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| 00:21.51 | [[thufir]] | sipsak is exactly what I was looking for. how do I know the SIP URI? I know the IP address of the server on the LAN, and can SSH in and get into the asterisk CLI -- but don't know the SIP URI. |
| 00:23.09 | WIMPy | sip:user@host |
| 00:23.59 | WIMPy | When in Asterisk Terms user is an extension. |
| 00:24.13 | [[thufir]] | right, yeah, I got that. sip show users, and pass That. |
| 00:24.16 | WIMPy | Where... |
| 00:24.59 | [[thufir]] | in the console, "sip show users" and username |
| 00:25.09 | WIMPy | No. users are accounts and not URIs. |
| 00:25.38 | WIMPy | (or part thereof) |
| 00:26.16 | WIMPy | They'd be the source URI in requests to Asterisk. |
| 00:28.12 | [[thufir]] | http://pastebin.com/n88Qf5zh for my attempt. 404 error. |
| 00:29.13 | WIMPy | see above |
| 00:32.43 | [[thufir]] | I'm sorry, I'm not understanding. a user is an account: a SIP account? Ok. The URI would be user@host and in this case the user is thufir101 and the host is the ip address (no fqdn right now). so, the uri would then be thufir101@192.168.1.2 right? |
| 00:33.37 | WIMPy | No. That would be a possible sourece URI that could be authenticated. |
| 00:34.01 | WIMPy | The user part of a destination URI is an extension as in extensions.conf. |
| 00:43.39 | [[thufir]] | ok. thinking. |
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| 00:46.50 | Mango45 | Question for anyone who's used Asterisk with PJSIP. Is it possible to negotiate the highest priority codec that both endpoints have in common, to avoid transcoding? |
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| 00:50.36 | [[thufir]] | when tkfender wrote (02:43:07 PM) [TK]D-Fender: ist is a logical separation in your dialplan this means that an "include=>vicidial-auto-phone" is just a **section** in extensions-vicidial.conf which, I'm 99% sure, is being appended or included in extensions.conf? http://pastebin.com/bhcUYZB5 for my extensions.conf (or a small bit of). |
| 00:52.11 | WIMPy | "include =>" and "#include" are very different things. |
| 00:53.37 | [[thufir]] | yes. "#include" is, I think(?) an incude, literal "put this here". for some reason it's commented out. whereas, "include =>" is saying "load this section now"? |
| 00:54.04 | WIMPy | No. It's not commented out. That's the syntax. |
| 00:54.20 | [[thufir]] | ohhh. ok. good to know. |
| 00:54.33 | WIMPy | #include inserts the contents of the referenced file as if they were placed at that position. |
| 00:54.57 | [[thufir]] | that was exactly my understanding (except I thought the # was comment, but otherwise, yes) |
| 00:55.01 | WIMPy | "include =>" adds the contents of another context to the current one. |
| 00:58.12 | [[thufir]] | in this case, when I look in extensions-vicidial.conf there's a section headed [vicidial-auto-internal] and then at the bottom include => vicidial-auto-internal which is kind of like (?) declaring "here's this function, auto-internal, let's just declare it, and then, at the bottom, ok "put this function here" add the content of this context (auto-internal) here. Or, am I off base? |
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| 00:59.19 | resno | im trying to troubleshoot an issue im having, that doesnt occur consistently. people that im calling through my trunk hear me breaking up, i can hear them just fine. |
| 01:00.16 | WIMPy | Not really. It adds the fextensions defined in the other context. However extensions directly defined in a context will always take precedence over any included oned. |
| 01:00.21 | WIMPy | ones. |
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| 01:01.44 | [[thufir]] | hmm. ok. |
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| 02:06.51 | [[thufir]] | weird, but good. I added "tleilax" to my /etc/hosts file, and now sipsak connects with 200! it works! :) |
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| 09:13.24 | makmak78 | Hi all. Is it possible to have Variables in sip.conf #include ? like #inlude /sip_{variable1}.conf |
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| 12:01.29 | RPerre | Hello there. I've been here yesterday with a question. I've posted on Asterisk forum, if any of you would like to help here is the link: http://forums.asterisk.org/viewtopic.php?f=13&t=92531&p=206568 |
| 12:01.34 | RPerre | thanks :D |
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| 12:16.00 | yun1989 | hello all |
| 12:16.06 | yun1989 | i have one doubt |
| 12:16.42 | yun1989 | it's possible this scenario Asterisk => PABX Analogic => Phone Analogic |
