| 00:09.09 | *** join/#asterisk serafie (~erin@user-24-214-173-250.knology.net) |
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| 00:09.59 | kfife | hey guys. |
| 00:10.59 | kfife | After upgrading from 11.0.0 to Certified 11.2. I get the following error on startup |
| 00:11.00 | kfife | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 00:11.16 | kfife | However everything seems to run fine. |
| 00:12.19 | kfife | I've updated the /etc/asterisk/asterisk.ctl to refer to the asterisk.ctl pipe in its new location, and it does exist. |
| 00:12.23 | kfife | Stumped. |
| 00:17.02 | [TK]D-Fender | it shouldn't have a new location |
| 00:17.09 | [TK]D-Fender | it loks where asterisk.conf tells it to.... |
| 00:17.22 | [TK]D-Fender | Go check your configs ... and then verify that asterisk is even starting. |
| 00:20.08 | [TK]D-Fender | heads out for a while... |
| 00:25.59 | kfife | if i restart the service, asterisk is running even after the error. Configs match the actual location of /var/run/asterisk.ctl |
| 00:26.23 | kfife | I don't see any similar complaints in /var/log/asterisk/messages |
| 00:26.59 | kfife | [TK]D-Fender: (sorry, back now) |
| 00:28.43 | kfife | I actually get two similar messages on service restart. Both the aformentioned one, and then on the next line I get |
| 00:28.44 | kfife | Starting asterisk: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 00:34.16 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 00:42.28 | kfife | so this is what it's like to have a threesome? |
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| 00:51.55 | Shariff | Hi there |
| 00:52.55 | Shariff | I have setup a new installation of asterisk 11 according to a tutorial and am trying to contact an extension which no client has registered to yet, hoping to get voicemail.. What I got was no dial tone.. how do get a dial tone? |
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| 00:53.13 | Shariff | It is supposed to be a voip-only setup |
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| 01:18.07 | puzzled | anyone know if MeetMe sets a var with exit status when a MeetMe fails e.g. due to wrong pin? |
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| 02:31.03 | Spengler1 | does anyone here us asterisk behind a nat with sip clients connecting externally? |
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| 03:07.28 | Spengler1 | does anyone here us asterisk behind a nat with sip clients connecting externally? |
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| 03:35.59 | p7ank5te7 | Is there any free dialplan builders out there that are relatively easy to use at all? I know apstel has a paid version. I'm just looking for something relatively simple, but I still don't quite get all the context and stuff like that and keep failing miserably. I think reading has helped but not 100%. Seeing a working config that I can build onto would help. |
| 03:44.21 | WIMPy | [sr]: Looks like they have some interesting stuff going on there. I wonder what they do in that BRI to PRI scenario. |
| 03:49.21 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 03:49.58 | WIMPy | And to all: That thing obviousely IS a SIP gateway. |
| 03:53.24 | WIMPy | (so that answers my question) |
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| 07:47.04 | Utente | Hi guys |
| 07:48.50 | drenda | I've a simple question, I think. I'm using Asterisk 11.4 with tls and srtp. Before I used upd and I configured qos. Now with tls and sips which ports are used? So I can set an appropriate qos policy. Thanks |
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| 07:59.09 | phillcz | Hi, |
| 07:59.09 | phillcz | I ran into some difficulties with my asterisk, I'm hoping someone could help. |
| 07:59.09 | phillcz | In a sip INVITE from a SIP provider I receive header: Contact: "Anonymous"<sip:203.160.8.74:5061> |
| 07:59.09 | phillcz | Asterisk fails to parse it and further requests sent out by asterisk are malformed: Contact: <sip:203.160.8.74>@10.1.1.250:5060> |
| 07:59.09 | phillcz | Have I found an asterisk bug or is it a malformed header? I can't find the space in the RFC3261. It seems like it shouldn't even be there. |
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| 08:09.26 | bulkorok | guten morgen |
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| 08:23.05 | XnOSX | hello, I have a question, how I can set two conditions in a GotoIf, Im triying separate with &, &&, (,), etc etc etc and nothing work! |
| 08:23.27 | XnOSX | I need to validate two conditions in a GotoIf |
| 08:24.27 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 08:26.35 | phillcz | GotoIf($[$["${TMP_FALLBACK_NUMBER_CLEANUP}" = "10"] | $["${TMP_FALLBACK_NUMBER_CLEANUP}" = ""]]?busy) |
| 08:26.56 | XnOSX | phillcz: but this is an OR condition isnt it? |
| 08:27.24 | XnOSX | I need set an AND conditions to validate two conditions at the same time |
| 08:27.43 | phillcz | ten use & |
| 08:28.04 | XnOSX | GotoIf (both conditions are true) |
| 08:28.29 | XnOSX | I already set with &, && etc but does not work correctly |
| 08:28.38 | kaldemar | XnOSX: show what you did. |
| 08:28.56 | XnOSX | kaldemar: the code? |
| 08:29.09 | XnOSX | ok hold on |
| 08:29.11 | kaldemar | XnOSX: and the CLI output |
| 08:30.16 | XnOSX | the code http://pastebin.com/CgLfWZkS |
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| 08:30.57 | kaldemar | $["${var1}" = "123" & "${var2}" = "321"] would also do, i.e. without surrounding both expressions with $[]. |
| 08:31.23 | kaldemar | "${FWD-NUM:0:1}" = 9 <--- the quotes are literal characters |
