | what is a typical ballpark local loop charge for a channelized T3 (DS3) when you're racked up with the telco? |
| 22:17.59 | brunner | I mean, if my server is co-located with the telephone company, what's a typical charged for a channelized T3? |
| 22:18.10 | *** join/#asterisk pagec (n=pagec@96.57.210.34) |
| 22:18.27 | *** join/#asterisk pc500 (n=fewa@208.39.251.21) |
| 22:18.44 | pc500 | Do you guys have your VoIP system on your voice vlan with your phones, or on a sepeate vlan? |
| 22:19.17 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-55.dyn.centurytel.net) |
| 22:19.18 | harryv | deadpigeon: not fiber |
| 22:19.19 | lesouvage | harryv: I'm not really an E1 expert but I have my share of problems once. What I remember is that there where much more lines in the config. (it was a Sangoma card) Are you sure there shouldn't be more values set? |
| 22:19.22 | deadpigeon | span=(spannum),(timing),(LBO),(framing),(coding) |
| 22:19.28 | jblack | I would keep all phone traffic on dedicated lines, inasmuch is possible. |
| 22:19.33 | deadpigeon | harry that's the span definition. |
| 22:19.37 | deadpigeon | your lbo is set to 0. |
| 22:19.39 | pagec | hey, i am currently using asterisk 1.4 but want to upgrade to 1.6 (on gentoo so I self compile). The current way is zaptel-libpri-asterisk. With 1.6 is it DAHDI-libpri-asterisk? where does DHADI tools fit into this? |
| 22:19.50 | deadpigeon | harryv: 0: 0 dB (CSU) / 0 - 133 feet (DSX-1) |
| 22:20.09 | deadpigeon | Our softswitch is 2 feet away from our Asterisk server, so 0 is fine for us. |
| 22:21.15 | deadpigeon | You might want to consider raising your build out, that may very well be your problem -- and if it isn't, it would have been a problem anyways. |
| 22:22.20 | deadpigeon | Your connections from your server to your other servers have lbo settings as well, and 0 may possibly be fine. But coming in from the PSTN, 0 is not gonna cut the mustard if you're more 300 feet away. |
| 22:22.58 | deadpigeon | Perhaps contacting your telco and requesting some information on what the LBO should be, might help you a bit. They very well may need to tweak their end anyways. |
| 22:23.10 | harryv | reading up on it. 300 feet away from what? |
| 22:26.21 | *** join/#asterisk telecos (n=sergio@76.167.219.87.dynamic.jazztel.es) |
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| 22:28.56 | NovceGuru | awesome...45 calls from people asking me why I called telling them their bank account was at risk |
| 22:28.59 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
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| 22:30.05 | deadpigeon | harryv: sry, just slightly confused. how far away is your dmark from your asterisk? |
| 22:31.16 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 22:31.33 | harryv | ah. ~10m |
| 22:32.23 | deadpigeon | lbo should be fine then. sry. turn debugging on and save the logs |
| 22:33.35 | deadpigeon | if you had a t-berd you'd be stylin. heh... need me one of those. |
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| 22:36.47 | harryv | pri debug, right? |
| 22:38.13 | harryv | now .. to wait for the next drop |
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| 22:42.20 | jeff_phillips | NovceGuru: you really shouldn't scare people like that. :) |
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| 22:46.42 | harryv | deadpigeon: http://sprunge.us/KUDc - I don't understand a word :) |
| 22:56.14 | Linuturk | hugglez Asterisk Advanced class |
| 22:56.23 | Linuturk | first day was fun :) |
| 22:56.29 | Linuturk | my brain hurts though |
| 22:57.03 | Qwell | Linuturk: fluxbuntu, huh? |
| 22:57.11 | Linuturk | heh, kinda |
| 22:57.23 | Linuturk | I'm not really active with that |
| 22:57.25 | Linuturk | anymore |
| 22:57.26 | Qwell | oh |
| 22:57.48 | Linuturk | assuming you are @ * |
| 22:57.58 | Linuturk | yes? |
| 22:57.59 | Qwell | mmhmm |
| 22:58.07 | Linuturk | in Huntsville? |
| 22:58.08 | jaytee | Linuturk, do you have Jared as an instructor? |
| 22:58.11 | Linuturk | yeah |
| 22:58.16 | Linuturk | Jared |
| 22:58.17 | Qwell | Linuturk: yes |
| 22:58.24 | jaytee | he's very good, I took that class in November |
| 22:58.42 | Linuturk | yeah, he's good. I enjoyed his book too |
| 22:58.45 | jaytee | that's when I got to meet Qwell in person |
| 22:59.03 | jaytee | he's not as tall in person as I thought he'd be :-) |
| 22:59.37 | Qwell | throws paper airplanes at jsmith-teaching |
| 22:59.37 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:59.41 | Linuturk | heh, who are you Qwell ? |
| 22:59.48 | Qwell | Linuturk: I am Qwell |
| 23:00.13 | Linuturk | elusive :) |
| 23:00.17 | jaytee | very |
| 23:00.21 | Linuturk | well, I'm Justin |
| 23:00.25 | Linuturk | easy to find in the class :) |
| 23:00.38 | jaytee | but you might catch him on a smoke break, if he hasn't quit yet |
| 23:00.46 | umpc | is Justin |
| 23:00.53 | Linuturk | shuns cigs |
| 23:01.24 | Linuturk | heh, my claims adjuster for a stolen AC is named Justin too |
| 23:01.35 | umpc | heh |
| 23:01.58 | jaytee | Konichiwa, Sensei!!! |
| 23:02.05 | Linuturk | waves @ jsmith |
| 23:02.32 | jsmith | Qwell: Hey! |
| 23:02.38 | jsmith | waves back to Linuturk |
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| 23:05.20 | Linuturk | jsmith: my brother's name is Jarod |