| 12:17.09 | yun1989 | without ATA between Asterisk and PABX |
| 12:17.13 | yun1989 | ? |
| 12:21.30 | matthew-moretalk | you could do it with an analogue card in the asterisk box |
| 12:23.25 | yun1989 | ok but for exemple I try to implement one case when exist N Analogic phones |
| 12:24.35 | yun1989 | when i make a call from asterisk to specified Analogic Phone |
| 12:24.53 | matthew-moretalk | Im unsure what you mean? |
| 12:25.22 | matthew-moretalk | you will be able to place the call down the analogue line. Its upto the PABX on how to route the call from there... |
| 12:28.21 | yun1989 | yes i don't know how PABX work |
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| 13:40.23 | matthew-moretalk | Hi whats the best way to call an async function? I dont want it to stop dialplan excution? |
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| 13:48.35 | RPerre | Hi, I have a question about ARI. I've posted on the forums: http://forums.asterisk.org/viewforum.php?f=13 any help would be very welcome thanks |
| 13:48.51 | RPerre | http://forums.asterisk.org/viewtopic.php?f=13&t=92531 |
| 13:48.56 | RPerre | sorry, bad link |
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| 13:58.04 | mjordan | RPerre: I'll be back online in an hour or so. In the meantime, it'd be helpful to see a verbose 5 trace of the CLI when you initiate your bridge recording |
| 13:58.42 | RPerre | mjordan, thans! will do |
| 13:58.45 | RPerre | thanks* |
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| 14:02.00 | RPerre | mjordan, here it is http://pastebin.com/QZXnpXVA |
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| 14:12.51 | ipalmer | nice easy one hopefully, I need to have an alphanumeric extension for someone to dial in but it says it's not found I have exten => monkeyXXXX,1 the XXXX can be any number passed in, is this possible and how |
| 14:20.42 | ipalmer | ignore its was the n in monkey causing the problem |
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| 14:47.19 | stefan27 | can local channels not be picked up with Pickup(20@context) ? |
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| 14:50.08 | stefan27 | I have [20@internal-extensions:106] Dial("Local/20:108@wx3trunk-queuedcall-000001ad;2", "Local/mollypbxdev3@internal-extensions-to-device/n,,tTkK") in new stack and then another SIP phone executes Pickup(20@internal-extensions) but gets response app_directed_pickup.c:367 pickup_exec: No target channel found for ... |
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| 15:03.52 | acidfu | when I do "sip show channelstats", the Jitter value (I get something like 0.0145) what does this value means, is it a ratio of something ? thanks ! |
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| 15:11.42 | mjordan | RPerre: so, the Verbose trace shows recording getting started on the bridge, and then something instructs the recording to stop |
| 15:12.07 | mjordan | bridge recording is done via a special channel that external APIs can't see. You can see the channel entering the bridge here: |
| 15:12.07 | mjordan | Â Â -- Channel Recorder/ARI-00000009;2 joined 'simple_bridge' stasis-bridge <5:1:515|5:2:516> |
| 15:12.11 | mjordan | And leaving it here: |
| 15:12.15 | mjordan | Â Â -- Channel Recorder/ARI-00000009;2 left 'softmix' stasis-bridge <5:1:515|5:2:516> |
| 15:12.30 | RPerre | i see |
| 15:12.36 | RPerre | nothing on my code does that |
| 15:12.38 | mjordan | What is the name of the recording you are attempting to make? |
| 15:12.44 | RPerre | debug |
| 15:12.47 | mjordan | RPerre: nope, that's the magicry of ARI |
| 15:12.53 | mjordan | k, and it creating that file: |
| 15:12.53 | mjordan | Â /var/spool/asterisk/recording/debug format: sln, (nil) |
| 15:12.59 | mjordan | The (nil) is interesting. |
| 15:13.00 | mjordan | looks |
| 15:13.11 | RPerre | what does that nil means? |
| 15:14.29 | mjordan | do you have WARNINGs going to your CLI as well? |
| 15:14.40 | RPerre | no, thats the full trace |
| 15:15.02 | mjordan | not quite what I meant :-) |
| 15:15.02 | RPerre | want to see the dialplan? |
| 15:15.06 | mjordan | nope |
| 15:15.16 | mjordan | In your logger.conf, do you have console => verbose,notice,warning,error? |
| 15:15.28 | RPerre | gimme a sec |
| 15:15.31 | mjordan | the (nil) means that the filestream failed to open for the recording |
| 15:15.38 | mjordan | which is why the recording immediately gave up |
| 15:15.47 | RPerre | OH! that might be permissions |
| 15:15.50 | mjordan | I'd expect a WARNING to get thrown out when that happens |
| 15:15.50 | mjordan | yup |
| 15:15.53 | RPerre | lemme check the conf tho |