| 08:31.53 | XnOSX | the CLI output http://pastebin.com/tfp70mp1 |
| 08:33.07 | XnOSX | I need to validate two conditions 1: the start number of the exten (start with 9 number) & have a 9 digits of lench |
| 08:33.42 | XnOSX | example: if 9XXXXXXXX Goto fwd-rout-to-landline |
| 08:33.58 | kaldemar | the syntax error is for an empty variable, FED-NUM |
| 08:34.22 | XnOSX | and the second GotoIf that I need to set is: If 1XX Goto fwd-rout-to-internal |
| 08:34.37 | kaldemar | why are you not using patterns for this? |
| 08:34.38 | XnOSX | 3 digits and start with 1 number |
| 08:35.32 | XnOSX | hooo this is an error by me, sorry must be FWD no FED |
| 08:35.55 | kaldemar | also, your CLI output does not match the extension in your pastebin. |
| 08:36.50 | XnOSX | seems to work now, hold on |
| 08:39.04 | XnOSX | kaldemar: already work perfect |
| 08:39.06 | XnOSX | ;) |
| 08:39.22 | XnOSX | same => n,GotoIf($[$[${FWD-NUM:0:1} = 9] & $[${LEN(${FWD-NUM})} = 9]]?fwd-rout-to-landline,1) |
| 08:39.35 | XnOSX | same => n,GotoIf($[$[${FWD-NUM:0:1} = 1] & $[${LEN(${FWD-NUM})} = 3]]?fwd-rout-to-internal,1) |
| 08:40.13 | XnOSX | my mistake was in the declaration of the variable |
| 08:40.31 | XnOSX | already fixed and now work correctly, thank you so much |
| 08:41.59 | kaldemar | $[${FWD-NUM:0:1} = 1 & ${LEN(${FWD-NUM})} = 3] would also do if you want to keep it simpler. |
| 08:42.45 | kaldemar | if you use quotes on both sides on the first comparison, you won't see a parser error in a case when FWD-NUM is empty. |
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| 09:03.12 | XnOSX | kaldemar: excellent Ill change it! |
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| 09:19.25 | [sr] | WIMPy: ya, the ISDN over IP may be useful to integrate older pbx's that use digital telefones that only work with that PBX |
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| 10:33.53 | drenda | I've a simple question, I think. I'm using Asterisk 11.4 with tls and srtp. Before I used upd and I configured qos. Now with tls and sips which ports are used? So I can set an appropriate qos policy. Thanks |
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| 10:42.41 | kaldemar | drenda: did you configure asterisk yourself? |
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| 10:43.43 | drenda | kaldemar, yes |
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| 10:44.27 | phillcz | Hi, I have a question - how reliable is CALLERID(num) in general? Do you use it or do you rather parse the FROM sip header? |
| 10:45.06 | Greenlight | I use it, and never have issues with it |
| 10:45.18 | Greenlight | I think some carriers can be odd though |
| 10:46.11 | kaldemar | drenda: if you did not specify a port in tlsbindaddr, the default port (5061) is used for SIP. |
| 10:46.56 | drenda | kaldemar, but that port is used also for media stream (sRtp)? |
| 10:47.05 | kaldemar | drenda: why? |
| 10:47.29 | phillcz | In my current asterisk configuration I have some kind of magic with the headers. A lot of cutting, etc. Unfortunately it was coded by a former employee and it fails with some sip providers. I was thinking perhaps the callerid(num) might be fine nowdays. |
| 10:47.52 | drenda | kaldemar, for implement policy for qos in order to prioritize voip traffic |
| 10:48.35 | kaldemar | drenda: 5061 is not used for SRTP unless you configure it so. |
| 10:49.20 | kaldemar | drenda: so use other ports for SRTP, just like when unencrypted. |
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| 10:52.22 | drenda | kaldemar, but which are the default ports used in srtp? For example in unencrypted traffic I know that are upd ports from 5000 to 15000 |
| 10:53.03 | kaldemar | drenda: 10000-20000 is the default range, not 5000-15000. the range is configured in rtp.conf. |
| 10:55.21 | drenda | kaldemar, yes you're right I modified that in my conf file. But I can't see a range for encrypted rtp traffic. |
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| 10:59.06 | skrusty | can anyone recommend a sip provider for US numbers? I am located in the UK. |
| 10:59.12 | kaldemar | drenda: both use the same range, iirc. |
| 10:59.37 | drenda | kaldemar, in udp o tcp? |
| 11:00.00 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:00.30 | drenda | kaldemar, with unencrypted I know that is UDP...but with sRTP traffic? |
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| 11:01.36 | sekil | srtp is udp no? |
| 11:03.29 | kaldemar | drenda: udp |
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| 11:05.14 | drenda | kaldemar, thanks very much |
| 11:05.30 | drenda | sekil, thanks also to you |
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| 11:18.24 | puzzled | skrusty: have a look at flowroute.com or voip.ms |
| 11:19.47 | skrusty | cheers |
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| 11:44.14 | geeksteve | # asterisk -rx'core show uptime' |
| 11:44.14 | geeksteve | System uptime: 4 years, 17 weeks, 2 days, 38 minutes, 37 seconds |
| 11:44.17 | geeksteve | is that a record? |
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| 11:52.17 | biomorph | geeksteve: Beats me |
| 11:52.19 | biomorph | System uptime: 3 days, 1 hour, 2 minutes, 55 seconds |
| 11:55.18 | geeksteve | well that's a start ;) |
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| 12:15.48 | bulkorok | geeksteve: well.. that tells me that you don't updates since 4 years... |
| 12:16.07 | geeksteve | in the situation it's used in, that isn't so much a problem |
| 12:16.13 | geeksteve | but general end-user use - yea that'd be bad |