| 23:05.34 | jsmith | Linuturk: Cool... |
| 23:07.44 | jeff_phillips | Linuturk: sounds like a utah spelling of Jared |
| 23:08.16 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-133-195.dsl.stlsmo.sbcglobal.net) |
| 23:08.47 | LemensTS | hey you got a favorite server for asterisk? got like 9 ip phones, and 4 incoming analog lines with no echo cancelation card. Maybe 2 itsp sip lines..dell power edge preferablly..im not much of a hardware guy |
| 23:09.10 | LemensTS | love you for ever if you help me choose one |
| 23:09.45 | jeff_phillips | LemensTS: I don't care for Dell anymore |
| 23:09.50 | Linuturk | jeff_phillips: southern Florida |
| 23:09.57 | harryv | when sending a mail to asterisk-users and I have a large chunk of debug output, is it best to attach it? |
| 23:09.58 | LemensTS | why not |
| 23:10.08 | jsmith | LemensTS: I'm not a big Dell fan, but the 2950 makes is a rock-solid Asterisk box |
| 23:10.12 | jsmith | s/makes is/makes/ |
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| 23:10.22 | jeff_phillips | I had problems with the last few Dell's we had |
| 23:10.41 | jsmith | jeff_phillips: If you buy one you're fine... if you buy two, they'll have to different motherboards and RAID cards :-/ |
| 23:10.53 | jsmith | LemensTS: I prefer the HP and IBM servers... |
| 23:11.37 | LemensTS | jsmith: ok what hp or ibm server do you recommend? i dont care about raid, ive had more problems with losing data on raid than without |
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| 23:12.09 | jeff_phillips | We bought a bunch of IBM's off lease cheap and haven't killed a single one yet, even the ones we put in a really harsh dusy manufacturing shop that normally causes everything with a cooling fan to die after a couple months |
| 23:12.21 | *** part/#asterisk jcoffi (n=jcoffi@75.147.155.89) |
| 23:12.28 | LemensTS | yea id prefer ibm over hp |
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| 23:13.01 | jeff_phillips | yeah the guys that were using dells in our shop robbed me of my spare IBMs to put in place of their dells |
| 23:13.24 | jeff_phillips | In two months I had 3 of the other machines die & their users stole my IBM's to replace them. |
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| 23:16.48 | jaytee | I've got two identical 2950's, one as an active server, the other as a failover |
| 23:22.06 | jsmith | jaytee: And to they have the same motherboards? Same RAID controllers? |
| 23:22.37 | jaytee | yep, bought at the same time so they came with the same chipset, SAS controller and mobo |
| 23:22.45 | jsmith | jaytee: (My last job was managing 6500 HP boxes, so I know them a lot better than the Dells) |
| 23:23.22 | jaytee | I think the 2950 is a decent machine but I used to swear by Compaq Proliant servers. |
| 23:23.34 | jsmith | Yeah... those are the HPs :-) |
| 23:23.39 | jsmith | DL360 and DL380, baby! |
| 23:23.45 | jaytee | I know, after the merger |
| 23:24.04 | Chainsaw | DL365 G1 & DL385 G2 here. |
| 23:24.09 | jaytee | but most of the tech in the architecture came from Compag |
| 23:24.10 | Chainsaw | But yeah, that is good kit. |
| 23:24.15 | Chainsaw | You can tell by the screws. |
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| 23:25.23 | jaytee | my boss is a big Dell fan so I don't have any choice in what I have to work with. |
| 23:25.46 | s14ck | hi |
| 23:25.56 | s14ck | i have this warning [Mar 23 18:22:36] WARNING[9279] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 23:25.58 | jaytee | I mentioned HP when I first started and got a half hour lecture that was basically, "You're not paid to think" kind of bullshit |
| 23:26.19 | s14ck | now I can't make call |
| 23:27.14 | jaytee | so every Optiplex SX-280 system that we ordered (over 40 of them) had to have motherboards replaced within the first year due to production bug with bad capacitors. |
| 23:27.52 | s14ck | moy i modify mfcr2 anis and and mfcr2_max_dnis |
| 23:28.01 | jsmith | s14ck: It sounds like your chan_dahdi.so module isn't loading. Try "core set debug 9", "module unload chan_dahdi.so" and then "module load chan_dahdi.so", and report any errors/warnings/notices |
| 23:28.15 | Chainsaw | jaytee: Shame really. My opinion is respected. (Which is how we ended up with DL365) |
| 23:28.46 | jaytee | my opinion isn't respected which is why I've been looking for a different job. |
| 23:29.03 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
| 23:29.41 | jaytee | tired of working for someone who micromanages what he doesn't grasp, throws temper tantrums at random and comes unglued in a crisis |
| 23:31.17 | jsmith | wanders off |
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| 23:34.16 | jblack | jaytee: This sucks,but the only thing between you and 9% unemployement is him |
| 23:34.40 | jaytee | that's true, so I'm not leaping before I look |
| 23:34.55 | jaytee | and I have to eat alot of "humble pie" if you know what I mean |
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| 23:41.25 | jasonwoot | in queues.conf, can I set more than one 'periodic-announcement' per queue? |
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| 23:46.20 | LemensTS | @jsmith: what ibm server do you recommend than |
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