| 15:16.30 | RPerre | console => notice,error messages => notice,warning,error |
| 15:17.08 | RPerre | oh, i never though of that |
| 15:17.17 | Mango45 | Is it possible to write sip debug to a file, one file for each call? |
| 15:17.35 | mjordan | RPerre: did the file get created? |
| 15:17.58 | mjordan | if so, then it probably isn't a permissions issue, but if you don't see the file in the location in the CLI, that's a sign that something went wrong when Asterisk attempted to write to that location |
| 15:18.04 | mjordan | Mango45: no |
| 15:18.06 | RPerre | mjordan, no. I'll change the permission of the folder (wich is owned by root not asterisk) and I'll get back to you. |
| 15:19.15 | Mango45 | mjordan, thanks. |
| 15:19.38 | RPerre | mjordan, you are absolutly right! |
| 15:19.47 | RPerre | i changed the ownsership of the folder |
| 15:19.54 | RPerre | and here it is, the recording! :D |
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| 15:20.09 | RPerre | mjordan, i owe you one. |
| 15:20.14 | RPerre | i really do |
| 15:21.23 | mjordan | not a problem |
| 15:21.34 | mjordan | glad to see ARI is (mostly) working for you :-) |
| 15:24.54 | RPerre | it is working very well! |
| 15:24.57 | RPerre | i'm loving ARI |
| 15:25.13 | RPerre | mjordan, if you need any colab just ask me |
| 15:25.28 | RPerre | I'm a programmer and sysadmin |
| 15:25.34 | RPerre | anything you need... |
| 15:26.33 | mjordan | RPerre: if you're interested in what is going on with ARI, I'd suggest subscribing to the app-dev mailing list |
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| 15:34.40 | kim_bruning | heh, I was just going "I'm singing the asterisk installation song" , and what do I see in ps? alt.asterisk.canary.tweet.tweet.tweet |
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| 15:46.05 | Kobaz | hey, anyone have a tool they would recommend for splitting out individual sip/rtp steams from a pcap into individual files |
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| 16:12.40 | ectospasm | I'm using Asterisk 1.8.15-cert2 on a busy system, and I have to request a hangup on some hung channels. Only, the "core show channels verbose" output truncates the channel name, so they're not unique in the list. Is there any way to make that column wider? |
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| 16:15.34 | Qwell | ectospasm: try concise instead of verbose |
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| 16:16.02 | ectospasm | That works, thanks Qwell! |
| 16:17.26 | kim_bruning | hum, I'm doing the asterisk hello world example on debian stable , but netstat -tulpen | grep asterisk doesn't show any open ports? |
| 16:17.46 | kim_bruning | (I'm using a clean /etc/asterisk , which SEEMS to work) |
| 16:18.59 | Qwell | kim_bruning: -t == tcp. What ports are you expecting to be open? |
| 16:19.29 | [TK]D-Fender | and what is a "clean" /etc/asterisk |
| 16:19.38 | kim_bruning | -u is udp IIRC |
| 16:19.58 | kim_bruning | means I started out mv /etc/asterisk/* /etc/asterisk/backup |
| 16:20.10 | Qwell | kim_bruning: I think they're mutually exclusive options |
| 16:20.18 | [TK]D-Fender | Nothing tells us what was ever in there to begine with |
| 16:20.30 | [TK]D-Fender | Also.... nothing proves Asterisk is RUNNING right now either |
| 16:20.42 | Qwell | netstat -tul shows only tcp ports open for me. remove the t, and suddenly udp |
| 16:20.50 | [TK]D-Fender | Perhaps you should back up a few steps |
| 16:21.23 | [TK]D-Fender | Qwell: His command worked on mine jsut fine. |
| 16:21.38 | kim_bruning | Qwell, I'm seeing other udp ports open, sooo ... anyway, rechecking hello world tut, says I need asterisk.conf and modules.conf |
| 16:21.43 | kim_bruning | snarfs and reads those |
| 16:21.50 | Qwell | [TK]D-Fender: Not in LISTEN though |
| 16:22.10 | Qwell | err, whoever said that. |
| 16:22.23 | [TK]D-Fender | udp 0 0 0.0.0.0:4569 0.0.0.0:* 499 13518 1963/asterisk |
| 16:22.35 | [TK]D-Fender | kim_bruning: If you have an empty config folder.. you're DOA |
| 16:22.41 | [TK]D-Fender | kinYou need all sorts of things |
| 16:22.48 | robmal | -tul works as you would expect it to, shows both tcp and udp. |
| 16:23.12 | Qwell | oh, weird, it shows the ports, but doesn't show UDP ports as state LISTEN, I guess. |
| 16:23.21 | kim_bruning | [TK]D-Fender, Yes, I'm currently following https://wiki.asterisk.org/wiki/display/AST/Hello+World#HelloWorld-HelloWorldwithAsteriskandSIP I'm using version 1.8.x |
| 16:23.38 | [TK]D-Fender | kim_bruning: You shouldn't be starting with 1.8 at this point. |
| 16:23.43 | [TK]D-Fender | 11 minimum |