| 12:16.33 | bulkorok | geeksteve: and, a lot of luck that there wasn't a segfault... yeah.. lots of luck... |
| 12:16.39 | geeksteve | 1.4 as well! |
| 12:16.50 | geeksteve | i must have the only stable 1.4 box that doesn't leak memory lol |
| 12:17.18 | bulkorok | well... maybe 1 call a day is possible for over 4 years :-D |
| 12:18.06 | geeksteve | averages about 15 concurrent calls, but can be anything between 0 and 30 in reality |
| 12:18.20 | geeksteve | I'm rather impressed - and too scared to touch it ;) |
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| 12:42.35 | *** join/#asterisk lukhas (~lucas@bearstech/lukhas) |
| 12:42.40 | lukhas | hello |
| 12:43.18 | lukhas | I was wondering if there was an easy way, in the dialplan, to know which SIP account/channel is emitting the call |
| 12:44.32 | lukhas | $CHANNEL is very helpful, but it includes channel identifier at the end, my dialplan would be cleaner if I could just use a variable, instead of having to match parts of it :) |
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| 12:55.21 | [TK]D-Fender | lukhas: So chop of the end part. |
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| 12:55.33 | lukhas | yes that's what I'm doing right now |
| 12:55.33 | [TK]D-Fender | lukhas: or use SeVar in your SIP entry |
| 12:55.39 | [TK]D-Fender | SetVar* |
| 12:56.19 | *** join/#asterisk serafie (~erin@nat/digium/x-iqxfvaljvunwiifh) |
| 12:56.27 | lukhas | ah, where can I use SetVar this way? |
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| 12:56.46 | seik0 | Hi, guys. We had a talk few months ago about one problem. Problem is that if we use SIP Registry for incoming calls from outside and loosing connection to outside, then asterisk is stuck (seems, trying to get registry) |
| 12:56.59 | [TK]D-Fender | [08:55][TK]D-Fenderlukhas: or use SeVar in your SIP entry <- |
| 12:57.00 | seik0 | here was said, that the problem is in dns |
| 12:57.30 | seik0 | and i tried to statically fix dns names needed for registry |
| 12:57.34 | seik0 | but that failed |
| 12:59.25 | lukhas | [TK]D-Fender: I'll have a look, thanks |
| 12:59.49 | lukhas | though I'm not sure our system is configured to have some code executed for every SIP account |
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| 13:06.26 | [TK]D-Fender | lukhas: This is a parameter for your peer ... not "code" |
| 13:06.38 | [TK]D-Fender | lukhas: Go read the sip.conf sample |
| 13:06.46 | lukhas | ah, ok |
| 13:06.50 | kaldemar | lukhas: ${CHANNEL(peername)} |
| 13:06.55 | lukhas | now that makes sense :) |
| 13:08.39 | [TK]D-Fender | There's that too... |
| 13:10.51 | seik0 | probably, i have the same issue: https://issues.asterisk.org/jira/browse/ASTERISK-14675 but i'm not sure. we have asterisk 1.8, but on jira it's not clear in which version bug was fixed (if really was) |
| 13:10.53 | LieutPants | [ASTERISK-14675] [Status: Closed] chan_sip lockup by DNS and connect timeouts - https://issues.asterisk.org/jira/browse/ASTERISK-14675 |
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| 15:11.07 | BorjaGVO | Hi. We want a system holding around 80 queues. It is working with FreePBX on top of it. We would like to study the viability of transtition from this to a vanilla Asterisk platform. I would like to know if anyone has experience on this and what tools can be useful here. For example, FreePBX makes way easy to add members to queues. Waht approach do you take to manage big systems just by using Asterisk? |
| 15:12.47 | [TK]D-Fender | BorjaGVO: vi <- |
| 15:12.50 | *** join/#asterisk Shariff (~chatzilla@2001:980:167b:1:4de0:d755:2410:d9bd) |
| 15:12.53 | Shariff | Hi there.. |
| 15:13.21 | ChannelZ | FreePBX doesn't make Asterisk do anything it couldn't already. |
| 15:13.22 | [TK]D-Fender | BorjaGVO: of vim, emacs, nano, pic, gedit, kate, Writer, mc, or whatever |
| 15:13.26 | Shariff | How do I perform a SIP test to see if a phone is in fact registered? Right now my phone gui tells me it is, but when I place a call there is no dial tone and I get the message (on the phone) that VOIP is not available... |
| 15:13.40 | BorjaGVO | [TK]D-Fender: Really? Just vi? I mean...it can get unmanagable when growing |
| 15:13.45 | [TK]D-Fender | BorjaGVO: And it isn't a transition as much as a "hard turn left" |
| 15:13.56 | [TK]D-Fender | BorjaGVO: Who says "unmanageable? |
| 15:14.21 | [TK]D-Fender | BorjaGVO: You haven't described a specific scenario I would call unmanageable yet. |
| 15:14.39 | Greenlight | BorjaGVO: I think people have used databases to generate the conf files as well as AMI to add remove memberrs |
| 15:14.57 | *** join/#asterisk navaismo (~navaismo@189.191.246.128) |
| 15:15.22 | [TK]D-Fender | Greenlight: Which in the end is really the same work... jsut a question of storage.... |
| 15:15.39 | Greenlight | Same work as what? |
| 15:15.41 | [TK]D-Fender | Greenlight: Debateably more work in fact |
| 15:15.46 | [TK]D-Fender | Greenlight: text config |
| 15:16.03 | Greenlight | I would not let half my call centre managers anywhere near vi on my servers. |
| 15:16.15 | Greenlight | I would let them use a web frontend |
| 15:16.39 | [TK]D-Fender | Greenlight: Oh now you're being picky about who you give access to? :p |
| 15:16.58 | Greenlight | Well I was reading into his situation :) |
| 15:17.10 | [TK]D-Fender | Greenlight: I'm comparing support load, not what level of idiot to accommodate ;) |
| 15:17.34 | [TK]D-Fender | Greenlight: I think you've moved somewhat into "writing between the lines" territory.... |
| 15:17.45 | Greenlight | Yea, I do that :) |