| 16:23.46 | kim_bruning | [TK]D-Fender, so first I emptied /etc/asterisk, and then I started filling it. |
| 16:23.52 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:24.10 | kim_bruning | Upside and downside of using debian stable |
| 16:25.03 | [TK]D-Fender | kim_bruning: there are a ton of really basic necessary files in there including indications.conf, and usually some channel-driver configs as well. |
| 16:25.16 | kim_bruning | hum |
| 16:25.31 | [TK]D-Fender | kim_bruning: "If you have no configuration files in /etc/asterisk/ then grab the sample config files from the source directory by navigating to it and running "make samples"." |
| 16:25.38 | kim_bruning | Yup |
| 16:25.46 | kim_bruning | and then it says "you need at least ..." |
| 16:25.51 | kim_bruning | well, I'll try the other way |
| 16:25.56 | [TK]D-Fender | kim_bruning: That guide you're following expects you to have some of the basics taken care of as-is. Not a empty folder. |
| 16:26.01 | kim_bruning | I always hate the first day using a new piece of software ;) |
| 16:26.06 | *** join/#asterisk darkbasic (~quassel@host37-245-static.119-2-b.business.telecomitalia.it) |
| 16:26.08 | kim_bruning | Suddenly you're back to being a n00b |
| 16:26.50 | kim_bruning | okay, old conf is back |
| 16:27.02 | [TK]D-Fender | kim_bruning: that page skips "indications.conf", and "rtp.conf" which you should really have as well |
| 16:27.20 | *** join/#asterisk linocisco (~linocisco@111.84.193.196) |
| 16:27.48 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 16:28.13 | kim_bruning | does the hello world work without those? |
| 16:28.41 | [TK]D-Fender | not sure how * reacts to not having any rtp.conf at all and trying to use SIP |
| 16:29.07 | kim_bruning | okay, well, with absolute default conf, netstat doesn't show asterisk |
| 16:29.56 | robmal | Can you start asterisk -vvvvr ? |
| 16:30.12 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 16:30.15 | [TK]D-Fender | Robthta doesn't start Asterisk |
| 16:30.23 | [TK]D-Fender | robmal: that doesn't start Asterisk |
| 16:30.55 | robmal | No it doesn't, but it might be running without listening to tcp/udp ports. |
| 16:31.01 | kim_bruning | indeed |
| 16:31.25 | kim_bruning | ok, did -vvvvr ... |
| 16:31.45 | [TK]D-Fender | robmal: which would certainly fail his test |
| 16:31.57 | robmal | [TK]D-Fender: It already did ;-) |
| 16:32.05 | kim_bruning | scratches head... |
| 16:32.11 | [TK]D-Fender | robmal: Now it can fail DIFFERENT! |
| 16:32.20 | [TK]D-Fender | kim_bruning: is * RUNNING? |
| 16:32.24 | robmal | Exactly. |
| 16:32.27 | kim_bruning | yes, it is |
| 16:32.32 | kim_bruning | it shows up in ps :-) |
| 16:32.35 | robmal | Ha, 1:0 for me ;-) |
| 16:32.46 | [TK]D-Fender | kim_bruning: and when you try to connect? |
| 16:32.52 | [TK]D-Fender | (to CLI) |
| 16:32.56 | kim_bruning | it connects |
| 16:33.10 | kim_bruning | Asterisk 1.8.13.1~dfsg1-3+deb7u3, Copyright (C) 1999 - 2012 Digium, Inc. and others. |
| 16:33.10 | kim_bruning | <PROTECTED> |
| 16:33.25 | kim_bruning | I'm a little suspicious of modules.conf |
| 16:34.31 | kim_bruning | sip show settings -> no such command |
| 16:34.34 | kim_bruning | Ahuh ... |
| 16:35.02 | robmal | Ok, killall asterisk && asterisk -c |
| 16:35.51 | linocisco | hi all, I am checking which server brand and specs to choose for my asterisk setup with 4 port FXO and 4 port GSM gateway, and total 30 users inside. they will just use PSTN and GSM line. and grandstream DECT phone |
| 16:36.18 | linocisco | which cheap economy server will work? preferabbly 1U size and tower type both |
| 16:37.31 | robmal | linocisco: What server brand are you used to? |
| 16:37.37 | kim_bruning | robmal, ok, did that... get a lot of loading messages .. am I looking for something specific? |
| 16:37.52 | kim_bruning | eh, now it does have sip |
| 16:37.57 | kim_bruning | what's going on |
| 16:38.24 | linocisco | robmal, I am used to with old HP DL Gen5. it is too noisy. but here it is not big office setup. and want to minimize cost |
| 16:38.26 | robmal | kim_bruning: Start another terminal and check netstat. |
| 16:39.25 | kim_bruning | robmal, yup, works ! |
| 16:39.28 | kim_bruning | so what's going wrong? |
| 16:39.41 | kim_bruning | or what's going wrong before (when I was using /etc/init.d ?) |
| 16:39.57 | kim_bruning | and my phone is now registered too |
| 16:40.31 | kim_bruning | and it just said "HELLO WORLD!" \o/ |
| 16:41.09 | robmal | kim_bruning: I'm sorry, i was hoping it wouldn't work as well ;-) It seems your config is ok, there might be something f**ked up with the startup scripts. |