| 15:17.48 | lukhas | [TK]D-Fender: $CHANNEL(peername) does the trick, thanks! (2hrs late) |
| 15:18.14 | BorjaGVO | Greenlight: What procedure would you use to generate conf files from databases? |
| 15:18.26 | Greenlight | The lanuage of your choice |
| 15:18.46 | Greenlight | ALso a lot depends on your needs |
| 15:19.46 | Greenlight | If you want a db of sip peers for your users, you could have a script that reads though it and outputs the defenitons to sip_dbpeers.conf, which is then included inside your handwritten sip.conf |
| 15:20.07 | Greenlight | If you update a password, for example, the script regenerates the file and executes a sip reload |
| 15:20.35 | [TK]D-Fender | Of you could jsut USE a database for your * configs. |
| 15:20.47 | [TK]D-Fender | Instead of generating flat-files. |
| 15:20.59 | Greenlight | I hear realtime is a mixed bag |
| 15:21.08 | [TK]D-Fender | Though the generation method does help you survive connection issues. |
| 15:21.11 | Shariff | Is there a command to use from the asterisk cli that makes a phone ring? |
| 15:21.36 | Greenlight | Shariff; originate |
| 15:21.48 | Shariff | Greenlight: Thanks! |
| 15:21.54 | BorjaGVO | [TK]D-Fender: You mean using database function inside .conf files or what? |
| 15:22.13 | Greenlight | He means you can use what's called "realtime" and have asterisk refer directly to the database |
| 15:22.13 | [TK]D-Fender | I mean using DB's FOR your configuration. |
| 15:22.17 | [TK]D-Fender | ^ |
| 15:22.44 | Greenlight | Hell, you can make it use Active Directory from what I read - which sounds really cool, but I've not tried it yet |
| 15:23.42 | BorjaGVO | Greenlight: Yeah, that's pretty cool. However, [TK]D-Fender , can you explain your point? |
| 15:24.05 | BorjaGVO | [TK]D-Fender: isn't it the same as Greenlight pointed out? |
| 15:24.30 | Greenlight | I meant using db and script to generate your conf files, he means using realtime |
| 15:25.06 | Greenlight | Whereby if a peer sends a request, and asterisk can't find it, it'll query your database directly for the informationm |
| 15:25.07 | BorjaGVO | hmmm...alright, sorry I didn't get it as I'm not familiar with realtime |
| 15:25.10 | igcewieling | example of generating sip.comf from a DB http://pastebin.ca/2388277 |
| 15:26.09 | BorjaGVO | thanks igcewieling, i'll take a look :-) |
| 15:27.21 | Greenlight | I would certainly recommand sticking with the config files route until you're happy with everything or have a real *need* to use realtime |
| 15:28.44 | Shariff | Why doesn't this do anything: ASTERISK*CLI> originate SIP/7001@general 7001 s@internal |
| 15:29.05 | Shariff | Did I mess up the syntax? |
| 15:29.23 | Greenlight | Shariff: What does it say? Is that peer registered? |
| 15:29.34 | Shariff | Greenlight: I get a warning |
| 15:29.41 | BorjaGVO | Greenlight: AI see. Why is that? I guess realtime is not very easy.. |
| 15:29.51 | Shariff | Greenlight: WARNING[21519]: res_clioriginate.c:167 handle_orig: else |
| 15:30.13 | [TK]D-Fender | [11:29]ShariffDid I mess up the syntax? <- yes |
| 15:30.18 | Greenlight | BorjaGVO: Yea, it adds another layer of complexity and introduces more neunces |
| 15:30.23 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:30.33 | Shariff | [TK]D-Fender: Any tips to resolve it? |
| 15:30.35 | [TK]D-Fender | Shariff: originate SIP/7001 7001 s@internal |
| 15:30.41 | Shariff | Aahh |
| 15:30.43 | Shariff | Will try |
| 15:30.48 | [TK]D-Fender | Shariff: You pass it the deivce the same way you would in a Dial() |
| 15:30.54 | BorjaGVO | Greenlight: Alright, I'll explore the solution you proposed then...thanks as always :-) |
| 15:31.03 | Shariff | [TK]D-Fender: Thanks ! |
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| 15:57.42 | atan | By chance is there a way on one of the Digium phones to set a chan var during the call from the phone itself? Say a customer call it, and the person taking the call wants to flag the call so the record can be added to the customers account. |
| 15:58.45 | [TK]D-Fender | atan: Not directly, but you could have it execute a request with the API that you could script to do that action via AMI |
| 15:58.45 | Elleni | Hi all, a short question. When trying to videocall after updating from Version 1.8 to 11 Version of Asterisk I get an error in asterisk console: Ignoring video stream offer because port number is zero |
| 15:58.57 | Elleni | did anything change in syntax? |
| 15:59.11 | [TK]D-Fender | Elleni: Clearly a wrong port is being specified by a device. You need to fix that device |
| 15:59.45 | Elleni | hm, with 1.8 it works but after upgrade I get this message |
| 15:59.53 | Elleni | how would I fix this? |
| 16:00.15 | [TK]D-Fender | Fix the end-point. |
| 16:00.22 | [TK]D-Fender | You need to be looking at your actual call. |
| 16:01.28 | jeffspeff | someone is attempting to get through to one of our servers, in the asterisk logs I'm seeing that it's sending fake auth rejections for different non-existant devices. but the thing that i don't understand is the IP address with each rejection is the external IP of the the asterisk server |
| 16:01.38 | Elleni | you mean the sip app on the cellphone for example (i am searching through its settings) |
| 16:03.07 | Elleni | [TK]D-Fender I cannot define any port on the app :( What I dont understand is that it worked like a charm on 1.8 version only after update on 11 I get this msg |
| 16:05.49 | [TK]D-Fender | Elleni: Go deal with your app. |
| 16:06.01 | [TK]D-Fender | Elleni: If it's specifying an illegal port then it has an issue |
| 16:06.09 | [TK]D-Fender | Elleni: Go look at the call debug |