| 16:41.13 | [TK]D-Fender | linocisco: Practically anything you can buy off the shelf could do the job |
| 16:41.16 | kim_bruning | Joy |
| 16:41.33 | kim_bruning | you'd expect debian stable to be -like- stable |
| 16:42.27 | kim_bruning | ok, so doing service asterisk restart borks it again |
| 16:42.28 | robmal | linocisco: HP is a great choice. If its too loud move it to a dedicated room. If you need realy tiny PBX, you can try RPi implementations, but that depends on your needs. |
| 16:43.14 | linocisco | robmal, I checked grandstream UCM6100 series, but that lack billing software feature |
| 16:43.37 | [TK]D-Fender | linocisco: Build a cheap PC. Anything will do. |
| 16:43.57 | linocisco | robmal, if i choose device based, low power, less space, less noise, all GUI integrated. |
| 16:44.06 | *** join/#asterisk marceloamorim (~marcelo@189-90-194-34.isimples.com.br) |
| 16:44.09 | kim_bruning | robmal, well, time for me to read the startup scripts |
| 16:44.13 | kim_bruning | not my plan for today :-P |
| 16:45.06 | linocisco | linocisco, for server, here HP HP DL320e Gen8 1U size Hot Plug 8SFF CTO Server cost 4654 USD, and too much for this small setup |
| 16:45.33 | Kobaz | hey tk |
| 16:45.35 | linocisco | can anybody suggest cheap server or |
| 16:45.36 | linocisco | ? |
| 16:45.46 | robmal | kim_bruning: At least you know your config works and you can run it, now its only a matter of running it on startup. |
| 16:45.47 | linocisco | acer is also good? |
| 16:45.49 | [TK]D-Fender | [11:43][TK]D-Fenderlinocisco: Build a cheap PC. Anything will do. |
| 16:46.01 | Kobaz | know any good tools for splitting out sip calls from giant pcap files (sip+rtp into individual files) |
| 16:46.07 | robmal | linocisco: Supermicro is cheap and reliable. |
| 16:46.17 | Kobaz | +1 supermicro |
| 16:46.24 | Kobaz | ipmi ftw |
| 16:46.29 | linocisco | robmal, is it chinese brand? |
| 16:46.36 | linocisco | robmal, i have never used that |
| 16:46.40 | Kobaz | it's all from china |
| 16:46.46 | Kobaz | your hp and dells are made in china |
| 16:46.56 | Kobaz | probably at the same factory as supermicro |
| 16:46.58 | coppice | "Supermicro is cheap" is something you don't often hear |
| 16:46.59 | rrittgarn1 | +1 for most supermicro boards / systems |
| 16:47.08 | rrittgarn1 | also true on the not always cheap |
| 16:47.26 | kim_bruning | robmal, which is kinda important |
| 16:47.30 | Kobaz | rrittgarn1: if you get a supermicro server that can do 1tb of memory and 4 processors, then yeah, it's not cheap |
| 16:48.12 | linocisco | supermicro which model is ok for my setup with 30 user max? |
| 16:48.14 | robmal | kim_bruning: It's just cheap good hardware. I don't have too much to say about the support, but from what i heard its acceptable. |
| 16:48.31 | robmal | For 30 users max you can use raspberry pi. |
| 16:48.35 | robmal | It's dead silent. |
| 16:48.53 | kim_bruning | robmal, I think you were answering some else ;-) |
| 16:48.59 | coppice | HP and Dell don't usually use people like Supermicro to make their stuff. They go for the cheapest option |
| 16:49.19 | kim_bruning | [TK]D-Fender, I built a cheap pc in a 19" case .... note that case width and height is standardised, but not case depth ... |
| 16:49.28 | robmal | Supermicro is to HP as is Asus to Dell ;-) |
| 16:49.42 | kim_bruning | (so now it's sitting on top of my cabinet until I have enough round tuits to buy a new cabinet :-p) |
| 16:50.42 | kim_bruning | robmal, the command in ps is listed as /usr/sbin/asterisk -p -U asterisk |
| 16:50.50 | kim_bruning | robmal, is that maybe a clue? |
| 16:51.15 | linocisco | robmal, we dont have RasPI device locally available to buy. moreover, it will take time to compile and less flexiblity to grow for future or add future recordings which will need more HDD space |
| 16:51.17 | robmal | kim_bruning: Check user/group on /etc/asterisk, maybe the daemon can't read them on startup. |
| 16:51.40 | kim_bruning | robmal, great minds think alike |
| 16:52.09 | robmal | Well, that's sure ain't me ;-) |
| 16:52.18 | robmal | linocisco: So go with HP. |
| 16:53.22 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-cllnqmmugltdxdbx) |
| 16:53.34 | kim_bruning | robmal, maybe it' my fault, since I HAD been messing with that dir before working on it |
| 16:53.41 | linocisco | robmal, I am using HP Desktop CMT Elite 8300 and8200 daily as desktop . no problem. Can I install asterisk for my production with 30 user max? |
| 16:53.43 | kim_bruning | BINGO! |
| 16:53.48 | kim_bruning | hugs robmal |
| 16:53.53 | robmal | linocisco: Sure. |