| 16:07.10 | Elleni | hm, ok. I see... thanks anyway :) |
| 16:07.24 | navaismo | hi all, [TK]D-Fender about the hidden dialplan you mentioned to use an AGI so i was testing and yes agi can be encrypted but unless you patch the asterisk all the output and dialplan can be obtained via agi set debug on, and by default using phpagi class it show every piece of code in the cli. So i guess asterisk by default can hide the dialplan |
| 16:08.39 | [TK]D-Fender | navaismo: Yes you'll see the AGI actions take, but not the logic around it. <- |
| 16:09.50 | [TK]D-Fender | navaismo: So the AGI is protected, but the specific things it tells * to do are not "secret" |
| 16:11.13 | navaismo | yup, so the issue is anyone with little knowledge can decrypt it via cli :( |
| 16:13.41 | [TK]D-Fender | How so? |
| 16:14.05 | [TK]D-Fender | Asterisk CLI only sees what gets pout through STDIN/STDOUT/STDERR. |
| 16:14.13 | [TK]D-Fender | Nothing more.... none of the actual "logic" |
| 16:15.08 | atan | Thank you [TK]D-Fender :-) had hopes I could build some app on the phone to enter a unique ID, but, oh well :D |
| 16:15.09 | navaismo | one sec Ill show you |
| 16:15.18 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 16:16.52 | [TK]D-Fender | atan: You can... but it doesn't go "direct". You till need to use a script on the server to to the variable set via AMI. There is simply no direct integration betwnn the phone and * on that level |
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| 16:23.50 | navaismo | [TK]D-Fender, this is the output of the AGI http://pastebin.com/Pm8EUG9P, so basically you can copy the dialplan without problem from cli output |
| 16:24.39 | [TK]D-Fender | navaismo: for THOSE bits... depends what you're doing obviously. |
| 16:25.27 | [TK]D-Fender | navaismo: You are are doing call processing based on DB values, etc... that is no exposed. If all you are doing is stuff you could be doing from * base dialplen.. then DUH of course it can be seen. |
| 16:25.59 | [TK]D-Fender | navaismo: Now if it's pulling those bits from other places they won't know where or how. |
| 16:26.13 | navaismo | well im trying to hide the dialplan only no matter what simple process so far using phpagi class cant be hidden |
| 16:26.18 | [TK]D-Fender | navaismo: They'll see the effect, but not the source & cause |
| 16:26.27 | [TK]D-Fender | navaismo: NONE CAN. |
| 16:26.39 | [TK]D-Fender | navaismo: This has nothing to dow ith PHPAGI. This is ***AGI*** |
| 16:26.58 | navaismo | i see |
| 16:27.17 | [TK]D-Fender | navaismo: AGI commands can be seen from CLI. This has nothing to do with the language. If you have a lot of PROGRAMMING logic that isn't an actual * SYSTEM CALL then THAT will not be visible |
| 16:27.59 | Shariff | I have set up asterisk quite basic as per a tutorial, my phone says it's registered. Asterisk has a host assigned in sip show peers.. But the phone also says VOIP is unavailble.. any tips on how to troubleshoot this? I only get warnings with an originate |
| 16:27.59 | [TK]D-Fender | navaismo: If all you are doing is trying to do boring stuff you'd just do in regular dialplan anyways then you aren't hiding anything and this is a waste of time. |
| 16:28.07 | [TK]D-Fender | navaismo: And you should have known that from the start. |
| 16:28.52 | navaismo | unless you patch the asterisk :) |
| 16:28.55 | [TK]D-Fender | Shariff: show us.... you are talking about multiple point incompletely. |
| 16:29.08 | [TK]D-Fender | navaismo: You have the source. Go for it. |
| 16:29.19 | [TK]D-Fender | navaismo: And be prepared to make that a continual effort |
| 16:29.29 | Shariff | [TK]D-Fender: I guessed as much, sorry for that.. I have no knowledge (yet) .. I'll pastebin the conf files |
| 16:29.49 | [TK]D-Fender | Shariff: Show us actual calls as well. |
| 16:29.57 | Shariff | I have none |
| 16:29.58 | [TK]D-Fender | Shariff: Configs are only a small part of the story. |
| 16:30.07 | [TK]D-Fender | Shariff: Your CALLS have issues. |
| 16:30.29 | Shariff | [TK]D-Fender: It would appear that getting the phone to talk with asterisk is already an issue |
| 16:30.36 | Shariff | If you tell me what to pastebin.. I will :D |
| 16:30.37 | navaismo | [TK]D-Fender, I did it, doesnt worth the effort, it hides the agi ouput but so boring to do every upgrade |
| 16:30.43 | [TK]D-Fender | Shariff: I can look at your call all day and say "it should drive fine" until I see that you are driving it through a road with 2-foot wide potholes I won't know where the problem is. |
| 16:31.10 | [TK]D-Fender | navaismo: Then you already had your answer and went through all this anyway... |
| 16:31.26 | Shariff | [TK]D-Fender: I get that.. and I aprpeciate your help! You tell me what you need, I'll find it and show you :) |
| 16:31.45 | Shariff | [TK]D-Fender: That's not to let you do the work, but I don't know what else (besides the conf files) to show you ) |
| 16:31.47 | Shariff | :) |
| 16:32.36 | [TK]D-Fender | Shariff: Show the actual problem. |
| 16:32.48 | [TK]D-Fender | Shariff: Then we'll start looking for the cause. |
| 16:33.40 | Shariff | The problem is: PHone is registered (according to phone) asterisk has a host in sip show peers. Phone has no dial tone and tells me when before I can dial that VOIP is unavailable.. |
| 16:34.23 | [TK]D-Fender | Shariff: now SHOW it. |
| 16:34.43 | [TK]D-Fender | Shariff: enable SIP DEBUG from * CLI and restart the phone so it registers so we can see if that looks fine. |
| 16:34.48 | navaismo | [TK]D-Fender, just to let you know and see if you have more ideas than edit the source code |
| 16:34.57 | navaismo | [TK]D-Fender, thanks anyway |
| 16:35.06 | Shariff | [TK]D-Fender: Thanks for that! will do |
| 16:35.20 | [TK]D-Fender | navaismo: do YOU see an option to lock down the disply that they can't override? No, it has to be in the binary hard-coded |
| 16:35.27 | [TK]D-Fender | navaismo: Common logic. |
| 16:38.03 | drmessano | Asterisk wouldn't be any fun if you didn't have to keep a set of patches to apply to every new release |
| 16:41.28 | navaismo | :S |
| 16:43.52 | Shariff | [TK]D-Fender: http://pastebin.com/QgcYScPS |
| 16:45.09 | *** join/#asterisk mikeones (~jkimbell@99-99-189-254.lightspeed.rcsntx.sbcglobal.net) |
| 16:50.33 | [TK]D-Fender | Shariff: Looks like a local phone and seems to register fine... |
| 16:50.47 | igcewieling | Can anyone recommend a way to install DAHDI without needing Internet access for downloading firmware files when running "make" |
| 16:51.00 | Shariff | [TK]D-Fender: http://pastebin.com/npHcLdCU for conf files |
| 16:51.13 | [TK]D-Fender | Shariff: So far it should work fine. |
| 16:51.20 | [TK]D-Fender | Shariff: try an actual call immediately |
| 16:51.32 | Greenlight | igcewieling: Yea I've had a nightmare with that before when stuck behind a proxy |
| 16:51.37 | Shariff | [TK]D-Fender: On the phone or using some command from asterisk? |
| 16:51.51 | [TK]D-Fender | from the phone |
| 16:52.22 | Shariff | [TK]D-Fender: I do not have a dial tone and the phone (siemens gigaset A580 hard phone) tells me "VOIP is unavailable" |
| 16:53.05 | [TK]D-Fender | Shariff: Perhaps you have not finished configuring some other aspect on it... |
| 16:53.06 | Shariff | [TK]D-Fender: When I make a call, I don't see any debug messages in asterisk (from sip debug) |
| 16:53.13 | [TK]D-Fender | Shariff: But the registration seems fine. |
| 16:53.24 | Shariff | Is there more to do? :) |
| 16:53.33 | Shariff | The tutorial did not mention that :D |
| 16:53.39 | [TK]D-Fender | Check your phone's manual, this doesn't appear to be a comm issue |
| 16:53.54 | Shariff | Ok.. so this is (as far as you can see) a phone issue.. not an asterisk issue? |
| 16:53.59 | navaismo | igcewieling, have you tried Downloading the files before compile? |
| 16:54.29 | igcewieling | navaismo: the solution was pretty easy -- unbreak internet access |
| 16:54.37 | navaismo | hahaha |
| 16:54.53 | jmetro | i had a user that wanted to do a full system restore from backup because his shortcuts werent working |
| 16:58.33 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 16:59.10 | MLNoah | How does the && operator behave? If I have $[DBLOOKUP1 = CONDITION1 && DBLOOKUP2 = CONDITION2], will Asterisk perform the second lookup even if the first lookup failed the condition check? |
| 16:59.56 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 17:00.02 | MLNoah | well, I guess it's a single &. but the same question remains. |
| 17:00.54 | [TK]D-Fender | MLNoah: I'd put odds on it.... |
| 17:01.53 | MLNoah | hrm. so is it worth making the dialplan a little more convoluted with a branch to avoid the second lookup... |
| 17:04.32 | [TK]D-Fender | MLNoah: It could be. |
| 17:05.04 | MLNoah | yeah, probably should stick with better database practice. |
| 17:05.17 | MLNoah | boo languages that don't handle and/or intelligently :P |
| 17:05.17 | MLNoah | thanks |
| 17:06.07 | [TK]D-Fender | MLNoah: Oh lets not open that can of worms, k? :) |
| 17:06.54 | MLNoah | he he, yeah, I should avoid a religious debate, hmmm? |
| 17:10.09 | [TK]D-Fender | MLNoah: No, telephony is not faith-based :) |
| 17:10.50 | WIMPy | Are you sure about that? |
| 17:10.58 | *** join/#asterisk geeksteve (~geeksteve@195.110.168.123) |
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| 17:17.57 | [TK]D-Fender | WIMPy: I BELIEVE so ;) |
| 17:18.31 | WIMPy | :-) |
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| 17:19.19 | WIMPy | has the feeling that you need a lot of faith lately to make phone calls. |
| 17:20.07 | [TK]D-Fender | WIMPy: Or give up actual hope, and just keep trying regardless |
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| 17:21.12 | WIMPy | yeah |
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| 17:56.58 | jmetro | is there dialplan logic to strip characters? |
| 17:57.03 | jmetro | im so tired of having to accomodate for +'s |
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| 18:00.40 | trdillon1 | how can i limit a sip extension to 1 concurent call? |
| 18:03.37 | trdillon1 | anyone? |
| 18:03.47 | jmetro | only give them one line on their phone |
| 18:03.52 | [TK]D-Fender | trdillon1: call-counter=yes |
| 18:04.22 | igcewieling | jmetro: you mean like ${EXTEN:2) |
| 18:04.46 | [TK]D-Fender | trdillon1: call-limit=1 |
| 18:05.05 | jmetro | igcewieling: i get numbers like "NXX-XXXX" and "+1NXX-XXXX" so i would have to count the exten first. |
| 18:05.20 | igcewieling | jmetro: accomodate for +s? We do a exten => _+1NXXNXXXXXX,1,Goto($EXTEN:1},1) then handle everything in exten => _1NXXNXXXXXX |
| 18:05.24 | jmetro | in which case, bloody nightmare |
| 18:05.24 | [TK]D-Fender | jmetro: You do have to accommodate for them... |
| 18:06.15 | jmetro | i would have to make a new incoming context with just _. and strip the characters based on length |
| 18:06.22 | jmetro | filter incoming |
| 18:06.35 | igcewieling | my way is more simple |
| 18:06.52 | [TK]D-Fender | trdillon1: These were both in the sample config. |
| 18:07.27 | jmetro | igcewieling: if you handle the longest one the shortest one falls through, if you handle the shortest the longest falls through. seems like my only option is to have 4 entries for every inbound |