| 16:54.03 | kim_bruning | I do feel stupid now. It's all just permissions stuff, and wasn't even asterisk |
| 16:54.15 | robmal | kim_bruning: Always a pleasure to help ;-) |
| 16:54.21 | kim_bruning | HELLO WORLD! |
| 16:54.36 | kim_bruning | That's what the phone said to me ;-) |
| 16:54.41 | adoodle | i'm having trouble getting my odbc cdr's to write to the database....is there any troubleshooting docs? for example, is there a way to get * / odbc to log out the SQL statement it's trying? |
| 16:54.47 | linocisco | robmal, asterisk 32bit is ok though ? |
| 16:54.56 | robmal | kim_bruning: If you need anything - msg me ;-) |
| 16:54.56 | kim_bruning | Ok, well once again into the breach |
| 16:55.14 | robmal | linocisco: Sure, why not? |
| 16:55.50 | kim_bruning | so now I need to figure out how to do this for reals |
| 16:55.58 | linocisco | robmal, because most server Linux recommend 64bit, I am doubting whether I can install asterisk 32bit on 64bit Linux OS. |
| 16:56.15 | robmal | You can, always. |
| 16:56.23 | robmal | Just instal ia32-libs along the way. |
| 16:56.56 | linocisco | robmal, ia32-libs? I am not quite sure what it is |
| 16:57.19 | *** join/#asterisk Valduare (~Valduare@108-198-116-80.lightspeed.mdsnwi.sbcglobal.net) |
| 16:57.23 | linocisco | robmal, i will find GUI based which include billing software |
| 16:57.37 | robmal | Shared libs, to run 32bit software on 64bit architecture. |
| 16:58.02 | robmal | linocisco: If you would skip the hard 'billing' part i'm sure i could help ;-) |
| 16:58.58 | linocisco | robmal, i see. I am not sure which one (Elastix or Vicidial or Freepbx)include billing feature. It is most important feature every company wants |
| 16:59.06 | *** join/#asterisk imox (~imox@p4FC5C6F2.dip0.t-ipconnect.de) |
| 16:59.11 | kim_bruning | kay, it's closing time at this office |
| 16:59.15 | kim_bruning | I'll have to continue in the morning |
| 16:59.17 | kim_bruning | laters! |
| 17:00.26 | robmal | linocisco: As long as you only need billing time, without adding money to the equation (prepaid/postpaid, free minutes/seconds etc.) i'm sure i can help. |
| 17:02.36 | linocisco | robmal, Billing software is not for calling card business. Just for management purpose, to control unnecessary outgoing calls of each staffs. Just to see how many calls per month, Incoming, outgoing, from which extesions, users, and how long and how much it will cost. We will define rates first |
| 17:03.31 | robmal | I'm aware of that, only the cost factor is complicated. |
| 17:04.27 | robmal | Everything else is pretty much 'excel-able' |
| 17:05.46 | *** join/#asterisk marceloamorim (~marcelo@189-90-194-34.isimples.com.br) |
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| 17:53.33 | *** join/#asterisk Geriatrix (~kvirc@mail.vancouveronthenet.com) |
| 17:53.53 | Geriatrix | hey guys - can someone give me a hand with alsa MoH ? |
| 17:54.05 | pabelanger | ~ask |
| 17:54.05 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:54.45 | Geriatrix | i have /bin/ast-playlinein script in custom MoH - however when executed i get monmp3thread: poll() failed: Interrupted system call |
| 17:55.09 | Geriatrix | the script executes /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw |
| 17:57.29 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 17:57.44 | SpaceInvaders | If I plug in an OBI110 and set the Proxy Server: to my asterisk IP address do I need to do anything on asterisk or does it magically "autoconfigure" to support that OBI110? FYI I have tried Google. I found instructions on config for the OBI110 but haven't found anything on configuring asterisk. |
| 18:00.10 | [TK]D-Fender | ~book |
| 18:00.10 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:00.13 | [TK]D-Fender | ^^^ |
| 18:00.22 | [TK]D-Fender | Thre is nothing "automatica" about configuring Asterisk |
| 18:00.39 | [TK]D-Fender | How you process calls from it, or send calls to it are up to you |
| 18:00.54 | [TK]D-Fender | You'll need to setup SIP device definition, configure your dialplan, etc |
| 18:02.28 | *** join/#asterisk roramirez (~rodrigo@201.220.242.180) |
| 18:02.38 | SpaceInvaders | OK I have instructions for that which I read through when I installed asterisk. I will go through that again to refresh my memory. |
| 18:02.54 | SpaceInvaders | Thanks! |
| 18:10.43 | *** join/#asterisk d00gster (~doughant@184.151.190.51) |
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| 19:00.57 | Geriatrix | can anyone assist me with monmp3thread: poll() failed: Interrupted system call |
| 19:01.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