| 18:07.51 | [TK]D-Fender | trdillon1: http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=markup |
| 18:08.06 | igcewieling | jmetro: just how many different patters are you expecting? |
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| 18:08.24 | jmetro | area code, 1+area, +area, +1+area |
| 18:08.54 | igcewieling | that is 3 lines to normalized the DNIS. simple |
| 18:08.55 | trdillon1 | i should mention that this is for the queue app |
| 18:10.02 | [TK]D-Fender | trdillon1: Yes, you really should. Somewhere along the lines of "first" |
| 18:10.22 | [TK]D-Fender | trdillon1: If you don't want it to send calls to busy memebers... then pass the STATE DEVICE when you add them |
| 18:11.10 | jmetro | or ringinuse=no? |
| 18:11.24 | [TK]D-Fender | that also |
| 18:11.33 | [TK]D-Fender | both together = success |
| 18:13.36 | WIMPy | jmetro: +area is obviousely wrong. |
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| 18:30.12 | jmetro | ptankstet? |
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| 18:34.13 | leifmadsen | jmetro: use FILTER() |
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| 18:34.53 | trdillon1 | how do I setup STATE DEVICE |
| 18:40.33 | [TK]D-Fender | trdillon1: its right were you specify the member. |
| 18:42.02 | jmetro | leifmadsen: Nice. I think i still need to make a separate context for it. |
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| 19:21.24 | AkkerKid | I like Bacon! |
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| 19:28.07 | [TK]D-Fender | AkkerKid: Then we like you! |
| 19:31.09 | navaismo | I never understands the US people fascination about a dry piece of meat |
| 19:32.12 | jmetro | :-* |
| 19:32.28 | igcewieling | I'm almost a vegetarian and even I like bacon. |
| 19:32.49 | jmetro | there's a jewish deli near us that makes "shmacon" which is beef bacon |
| 19:35.37 | Qwell | err, what? |
| 19:35.38 | navaismo | mieh its only a thin salad dry meat nothing fancy at all. |
| 19:35.51 | Qwell | That's called slices of beef. |
| 19:36.41 | jmetro | Its slices of beef but its crispy and bacon-y tasting |
| 19:36.45 | jmetro | not sure how they baconize it. |
| 19:37.55 | navaismo | Argentinian/Uruguayan people make the best meat dishes |
| 19:38.36 | jmetro | i dont know how but i grilled up the most delicious steak i've ever eaten but gave it to my dog who was dying the next day |
| 19:39.02 | navaismo | :S |
| 19:40.09 | igcewieling | IIRC Baconization was invented by Thomas Hancock in 1844 |
| 19:41.32 | navaismo | sounds like a superhero movie, |
| 19:41.53 | navaismo | http://9gag.com/gag/aPv680K |
| 19:44.45 | igcewieling | all things bacon http://www.thinkgeek.com/brain/whereisit.cgi?t=bacon&x=-186&y=-282 |
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| 21:43.09 | danfromuk | Hi, Is there a variable thats available in the dialplan that contains the dialling sip peer's IP address? |
| 21:44.10 | WIMPy | CHANNEL(peerip) |
| 21:44.44 | danfromuk | Does that work for guests too? |
| 21:44.55 | WIMPy | Sure |
| 21:46.18 | danfromuk | It appears to not work. |
| 21:48.07 | WIMPy | I believed it worked for me. |
| 21:48.09 | danfromuk | Nevermind. Typo |
| 21:48.13 | danfromuk | Thanks |
| 21:48.17 | WIMPy | ok |
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| 21:53.12 | danfromuk | I know this sounds like a rookie question. Currently, the incoming context is quite open. I want to secure it a little by only allowing incoming calls from our providers. I have a peer defined and the context=securedincomingcontext. But calls seem to still be coming in to [insecurecontext] |
| 21:54.08 | danfromuk | Currently the provider's peer details only have one host=sip.provider.com line. Since it isn't working, its possible they arent sending incoming calls from sip.provider.com but from another ip. |
| 21:54.32 | danfromuk | Is it simply a case of adding multiple host= lines for each ip they list? |
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| 21:58.34 | WIMPy | No, you can only have one host(IP) per peer. You need to define one peer per IP. |
| 21:59.35 | danfromuk | Does everyone have multiple peer's per provider? Sounds a little unusual. |
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| 22:04.58 | leifmadsen | yes everyone does |
| 22:05.17 | carrar | EVERYONE |
| 22:05.18 | leifmadsen | or they send it to a proxy to deal with |
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| 22:06.20 | carrar | danfromuk, not everyone |
| 22:07.15 | carrar | usaully a uniq peer for each type of service |
| 22:07.24 | carrar | like LD vs 2-way |
| 22:07.41 | carrar | and then there is verizon |
| 22:07.56 | carrar | 9 IP's |
| 22:08.10 | WIMPy | Not that many :-) |
| 22:08.22 | carrar | yeah plus they are in a SRV record |
| 22:08.25 | carrar | so it's easy |
| 22:08.27 | danfromuk | Magrathea have 20 IPs that can send sip packets. So I need to create 20 peers? |
| 22:08.44 | carrar | yeah |
| 22:08.51 | WIMPy | I have one ITSP with 36 IPs. |
| 22:08.57 | carrar | 20 seems like a lot |
| 22:09.04 | carrar | why so many |
| 22:09.09 | danfromuk | They are removing 4 in a few weeks. |
| 22:09.20 | carrar | seems excessive |
| 22:09.39 | WIMPy | Some take HA serious. |
| 22:09.48 | carrar | You don't need 2 IP's to be HA |
| 22:09.51 | *** join/#asterisk PaybackTonyB (1815f748@gateway/web/freenode/ip.24.21.247.72) |
| 22:10.12 | WIMPy | At least 2. |
| 22:10.14 | PaybackTonyB | Finally. |
| 22:10.17 | carrar | no |
| 22:10.28 | carrar | let the carrier do the redundency and offer 1 ip |