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| 20:08.30 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 20:08.41 | *** join/#asterisk madax (~madax@c187-193.i06-15.onvol.net) |
| 20:11.20 | roler | Hey guys - trying to get a Sangoma B601DE working with Time Warner PRI. Everything points to a successful connection, yet incoming and outgoing calls don't work. dahdi show status shows no errors, Alarms says OK. running wanpipemon -i w1g1 -c Ta shows no alarms either. I called TIme Warner and they said "During the communication sequence there are a certain number of steps. one of those steps is not complete". I don't know how to parse logs or run |
| 20:11.20 | roler | <PROTECTED> |
| 20:12.21 | WIMPy | pri set debug ... |
| 20:25.36 | roler | WIMPy; thanks (again). I used pri set debug on (and got an error) but after googling, pri set debug on span 1 starts showing some output. Is this adequate ? |
| 20:27.02 | WIMPy | I don't know if it is to you. I prefer to use set debug hex and use tracI to decode the result. |
| 20:28.32 | roler | WIMPy; I am seeing this a lot. Does this mean anything to you ? http://pastebin.com/iBtKsVrC |
| 20:29.53 | WIMPy | Line 11 tells you the two endpoints don't agree on the state of the connection. |
| 20:38.13 | roler | WIMPy; thanks. That returned some google results... Will keep plugging away |
| 20:47.55 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
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| 21:34.52 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 21:35.34 | roler | WIMPy; just wanted to chime in and say it's working. for the most part. I can get incoming calls, not outgoing. I am seeing "Cause: Bearer capability not authorized (57), class = Service or Option not Available" ... so close :) |
| 21:35.50 | roler | at least I know the circuit is up, I can play around with it this weekend |
| 21:36.51 | WIMPy | Ooops. Are you requesting additional features? Seems unlikely that you're not allowed to make speach calls. |
| 21:37.27 | roler | hmm.... |
| 21:38.27 | WIMPy | Err. Speech. Need to get that right. |
| 21:45.56 | roler | WIMPy; I am trying not to be a bother too much here - been searching for where these additional features may be set. I tried dialing out using my SIP app, and have the same issue so I don't think my desk VoIP phone is requesting additional features... Is there a setting or option somewhere? |
| 21:46.06 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-jdgfepviqlyitszl) |
| 21:46.36 | WIMPy | It can't it would be your asterisk configuration, if any. |
| 21:47.50 | WIMPy | Look at the actual request sent. |
| 21:52.23 | roler | Ok, I see a line that says "Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) \n Ext:1 Trans/mode/rate 64kbps, circuit-mode (16) \n User information layer 1: u-Law (34) |
| 21:52.35 | roler | does (0) mean it's not supported? Or 0 is just the identifier of Speech? |
| 21:52.48 | WIMPy | The later. |
| 21:52.55 | WIMPy | So that part ist surely not the issue. |
| 21:54.41 | roler | yea... I'm just having trouble finding the line that tells me what the phone itself is requesting |
| 21:54.46 | roler | there is a lot of info in intense mode :) |
| 21:55.21 | *** join/#asterisk Shaan (~shaan@unaffiliated/shaan) |
| 21:55.50 | WIMPy | The phone part is not of interest here. |
| 21:57.04 | Shaan | Hi guys im being given some misleading information im told i can use anoter port for the sip phone to connect since port 5060is blocked on my clients ISP so i went ahead changed it to another port but obviously the phone is not registering so that leads me to ask i obviously haveto change some settings in asterisk for it to accept sip registerations on another port other then 5060 |
| 21:57.36 | robmal | Indeed. |
| 21:57.54 | Shaan | robmal: so can i have 2 ports used lets say like 5060 and 5090 |
| 21:58.07 | Shaan | because obviously my carrier only accepts traffic on 5060 |
| 21:58.12 | WIMPy | No |
| 21:58.22 | WIMPy | Not from Asterisk that is. |
| 21:58.29 | WIMPy | You could do some iptables magic. |
| 21:58.38 | Shaan | WIMPy: explain please? |
| 21:58.55 | robmal | There is port=xxxx in [general] which sets the port. |
| 21:59.10 | roler | WIMPy; is it this? Called Party Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'XXXXXXXXXX' ] ? |
| 21:59.48 | WIMPy | roler: pastebin the whole SETUP message. |
| 21:59.56 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:00.29 | Shaan | WIMPy: so if i want to use multiple ports for accepting sip traffic my only solution is Iptables? |
| 22:00.39 | Shaan | WIMPy: so can you point me in the right direction please |