| 22:10.41 | WIMPy | If you want to care fo routing issues as well, you need at least 2 IPs. |
| 22:10.55 | carrar | well at 1 site you can do just 1 IP |
| 22:11.00 | carrar | with vrrp |
| 22:11.01 | carrar | or whatever |
| 22:11.22 | WIMPy | Exactely. One site isn't much. |
| 22:11.54 | carrar | but 36 is nuts |
| 22:11.56 | carrar | heh |
| 22:11.58 | WIMPy | They send every call from two different networks. |
| 22:12.00 | carrar | me thinks |
| 22:12.08 | WIMPy | HA taken serious. |
| 22:12.10 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qxrmwazuevdranlm) |
| 22:12.14 | PaybackTonyB | So I've been digging around the wiki, google, everywhere and can't find the answer. This is my last resort. Can anyone point me in the right direction on how to change the audio file played for music on hold (for ConfBridge) in the dialplan (or AGI)? |
| 22:12.34 | carrar | So you're getting two invites from two different places for 1 call? |
| 22:12.44 | WIMPy | yes |
| 22:12.46 | carrar | ugg |
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| 22:17.10 | danfromuk | Ok, multiple peers seems to have resolved it. |
| 22:17.38 | carrar | Can you round robbin calls back out to them accross all those same peers? |
| 22:18.20 | WIMPy | Their DNS RR doesn;t list all of them. I think some are spare. |
| 22:18.58 | navaismo | PaybackTonyB, for confbridge settings use music_on_hold_class in dialplan logic you can Set(CHANNEL(musicclass)=class) |
| 22:22.54 | PaybackTonyB | @navaismo - I appreciate the help. I have seen that I can change the class. In our app, we let users specify what audio file to play (which resides off our servers, we grab it in the agi), so no class is setup in the conf. Right now, from my limited knowledge, my only option is to use asterisk realtime and build a moh class each time. |
| 22:26.35 | navaismo | yes thats right, or if you have already classes created just upload & override the audio file, or upload & edit the files, or add another channle streamig the new file |
| 22:26.51 | navaismo | oh well... |
| 22:28.02 | navaismo | guess thats not the answer he looked at |
| 22:28.29 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:29.49 | WIMPy | You scared him. |
| 22:29.50 | *** join/#asterisk PaybackTonyB (~PaybackTo@c-24-21-247-72.hsd1.or.comcast.net) |
| 22:30.02 | PaybackTonyB | Was using webchat, it failed on me |
| 22:30.10 | raidghost | [TK]D-Fender: CDR cost 0.4 not supposed to work in latest freepbx? |
| 22:30.38 | raidghost | I guess a 6 year old module have done its work. |
| 22:30.44 | WIMPy | PaybackTonyB: Oh, we thought you got scared. |
| 22:30.49 | PaybackTonyB | @navaismo: Did you have a follow up to my reply? |
| 22:30.58 | [TK]D-Fender | raidghost: Wrong channel to ask that..... |
| 22:31.01 | PaybackTonyB | I was scared, as soon as I got help it told me I lost connection |
| 22:31.02 | raidghost | ups |
| 22:31.12 | raidghost | sorry, i thought it was the correct one, abit tired:P |
| 22:31.44 | navaismo | PaybackTonyB, I wrote " yes thats right, or if you have already classes created just upload & override the audio file, or upload & edit the files, or add another channle streamig the new file" |
| 22:32.30 | PaybackTonyB | By upload and override I assume that means just overwrite the one the class points to? |
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| 22:34.43 | navaismo | override the uploaded sound file with the old one. |
| 22:36.08 | PaybackTonyB | If only there was a way to change the moh audio file from Set or something. Anyways, thank you for the help. It sounds like my best bet is to use RealTime and just build out moh classes based on the audio file the agi is told to use / the conference name. |
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| 22:49.37 | Shariff | A book on asterisk is saying: ".. change runuser and rungroup to have values of asteriskpbx." .. do they mean chown asteriskpbx:asteriskpbx or do they mean something else? |
| 22:50.21 | WIMPy | Look at asterisk.conf. |
| 22:50.41 | robl^ | they mean to change the script that starts asterisk so it actually runs AS another user. |
| 22:50.42 | Shariff | WIMPy: Aahhhh I see it.. the user and group in [files] |
| 22:50.52 | Shariff | Thanks! |
| 22:51.18 | WIMPy | No. [options] |
| 22:51.53 | Shariff | You are completely right.. I missed that one.. I saw a user and group and assumed (wrongly) thanks!! |
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| 23:06.11 | Shariff | Do the conf files in /etc/asterisk need to be owned by the user/group that asterisk is run as ? |
| 23:06.21 | Shariff | Or will root/root do? |
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| 23:06.41 | [TK]D-Fender | Shariff: * clearly needs to be able to read them. |
| 23:06.44 | WIMPy | They need to be readable by asterisk, obviousely. |
| 23:06.47 | [TK]D-Fender | Shariff: This is *NIX 101 |
| 23:07.11 | Shariff | [TK]D-Fender: yeah.. world can read :) And it's been a while.. I'm a winbox user by default :D |
| 23:07.29 | Shariff | Last time I worked on linux was slackware 6 I believe ) |
| 23:07.31 | Shariff | :) |
| 23:07.36 | [TK]D-Fender | Shariff: You use Mikrotik too? Excellent... |
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| 23:08.22 | Shariff | `Not familiar with that at all :) |
| 23:08.24 | Shariff | So Nope :D |
| 23:12.27 | [TK]D-Fender | winbox = mikrotik router management tool |
| 23:12.38 | Shariff | Ahhh.. I meant a windows machine :) |
| 23:12.59 | Shariff | win7 (for now) |
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| 23:18.42 | endre | stf |
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