| 22:01.55 | WIMPy | Shaan: Yes, "REDIRECT" |
| 22:02.07 | Shaan | WIMPy: awesome thanks! |
| 22:03.24 | Shaan | WIMPy: iptables -t nat -A PREROUTING -i eth0 -p upd --dport 8887 -j REDIRECT --to-port 5060 |
| 22:03.39 | Shaan | does that look correct to you?, and do i do samething for outgoing too? |
| 22:04.43 | WIMPy | yes |
| 22:04.48 | WIMPy | And no. |
| 22:05.22 | WIMPy | YOu might have to explicitely configure the other port as proxy in your client. |
| 22:05.38 | Shaan | other then the client aspect ie the softphone |
| 22:05.48 | Shaan | do i haveto add anything else to my Firewall? |
| 22:06.02 | WIMPy | no |
| 22:06.23 | robmal | Shaan: Without -d WAN_IP it will surely fu*k up your internet usage :-) |
| 22:06.29 | WIMPy | The answers will come from 5060. |
| 22:07.05 | WIMPy | robmal: I think the -i will do the same. |
| 22:07.18 | Shaan | robmal: come again pleasE? |
| 22:07.22 | robmal | WIMPy: If you have 1 IP - sure. But why take the risk? |
| 22:07.27 | Shaan | WIMPy: pardon answers from 5060? |
| 22:07.40 | *** join/#asterisk moke (~moke@unaffiliated/moke) |
| 22:07.44 | robmal | Also, go with -j DNAT --to .... |
| 22:07.50 | Shaan | i need to send output traffic backout via port 8887 |
| 22:08.10 | WIMPy | That won't happen. |
| 22:09.10 | WIMPy | I don't know if chan_pjsip offers additional possibilities here. |
| 22:09.25 | Shaan | WIMPy: are you reffering to sending traffic out via 8887? |
| 22:09.41 | WIMPy | yes |
| 22:09.56 | Shaan | so how will it know where tosend trafic back to if sip client registers on 8887 |
| 22:10.04 | Shaan | wont it have a routing issue? |
| 22:10.08 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 22:10.36 | WIMPy | It depends on the client and its configuration. |
| 22:10.47 | roler | WIMPy; http://pastebin.com/bg2uiDes |
| 22:11.23 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 22:11.31 | WIMPy | Receiving answers from another port could be filtered, while transferred URIs might cause it to change port itself for the next request. |
| 22:12.30 | Shaan | so WIMPy what do you suggest my client is on a non voip friendly ISP |
| 22:12.35 | Shaan | how can i make this work |
| 22:13.07 | WIMPy | shaan: It's probably the Display element that isn't allowed. |
| 22:13.26 | WIMPy | Shaan: Use IAX instead. |
| 22:13.41 | WIMPy | sorry |
| 22:13.50 | WIMPy | roler: It's probably the Display element that isn't allowed. |
| 22:14.00 | Shaan | nope IAX wont work its Pakistan there call centers apparently always have issues |
| 22:14.02 | WIMPy | roler: Also your caller ID isn't valid. |
| 22:15.03 | WIMPy | Shaan: Didn't you say softphones? |
| 22:15.10 | Shaan | WIMPy: yes its a softphone |
| 22:15.14 | Shaan | x lite |
| 22:19.07 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 22:20.24 | Shaan | WIMPy: can you tell me if i need to do anyting else or note please? |
| 22:23.24 | WIMPy | Maybe pray? |
| 22:25.26 | Shaan | lol |
| 22:33.11 | AnonGirl | lol |
| 22:47.20 | roler | WIMPy; caller id was the ticket. thank you so much! |
| 22:49.31 | WIMPy | Bad error messages from bad software :-( |
| 22:59.22 | gusto | WIMPy, hi |
| 22:59.55 | gusto | WIMPy, I discovered that I have one bad sector (or more precisely an unreadable sector) on my 2 TB external USB HDD |
| 23:01.17 | roler | WIMPy; I agree. I'm using Elastix. I find the interface a little clunky to use, especially when setting up IVR's and what not. But when I have to go to command line and debug stuff, that's when I think you'll lose most customers. Of course, asterisk is free so no one has a right to complain :) |
| 23:06.10 | WIMPy | roler: I was referring to the software on your telcos switch. |
| 23:06.54 | roler | WIMPy; heh... those messages were specific to time warner? I assumed the debugging messaging was part of asterisk |
| 23:08.12 | AnonGirl | Shaan: are you alex bkash? |
| 23:08.43 | WIMPy | roler: The messages, but not the contents. |
| 23:09.44 | roler | WIMPy; got it. |
| 23:09.55 | WIMPy | roler: "Cause: Bearer capability not authorized (57)" that was the message from the CO, just decoded by Asterisk. |
| 23:10.00 | WIMPy | And it was clearly wrong. |
| 23:15.51 | *** join/#asterisk infobot (ibot@rikers.org) |
| 23:15.51 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.2.0 (2015/02/06), 11.16.0 (2015/02/06), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 23:39.45 | Shaan | AnonGirl: no, but i can be if you want :-) |
| 23:45.36 | AnonGirl | Shaan: lol |
| 23:52.08 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
| 23:55.27 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |