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| 00:35.57 | p3nguin | Interesting. I think chicago.voip.ms just went offline. |
| 00:37.40 | gordonjcp | can externip be passed a hostname rather than an IP address? |
| 00:38.39 | p3nguin | No. |
| 00:38.46 | gordonjcp | hm |
| 00:38.50 | p3nguin | But fortunately, there's externhost ! |
| 00:39.03 | gordonjcp | ah, makes sense |
| 00:40.03 | p3nguin | Usually externaddr (formerly externip) is used with a static public IP address, and externhost is used with a dynamic host name. |
| 00:40.11 | gordonjcp | yup |
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| 01:02.35 | s[X] | p3nguin, you familiar with nsupdate |
| 01:02.43 | p3nguin | Only a little. |
| 01:03.01 | s[X] | cant get my key pairs to auth |
| 01:03.07 | s[X] | its painfully frustrating |
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| 01:08.57 | JerJer | yawns |
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| 01:33.05 | libryder | anyone know how to control what order hunt members are dialed in a linear queue strategy? i tried penalty but it just keeps calling the member with the lowest penalty every time |
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| 02:34.49 | tzanger | https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration |
| 02:34.55 | F2Knight | Just did an svn update on a test box and got an error about NAT http://pastebin.com/WkADHDtC anyone encounter this yet? |
| 02:35.03 | F2Knight | and what might be the fix |
| 02:35.19 | tzanger | is that link accurate for 1.8.x? thinking of using sqlite for sip realtime but that suggests that unless unixodbc has support for it, it doesn't exist |
| 02:36.21 | F2Knight | tzanger, the contribs/realtime directory has some defined SQL statements for different supported DB engines. |
| 02:36.57 | F2Knight | but the primary reason most people would use realtime is because you can have a centurally located database with your accounts. |
| 02:37.24 | F2Knight | using SQLITE ... while i believe technically possible, has no real value. |
| 02:38.08 | tzanger | F2Knight: yes, I know sqlite has no network support and more than one accessor is... not recommended |
| 02:38.14 | tzanger | I'll take a look there and see |
| 02:38.15 | F2Knight | SQLLite is very I/O heavy as it is not a proper database.. well not a Relational Database anyways.. its more like MS ACCESS. where its all in one file. |
| 02:38.17 | tzanger | thank you |
| 02:38.33 | tzanger | F2Knight: yeah, I'm pretty familliar with sqlite, just not with asterisk :-) |
| 02:38.40 | F2Knight | I run realtime.. MySQL DB |
| 02:39.40 | F2Knight | there are some 'fixes' you might need to apply to the sql scripts.. look at it and adjust to your needs |
| 02:39.53 | F2Knight | 1.8 does use sqlite3 now for its internal astdb |
| 02:40.27 | F2Knight | but if you are running a single node setup. your more often then not best to just stick with the static sip.conf file |
| 02:40.45 | F2Knight | esp if your not fimular with * at all |
| 02:40.53 | tzanger | oh I'm quite familiar with asterisk |
| 02:41.02 | tzanger | been running it since 1.0 |
| 02:41.12 | tzanger | compiled dialplans ftw. :-) |
| 02:41.23 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 02:41.26 | F2Knight | <tzanger> F2Knight: yeah, I'm pretty familliar with sqlite, just not with asterisk :-) <--- seems there is a misunderstanding then |
| 02:41.46 | tzanger | F2Knight: familliar with sqlite3, familliar with asterisk, just not the two together |
| 02:41.49 | tzanger | and not with realtime at all |
| 02:42.15 | F2Knight | ah well they have nothing to do with each other really |
| 02:42.31 | F2Knight | one is a DB access using common everyday ODBC |
| 02:42.40 | F2Knight | the other is a differnt ball of wax |
| 02:42.44 | tzanger | eh? |
| 02:42.51 | tzanger | what's odbc got to do with sqlite3 use in asterisk? |
| 02:43.01 | F2Knight | asterisk Realtime. |
| 02:43.19 | tzanger | asterisk realtime isn't locked to odbc, unless I'm mistaken |
| 02:43.21 | tzanger | which is possible |
| 02:43.24 | F2Knight | ODBC. must use it if you want things to work right, esp things like voicemail |
| 02:43.38 | tzanger | right right |
| 02:43.54 | F2Knight | the older versions were not at all but they are moving everything to ODBC support pretty much entierly |
| 02:44.10 | tzanger | I'm still in the design stage, might end up using Kamailio in front of * to isolate it but I am not sure I'll end upw tih that |
| 02:44.18 | tzanger | voicemail on the handsets isn't really coming from this * box anyway |
| 02:44.41 | F2Knight | if the system supports an ODBC interface asterisk can interface with that and let the system deal with interfacing to what ever the DB is.. flat file sqlite3 mysql firebird DB3 etc asterisk just dont care at that point its ODBC for it |
| 02:44.57 | tzanger | understood |
| 02:45.37 | tzanger | we're using sqlite3 for other aspects of the system, but there are going to be very few (i.e. under 30) devices registered, but they're all dynamically allocated |
| 02:45.43 | F2Knight | Kamillio is a big learning curve.. been trying to work with it my self... and plan to use my RealtimeDB to fetch account info to kamaillio.. but remember. Kamaillio does not media. |
| 02:45.53 | tzanger | or at least the bulk of them. autocreatepeer is being used to handle that though |
| 02:46.40 | tzanger | no I know kamailio has no media support. that's fine. this * box is always in the loop. I might be playing with some forced reinvites to send the call over another bearer mid-call based on outside-of-asterisk routing decisions |
| 02:47.03 | tzanger | just looking at kamailio to do the funky stuff if Asterisk proves too difficult |
| 02:47.16 | F2Knight | so if your PBX is mostly going to be doing media stuff IVR/ voicemail etc. You may want to reconsider ... also unless you have thousands of calls persecond I think Kamailio might be over kill. its fail over support is nice.. for load balancing .. but I think you need several thousands of calls to warrent that.. vs. just asterisk with HA setup |
| 02:47.29 | tzanger | no no this is very low volume, just complex |
| 02:47.46 | tzanger | no need for HA in this application either, which is nice |
| 02:47.58 | tzanger | the other end is all high availability, I'm not on that team though |
| 02:48.15 | tzanger | anyway, I should get back to the hotel. thanks for the info, I appreciate it |
| 02:48.24 | F2Knight | enjoy |
| 02:48.43 | F2Knight | happy )*( hacking |
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| 03:02.26 | *** join/#asterisk sogi (sogi@triton.intrak.tuke.sk) |
| 03:02.31 | sogi | hey guys. |
| 03:03.16 | sogi | somebody with problem with DTMF on asterisk 1.8 - couple of DTMF are being ignored |
| 03:03.39 | sogi | + I can only enter 3 digits in IVR... |
| 03:03.42 | sogi | ? |
| 03:03.48 | sogi | that was question :D sorry |
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| 03:13.27 | SeRi|zzZZzz | p3nguin_: !!!!!! |
| 03:14.16 | SeRi | rofl @ the msg |
| 03:17.58 | SeRi | what a day |
| 03:18.21 | p3nguin_ | What message? |
| 03:19.01 | SeRi | jelapanos |
| 03:19.07 | SeRi | vmail |
| 03:19.11 | p3nguin | Oh. |
| 03:19.15 | SeRi | hahahahaha! |
| 03:19.18 | p3nguin | :) |
| 03:19.27 | SeRi | d00d I just got home :/ |
| 03:19.41 | SeRi | went to a customers after work... |
| 03:20.19 | SeRi | Ill be working on saturday so no boxing for me :( |
| 03:25.38 | p3nguin | No ground 'n pound? |
| 03:28.36 | sogi | well |
| 03:28.47 | sogi | anybody to save my ass? :D |
| 03:30.39 | SeRi | p3nguin: nope :( |
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| 03:35.14 | s[X] | woot got nsupdate working |
| 03:35.17 | s[X] | stupid fkn permissions |
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| 03:43.29 | SeRi | p3nguin: I have a small problem when parking calls. |
| 03:43.45 | SeRi | when I park the call and retrieve it I can hear them but they can not hear me. |
| 03:43.57 | SeRi | only when parking calls |
| 03:44.31 | SeRi | p3nguin: can you give me a hand with this issue? |
| 03:44.42 | SeRi | I cant seem to see anything on the logs |
| 03:53.28 | SeRi | any body in? :( |
| 03:56.52 | SeRi | it's only with pstn calls |
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| 03:58.35 | WIMPy | I had an issue with one-way-audio in ConfBridge after putting a call on hold. |
| 03:59.00 | SeRi | WIMPy: I do have confbridge setup |
| 03:59.20 | WIMPy | Maybe there's a general issue with inactive calls. |
| 03:59.23 | SeRi | but its just regular calls coming in and been parked |
| 03:59.41 | SeRi | I see. let me try and disable confbridge |
| 04:00.15 | WIMPy | If you're not using it, I don't think it will do you any harm. |
| 04:00.47 | WIMPy | I was more thinking it could be two symptoms of another issue. |
| 04:01.11 | SeRi | I see |
| 04:01.16 | SeRi | well is odd. |
| 04:01.23 | SeRi | I can find anything on the logs... |
| 04:01.27 | SeRi | cant* |
| 04:01.35 | SeRi | that would say there is an issue |
| 04:02.13 | WIMPy | Might be interesting to debug hold states. |
| 04:02.23 | SeRi | how can I do that? |
| 04:02.49 | WIMPy | Depends on the channels. |
| 04:03.32 | SeRi | sip? |
| 04:05.46 | SeRi | mhhh just in case I just turn off the shit shaper |
| 04:08.05 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 04:09.13 | SeRi | ok is not that |
| 04:09.22 | SeRi | I just did a regular hold and it works fine |
| 04:09.28 | SeRi | its only with parked calls |
| 04:10.03 | WIMPy | A hold flag in core show channels would be a good thing. |
| 04:10.44 | p3nguin | How will you "disable confbridge"? |
| 04:12.18 | WIMPy | BTW: Is there some add-on for SIP to display hold status? |
| 04:13.03 | SeRi | p3nguin: well per say I didnt disable it. but the way to disable it is by unloading the module |
| 04:13.11 | p3nguin | it's "per se" |
| 04:13.25 | SeRi | Yes sr :) |
| 04:13.29 | JerJer | anyone have any ideas why I can telnet to tcp port 5060, paste in a register and get a 401 unauth response but absolutely nothing when using asterisk ? (asterisk just retransmits) |
| 04:13.29 | Nugget | telnet is eeeeeeevil! |
| 04:13.50 | SeRi | p3nguin: It only happens with parked calls only |
| 04:13.56 | JerJer | no firewalls or nat |
| 04:13.57 | p3nguin | SIP doesn't use TCP 5060. |
| 04:13.59 | WIMPy | Did you tell it to use tcp? |
| 04:14.40 | SeRi | p3nguin: you want to take a look at the logs and see if you spot something my untrain eyes cant see? |
| 04:14.47 | p3nguin | Not right now. |
| 04:15.01 | JerJer | i see packets in tcpdump - but nothing gets to the other end |
| 04:15.08 | SeRi | :( ok. |
| 04:15.15 | [TK]D-Fender | "when using asterisk" also tells us nothing. We have no idea how these 2 scenarios relate to each other networking wise. What you are connecting to, how, etc |
| 04:15.38 | [TK]D-Fender | I suppose you might be trying to imply that * should be trying to register... |
| 04:15.56 | [TK]D-Fender | Which we don't see configs for, status dumps to iundicate attempts, SIP debug from CLI.... |
| 04:16.00 | JerJer | [TK]D-Fender: i see the same attitude is still in this channel |
| 04:16.12 | [TK]D-Fender | JerJer: Just filling in what I don't see :) |
| 04:16.29 | JerJer | obviously i've tried the common shit if i'm askign here |
| 04:16.45 | [TK]D-Fender | JerJer: You can pony up something at any time. Our callers are waiting to take your orders now! |
| 04:16.56 | SeRi | lol |
| 04:17.02 | [TK]D-Fender | JerJer: Well you aren't showing us anything.... Really.. you know we aren't psychic. |
| 04:17.20 | JerJer | bye |
| 04:17.23 | SeRi | [TK]D-Fender: I am having issues when parking calls. and I cant seem to find anythong on the calls |
| 04:17.34 | SeRi | s/calls/logs/ |
| 04:17.37 | *** part/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 04:17.45 | SeRi | dam it |
| 04:17.51 | SeRi | shit |
| 04:17.53 | SeRi | lol |
| 04:18.00 | F2Knight | crap? |
| 04:18.07 | [TK]D-Fender | Lots 'o' |
| 04:18.24 | SeRi | any who I get garble audio and sometimes only one way audio. |
| 04:18.24 | F2Knight | fudge brownies w/ wallnuts |
| 04:18.28 | SeRi | only with a parked calls |
| 04:18.53 | F2Knight | SeRi, is one of the devices behind nat? |
| 04:19.04 | SeRi | everythign else works fine. as soon as I park a call and retrive it I can hear them fine but the cant hear me |
| 04:19.25 | SeRi | F2Knight: obiously not the issue if all calls can come in just fine |
| 04:19.39 | SeRi | but yes my * is behind nat |
| 04:19.59 | SeRi | It all works well. It is only when I park a call. |
| 04:20.32 | F2Knight | I had a simiular issue where calls came in fine but when going to a queue it would come out all mucked up one way and stuff but worked fine if all callers were on the same LAN. can you see if parking works from a lan to lan side? |
| 04:20.37 | SeRi | so I have to asume the issue is internally |
| 04:20.58 | SeRi | F2Knight: one sec |
| 04:21.18 | F2Knight | for me the issue turned out to be that I was not handling re-invites correctly |
| 04:24.00 | SeRi | F2Knight: in the lan seems to work ok |
| 04:24.16 | SeRi | Mhhhhh so what did you do for re invites? |
| 04:24.49 | F2Knight | looking through my notes now to see if I can find it. |
| 04:24.55 | SeRi | F2Knight: Thanks |
| 04:24.58 | F2Knight | but now you know its a NAT issue :) |
| 04:25.17 | SeRi | Well it puzzles me because it was working fine before :/ |
| 04:25.27 | SeRi | Out of no where I have this issue |
| 04:25.36 | SeRi | but.... mhhhhh one sec |
| 04:27.39 | SeRi | nope that was not it.... |
| 04:27.42 | SeRi | shit |
| 04:27.45 | SeRi | :( |
| 04:27.59 | SeRi | I made some changes on the ports but that was not it |
| 04:28.36 | SeRi | hold works fine from pstn or internally |
| 04:28.46 | F2Knight | what is your sip.conf entry for the channel in quesiton.. ? look at the canreinvie=no |
| 04:29.05 | SeRi | Thats set |
| 04:29.55 | SeRi | I dont think the inernal phones need to have that option... correct? |
| 04:31.23 | WIMPy | just wonders how to actually use the park app. |
| 04:32.29 | SeRi | Its defently with parked calls only |
| 04:33.06 | WIMPy | If I transfer a call to Park() it becomes inactive indeed. |
| 04:33.49 | WIMPy | But what's the point of telling the parked part where they have been parked? |
| 04:34.00 | *** join/#asterisk irroot (~gregory@197.105.18.172) |
| 04:34.37 | [TK]D-Fender | You don't tell the parked party where they are parked. |
| 04:34.54 | WIMPy | No, but Asterisk does. |
| 04:35.01 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:35.05 | [TK]D-Fender | not if you're doing it right |
| 04:35.13 | WIMPy | But then I'm not sure how it coult tell me. |
| 04:35.23 | WIMPy | d |
| 04:35.57 | [TK]D-Fender | Stop doing blind transfers to parking |
| 04:36.57 | WIMPy | But when I do an attended transfer I only get MOH. Obviousely Asterisk can't know it will become a transfer at that point. |
| 04:38.50 | [TK]D-Fender | ... |
| 04:39.12 | [TK]D-Fender | Attended transfer stat. Listen to lot #. Finished attended transfer. |
| 04:39.31 | [TK]D-Fender | You don't sit there forever... |
| 04:39.53 | carrar | I like to sit here forever |
| 04:40.04 | WIMPy | I don't get an announcement, I only get MOH. |
| 04:40.07 | carrar | I'm you're huckleberry |
| 04:40.19 | carrar | err |
| 04:40.22 | carrar | your |
| 04:40.30 | [TK]D-Fender | WIMPy: Show us configs and attempts. |
| 04:42.14 | WIMPy | Well the log says playing digits/1, but I don't hear it. |
| 04:42.25 | WIMPy | Do I need to Answer() before Park()? |
| 04:42.43 | [TK]D-Fender | sits and waits |
| 04:43.37 | WIMPy | Ok, that makes a yes. |
| 04:43.42 | WIMPy | Bad. |
| 04:48.03 | [TK]D-Fender | And another no-show bites the dust... |
| 04:48.26 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 04:49.10 | SeRi | well that was no fun |
| 04:49.41 | carrar | Thats comcastic |
| 04:50.12 | SeRi | indeed! |
| 04:52.47 | SeRi | well I cant seem to figure it out. and looks like today every body is bussy.... :( |
| 04:54.30 | [TK]D-Fender | SeRi: I would chime in if there was anything to comment on... |
| 04:54.51 | SeRi | [TK]D-Fender: Thanks :) |
| 04:55.47 | SeRi | I cant seem to nail this one out and seem to be an internal issue. F2K claims a nat issue but Its not since holds work just fine. park uses a different module than hold.... |
| 04:56.21 | SeRi | though technically is identical in some aspects |
| 04:57.46 | SeRi | ok found something |
| 04:58.07 | SeRi | if the same phone that puts the call on park retrievs it the issue does not present it self |
| 04:58.16 | p3nguin | Instead of using Park, what happens if you use the parking feature? |
| 04:58.31 | SeRi | p3nguin: Thats what I am doing |
| 04:59.00 | SeRi | using the park feature and retreving it as the announce ext |
| 04:59.07 | p3nguin | Not using Park()? |
| 04:59.10 | SeRi | in all cases 701 |
| 05:00.14 | SeRi | p3nguin: I am using features.conf and including the context include => parkedcalls |
| 05:00.41 | SeRi | I send the calls to 700 to be parked |
| 05:00.56 | SeRi | and it anounces where is been parked |
| 05:01.03 | SeRi | 701 |
| 05:01.40 | SeRi | If I dal 701 from the same phone I sent the call to be parked to retrive it everything is fine |
| 05:02.26 | [TK]D-Fender | Yup... nothing to see here... |
| 05:02.32 | SeRi | but if I dial 701 from my home phone "linksys pap2" I get nothing but garble audio from my side to her and some times they dont even get audio... Though I can hear them just fine |
| 05:02.37 | [TK]D-Fender | goes back to watching the Colbert Report |
| 05:05.01 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 05:08.16 | SeRi | ok the issue is only with the pap2 |
| 05:08.31 | SeRi | I can retrive calls fine from the polycom |
| 05:26.55 | SeRi | if any body can look at this log for an issue with park calls and retrivial I grately appreciated.: http://pastebin.com/raw.php?i=SqmD6QBZ |
| 05:35.52 | SeRi | fixed |
| 05:35.56 | SeRi | bastard |
| 05:36.05 | SeRi | pap2 you are a bastard |
| 05:36.17 | SeRi | I am happy now :) |
| 05:37.31 | ChannelZ | good your log was a mess |
| 05:38.18 | SeRi | ChannelZ: Thanks is my specialty |
| 05:38.20 | SeRi | lol |
| 05:38.50 | SeRi | well some how the pap2 reverted some features back and I didnt know how tha f it happen. |
| 05:38.58 | SeRi | :/ |
| 05:39.14 | SeRi | maybe I did it drunk or something |
| 05:42.42 | SeRi | rtp packet size/jitter/rtp ports where all some how all fucked up and not matching my settings :/ |
| 05:52.04 | [TK]D-Fender | ok, checking out for the night, later all |
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| 06:40.01 | _omer | hello, I need to dial 100 calls in a single loop in my AGI Script but loop get stopped at Dial command until call is not answered or timed out..... |
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| 06:47.23 | irroot | my challenge is to see if i can get Samba 3 and 4 to coexist on same server different domains |
| 06:48.03 | WIMPy | I see the S&M scene is alive. |
| 06:48.28 | irroot | WIMPy LOL |
| 06:49.09 | irroot | want to get openchange working it requires Samba 4 its a native replacement for exchange bug for bug compatibility :P |
| 06:49.40 | SeRi | irroot: I dont see why not. |
| 06:49.57 | SeRi | If you compile the binary with a user prefix I dont see why the two can not run together |
| 06:50.00 | irroot | ill bind it to a "spare" ip |
| 06:50.08 | irroot | or alias |
| 06:50.44 | irroot | the prefix is not the issue samba4 is "samba" samba3 is "smbd/nmbd" |
| 06:51.57 | WIMPy | _omer: You're using the wrong tool. You want AMI, call files or a shell script. |
| 06:51.58 | SeRi | ok so no troubles with the bin.... so spun a new ip for ether a second interface or as you posted an alias... Though I am not sure if that works since it has to resolve back to the same IP :/ |
| 06:52.28 | SeRi | samba can not bind to the same IP twice |
| 06:53.13 | SeRi | well two different ones.... unless there is something that I dont know. |
| 06:55.05 | *** join/#asterisk devops (~devops@115.111.177.122) |
| 06:55.39 | irroot | SeRi yeah i will have a extra ip the namespace will be tricky may need to do some ju-ju with ldap config to get the domains seperated |
| 06:55.55 | devops | Is dahdi providing API along with the driver |
| 06:56.36 | irroot | devops yes indeed the dahdi driver and then chan_dahdi and chan_ss7 [libpri/libss7] |
| 06:56.43 | SeRi | irroot: I see. let me know how it works out.. |
| 06:57.03 | s[X] | hey irroot |
| 06:57.05 | irroot | seri maybe delete outlook and install thunderbird :P |
| 06:57.23 | irroot | s[X] morning there |
| 06:57.25 | SeRi | lol |
| 06:57.29 | s[X] | hey SeRi |
| 06:57.41 | SeRi | s[X]: waz up. |
| 06:57.55 | s[X] | Replacing my noisy as fuck DL380 G3 with a laptop lol |
| 07:00.09 | irroot | hehe the noise that bad ? |
| 07:00.28 | irroot | s[X] i use a net book loving it took bit to get used too |
| 07:00.29 | s[X] | You wouldnt want to be in the same room as it for too long |
| 07:00.44 | _omer | WIMPy: yes, I am using AMI to initiate calls...but I thought if I could do that in a loop in AGI Script. |
| 07:00.45 | s[X] | Its just for my asterisk box |
| 07:01.15 | irroot | s[X] depends on your requirements |
| 07:01.22 | WIMPy | _omer: You can't parallise dialplan. |
| 07:01.22 | s[X] | home asterisk box |
| 07:01.23 | s[X] | lol |
| 07:01.31 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 07:01.34 | s[X] | i have a 45RU cabinet at hom |
| 07:01.38 | irroot | i use a small atom micro pc |
| 07:02.10 | irroot | hehe double points for a aircon in the cab and double down if you have a pyro pack |
| 07:02.32 | _omer | WIMPy: yep, I got it...thanks |
| 07:02.42 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 07:03.04 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:03.10 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:03.20 | s[X] | irroot: busy constructing a custom thermoelectric cooler |
| 07:03.33 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:03.46 | irroot | s[X] lol |
| 07:03.51 | s[X] | :P |
| 07:04.07 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 07:04.15 | irroot | p3nguin yo |
| 07:04.22 | s[X] | i got a photo here somewhere |
| 07:05.10 | s[X] | http://themicroserver.com/images/server_rack.jpg |
| 07:05.37 | s[X] | the HDD Array is probably the only piece of modern equipment |
| 07:05.55 | s[X] | Its 22TB of goodness :P |
| 07:09.01 | irroot | bangs head just got asked what to do when configure says no acceptable cc .... |
| 07:09.08 | irroot | s[X] mmm super cool |
| 07:10.18 | irroot | aint seen the bay stuff in a while |
| 07:10.31 | s[X] | yeah she be old :P |
| 07:10.35 | irroot | they were swallowed by nortel ?? |
| 07:10.40 | s[X] | yeah |
| 07:11.12 | irroot | remeber global internet ?? they used bay all over got x2 working on it back in the day |
| 07:11.26 | s[X] | nah dont |
| 07:11.46 | devops | irroot: I am researching on ss7 stack . for developing ss7 stack |
| 07:12.12 | devops | irroot: I didn't found any API doc for communicating with the card or the driver |
| 07:12.58 | irroot | s[X] that was circa '97/8 one of the first ISP's here in ZA |
| 07:13.13 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:13.14 | schmidts | good morning |
| 07:13.18 | s[X] | Iafrica rings a bells |
| 07:13.22 | irroot | devops get asterisk source dahdi source and libss7 source |
| 07:13.43 | irroot | the libss7 docs [doxygen] should help |
| 07:13.57 | irroot | s[X] yeah about same time |
| 07:16.07 | *** join/#asterisk jkroon (~jkroon@196.25.195.42) |
| 07:19.54 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:24.17 | s[X] | Hey irroot i wouldnt chatting to you about the 7200s - > asterisk setup |
| 07:24.24 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:24.39 | s[X] | I'm about to head home but if ur available later wouldnt mind a quick chat |
| 07:24.52 | irroot | s[X] i dont get too involved on the samsung setup |
| 07:25.06 | irroot | yeah should be arround 9am here |
| 07:25.08 | *** join/#asterisk Akuma (~Akuma@modemcable131.103-179-173.mc.videotron.ca) |
| 07:25.36 | SeRi | cya guys. g/n |
| 07:26.29 | s[X] | Isnt it already 9:30 ? |
| 07:26.38 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 07:26.41 | irroot | SeRi hehe i just got here lol want to chase me |
| 07:26.55 | SeRi | lolno |
| 07:26.57 | irroot | yeah 9:26 SAST |
| 07:26.58 | SeRi | I am leaving for the night |
| 07:27.04 | SeRi | 130AM here :P |
| 07:27.12 | SeRi | cya! |
| 07:27.14 | irroot | SeRi ah have a good morning then |
| 07:27.16 | irroot | cheers |
| 07:27.18 | s[X] | Cya Seri |
| 07:27.20 | s[X] | im heading home |
| 07:27.21 | s[X] | cya all |
| 07:35.23 | irroot | ok genius with out CC wants to now get gcc |
| 07:35.32 | irroot | but thats not all he wants to use "source" |
| 07:35.39 | irroot | so i gave him link |
| 07:35.57 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:36.03 | irroot | but now who is going to tell him that you cant compile a compiler without a compiler |
| 07:36.10 | irroot | ollii o/ |
| 07:36.31 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:40.35 | kaii | morning everybody |
| 07:42.05 | irroot | kaii morning |
| 07:43.44 | ollii | heyho |
| 07:46.08 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:47.22 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 07:47.26 | ollii | way to early...i should be in bed right now :/ |
| 07:47.54 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 07:48.14 | kaii | ollii: i was here at work half an hour before you, dont complain :P |
| 07:49.56 | irroot | is at home in bed ... working best of both |
| 07:50.11 | ollii | -.- ! |
| 07:51.14 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 08:02.33 | *** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net) |
| 08:02.50 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
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| 08:11.35 | *** join/#asterisk Haraken (~ryuk@unaffiliated/haraken) |
| 08:24.50 | zkn | Hello, could someone briefly explain how is "pickupexten" in features.conf supposed to work? What I require at this point is to be able to pick up a ringing extension from another extension, would this allow this? |
| 08:27.11 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:27.15 | kaldemar | zkn: it won't allow it, but it only defines the combination that is used to pick up calls. |
| 08:27.22 | *** join/#asterisk mandla (~quassel@168.167.180.161) |
| 08:27.49 | zkn | ok, so it's specifically for picking up calls from the parking lot after they have been parked |
| 08:28.11 | kaldemar | zkn: allowing pickup is configure with pickupgroup and callgroup in channel configuration files. |
| 08:28.18 | zkn | i see |
| 08:28.21 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:28.25 | kaldemar | zkn: no, it is not related to parking. |
| 08:28.53 | kaldemar | it is for picking up a ringing channel. |
| 08:29.10 | zkn | oh, still |
| 08:29.44 | zkn | okay, so what I need is to set up call groups and pickupgroups |
| 08:30.47 | kaldemar | if you have callgroup=1 for channel A, any channel that has pickupgroup=1 can pick up a ringing call to A up by using *8 or someting else defined in features.conf. |
| 08:31.16 | kaldemar | see the sample sip.conf for an example. |
| 08:31.53 | zkn | yesyes |
| 08:31.57 | zkn | already checking |
| 08:32.02 | zkn | thanks, kaldemar |
| 08:42.30 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
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| 08:47.09 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 09:00.17 | *** join/#asterisk s[X] (~mark@ppp118-208-95-198.lns20.bne4.internode.on.net) |
| 09:00.33 | s[X] | Hey all |
| 09:01.06 | ppc | yo |
| 09:01.12 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:01.37 | *** join/#asterisk irroot (~gregory@197.174.253.206) |
| 09:10.30 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 09:10.46 | angryuser | Hello anyone from sangoma here ? |
| 09:10.55 | angryuser | Marc ? |
| 09:15.31 | Faustov | Marc ftw, fixed my sangoma \o/ |
| 09:15.41 | Faustov | but I don't think he's here |
| 09:16.26 | angryuser | I have a very wierd stuff, installed a 101 in centos 6, card send nothing |
| 09:16.37 | angryuser | changed, card, same result |
| 09:16.48 | angryuser | It does not even detect loopback |
| 09:17.24 | Faustov | well, are the modules up? |
| 09:17.34 | Faustov | what do you get in dmesg when you load them? |
| 09:17.41 | angryuser | Faustov, yea, i can see spans, ect |
| 09:17.49 | Faustov | which version of wanpipe are you using |
| 09:17.51 | Faustov | and which kernel |
| 09:17.57 | angryuser | Faustov, i tried also install it with Freswitch |
| 09:18.13 | angryuser | Both see the card no problem |
| 09:18.33 | angryuser | Faustov, wanpipe-3.5.24 |
| 09:18.50 | angryuser | Faustov, 2.6.32-71.29.1.el6.x86_64 #1 SMP |
| 09:19.12 | Faustov | ok, from my experience 32 is the last version that lets wanpipe compile |
| 09:19.28 | angryuser | The only difference i saw with asterisk are the unnumbered frames |
| 09:19.53 | angryuser | But the interface w1g1 RX bytes:0 (0.0 b) TX bytes:0 (0.0 b) |
| 09:20.00 | Faustov | but I never used freswitch (whatever that is) so I can't really help, I've installed wanpipe manually and the configuration was quite easy over an interactive shell script |
| 09:20.21 | Faustov | what aboud dahdi? do you have it running, modules loaded? |
| 09:20.23 | angryuser | Faustov, its the same stuff, you think a kernel too high ? |
| 09:20.30 | Faustov | no, your kernel should be ok |
| 09:20.42 | Faustov | its the network abi that was causing problems |
| 09:20.45 | angryuser | Faustov, card detects, asterisk starts, i can see the span and channels |
| 09:20.45 | s[X] | irroot u around bud ? |
| 09:20.46 | Faustov | in later versions |
| 09:20.59 | Faustov | hmm |
| 09:21.27 | Faustov | angryuser: and you can get dahdi show channels to list your ports? |
| 09:21.38 | angryuser | Faustov, but loopback doing nothing, i plugged that loopback to the patton GW, the port goes up |
| 09:21.39 | irroot | yip |
| 09:21.58 | angryuser | angryuser, sure, i have the qsig debug even, ect |
| 09:22.28 | angryuser | Faustov, well, i will try with centos 5.6 just to be sure |
| 09:22.44 | irroot | s[X] what up |
| 09:23.11 | angryuser | Faustov, with debian sqeeze latest sangomas drivers do not compile btw |
| 09:23.14 | s[X] | Hey just was curious if i were to go down the Asterisk route between my ITSP and 7200s |
| 09:23.28 | s[X] | I was going to virtualize the * box |
| 09:24.09 | irroot | yeah |
| 09:24.31 | s[X] | I didnt setup the 7200 |
| 09:24.37 | irroot | should be ok may affect timing |
| 09:24.38 | s[X] | You reckon it would be a big job |
| 09:25.23 | s[X] | Half tempted to get the guys that installed it to do it but they arent the sharpest |
| 09:26.04 | Faustov | angryuser: sorry I have no better ideas |
| 09:28.25 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 09:28.50 | joobie | hey guys.. i have a queue setup where there are multiple SIP members registered |
| 09:29.16 | joobie | i want to set this up so that a member is actually a telephone number |
| 09:29.22 | joobie | how would this work? is it possible? |
| 09:29.57 | joobie | i basically have a bunch of SIP phones registered in the office |
| 09:30.25 | joobie | but for this one period of the day, those phones won't be serviced.. rather one user is going home and i want their home phone to ring as calls come through |
| 09:30.41 | joobie | their home phone is just a standard home phone, disconnected from the voip.. so we need to ring it via the PSTN network |
| 09:30.55 | joobie | we can use a sip peer of ours to do it.. just not sure how it would all integrate to the queue though |
| 09:32.09 | kaldemar | joobie: use local channels. |
| 09:32.53 | joobie | i use 1.4 |
| 09:32.54 | kaldemar | joobie: then the members are just extensions in your dialplan and dial what you want based on a logic that you define. |
| 09:33.04 | joobie | is this compatible? |
| 09:33.05 | *** join/#asterisk chasing`Sol (~cS@41.206.150.61) |
| 09:34.12 | kaldemar | yes. remember to use /n in the channel name so the local channel stays in the path. |
| 09:37.00 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 09:37.04 | joobie | what would i use in the 'member' declariation within queues.conf kaldemar ? |
| 09:37.17 | joobie | currently i use 'member => SIP/1000' for example |
| 09:37.35 | kaldemar | joobie: like i said, a local channel. Local/exten@context/n |
| 09:39.02 | joobie | ahh k.. what about concurrent calls - like if i have 2 in queue |
| 09:39.17 | joobie | 1 is answered.. will it try and dial the number over n over even though it's in use? |
| 09:39.33 | joobie | do i need some sorta dialplan logic to limit it to 1 call? |
| 09:41.06 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 09:41.48 | kaldemar | why would it? |
| 09:42.20 | *** part/#asterisk ihor (~Miranda@194.44.15.90) |
| 09:42.53 | joobie | i dont know |
| 09:42.59 | joobie | never used chan_local |
| 09:43.05 | joobie | not sure how it handles |
| 09:46.13 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:51.02 | joobie | kaldemar, i got an interesting warning back |
| 09:51.26 | joobie | [Nov 30 20:49:56] WARNING[26317]: app_queue.c:3137 try_calling: The device state of this queue member, Local/200@context/n, s still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 09:51.43 | joobie | it works ok.. just i get that warning where it's not in use |
| 09:52.16 | joobie | it comes up when it is in use |
| 09:52.18 | joobie | any ideas? |
| 09:54.08 | *** join/#asterisk Pio (~pio@reyes.longstair.com) |
| 10:03.40 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:03.50 | *** join/#asterisk maxhbp204 (~chatzilla@122.179.186.94) |
| 10:05.12 | maxhbp204 | Hi, i am having digium card for E1 line 4 ports in it, now i want to configure ss7 on it for making calls, so do i need for each E1 link we shall use a separate signaling channel |
| 10:05.18 | maxhbp204 | or it should be 1 for all |
| 10:05.22 | dym | Hey - im now running a Digium TE220 connected and dahdi configured. But when i have a call incoming i dont see anything on the CLI. any idea? http://pastebin.com/fS2QNJzv |
| 10:05.25 | *** join/#asterisk timahvo1 (~rogue@197.178.131.64) |
| 10:05.25 | maxhbp204 | can anybody help me for that |
| 10:05.36 | kaldemar | joobie: that's probably because you have ringinuse=no configured for the queue and chan Local does not support device state. |
| 10:06.20 | joobie | can i make it support device state |
| 10:06.27 | joobie | or put something in the dialplan to maket his work |
| 10:06.51 | maxhbp204 | dym: i think you have not included dahdi-channels.conf in chan-dahdi,conf file |
| 10:07.02 | dym | maxhbp204: checking... |
| 10:07.08 | maxhbp204 | ok |
| 10:07.32 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:5d7b:425a:59c5:fef4) |
| 10:07.39 | kaldemar | joobie: i thought you said it works already? that's just a line of debug. |
| 10:07.45 | maxhbp204 | I am having digium card for E1 line 4 ports in it, now i want to configure ss7 on it for making calls, so do i need for each E1 link we shall use a separate signaling channel??? can any body help me?? |
| 10:07.58 | *** join/#asterisk wannaknow (~realesnam@213.8.76.179) |
| 10:08.13 | wannaknow | Hi everyone |
| 10:08.54 | kaldemar | maxhbp204: https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7 |
| 10:09.43 | maxhbp204 | kaldemar: yes thanks for the link, i just want to know do i need seperate signalling lines or can i use the working first one? |
| 10:12.04 | wannaknow | i have a question: every 5 minutes exactly I see on my CLI "Manager 'admin' logged on from 127.0.0.1" and immediately afterwards "Manager 'admin' logged off from 127.0.0.1". Can anyone tell me what it means or what process originates it? I'm using version 1.8.4 and these manager connections often result in a "Broken pipe" error. |
| 10:12.50 | *** join/#asterisk hajekd (~hajekd@82.208.11.91) |
| 10:13.03 | kaldemar | maxhbp204: i think the sigchan is set per linkset, not per span. |
| 10:13.13 | kaldemar | maxhbp204: not entirely sure though. |
| 10:13.40 | hajekd | Is it save to run AGI() in h extension? |
| 10:13.47 | maxhbp204 | kaldemar: ok so i have to take 4 lines right, thanks for your suggestion, i will try with that as well |
| 10:13.53 | *** join/#asterisk irroot (~gregory@197.104.105.11) |
| 10:14.27 | kaldemar | wannaknow: something or someone from the local host is logging in via the manager interface. if you get broken pipe errors, sounds like it is a broken script that does not read asterisk's responses properly. |
| 10:15.21 | kaldemar | hajekd: as safe as anywhere else. depends on what the AGI does. |
| 10:17.06 | hajekd | kaldemar: We are doing some cleanup code in AGI() in h extension, but dialplan is not processed when dst channel hangup first - same as this issue states: https://issues.asterisk.org/jira/browse/ASTERISK-18811 |
| 10:24.17 | wannaknow | kaldemar: Thanks for your reply. How can I tell what process is logging in every 5 minutes? We run different versions of Asterisk on quite a few servers, and the same thing happens on all of them. I'd like to fix the broken pipe error, but first I need to find that script. |
| 10:26.26 | *** join/#asterisk hetii (~hetii@194.181.154.25) |
| 10:28.13 | kaldemar | wannaknow: did you build the system yourself? |
| 10:31.59 | wannaknow | kaldemar: No, but as far as I know everything is pretty standard |
| 10:32.04 | kaldemar | hajekd: does your AGI hang up the channel? |
| 10:33.28 | kaldemar | wannaknow: pretty standard means pretty much nothing. nothing is plain asterisk itself will use AMI by itself, that's for sure. try to monitor what makes a connection to the tcp port that is defined as a bind port in manager.conf. |
| 10:37.42 | wannaknow | kaldemar: Thanks, I'll give it a shot |
| 10:39.26 | *** join/#asterisk qakhan (~qakhan@182.185.242.110) |
| 10:39.38 | kaldemar | hajekd: the AGI hanging up the channel is not it. i can't reproduce your issue with a simple AGI script that only sets variables and executes a NoOp. |
| 10:39.59 | kaldemar | hajekd: i'm on asterisk 10 though, but the core snippet you put in jira has not changed. |
| 10:40.49 | hajekd | kaldemar: I didn't test on 10, but on 1.8 and 1.6. It works fine with 1.6 and 1.2. The diaplan after AGI is processed. But not in 1.8. |
| 10:42.16 | hajekd | kaldemar: And it depends who hangup first. Make sure dest channel (callee) hangup first |
| 10:42.45 | hajekd | kaldemar: It works fine when caller hangup first |
| 10:43.20 | kaldemar | hajekd: the callee did hang up first. |
| 10:43.42 | hajekd | kaldemar: 1.10? |
| 10:43.54 | kaldemar | 10. there is no 1.10. |
| 10:43.57 | joobie | burp |
| 10:44.10 | hajekd | ;), ok |
| 10:44.11 | joobie | so when is asterisk 10 going to be "stable" ? |
| 10:44.16 | joobie | is there a release date yet? |
| 10:44.54 | wdoekes2 | joobie: it's ready when it's ready |
| 10:45.00 | joobie | that's nice |
| 10:45.03 | joobie | when is that? |
| 10:45.17 | wdoekes2 | and even then, you shouldn't expect version 10.0 to be totally bug free |
| 10:45.26 | joobie | im considering doing an upgrade from 1.4 to 1.8 over the christmas period.. dont want to go to the effort if 10 is coming out the following month |
| 10:45.42 | qakhan | i have setup a queue with 4 agents. when i transfer call from 1 agent to other agent, call transfer successfully, but when a new call comes in queue then call also goes to that agent to whom old call was transfered. |
| 10:45.52 | wdoekes2 | it's in rc mode, so you could use that, unless you're affected by one of the open bugs |
| 10:45.52 | kaldemar | joobie: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 10:45.59 | qakhan | why it is happening? |
| 10:46.50 | joobie | kaldemar, that says on the 12th of last month it shoudl have been released |
| 10:46.52 | joobie | or is that RC ? |
| 10:46.57 | joobie | what about stable... |
| 10:47.08 | kaldemar | joobie: yes, it does say that. :) |
| 10:47.20 | qakhan | hi all |
| 10:47.32 | joobie | qakhan, what ring type mode are u using |
| 10:47.35 | wdoekes2 | joobie: https://issues.asterisk.org/jira/browse/ASTERISK-18847 <-- no open blockers |
| 10:47.42 | wdoekes2 | I suggest you try the latest rc version |
| 10:48.21 | qakhan | joobie i am simply usind exten => queue(abc,t) |
| 10:48.34 | qakhan | is it correct? |
| 10:49.01 | joobie | all those blockers are resolved though wdoekes2 |
| 10:49.24 | joobie | qakhan, what about your quque.conf |
| 10:49.25 | wdoekes2 | "no open blockers" |
| 10:49.34 | qakhan | ringall |
| 10:49.37 | joobie | so.. why is it still rc2? |
| 10:49.49 | joobie | qakhan, so ur next call after the transfer does not ringall? |
| 10:49.53 | wdoekes2 | so people get to test it |
| 10:49.54 | joobie | it rings only one extension? |
| 10:50.10 | wdoekes2 | the latest rc becomes the final version if no bugs are found within a reasonable time |
| 10:50.10 | joobie | sounds like 10 is almost htere |
| 10:50.14 | joobie | it supports skype ya? |
| 10:50.26 | joobie | ahh |
| 10:50.51 | qakhan | yes it rings all, but it not suppose to ring that agent while he is in call |
| 10:50.52 | schmidts | joobie AFAIK skype support is allready over |
| 10:50.54 | kaldemar | joobie: forget about skype. |
| 10:51.42 | joobie | already over? |
| 10:51.45 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 10:51.45 | joobie | forget about skype? |
| 10:51.50 | joobie | it's not supported in 10? |
| 10:51.54 | joobie | i read it was a feature of 10....... |
| 10:52.25 | joobie | qakhan, when u say it rings the agent while he's in a call.. how does the agent see this? |
| 10:52.47 | joobie | does he have a multi-line phone |
| 10:52.56 | qakhan | yes |
| 10:53.04 | joobie | and multiple lines are registered |
| 10:53.06 | joobie | to his single extensino |
| 10:53.11 | joobie | fuk im out of booze :/ |
| 10:53.12 | qakhan | and a popup shows in our application |
| 10:53.13 | kaldemar | joobie: you read wrong. asterisk 10 supports SILK, the codec that skype uses. |
| 10:53.48 | qakhan | no only 1 ext is registered |
| 10:53.56 | joobie | kaldemar, so does this mean we still cant integrate to skype? just use the shitty skype codec? |
| 10:54.07 | joobie | qakhan, do a test |
| 10:54.16 | joobie | qakhan, ring the queue and get that dood to answer |
| 10:54.21 | joobie | and do 'queue show' |
| 10:54.28 | joobie | does it say he's in use |
| 10:54.29 | joobie | ? |
| 10:54.34 | qakhan | yes |
| 10:54.38 | joobie | he is in use? |
| 10:55.16 | qakhan | ok agent A transfer call to Agent B |
| 10:56.06 | qakhan | after trans call it shows agent A is free and Agent B also free, while Agent B is on call |
| 10:56.52 | qakhan | i think i am doing something wrong with transfer calls in queue |
| 10:56.59 | qakhan | but dont know what? |
| 10:57.14 | kaldemar | hajekd: can't reproduce your issue on 1.8.7.1 either. |
| 10:57.21 | gordonjcp | yay, my phones arrived |
| 10:57.23 | joobie | i havent debuged this issue before qakhan |
| 10:57.31 | joobie | but your problem is that it shoudl say "in use" |
| 10:57.38 | joobie | if the queue module doesnt see ur ext in use |
| 10:57.39 | qakhan | yes |
| 10:57.44 | joobie | then u are fuked, and it willt ransfer calls to it |
| 10:57.56 | hajekd | kaldemar: You get NoOp message after the AGI(sleep)? |
| 10:58.15 | joobie | gordonjcp, what phone? |
| 10:58.27 | qakhan | can u tell me how tranfser a call to other agent in queue |
| 10:58.35 | kaldemar | hajekd: yes, i have two noops after the AGI call, and i see them. |
| 10:58.40 | joobie | it's just a normal transfer |
| 10:58.48 | joobie | queue module should pick up state for the sip extension |
| 10:58.58 | joobie | like there's no funky "tell the queue i'm busy" type thing u need to do |
| 10:59.14 | joobie | if u dial out |
| 10:59.17 | joobie | from that phone |
| 10:59.21 | joobie | does it register and inuse |
| 10:59.21 | joobie | ? |
| 10:59.23 | joobie | in queue show |
| 10:59.32 | joobie | sweet |
| 10:59.35 | joobie | i found sum scotch |
| 10:59.53 | qakhan | ok |
| 10:59.58 | qakhan | yes plz |
| 11:00.37 | joobie | guys i use 1.4 now |
| 11:00.43 | joobie | if i bump up to 10 or 1.8 |
| 11:00.55 | joobie | would i need to redo my extensions.conf / extensions.ael ? |
| 11:00.59 | hajekd | kaldemar: What do you have in your AGI( sleep )? |
| 11:01.12 | hajekd | kaldemar: will try the same |
| 11:01.17 | qakhan | what joobie |
| 11:01.19 | qakhan | ? |
| 11:01.27 | joobie | qakhan, huh? |
| 11:02.05 | qakhan | you said you got something |
| 11:02.38 | joobie | eh? |
| 11:02.44 | kaldemar | hajekd: some exec noops, a few variable sets and gets, with proper response reads. |
| 11:02.47 | joobie | try dial out bro from that phone |
| 11:02.53 | joobie | and see if it comes up as in use in the queue show |
| 11:03.42 | qakhan | ok |
| 11:05.50 | qakhan | joobie it shows Agent/1003 (Not in use) has taken no calls yet |
| 11:06.07 | qakhan | while i made a call from 1003 |
| 11:06.41 | irroot | joobie the answer is not yes/no there are changes that may affect you some fuctions / apps have been changed or merged |
| 11:06.48 | gordonjcp | joobie: Cisco 7910 |
| 11:07.12 | gordonjcp | joobie: I bought two just to play about with, for 20 quid each I can't really go wrong |
| 11:07.42 | irroot | the one that sticks out is change from | to , as a seperator and changes to SET / MSET |
| 11:07.42 | irroot | its not painfull |
| 11:07.53 | joobie | ahh nice gordonjcp |
| 11:08.02 | joobie | qakhan, that is ur issue |
| 11:08.09 | joobie | qakhan, the state is not coming up |
| 11:08.21 | qakhan | yes i know |
| 11:08.23 | joobie | qakhan, do you get any warnings / errors in console when u dial out? |
| 11:08.26 | *** join/#asterisk frawd (~francois@19.Red-81-39-176.dynamicIP.rima-tde.net) |
| 11:08.30 | qakhan | no |
| 11:08.35 | joobie | what version asterisk |
| 11:08.57 | qakhan | 1.4.38 |
| 11:09.02 | qakhan | Executing [1002@agent:1] Dial("SIP/1003-00000014", "SIP/1002") in new stack |
| 11:09.02 | qakhan | <PROTECTED> |
| 11:09.02 | qakhan | <PROTECTED> |
| 11:09.10 | qakhan | when i called to 1002 |
| 11:10.20 | joobie | can u dump ur sip.conf for 1003 ? |
| 11:10.59 | qakhan | u want to see my 1003 sip.conf? |
| 11:11.04 | joobie | yes |
| 11:11.18 | qakhan | 1 min plz |
| 11:11.31 | joobie | including username and password if it will let me register from my location |
| 11:11.39 | joobie | ;P |
| 11:11.46 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:12.08 | qakhan | [1003] $ |
| 11:12.09 | qakhan | type=friend $ |
| 11:12.09 | qakhan | context=agent $ |
| 11:12.09 | qakhan | username=1003 |
| 11:12.09 | qakhan | callerid=<1003> |
| 11:12.09 | qakhan | host=dynamic $ |
| 11:12.10 | qakhan | secret=1234 $ |
| 11:12.11 | qakhan | limitonpeers=yes |
| 11:12.11 | qakhan | notifyhold=yes $ |
| 11:12.11 | qakhan | call-limit=2 $ |
| 11:12.12 | qakhan | accountcode=Agent |
| 11:14.50 | joobie | it's probably an issue with your limitonpeers / type |
| 11:15.13 | joobie | try remove limitonpeers |
| 11:15.29 | hajekd | kaldemar: really make sure both legs are hanguped when sleeping in that AGI |
| 11:15.55 | joobie | re-register ur sip |
| 11:15.59 | joobie | 1003 |
| 11:16.05 | joobie | after that |
| 11:16.06 | qakhan | accountcode=Agent? |
| 11:16.12 | hajekd | kaldemar: I'm trying now and it does not work. Put sleep 10s in that agi and you should be able to reproduce |
| 11:16.23 | qakhan | what its mean limitonpeers? |
| 11:16.31 | joobie | http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers |
| 11:16.43 | joobie | im gona boot qakhan |
| 11:16.48 | joobie | try remove that |
| 11:16.49 | qakhan | y |
| 11:16.51 | joobie | and re-register |
| 11:17.11 | joobie | also notifyhold |
| 11:17.15 | joobie | try remove that |
| 11:17.23 | joobie | if it doesnt work |
| 11:17.40 | joobie | i use 1.4 and my states are OK |
| 11:18.05 | joobie | i dont use those options.. maybe there is a bug or something becuase they dont seem to directly relate to your issue |
| 11:18.16 | joobie | limitonpeers i can see may affect ur issue, but it's a longshot |
| 11:18.18 | joobie | anyway im out |
| 11:18.20 | joobie | goodluck |
| 11:20.44 | kaldemar | hajekd: i see, that was the first time you said that the originating channel needs to hang up during the AGI execution. the hangup extension will not continue execution if the source channel hangs up during its execution. that has nothing to do with AGI. |
| 11:21.07 | hajekd | kaldemar: dial with g? |
| 11:21.42 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 11:22.02 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 11:22.45 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 11:23.03 | mirelab | l |
| 11:23.26 | mirelab | hello |
| 11:23.35 | kaldemar | hajekd: g has no effect. it makes the dialplan execution go on in the extension that does the dial, when the destination channel hangs up. |
| 11:24.16 | hajekd | kaldemar: ah, so you can reproduce now? |
| 11:24.21 | mirelab | does anyone know why my SIP/<exten>@<IP address> is seen as Unknown |
| 11:24.37 | mirelab | my Agent* |
| 11:25.02 | kaldemar | hajekd: yes, but it is not related to AGI. |
| 11:25.18 | mirelab | i need to see status of SIP agent in queue status |
| 11:32.21 | hajekd | kaldemar: i think it is related to AGI, replace the h,n,AGI(wait.php) with h,n,System(sleep 10) and try it. It will work just fine and the NoOp message at then end is shown in all cases. |
| 11:33.15 | *** join/#asterisk mintos (~mvaliyav@117.206.23.180) |
| 11:36.33 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 11:39.34 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 11:44.24 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 11:44.36 | *** join/#asterisk irroot (~gregory@197.106.243.133) |
| 11:48.08 | *** join/#asterisk tris (tristan@2001:1868:a00a::4) |
| 11:48.43 | kaldemar | hajekd: replace the agi call with a Wait(10) and try it for yourself. |
| 11:49.49 | IsUp | ehlo |
| 11:53.49 | kaldemar | hajekd: System does not work like other applications. |
| 12:00.52 | *** part/#asterisk giany (~giany@shifu.x83.org) |
| 12:06.55 | *** join/#asterisk CVirus (~GoD@41.233.84.9) |
| 12:07.03 | CVirus | Which codec shall I use ulaw or alaw ? |
| 12:07.31 | IsUp | CVirus: it depends |
| 12:07.34 | mirelab | Has anyone used member => SIP/1000 for example in queues? |
| 12:07.50 | CVirus | IsUp: on what exactly ? |
| 12:08.11 | IsUp | CVirus: on your provider, phone |
| 12:08.23 | CVirus | IsUp: I live in Egypt |
| 12:08.36 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 12:08.50 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 12:09.06 | mirelab | CVirus: alaw is for Europian standard and ulaw for US standard |
| 12:09.53 | hajekd | kaldemar: yes with wait(10) it does not work -> so it is a bug |
| 12:09.53 | mirelab | CVirus: now the question is which standard is used in Egypt |
| 12:09.59 | IsUp | CVirus: ask your provider to be sure, in my opinion |
| 12:10.05 | hajekd | kaldemar: wonder if it works in 10 |
| 12:11.34 | kaldemar | hajekd: it behaves the same in 10. |
| 12:12.02 | hajekd | kaldemar: try Dial with g option -> AGI will work and NoOp will be shown -> weird ;) |
| 12:12.34 | *** join/#asterisk Joker (joker@gentoo/developer/joker) |
| 12:13.31 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:17.35 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:20.55 | *** join/#asterisk StaRetji (~BigEight@80.93.240.171) |
| 12:23.13 | *** join/#asterisk mintos (~mvaliyav@115.242.190.235) |
| 12:23.26 | StaRetji | Folks, can someone help me out. I have to record all incoming calls to specific extension. I googled a bit, but so far I failed to make it work. Is there step by step guide for this feature? Thx |
| 12:24.35 | ollii | http://www.voip-info.org/wiki/view/MixMonitor ? |
| 12:25.47 | *** part/#asterisk alex_ole (~a.olehnov@86.57.158.78) |
| 12:26.34 | devmikey | Question: What cell phone service providers are there? I've thought of att, sprint, verizon, alltel, virgin mobile, tmobile, and metropcs. anybody else come to mind? |
| 12:27.05 | ollii | devmikey: a country would be helpful...but i assume your from america :P |
| 12:27.40 | wdoekes2 | :) |
| 12:28.09 | devmikey | us |
| 12:28.41 | devmikey | doesnt the fact that I assume you would know indicative enough? |
| 12:28.58 | StaRetji | ollii: thx dude, I'll check it out. So far I tried exten => 5551,1,Ringing()_________________exten => 5551,2,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)})______________exten => 5551,3,Monitor(wav,${CALLFILENAME},m)__________________exten => 5551,4,Dial(SIP/5551|20|trf) but in /var/spool/asterisk/monitor there is nothing |
| 12:28.58 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.130.16) |
| 12:29.36 | ollii | devmikey: so is america! ;> |
| 12:41.44 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 12:42.12 | dandate2 | skype allows free call termination to toll-free numbers, if i install skype2sip will that allow me to call toll-free numbers for free? |
| 12:46.37 | *** join/#asterisk mintos (~mvaliyav@117.206.23.182) |
| 12:46.44 | jacc0 | I want to send email from asterisk dialplan; can anyone point me to a good example? |
| 12:47.23 | ollii | System(echo "foo" | mail -s "subject" receiver@mail.com ) |
| 12:47.34 | jacc0 | ty |
| 12:47.48 | *** join/#asterisk joelsolanki (~joelsolan@124.125.149.22) |
| 12:47.53 | joelsolanki | Good morning |
| 12:48.02 | ollii | you could use postfix for example |
| 12:49.55 | jacc0 | ty again |
| 12:57.59 | qakhan | hi all |
| 12:58.21 | qakhan | how i setup call transfer to other agent in a queue |
| 13:01.25 | *** join/#asterisk joelsolanki (~joelsolan@124.125.149.22) |
| 13:02.12 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 13:03.23 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:11.45 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 13:12.13 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 13:14.02 | dandate2 | hmm i see skype 2 asterisk is no longer for sale, anyway to get this open source? |
| 13:17.01 | bulkorok | dandate2: freeswitch can: http://wiki.freeswitch.org/wiki/Skypopen |
| 13:18.09 | *** join/#asterisk irroot (~gregory@197.108.119.84) |
| 13:18.11 | bulkorok | dandate2: more infos: http://www.voip-info.org/wiki/view/Skype+Gateways |
| 13:18.15 | dandate2 | i never looked into freeswitch, will this require me to reinstall or scrap my current asterisk phone system? |
| 13:18.47 | bulkorok | dandate2: freeswitch is a seperate program... more like a softswitch than a pbx... |
| 13:19.34 | bulkorok | maybe you find some matching at voip-info.org |
| 13:21.27 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:22.59 | dandate2 | we have all kinds of custom coding for our asterisk build, is there any other way to get free or cheap termination to toll-free numbers? |
| 13:23.36 | dandate2 | all the voip providers are charging me 1 cent per minute; which is really lame since skype is free calls to toll-free |
| 13:25.40 | IsUp | dandate2: i dont think that Skype is totally 'free', probably theres a 'fair usage' policy |
| 13:27.56 | dandate2 | well if i have a skype account with $0 balance i can call toll-free as much as i like; mabye they would crack down on a pbx making tons of concurrents tho |
| 13:28.58 | IsUp | dandate2: i can recommend a provider, i'm pming u |
| 13:30.50 | dandate2 | cool |
| 13:32.08 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers |
| 13:32.40 | IsUp | hello [TK]D-Fender |
| 13:38.17 | devmikey | is there a site you trust to review wireless provider quality |
| 13:38.39 | dandate2 | wow they were truly free toll-free calls |
| 13:38.44 | dandate2 | i gotta setup this trunk asap |
| 13:42.07 | leifmadsen | qakhan: be more specific -- you can transfer in any number of ways |
| 13:46.23 | *** join/#asterisk joelsolanki (~joelsolan@124.125.149.22) |
| 13:46.28 | joelsolanki | good morning guys |
| 13:46.35 | joelsolanki | http://pastebin.com/H5c7kbR0 |
| 13:47.22 | joelsolanki | plz take a look. this is vps server. i want to use it for DIDs unfortunately. when registering eyebeam from private IP/nat IP calls doesnt come on eyebeam. |
| 13:47.30 | joelsolanki | it works on public ip. can recommendations plz |
| 13:48.14 | IsUp | joelsolanki: set your externip= and localnet= under [general] in sip.conf |
| 13:48.49 | leifmadsen | plus if asterisk is behind a firewall/nat you'll need to make the appropriate changes to allow external devices to communicate with it |
| 13:49.32 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 13:49.52 | joelsolanki | no asterisk is not on private ip. |
| 13:49.56 | joelsolanki | eyebeam is on private ip. |
| 13:50.04 | joelsolanki | asterisk has a public ip |
| 13:50.24 | joelsolanki | so no need to use externip stuff. |
| 13:51.07 | kaldemar | joelsolanki: explain "registered on NAT IP 192.168.1.52" some more. is that the address of the eyebeam? |
| 13:51.07 | joelsolanki | any solution for this problem ? |
| 13:51.17 | joelsolanki | sure |
| 13:51.45 | joelsolanki | eyebeam is installed on windows with ip of 192.168.1.52 |
| 13:51.51 | [TK]D-Fender | joelsolanki, You should have qualify=yes as a NAT keepalive on your peer <- |
| 13:52.01 | kaldemar | set qualify=yes for it. the router might be dropping the port. |
| 13:52.18 | joelsolanki | i see. let me try it one moment |
| 13:52.28 | s[X] | [TK]D-Fender: hey |
| 13:52.30 | [TK]D-Fender | joelsolanki, Correct this then if it failes (after restarting eyebeam for the test) then pastebin the SIP DEBUG of the registration attempt |
| 13:53.12 | [TK]D-Fender | joelsolanki, And now is a great time to stop using "|" as a parameter delimiter in extensions.conf .... |
| 13:53.30 | leifmadsen | [TK]D-Fender: now? :) |
| 13:53.32 | [TK]D-Fender | joelsolanki, Before it bites you in the behind when you upgrade |
| 13:53.41 | [TK]D-Fender | leifmadsen, "Then" too :) |
| 13:53.46 | joelsolanki | :) |
| 13:53.47 | joelsolanki | correct |
| 13:53.50 | leifmadsen | btw: "then" was about 6 years ago |
| 13:54.24 | [TK]D-Fender | leifmadsen, Well he wasn't bitten "then", so "now" would still be a good time :) |
| 13:54.45 | [TK]D-Fender | So when will "then" be "now"? ........ |
| 13:54.46 | [TK]D-Fender | SOON! |
| 13:54.53 | [TK]D-Fender | </spaceballs> |
| 13:55.29 | leifmadsen | :D |
| 13:56.05 | s[X] | system-config-network |
| 13:56.20 | s[X] | lol |
| 13:57.30 | joelsolanki | i am getting same issue |
| 13:57.36 | joelsolanki | let me put in sip debug |
| 13:59.18 | joelsolanki | here is the sip debug |
| 13:59.19 | joelsolanki | http://pastebin.com/c9UTwXP1 |
| 14:00.50 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 14:01.41 | [TK]D-Fender | joelsolanki, There is no registration attempt from eyebeam in there. |
| 14:01.56 | *** join/#asterisk TimeRider (~steve@92.40.254.189.threembb.co.uk) |
| 14:02.04 | [TK]D-Fender | <PROTECTED> |
| 14:02.05 | joelsolanki | you want me to register it again and do sip debug ? |
| 14:02.10 | joelsolanki | oh ok |
| 14:02.14 | joelsolanki | let me do it again plz |
| 14:02.14 | [TK]D-Fender | That's what I told you to do. |
| 14:02.25 | [TK]D-Fender | Please do |
| 14:02.34 | joelsolanki | ok sure |
| 14:02.44 | dym | [Nov 30 15:03:07] WARNING[2853]: pbx.c:8747 ast_pbx_run_app: No such application 'txfax' |
| 14:02.47 | dym | :( |
| 14:02.51 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:03.32 | dym | is it not called txfax anymore? |
| 14:03.53 | [TK]D-Fender | dym, You are probably missing prerequisites when you installed *. |
| 14:04.02 | dym | spandsp |
| 14:04.14 | [TK]D-Fender | dym, look in menuconfig to see if it's ***'d |
| 14:04.21 | [TK]D-Fender | That's usually it |
| 14:04.48 | dym | its app_txfax right? |
| 14:05.09 | *** join/#asterisk serafie (~erin@nat/digium/x-txytgnlcrqttydgh) |
| 14:05.23 | dym | res_fax and res_fax_spandsp are selectable |
| 14:06.22 | dym | [TK]D-Fender: so the spandsp one should be what im looking for, right? |
| 14:06.54 | [TK]D-Fender | IIRC that is the generic one, not FFA. |
| 14:06.56 | [TK]D-Fender | Go for it |
| 14:07.53 | dym | cant select - its in < > |
| 14:08.00 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:08.03 | dym | ah |
| 14:08.04 | dym | now |
| 14:09.54 | joelsolanki | here is it is http://pastebin.com/a36Q5K7u |
| 14:09.57 | joelsolanki | plz check. |
| 14:10.35 | dym | [TK]D-Fender: Do you know if, while transmitting a fax, it is possible to check the status, and/or cancel the sending? |
| 14:11.30 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:12.03 | [TK]D-Fender | dym, Don't think so... |
| 14:12.20 | [TK]D-Fender | dym, If that's what you're looking for I'd recommend IAXModem + Hylafax |
| 14:12.49 | *** join/#asterisk [1]joelsolanki (~joelsolan@124.125.149.22) |
| 14:12.54 | [1]joelsolanki | hi |
| 14:13.04 | [1]joelsolanki | sorry i got disconnecteed |
| 14:13.12 | [1]joelsolanki | here is it is http://pastebin.com/a36Q5K7u |
| 14:13.22 | wdoekes2 | joelsolanki: NAT/firewall doesn't stay punctured. either you need to lower the qualifyfreq (more often), or enable keepalived on the remote end |
| 14:13.35 | wdoekes2 | s/lived/lives |
| 14:14.04 | [1]joelsolanki | qualify=400 is fine ? |
| 14:15.00 | joelsolanki | let me know plz |
| 14:15.13 | wdoekes2 | or you could read the sample sip.conf |
| 14:15.18 | [TK]D-Fender | [1]joelsolanki, Looks registered.. |
| 14:15.36 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 14:15.42 | [TK]D-Fender | joelsolanki, However these retransmits make me wonder about something filtering the qualify packets. |
| 14:16.06 | [TK]D-Fender | don't lower qualify... that is the TTL, not the frequency |
| 14:16.23 | ollii | maybe an openvpn tunnel would be fine...nothing to worry about nat ?! |
| 14:16.38 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-igucirkubtoyngwx) |
| 14:17.12 | joelsolanki | i see |
| 14:17.31 | joelsolanki | asterisk is installed on virtual private server from www.server4you.com guys |
| 14:17.42 | joelsolanki | i doubt it has something to do with it ? |
| 14:17.54 | [TK]D-Fender | joelsolanki, I'm suspecting more on your eyebeam end |
| 14:18.12 | [TK]D-Fender | it isn't responding. Please describe in full detail that entire side of this picture. |
| 14:18.14 | CVirus | What is the difference between dahdi-complete and dahdi-linux ? |
| 14:18.28 | [TK]D-Fender | CVirus, complete + linux + tools |
| 14:18.30 | [TK]D-Fender | = |
| 14:18.39 | [TK]D-Fender | CVirus, complete = linux + tools |
| 14:18.47 | CVirus | aha |
| 14:18.49 | CVirus | [TK]D-Fender: Thanks |
| 14:18.54 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:19.08 | joelsolanki | ok then i should test on other private ip. what do you suggest ? |
| 14:19.18 | joelsolanki | i mean other than this network ? |
| 14:20.33 | dym | [TK]D-Fender: we just switched from hylafax :D |
| 14:20.41 | dym | we wanted a more versatile system |
| 14:20.54 | CVirus | [TK]D-Fender: I'm using dahdi-linux-2.5.0.1 and wanpipe-3.5.23 .. are they compatible ? |
| 14:21.45 | ollii | CVirus: yeah...somehow they are |
| 14:21.50 | CVirus | ollii: thanks |
| 14:22.02 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.2) |
| 14:22.02 | ollii | #sangoma could help you for detailed questions |
| 14:23.52 | dym | [TK]D-Fender: Even after compiling with fax_spandsp i dont get app_rxfax / app_txfax |
| 14:24.10 | dym | [Nov 30 15:23:58] WARNING[13426]: pbx.c:8747 ast_pbx_run_app: No such application 'txfax' |
| 14:24.25 | [TK]D-Fender | dym, "core show applications like fax" |
| 14:24.33 | [TK]D-Fender | -s |
| 14:24.49 | dym | <PROTECTED> |
| 14:24.49 | dym | <PROTECTED> |
| 14:24.49 | dym | <PROTECTED> |
| 14:24.51 | dym | oddd |
| 14:24.53 | dym | odd |
| 14:24.55 | [TK]D-Fender | Even :) |
| 14:25.09 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 14:25.38 | dym | So its Sendfax? Is that still txfax of spandsp? |
| 14:26.04 | joelsolanki | any suggestion D-Fender ? |
| 14:28.55 | SeRi | g/m all. |
| 14:30.34 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:33.45 | wdoekes2 | joelsolanki: did you try my suggestions already? |
| 14:35.44 | dym | How can I group a set of DAHDI Channels? |
| 14:35.54 | *** join/#asterisk lcat (~lcat@187.45.255.76) |
| 14:38.33 | Faustov | What do you recommend to achieve the following result: 1 extension rings 2 agents over IAX trunks simultaneously, until the first one answers? I'm having a problem when someone sets "DND" and due to that, the other agent does not ring at all (gets SIP response 483) |
| 14:39.26 | *** join/#asterisk screenn (~screenn@178.251.111.71) |
| 14:41.08 | *** join/#asterisk openstandards (~openstand@87.127.196.56) |
| 14:43.16 | leifmadsen | Faustov: use the flag in Dial() to ignore the busy message |
| 14:44.04 | *** join/#asterisk irroot (~gregory@197.169.46.11) |
| 14:47.08 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:50.02 | Faustov | leifmadsen: the I option? |
| 14:50.18 | leifmadsen | I don't know, does it work? does the description seem to make sense? |
| 14:50.20 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:50.52 | Faustov | not really, neither option's description seems to fit to exactly what you said, so I'm assuming one of the options includes this |
| 14:51.11 | Faustov | I - Asterisk will ignore any connected line update requests or redirecting party update requests it may receiveon this dial attempt. |
| 14:51.15 | Faustov | this one seems the closest |
| 14:51.25 | Faustov | from here: https://wiki.asterisk.org/wiki/display/AST/Application_Dial |
| 14:54.10 | [TK]D-Fender | joelsolanki, I'm waiting for the answer to my last request |
| 14:54.38 | leifmadsen | Faustov: my bad -- 'i' is for forwarding requests, not busy requests |
| 14:55.07 | leifmadsen | Faustov: you'll need to use a Local channel to call them independently then |
| 14:55.23 | leifmadsen | Dial(Local/100@phones&Local/101@phones) |
| 14:55.33 | [TK]D-Fender | Faustov, Dial each leg via local channels and add a long Wait() on the end to kill time in case of no answer. |
| 14:55.36 | leifmadsen | then you can handle call handling independently |
| 14:56.34 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 14:58.44 | Faustov | is processing |
| 14:59.51 | Faustov | ok, thanks guys, seems like this is what I need to try |
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| 15:11.39 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 15:12.02 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 15:12.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:12.21 | iprouteth0 | are g.729 license fees one-time? |
| 15:13.09 | jeffspeff | when i do an atxfer the CID doesn't update properly for the person that i transfered the call to; it still shows my phones CID. any suggestions? |
| 15:14.02 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:14.44 | *** join/#asterisk nanoha-sama (~nanoha-sa@curriegrad2004-3-pt.tunnel.tserv14.sea1.ipv6.he.net) |
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| 15:25.04 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:25.47 | *** join/#asterisk timahvo1 (~rogue@41.90.206.172) |
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| 15:29.12 | [TK]D-Fender | iprouteth0, Per server |
| 15:30.06 | SeRi | WIMPy: you in? |
| 15:30.13 | iprouteth0 | But no type of renewal needed like cisco licensing for instance |
| 15:30.15 | iprouteth0 | ? |
| 15:30.34 | WIMPy | SeRi: hi |
| 15:31.21 | SeRi | WIMPy: was it you that was talking about packet shaping/qos a week back with dijib? |
| 15:31.49 | WIMPy | I did take part at some time, yes. |
| 15:32.52 | SeRi | WIMPy: If my mind serves me correct I think it was you or flor that said that the goal was not to have packet drops or something similar I cant remember exactly what it was.... |
| 15:33.07 | *** join/#asterisk navaismo (~navaismo@201.123.84.9) |
| 15:33.15 | WIMPy | +not |
| 15:33.44 | WIMPy | I.e. to make sure you don't send more packets that will be transmitted by your modem. |
| 15:33.56 | WIMPy | than |
| 15:34.17 | SeRi | ok so something must be wrong on my setup. the qOthersHigh on LAN has over 768 packets droped |
| 15:34.30 | WIMPy | And additionally to reorder packets. |
| 15:34.37 | SeRi | while everything else seems ok |
| 15:34.59 | WIMPy | You mean your traffic control has dropped packets? |
| 15:35.32 | SeRi | <PROTECTED> |
| 15:35.48 | SeRi | ops |
| 15:35.50 | SeRi | one sec |
| 15:37.05 | SeRi | 0/pps 0 b/s 0 borrows 0 suspends 768 drops |
| 15:37.15 | SeRi | ^^ thats what is reporting |
| 15:37.27 | WIMPy | What? |
| 15:37.56 | SeRi | My Traffic Shaper |
| 15:38.21 | SeRi | The queue " qOthersHigh on LAN" |
| 15:38.30 | WIMPy | If you try to use more BW than you have, you will get dropped packets. The point is to make sure to only drop packets that aren't important. |
| 15:39.02 | WIMPy | Yes, you transmit scheduler will drop packets if neccessary. That's the idea. |
| 15:39.30 | WIMPy | If you don't, your modem will and that will just drop any random packets. |
| 15:40.32 | SeRi | ok. I understand. ill go back to the drwaing board and see what is consuming most of the bandwidth. and refine some of the queues |
| 15:40.41 | SeRi | Thanks |
| 15:41.25 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
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| 15:43.02 | devmikey | anybody here use 'fon'? |
| 15:44.56 | [TK]D-Fender | iprouteth0, Correct. One time per server |
| 15:47.47 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
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| 16:00.56 | Qwell | devmikey: the wireless thingie? I thought that died like 5 years ago. |
| 16:01.51 | devmikey | yes the wireless thingy |
| 16:05.50 | devmikey | Basically I want to buy something like att wifi or boingo access |
| 16:06.07 | devmikey | that looked like an interesting alternatvie |
| 16:07.00 | *** join/#asterisk moy (~moy@216.172.42.74) |
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| 16:43.22 | IsUp | hello |
| 16:43.27 | dym | Hi there! |
| 16:43.28 | devmikey | goodbye |
| 16:43.33 | dym | devmikey: oi! behave |
| 16:47.19 | *** join/#asterisk hfb (~hfb@pool-98-119-109-145.lsanca.dsl-w.verizon.net) |
| 16:47.48 | *** join/#asterisk timahvo1 (~rogue@197.179.49.72) |
| 16:51.18 | StaRetji | Need working example of http://www.voip-info.org/wiki/view/MixMonitor |
| 16:51.21 | StaRetji | thx ;) |
| 16:54.00 | *** join/#asterisk bullium (~wbradshaw@216.68.250.30) |
| 16:54.34 | bullium | How can I monitor/debug a single sip peer and only that peer from within the console |
| 16:55.18 | [TK]D-Fender | bullium, "sip set debug peer [peer or ip]" |
| 16:55.30 | leifmadsen | [TK]D-Fender: WOAH |
| 16:55.31 | bullium | [TK]D-Fender, thanks |
| 16:57.07 | [TK]D-Fender | leifmadsen, ... </keanu> ? |
| 16:57.14 | leifmadsen | [TK]D-Fender: sure! |
| 16:57.22 | [TK]D-Fender | PARTY ON! |
| 17:06.09 | p3nguin | seri: Bad news... |
| 17:06.25 | p3nguin | No shaper, calls still get dropped after a random amount of time. |
| 17:07.48 | p3nguin | I've made another change and rebooted the router once, so I'll see if that makes any difference. If it does not, I'll have to change out the cable modem. |
| 17:13.51 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 17:16.49 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 17:20.27 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 17:24.25 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-tqynaarblksvrvdm) |
| 17:24.26 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 17:24.59 | Qwell | I wonder what Cisco uses Asterisk for! |
| 17:25.15 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:26.42 | gordonjcp | ooh, a cisco guy, great |
| 17:27.05 | gordonjcp | wonder if he knows about 7910G+SW phones not working off PoE |
| 17:27.16 | p3nguin | Now if only he would actually participate in discussion and/or help with Cisco issues. |
| 17:27.24 | Qwell | p3nguin: if only |
| 17:27.49 | gordonjcp | if no-one turns up at the hackspace soon, I'm off home to play with my phones |
| 17:27.56 | p3nguin | I have a Cisco issue that no one seems to know how to overcome. |
| 17:28.24 | WIMPy | p3nguin: Do yu think it is possible to overcome? |
| 17:28.37 | p3nguin | Yes, but I don't know how to do it. |
| 17:28.39 | Qwell | WIMPy: Given that it's Cisco...no. |
| 17:29.05 | WIMPy | Qwell: That's what my experience tells me. |
| 17:29.07 | p3nguin | It's possible. Maybe not possible with Asterisk and chan_sccp-b, but it is possible. |
| 17:29.31 | p3nguin | I just want the phone to ring when active rather than give a call waiting tone. |
| 17:29.46 | p3nguin | There's a parameter for it when using call manager. |
| 17:30.13 | Qwell | yeah you'd need to patch the source for that |
| 17:31.02 | p3nguin | Where do I need to concentrate? I thought it might just work if I put the right settings in the xml file. |
| 17:31.05 | WIMPy | Yes, I also like it when the phone just rings. |
| 17:31.45 | p3nguin | something RingActive something |
| 17:33.19 | p3nguin | I think it would just give one single ring for a new call while I'm already on the phone. That's better than just the beep tone in the ear piece. |
| 17:33.46 | WIMPy | That's what my old phones used to do. |
| 17:34.14 | r0m|u | p3nguin: waz up |
| 17:34.34 | WIMPy | But maybe one day the VOIP world will catch up with the old stuff. |
| 17:34.44 | *** join/#asterisk AmirBehzad (~behzad@86.57.4.5) |
| 17:35.09 | p3nguin | (1106.24) <p3nguin> No shaper, calls still get dropped after a random amount of time. |
| 17:35.41 | WIMPy | Not for that part probably. |
| 17:35.52 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:35.53 | wcselby | o/ |
| 17:36.14 | r0m|u | p3nguin: Like I said before the issue seems past the shapers :/ |
| 17:36.21 | r0m|u | wcselby: hola! |
| 17:36.49 | wcselby | r0m|u did you ever get sorted with Comcast? |
| 17:37.01 | r0m|u | wcselby: Yes. |
| 17:37.06 | r0m|u | All ok now. |
| 17:37.29 | p3nguin | I'm concentrating on the cable modem now. |
| 17:37.34 | r0m|u | random drops still excist but my phone and my account has been untouched :) |
| 17:37.41 | wcselby | heh |
| 17:38.19 | r0m|u | p3nguin: can you log in to your cable modems status page? |
| 17:38.25 | p3nguin | Yes. |
| 17:38.42 | r0m|u | I have a route set in my firewall to allow me to reach 192.168.100.1 |
| 17:38.58 | p3nguin | Me too! It's called the "default gateway." |
| 17:39.08 | wcselby | lol |
| 17:39.11 | p3nguin | :D |
| 17:39.21 | p3nguin | Everyone has one. |
| 17:39.29 | *** join/#asterisk kikohnl (~kotis@ext-dip-166.hnl.cdsinc.com) |
| 17:39.46 | WIMPy | You don't filter private IPs? |
| 17:40.02 | r0m|u | p3nguin: not everybody can do that |
| 17:40.09 | p3nguin | Why not? |
| 17:40.20 | r0m|u | cm are set in bridge mode and people most of the time can not reach the status page |
| 17:40.29 | p3nguin | They're doing it wrong. |
| 17:41.09 | p3nguin | Even with a modem that is bridging, the modem's address is still available on the LAN port. |
| 17:41.10 | r0m|u | And is not the GW ether. my firewalls see the comcast gateway not my modems IP. |
| 17:41.46 | p3nguin | Uh, no, that's not what I meant. I mean that the default gateway is used when you try to access 192.168.100.1. |
| 17:41.53 | r0m|u | ah! :) |
| 17:41.54 | p3nguin | It is then sent via your router to the modem. |
| 17:41.55 | r0m|u | ok |
| 17:42.05 | r0m|u | I see :) |
| 17:42.47 | p3nguin | If your router does not allow 192.168.0.0/16 to be router to the WAN interface, that would prevent the modem from being reached. |
| 17:43.12 | p3nguin | s/be router/be routed/ |
| 17:43.47 | p3nguin | But I've never ever encountered that in any residential router/modem deployment. Ever. |
| 17:43.56 | r0m|u | p3nguin: can you PB RF Parameters? basically most issues come from high noise |
| 17:44.08 | r0m|u | low power |
| 17:44.16 | p3nguin | I'll have a look at it. |
| 17:45.20 | p3nguin | SNR is 37dB, which is way above the lower limit. |
| 17:45.27 | r0m|u | if the modem is the issues... the modem log will show you an "out of sync" in the logs. "out of syncs" happen very quick but applications that are sensitive to connection can detect a drop and take you off line |
| 17:45.28 | p3nguin | They're happy with 26. |
| 17:45.40 | r0m|u | if an "out of sync" happen for to long the modem reboots |
| 17:46.05 | IsUp | r0m|u: its cable or dsl? |
| 17:46.12 | r0m|u | cable |
| 17:46.23 | p3nguin | Downstream power is -4 dB and upstream is 45 dB. Both of those are very acceptable values. |
| 17:46.29 | wcselby | i'm trying to do a boolean evaluation inside a gotoif statement, if I want to check if something has returned false, can I just say GotoIf($[!${validDid}]... or do I need to use GotoIf($["${validDid}" = "false"]... ? |
| 17:46.51 | p3nguin | 0 and 50 is preferred, but downstream is good if it is between -10 and 10. |
| 17:47.19 | *** join/#asterisk vpopov (~happylife@149.62.3.49) |
| 17:47.23 | p3nguin | wcselby: You can use ! to negate it. |
| 17:47.55 | p3nguin | At least in other situations, like !${ISNULL}. |
| 17:48.12 | p3nguin | !${ISNULL(whatever)} |
| 17:48.24 | p3nguin | which would be the same as using EXISTS(). |
| 17:48.33 | r0m|u | p3nguin: Corrected/Uncorrectables? |
| 17:48.34 | wcselby | heh |
| 17:48.46 | p3nguin | !${EXISTS(whatever)} would be the same as using ISNULL(). |
| 17:48.53 | p3nguin | r0m|u: What? |
| 17:49.55 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:3c68:c9fa:9143:78c6) |
| 17:50.03 | r0m|u | when lots of noise happens the modem tend to correct mangled packet |
| 17:50.11 | r0m|u | p3nguin: ^^ |
| 17:50.21 | p3nguin | But my SNR is very high, so I doubt there's much noise. |
| 17:50.40 | r0m|u | when ALOT of noise happens the modem is unable to correct them and loged them |
| 17:50.48 | p3nguin | a lot, maybe? |
| 17:50.52 | r0m|u | your SNR seems ok though it can still happen |
| 17:50.58 | r0m|u | p3nguin: yes :P |
| 17:51.02 | p3nguin | (since alot isn't a word) |
| 17:51.10 | r0m|u | yes sr. |
| 17:51.20 | p3nguin | Did they have that on your test? |
| 17:51.25 | r0m|u | salutes sargent spell! |
| 17:51.34 | r0m|u | lol |
| 17:51.47 | r0m|u | p3nguin: no |
| 17:51.52 | r0m|u | Thank God |
| 17:51.53 | p3nguin | awww |
| 17:51.56 | r0m|u | I would probable fail |
| 17:52.02 | p3nguin | I wanted to be in your head for that one, too! |
| 17:52.16 | r0m|u | hahahaha |
| 17:52.25 | r0m|u | It will stick ;) |
| 17:52.40 | r0m|u | I am getting better though :) |
| 17:53.27 | p3nguin | I hope I can work out whatever is dropping calls so I can use the shaper again. |
| 17:54.10 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:55.17 | wcselby | afk |
| 17:57.29 | jeffspeff | on a cisco 504g, i'm trying to set a programmed softkey to dial a featurecode that i've defined in features.conf. however the softkey keeps passing the code through the dialplan instead of dtmf (like if i just dialed the number) which works. |
| 17:57.50 | IsUp | p3nguin, i dont know much about your case but are you able to run wireshark or anything to save a trace or something? |
| 17:57.59 | IsUp | p3nguin, maybe it helps |
| 17:58.13 | *** join/#asterisk rdahlin_1 (~rdahlin_1@78-73-19-238-no168.tbcn.telia.com) |
| 17:59.00 | p3nguin | I could run tshark on the router, I guess. |
| 17:59.44 | IsUp | also, i am using adsl on my home. i was having strange problems. my link was going up/down at random times. |
| 18:00.08 | IsUp | finally my telco techs replaced my cable, from my apartment to their street box or whatever its called |
| 18:00.26 | IsUp | and my noise levels was fine befure they replace anything |
| 18:00.52 | r0m|u | p3nguin: run it. |
| 18:01.10 | r0m|u | eliminate the router out of the equation |
| 18:01.31 | r0m|u | if the router is not the issue and asterisk is not the issue than there is only one thing left |
| 18:02.32 | IsUp | p3nguin, your girlfriend is hanging up phone randomly. |
| 18:04.46 | p3nguin | If I record the call to listen to what happens, both sides of the call still have audio hitting asterisk, but they are no longer talking to each other. Both sides questioning, "Are you still there?" at the same time. |
| 18:05.08 | p3nguin | SIP registrations also drop at that exact moment. |
| 18:06.29 | p3nguin | Anyone know a toll-free number not provided on VoIP.ms that would have an extremely long hold time? :) |
| 18:06.37 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 18:07.10 | p3nguin | I'd need about 45 minutes to see if it happens again. |
| 18:08.21 | IsUp | p3nguin: you can use my voxbeam account if needed. i have enough credits. |
| 18:08.37 | p3nguin | I can't get a good test if I change accounts. |
| 18:08.43 | devmikey | anyone ever use boingo? |
| 18:08.46 | p3nguin | I need to leave everything as it is. |
| 18:09.01 | p3nguin | Just need to call some number and sit on hold for an hour. |
| 18:09.08 | r0m|u | p3nguin: would it work if you call a conf? |
| 18:09.23 | p3nguin | If it's not on voipms, yes. |
| 18:09.30 | p3nguin | I need it to be on the PSTN. |
| 18:09.40 | r0m|u | you can dial my CC ID |
| 18:09.52 | IsUp | p3nguin: i can provide you my company DID, and i can put Wait() or whatever you need. its PSTN. |
| 18:09.52 | r0m|u | ooo |
| 18:10.12 | r0m|u | p3nguin: I also have a DID from CC but its not 1800 |
| 18:10.50 | p3nguin | I was going to make some larger company pay for the call to their toll-free so I don't pay for minutes... |
| 18:11.15 | r0m|u | Yea I figured.... sorry :( |
| 18:11.16 | p3nguin | but I could call non-toll-free. It would only cost a little bit. |
| 18:11.32 | p3nguin | Not big deal, I was just being a cheap ass. |
| 18:11.35 | IsUp | p3nguin: i have an Asterisk test server with public ip if you want |
| 18:11.37 | r0m|u | p3nguin: msg me ill set you up |
| 18:11.44 | IsUp | p3nguin: you can talk sip-to-sip if you needed |
| 18:11.51 | p3nguin | Again, it has to be on the PSTN. |
| 18:11.52 | IsUp | p3nguin: i can give you my root access |
| 18:11.53 | r0m|u | IsUp: he needs pstn |
| 18:12.09 | p3nguin | Changing the scenario does not give a good test. |
| 18:12.15 | p3nguin | The conditions must remain the same. |
| 18:12.26 | IsUp | p3nguin: yea just trying to help :p ok i have my Turkey PSTN DID. if you need anything just let me know |
| 18:13.15 | p3nguin | I have a feeling that will be routed differently, so I really need to use a US number not on my provider. |
| 18:13.22 | r0m|u | p3nguin: if it would work msg me and ill set you up with my CC DID that you can call. |
| 18:13.37 | p3nguin | Does it go into a conference? |
| 18:13.50 | r0m|u | I can have it go to what ever you tell me |
| 18:13.52 | p3nguin | with moh, preferably |
| 18:14.00 | r0m|u | ok |
| 18:14.07 | IsUp | or you can use Echo maybe |
| 18:14.21 | p3nguin | Just need an Answer() and a MeetMe() or ConfBridge() with moh. |
| 18:14.43 | r0m|u | I have meetme setup. |
| 18:14.52 | r0m|u | one sec while I make the change p3nguin |
| 18:14.56 | p3nguin | okay |
| 18:15.14 | r0m|u | ^^ROFL^^ |
| 18:15.22 | p3nguin | heh |
| 18:16.56 | p3nguin | 281 or 832 area code? I see two numbers on calls from you via CC. |
| 18:20.18 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 18:20.20 | r0m|u | 832 |
| 18:21.57 | *** join/#asterisk irroot (~gregory@197.104.117.199) |
| 18:30.36 | IsUp | p3nguin: i pmed u my account |
| 18:30.38 | IsUp | gotta go |
| 18:31.22 | *** join/#asterisk asilva (~andre@2801:88:1000:2::12) |
| 18:32.03 | asilva | Hello, i'm trying to enable atxfer on features.conf, but i when i press *2 when try to type the destination number for the transfer i can only type 1 digit and the transfer got broken |
| 18:33.21 | *** join/#asterisk Cesar_B (~chatzilla@201.200.175.218) |
| 18:35.18 | Cesar_B | hi all, i m using asterisk 1.4.21.2 , and i want to have the billsec in milliseconds, something like this: billsec=23.215 , is that possible? |
| 18:38.54 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 18:40.04 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:40.15 | p3nguin | I guess it would have been helpful for isup to let me know about a pm prior to doing so, since I block unsolicited messages. |
| 18:41.30 | p3nguin | So now I just sit and watch the console and tshark... and wait. |
| 18:42.57 | p3nguin | I'm only watching SIP and RTP on my WAN interface. Should that be good enough? |
| 18:43.17 | r0m|u | I belive so. |
| 18:43.18 | *** join/#asterisk irroot (~gregory@197.174.108.107) |
| 18:44.02 | r0m|u | p3nguin: I am about to start setting up a white list. for some reason I been getting pm's as well :/ |
| 18:44.19 | r0m|u | I see how that can be annoying for you guys |
| 18:44.22 | [TK]D-Fender | Cesar_B, Certainly not in that branch. |
| 18:44.23 | r0m|u | and wuick |
| 18:44.30 | [TK]D-Fender | Cesar_B, And that is a massively outdated release |
| 18:44.53 | *** join/#asterisk pietro (~pietro@88-149-227-118.dynamic.ngi.it) |
| 18:45.06 | r0m|u | s/wuick/quick/ |
| 18:46.04 | p3nguin | That will be good if it was the modem giving me a problem. I'll be able to put the shaper back on if it's fixed now. |
| 18:47.04 | Qwell | Cesar_B: Why such an old version? |
| 18:47.39 | r0m|u | p3nguin: That will be the easy way :) I hope for you that it is the modem having to track down issues with vyatta seems bit obscure... |
| 18:48.25 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
| 18:50.10 | [TK]D-Fender | Qwell, Looks like an old Debian stable # |
| 18:50.37 | Qwell | More like Debian insecure. |
| 18:51.31 | [TK]D-Fender | Qwell, SHHH!! Leave them their delusions... it's all they've got left ;) |
| 18:52.48 | Cesar_B | [TK]D-Fender: in what branch can have that feature? |
| 18:53.17 | [TK]D-Fender | Cesar_B, I'd start looking at 1.8.... not sure if it's there, but it is LTS at least |
| 18:53.24 | Cesar_B | Qwell: because i m using a2billing |
| 18:53.56 | Qwell | a2billing supports 1.8... |
| 18:54.55 | Cesar_B | ok, i will give a try, what its the setting in the conf to have milliseconds in the billsec? anyone knows? |
| 18:55.05 | Cesar_B | in the 1.8 branch |
| 18:56.28 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 18:58.56 | pdtpatrick1 | question .. is there a way to get clients that are registered? |
| 18:59.18 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
| 18:59.34 | wcselby | pdtpatrick1 sip show registration? |
| 18:59.37 | wcselby | sip show peers? |
| 18:59.51 | wcselby | depends on what technology your clients are using and if you ahve qualify set or not |
| 19:00.09 | pdtpatrick1 | clients that have successfully authenticated to the AMI |
| 19:00.21 | wcselby | manager show users I think |
| 19:00.25 | wcselby | or manager show sessions maybe |
| 19:00.55 | pdtpatrick1 | cool |
| 19:01.03 | pdtpatrick1 | there's no sessions |
| 19:01.12 | wcselby | manager show connected |
| 19:01.26 | pdtpatrick1 | ahh i c |
| 19:01.39 | wcselby | at least on 1.6.2.15 that works |
| 19:02.32 | pdtpatrick1 | yeah that's what i was looking for :) Thanks |
| 19:09.22 | wcselby | i'll bbl |
| 19:10.02 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 19:10.19 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:14.14 | pdtpatrick1 | is there a more detailed wiki for lua with asterisk besides the one in the wiki ? |
| 19:14.46 | jeffspeff | on a cisco 504g, i'm trying to set a programmed softkey to dial a featurecode that i've defined in features.conf. however the softkey keeps passing the code through the dialplan instead of dtmf (like if i just dialed the number) which works. any ideas on how to either get the softkey to dial through dtmf instead of extension or how to put the featurecode in extensions.conf instead of features.conf? |
| 19:16.02 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 19:18.37 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 19:18.50 | *** join/#asterisk irroot (~gregory@197.169.161.132) |
| 19:22.11 | p3nguin | tshark: The file "/tmp/wiresharkXXXXRHx506" could not be opened: Uncompression error: buffer error. |
| 19:23.47 | p3nguin | tshark broke. |
| 19:23.50 | p3nguin | I can't restart it. |
| 19:24.25 | WIMPy | Is it still running? |
| 19:28.16 | p3nguin | no |
| 19:28.22 | p3nguin | It died, and it won't restart. |
| 19:29.04 | p3nguin | Uncompression error: buffer error. I don't really get it. |
| 19:31.00 | p3nguin | I deleted all the files it created, now it starts again. |
| 19:32.47 | p3nguin | But now I don't see RTP. |
| 19:34.59 | WIMPy | What are the new features in Asterisk 11? |
| 19:36.20 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:36.54 | *** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net) |
| 19:37.16 | p3nguin | Ah, it doesn't know they are RTP packets since I lost the original stream. Now they are just showing up as UDP it seems. |
| 19:46.55 | r0m|u | p3nguin: how is it holding out? |
| 19:47.26 | p3nguin | It seems to be up still. I called in to the conf and I can hear audio from that phone. |
| 19:47.40 | p3nguin | I had her call the conf and just put down the phone. |
| 19:47.54 | p3nguin | It'd over 1 hour. |
| 19:47.58 | p3nguin | it's |
| 19:48.14 | r0m|u | I see. |
| 19:48.30 | p3nguin | It usually dropped between 1 and 40 minutes randomly. |
| 19:48.42 | r0m|u | Mhhhh and nothing yet..... |
| 19:48.50 | p3nguin | Now I just need to make sure the phone still has audio from the conf. |
| 19:48.52 | r0m|u | looking at my side everything seems ok |
| 19:49.06 | r0m|u | on warning of drop frames or anything |
| 19:49.13 | r0m|u | s/on/no/ |
| 19:49.27 | p3nguin | I'm going to figure it to be either my router or the modem, with a bias against the modem. |
| 19:49.48 | p3nguin | I'll run it like this for a few days, then turn the shaping back on. |
| 19:50.38 | p3nguin | If it doesn't keep dropping calls during the next few days, that is. |
| 19:50.53 | r0m|u | Seems like a plan. I am have a 50/50.... I dont know vyatta so I wont blame it yet.... but I wouldnt doubt is killing yout sip sessions :/ is your isp know for been reliable unlike comcastic? |
| 19:51.09 | p3nguin | If it still drops calls without the shaper and with the changes I've made, and even after I've done this test, I'll be at a loss. |
| 19:52.16 | p3nguin | I don't really know if they are known for being reliable or not, but until I changed asterisk to vyatta and a different modem, I didn't ever have this problem. |
| 19:52.58 | p3nguin | And I didn't notice it until I started doing shaping, so that was my first idea of the cause, which seems to be incorrect. |
| 19:54.05 | r0m|u | mhhhhh mhhhhh this sounds like a vyatta issue. The only problem a modem could be causing is just loosing sync and or random reboots caused by the lost of sync.... your numbers look ok so I wont blame packet corruption.... at this point I am at 1 80/20 that vyatta is the issue |
| 19:54.18 | p3nguin | IAX2 Mini packet, Raw mu-law data (G.711) |
| 19:54.27 | p3nguin | Why is it a mini packet instead of a packet? |
| 19:55.14 | r0m|u | Thats strange. I never sharked on IAX only sip and even than they where packets |
| 19:55.26 | Qwell | p3nguin: full frames have more stuff in them, they aren't needed all the time |
| 19:55.36 | p3nguin | Is that part of trunking? |
| 19:55.52 | Qwell | not sure |
| 19:56.26 | r0m|u | That makes sense |
| 19:59.00 | Cesar_B | i m looking at the configs in the 1.8 branch, looking for the setting to have microseconds in the billsec cdr field, they have a special setting to activate it, or its a default value now have the microseconds in the billsec field |
| 19:59.03 | Cesar_B | ? |
| 20:09.25 | r0m|u | how can I see who is ether dialing in and or out? |
| 20:09.34 | r0m|u | or active calls |
| 20:09.36 | p3nguin | core show channels |
| 20:11.03 | r0m|u | p3nguin: you been the only one in the conf is it normal I see 4 active channels? Wouldnt be 2? |
| 20:11.19 | p3nguin | four channels, two calls |
| 20:11.30 | p3nguin | Check again. |
| 20:11.46 | r0m|u | nobody is in |
| 20:11.51 | p3nguin | Test complete! |
| 20:11.52 | r0m|u | did you hung up? |
| 20:11.59 | r0m|u | o! |
| 20:12.07 | p3nguin | Duration 92.616667m |
| 20:12.07 | r0m|u | you got disconnected? |
| 20:12.09 | p3nguin | no drop |
| 20:12.14 | r0m|u | o wow |
| 20:12.29 | r0m|u | good news .... I guess? |
| 20:14.15 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 20:14.34 | p3nguin | I think it's good. |
| 20:14.47 | p3nguin | Calls would drop between 1 and 40 minutes. |
| 20:15.08 | p3nguin | over 92 = good news |
| 20:15.30 | p3nguin | So I'm going to run it like this for a few days before making any other changes. |
| 20:15.40 | r0m|u | but thats weird..... did you change anything before the call? |
| 20:15.48 | p3nguin | If calls are still dropping, I'll take additional steps. |
| 20:15.53 | p3nguin | Yes, I made changes this morning. |
| 20:16.00 | r0m|u | o ok. |
| 20:16.07 | p3nguin | That's why I wanted to do a new test. |
| 20:16.16 | r0m|u | I see. what did you do? |
| 20:16.40 | p3nguin | So if I don't get any more dropped calls over a few days, I'll start shaping again. |
| 20:17.34 | p3nguin | Made some adjustments on the modem. |
| 20:20.14 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 20:20.20 | timeshell | Hi |
| 20:20.33 | timeshell | Has anyone else found 1.8.x to be rather unstable? |
| 20:20.36 | p3nguin | I hate waiting days for things like this. I'd like to start shaping right now. |
| 20:20.53 | p3nguin | I run 1.8.7.1, and it seems pretty stable. |
| 20:21.04 | timeshell | We have really weird issues using it |
| 20:21.05 | anonymouz666 | timeshell: it seems is for some people. |
| 20:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-emxnntoxtcvrctaf) |
| 20:21.23 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 20:21.34 | r0m|u | cbwest: do you talk? |
| 20:21.40 | timeshell | Like phones continue ringing after you answer it. Half attended transfers failing over to voicemail. |
| 20:21.50 | leifmadsen | timeshell: nope |
| 20:21.58 | r0m|u | timeshell: all ok here |
| 20:22.09 | timeshell | We have had really weird issues ever since moving from 1.6 to 1.8 |
| 20:22.20 | leifmadsen | define: "really weird issues" |
| 20:22.27 | timeshell | See above |
| 20:22.49 | timeshell | In fact, it was most stable when were still on 1.6.0.18 |
| 20:23.06 | leifmadsen | see "what" above? |
| 20:23.12 | leifmadsen | you joined the room, then said you have weird issues |
| 20:23.15 | leifmadsen | that is all I've seen |
| 20:23.26 | leifmadsen | oh there it is |
| 20:23.58 | timeshell | I have some cases where the phones continue to ring on one side but the person on the other hears the person talking |
| 20:23.58 | leifmadsen | never run into that issue |
| 20:24.15 | leifmadsen | never experienced that |
| 20:24.19 | leifmadsen | have several deployments |
| 20:24.20 | timeshell | I have had issues where on my side I'd answer and hear the remote party talking but MY phone still keeps ringing while I'm talking to him |
| 20:24.33 | leifmadsen | what phone? |
| 20:24.47 | timeshell | Polycom IP 501's and Bria for iPhone. |
| 20:24.52 | leifmadsen | weird |
| 20:25.02 | timeshell | Very weird. |
| 20:25.04 | leifmadsen | have many Polycom IP335's deployed and have never experienced that |
| 20:25.11 | leifmadsen | everything just seems to work |
| 20:25.23 | leifmadsen | would have to see SIP traces and configurations to speak any more abu tit |
| 20:25.26 | timeshell | I have IP500's at home and experiencing the half attended transfer problems there too. |
| 20:25.34 | timeshell | So, two installations with the same problems. |
| 20:25.55 | timeshell | Both using 1.8.x |
| 20:26.01 | timeshell | Never seen these with 1.6 |
| 20:26.28 | timeshell | Suppose that the config written for 1.6 is incompatible with 1.8? |
| 20:26.38 | leifmadsen | impossible to say |
| 20:26.40 | timeshell | Causing weirdness? |
| 20:26.43 | leifmadsen | I'm still waiting for more information |
| 20:27.30 | timeshell | I'm going to try rewriting a clean config for one of them (eventually) and see if that clears it up. |
| 20:27.56 | timeshell | Either that or I have to revert back to one of the 1.6 versions. |
| 20:28.16 | timeshell | I have too many people complaining. |
| 20:28.50 | leifmadsen | you need to do more debugging |
| 20:28.58 | leifmadsen | actually look at the traces and determine what is happening |
| 20:29.08 | leifmadsen | rewriting the entire configuration without debugging just seems like a lot of effort |
| 20:29.17 | leifmadsen | without any guarantee that it changes anything |
| 20:29.38 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 20:29.40 | timeshell | I put the sip debug info on that one bug and was only told it didn't help.... |
| 20:29.48 | IsUp | hello |
| 20:30.10 | gordonjcp | anyone here familiar with Cisco 7910G+SW phones? |
| 20:30.16 | gordonjcp | or any similar Cisco phone? |
| 20:30.17 | leifmadsen | points at cbwest |
| 20:30.27 | gordonjcp | in particular, is there anything you need to do to make them work with PoE? |
| 20:31.04 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 20:31.57 | Qwell | leifmadsen: heh, I see what you did there. |
| 20:32.11 | leifmadsen | Qwell: :) |
| 20:32.25 | Qwell | gordonjcp: iirc, those phones only support the bastardized Cisco version of PoE. |
| 20:32.32 | Qwell | You need to make a special cable. |
| 20:32.32 | leifmadsen | ya that's what I remember as well |
| 20:33.21 | gordonjcp | Qwell: oh, no probs, what does that involve? |
| 20:33.30 | Qwell | got me |
| 20:33.35 | Qwell | swapping some pairs |
| 20:33.50 | timeshell | At any rate, I have to try to get that trace too when I get some time for it. |
| 20:44.27 | gordonjcp | Qwell: thanks, I've found an article that describes exactly the situation I'm in with a Cisco phone and 3Com PoE taps |
| 20:57.06 | gordonjcp | Qwell: awesome, it's basically the opposite of a crossover cable; you swap blue and brown instead of orange and green |
| 21:02.46 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 21:05.53 | akrohn | I have a silly question for you. I have an Asterisk 1.6.0.6 server that hosts around 20 or 30 customer businesses. usually around 25+ active calls. We had to create a cron script that runs every minute to see if asterisk has crashed and to restart it |
| 21:06.43 | akrohn | one of the last things in the logs is always "app_voicemail.c: Unable to read password" |
| 21:07.28 | akrohn | other than that, the logs give really no indication of why it's crashing. Are there (or can you tell me where to find) known bugs for voicemail that crashes asterisk in this version? |
| 21:07.37 | Qwell | upgrade |
| 21:07.59 | akrohn | heard that. that's my plan. but boss is insistent i resolve it |
| 21:08.18 | Qwell | Sometimes bosses are stupid. |
| 21:08.28 | akrohn | because we have to deal with the current system for a little while until the new one gets built |
| 21:08.34 | akrohn | haha right? |
| 21:08.38 | Qwell | and that somehow prevents you from upgrading? |
| 21:08.57 | akrohn | if i jump from 1.6 to 1.8, won't my dialplans get screwed up? |
| 21:09.14 | Qwell | I didn't say 1.8 |
| 21:09.28 | *** join/#asterisk KevinLynn (~klynn@161.253.143.80) |
| 21:09.29 | akrohn | the latest 1.6 then |
| 21:10.07 | KevinLynn | can someone here tell me a channel for use by the AMI that is always answered? (used to know this but my info is at home) |
| 21:10.10 | Qwell | You're over a year behind the latest version in the 1.6.0 series. |
| 21:11.02 | akrohn | thanks Qwell, I'll check that out |
| 21:13.03 | akrohn | is there an upgrade path I should know about? or can i go from 1.6.0.6 to 1.6.2.20 ? |
| 21:13.19 | Qwell | read the UPGRADE.txt included with the source? |
| 21:13.34 | akrohn | word |
| 21:15.33 | klynn | as per my question.. Local/s? |
| 21:16.47 | _Corey_ | klynn: You want to make an extension that immediately answers? |
| 21:17.22 | klynn | Corey: I'm hoping one exists that always answers |
| 21:17.28 | klynn | can be an external sip address too |
| 21:17.31 | klynn | it's for testing |
| 21:17.49 | _Corey_ | klynn: Well, no it depends on your dialplan... |
| 21:18.06 | _Corey_ | [always-answer] can have an "s,1,Answer" |
| 21:18.20 | _Corey_ | and you can dial Local/s@always-answer, etc. |
| 21:18.28 | klynn | nice let me try that |
| 21:20.47 | klynn | well.. that would probably work but I'm trying to work strictly within default contexts |
| 21:20.47 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
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| 21:31.33 | [TK]D-Fender | checking out for now, BBIAB |
| 21:31.38 | *** join/#asterisk timahvo1 (~rogue@197.179.49.72) |
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| 21:34.41 | rotten777 | does anyone here have some sample polycom configs I can dig around in? I'm trying to pick up the 330+ pages in the admin guide but I learn much better by example... |
| 21:34.51 | klynn | hmm.. maybe if I create a conference I can just drop this to that conference.. |
| 21:39.33 | timeshell | leifmadsen How do I do a backtrace in 1.8? |
| 21:39.47 | leifmadsen | ~asterisk-debugging |
| 21:40.02 | WIMPy | ~collectdebug |
| 21:40.03 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:40.08 | leifmadsen | there are pages on the wiki that explain -- same process as all versions of asterisk |
| 21:40.13 | leifmadsen | WIMPy: thanks |
| 21:41.00 | timeshell | That's it? |
| 21:41.18 | timeshell | Isn't that what I did in ASTERISK-18685 |
| 21:41.28 | timeshell | Why are they asking for another trace then? |
| 21:42.35 | Qwell | because that isn't a sip debug |
| 21:42.58 | timeshell | Really... |
| 21:43.00 | timeshell | What is it then? |
| 21:43.08 | Qwell | an asterisk debug log |
| 21:43.12 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 21:43.57 | timeshell | https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 21:44.02 | sawgood | Hi: any channel op available? |
| 21:44.04 | timeshell | ^^^^ that is more or less how I got it. |
| 21:44.07 | Qwell | sawgood: ? |
| 21:44.24 | sawgood | Hi Qwell ... I wanted to ask you if this would help the #asterisk channel ... |
| 21:44.33 | sawgood | can I send you a private msg? |
| 21:44.56 | Qwell | sure |
| 21:45.16 | timeshell | ~collecttrace |
| 21:45.18 | tuxxie | is thier software that like trixbox or fonality's hud client that will dlsplay users calls status's |
| 21:45.25 | timeshell | ~collectbacktrace |
| 21:45.33 | Qwell | tuxxie: core show channels ? |
| 21:45.33 | timeshell | meh |
| 21:45.53 | Qwell | sawgood: no. |
| 21:46.03 | tuxxie | Qwell: for my cleints |
| 21:46.06 | tuxxie | not my self |
| 21:46.10 | Qwell | sawgood: We don't do advertising here. |
| 21:46.22 | sawgood | no advertising ... a simple GIFT to the channel |
| 21:46.28 | Qwell | It's advertising. |
| 21:46.43 | sawgood | oh ok ... I am a technican, so I just wanted to give back something |
| 21:46.57 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 21:47.08 | sawgood | its cool ... I thought I would offer it ... no worries ... |
| 21:47.42 | r0m|u | learns from Qwell's kung fu techniques |
| 21:48.20 | _Corey_ | sawgood: Sponsor a cocktail hour at next year's Astricon... that would be nie |
| 21:48.24 | _Corey_ | nice even :) |
| 21:48.43 | sawgood | right on .. a side party ... |
| 21:48.43 | r0m|u | ill make sure to be there :P |
| 21:49.02 | sawgood | how was Astricon in Denver last month? |
| 21:49.41 | leifmadsen | pretty kick ass |
| 21:49.46 | leifmadsen | Qwell was really excited |
| 21:49.51 | Qwell | SO EXCITED |
| 21:49.58 | sawgood | I was so close to coming, but my skill set is not par with you guys ... |
| 21:50.08 | _Corey_ | a good time was had by all ;) |
| 21:50.09 | sawgood | I wanted more time in 'grade' before I met your team |
| 21:50.09 | Qwell | that's...kinda the reason for going. |
| 21:50.31 | sawgood | I will be there next year for sure (100%) as long as it is hosted in the US |
| 21:50.32 | Qwell | _Corey_: not I. I was stuck in my room during the party, writing my presentation. :p |
| 21:50.46 | Qwell | one of like 14 that I had to give. |
| 21:50.55 | WIMPy | leifmadsen: About the Park(): You're right. The number is actually just cut off. If you listen really carefully, you can hear a tiny fraction of the end of the number. |
| 21:51.12 | WIMPy | But is that the way it should be? |
| 21:51.13 | _Corey_ | Qwell: lol, they had me doing three in one day with a hangover... i don't want to hear about it ;) |
| 21:51.19 | Qwell | 3? wow. |
| 21:51.42 | sawgood | Is there a location set for 2012 yet? |
| 21:51.46 | leifmadsen | WIMPy: that happens all the time -- try not using answer or anything like that with a prompt into an auto-attendant |
| 21:51.55 | leifmadsen | sawgood: no, but likely Denver again |
| 21:52.13 | leifmadsen | WIMPy: that's why I always put Playback(silence/1) before any prompts |
| 21:52.22 | sawgood | perfect because I am Broncos fan! |
| 21:52.32 | leifmadsen | same |
| 21:52.37 | sawgood | leifmadsen: I really enjoyed your BOOKS |
| 21:52.43 | leifmadsen | does the Tebow pose |
| 21:52.43 | sawgood | I am on the 2nd pass of one of them |
| 21:52.51 | timeshell | snaps a shot |
| 21:52.54 | leifmadsen | sawgood: glad you're enjoying it :) |
| 21:52.57 | sawgood | I've been a Broncos fan since 1976 |
| 21:53.02 | leifmadsen | I was born in 1981 |
| 21:53.03 | WIMPy | leifmadsen: I have to admit that I usually use Answer(300), but in that case Answer() is enough. |
| 21:53.46 | WIMPy | I don't think I missed a whole file so far. |
| 21:53.50 | sawgood | 1981 Denver went 11-5 and was 'cheated' out of the playoffs |
| 21:54.14 | _Corey_ | I think my only regret from this year's event was sleeping through the dCAP breakfast... |
| 21:54.30 | _Corey_ | was it well attended? |
| 21:54.53 | Qwell | _Corey_: I was almost up until dCAP breakfast one of the nights... |
| 21:54.55 | sawgood | I remember that year specifically because in week 16 Denver had to hope for one of two games to go their way, and both teams lost (and they missed the playoffs by 1/2 a game) |
| 21:55.18 | leifmadsen | sawgood: sounds like the LEafs |
| 21:55.34 | leifmadsen | last time the Leafs were in the playoffs, I had just started college |
| 21:55.56 | sawgood | two tabby cats are fighting next to me |
| 21:56.03 | sawgood | cat fight .. |
| 21:56.49 | sawgood | the smaller kitten beat out the full grown tabby in this round |
| 21:58.23 | *** join/#asterisk Eitan (~Eitan@12.192.84.98) |
| 21:58.36 | _Corey_ | Qwell: They need to move it from the morning after the party... I've only managed to attend once in the 4 years I've been going to Astricon ;) |
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| 22:04.50 | vader-- | I was wonder if you guys could provide some suggestions/examples of how small store's pbx system are setup. I am putting together one for a friend running asterisk. He will have 4 Polycom 335 IP Phones. 2 of them will be at his front of store counters, 2 for his offices. Each phone has two lines. I think he will eventually want an IVR, but to begin with he will want it to ring all phones. I know he will want to be able to put calls on ho |
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| 22:09.11 | s[X] | hey p3nguin |
| 22:09.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:10.08 | s[X] | hey [TK]D-Fender |
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| 22:30.48 | paulc | vader--: You get any replies to your question? |
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| 22:33.06 | billy_ran_away | Anyone know what Ubuntu repo has Asterisk 10? |
| 22:33.43 | Qwell | none |
| 22:33.44 | [TK]D-Fender | I'd be betting on "none" |
| 22:33.50 | [TK]D-Fender | * is not released yet |
| 22:33.53 | Qwell | [TK]D-Fender: All bets are closed. |
| 22:33.55 | [TK]D-Fender | 10 |
| 22:34.12 | Qwell | Money has been forfeit. Please play again later. |
| 22:34.20 | Qwell | runs off to buy some stuff |
| 22:35.18 | [TK]D-Fender | That was "I would", not "I did" :) |
| 22:35.33 | Qwell | ambiguities go to Qwell. |
| 22:35.39 | [TK]D-Fender | clubs Qwell with a Big Book Of Contractions and takes his money back |
| 22:36.30 | billy_ran_away | anyone have a handy list of dependencies to compile asterisk 10 from source on ubuntu 11.10? |
| 22:36.31 | [TK]D-Fender | Chapter 8: Octo-mom (Or nursing post-doggy style) |
| 22:36.52 | [TK]D-Fender | billy_ran_away: In the tarball |
| 22:37.03 | billy_ran_away | [TK]D-Fender: Awesome thanks |
| 22:41.29 | vader-- | paulcpaulc not really |
| 22:42.35 | *** join/#asterisk fiesch (d95c43dc@gateway/web/freenode/ip.217.92.67.220) |
| 22:46.11 | fiesch | hi.. can sb give me a hand on pri? I have a OpenVox DE210E here on asterisk 1.8.7.1, dahdi 2.5.0.2, libpri 1.4.11.5 running on CentOS 5.6 which is driving me nuts.. 85 % of the time this card works beautifully, but when the remote line is "shaky" (distortion on the signal), the card does not get the full number - for instance when the remote party dials XXX-XXXYYYZZZ the card would only pick up XXX-XXXYYYZ or XXX-XXXYYYZZ |
| 22:46.29 | fiesch | has anyone in here seen behaviour like this? OpenVox themselves seem stumped |
| 22:47.09 | Qwell | fiesch: Welcome to the world of terrible clone cards. |
| 22:47.17 | WIMPy | fiesch: First of all I'd recommend more recent versions. |
| 22:47.37 | WIMPy | But if your line is borked, there isn;t much you can do about that. |
| 22:47.53 | fiesch | Qwell: thanks ;) |
| 22:48.09 | fiesch | WIMPy: Well now here's the tricky part |
| 22:48.29 | fiesch | a) OpenVox said to downgrade versions because the new ones haven't been tested by them |
| 22:48.36 | fiesch | so i did - without result |
| 22:48.37 | fiesch | and b) |
| 22:49.04 | fiesch | i have a legacy pbx working that same line normally with a cologne chip PRI card (SWYX) which handles the line just fine |
| 22:49.48 | fiesch | weird thing is that this appears to be at libpri level as i can see the number beeing clipped in intense span debug |
| 22:50.17 | fiesch | so i thought i might play around with line gain, to no avail |
| 22:50.34 | WIMPy | Well, in that case I'd go and bug OpenVox. |
| 22:51.13 | fiesch | i did, it actually took me a whole week to get the card running altogether with the CEO of Openvox ending up on my server for 2 hours trying to fix things |
| 22:51.31 | Qwell | refers to his previous comment. |
| 22:51.49 | WIMPy | For small numbers of "fix". |
| 22:51.58 | fiesch | now i have a forum entry on their site which hasn't seen a entry in a month or so wince recommending to downgrade |
| 22:52.14 | fiesch | *since |
| 22:52.23 | WIMPy | Well, I'd personally seer clear of hardware that isn't supported by Linux natively, But that's not alwyas possible. |
| 22:52.43 | fiesch | Qwell: I actually orderer a Digium dual San PRI, but I couldn't get the card shipped on time in germany |
| 22:52.53 | fiesch | *Span |
| 22:53.25 | fiesch | so the distributor offered OpenVox saying he had had tons of satisfied customers with these |
| 22:53.34 | WIMPy | Why didn;t you use th swyx card? |
| 22:53.36 | fiesch | which could be shipped on time |
| 22:54.09 | fiesch | The card needs to stay in the legacy PBX to have a hot fallback in case the new thing breaks |
| 22:54.47 | fiesch | it's a weird conglomerate of asterisk, lync and a third party fax server along with A/D Gateways and the like, a lot of ends that can break |
| 22:55.08 | fiesch | and 10 minutes of downtime is a real problem with the customer |
| 22:56.15 | fiesch | the whole thing is actually running smoothely if it weren't for the dropped end digits which, of course, totally screws up my call routing |
| 22:56.51 | WIMPy | Seems unlikely that's the only issue. |
| 22:57.57 | fiesch | i closely monitored the sys and haven't seen anything else in the logs.. I originally had the timing signal in mind as the bad guy but i guess calls would be all over the place if the base tick was screwed up |
| 22:58.39 | WIMPy | To me it sounds like a timing or IRQ issue. |
| 22:59.29 | fiesch | I have been considering scrapping the card, putting the paid cash on the "you'll learn" account and getting a digium card, but the problem is so hard for me to narrow down that I'm not a 100% confident that the problems would go away |
| 23:00.07 | WIMPy | Have you tried it in another PC? |
| 23:00.42 | WIMPy | And if you already know that a HFC-E1 works, you can always get one of those. |
| 23:00.45 | fiesch | not in this configuration, i had it in 5 Servers trying to get it to run altogether as requested by OpenVox.. |
| 23:01.12 | fiesch | They, too, first suspected an IRQ issue which mad them request the platform change |
| 23:01.37 | fiesch | though it turned out to be a simple misconfiguration in dahdi system.conf |
| 23:01.39 | WIMPy | I assume you did go through things like IRQ sharing? |
| 23:02.05 | fiesch | I did and disabled all onboard devices which were assigned the same irq |
| 23:02.25 | WIMPy | Hmmm. |
| 23:02.48 | fiesch | though i am not a 100% sure how the state of it is right now. Is there a easy way to investigate the current irq assignment via bash? |
| 23:02.51 | WIMPy | Even disabled devices can still be an issue. |
| 23:03.08 | WIMPy | cat /proc/interrupts |
| 23:03.42 | fiesch | looks good |
| 23:04.22 | fiesch | 169: 237208 0 13736139 509362 IO-APIC-level wct2xxp |
| 23:05.15 | fiesch | my problem with timing and irq as a source is that i can always reproducethe error from given line while being unable to generate the error from specific others |
| 23:05.18 | WIMPy | You could try to assign only that IRQ to one core of its own. |
| 23:05.37 | WIMPy | But I'm not sure that would help. |
| 23:05.51 | WIMPy | Please elaborate. |
| 23:06.07 | fiesch | thanks for your input, i apreciate it |
| 23:06.15 | fiesch | well.. |
| 23:06.43 | fiesch | if irq and / or timing issues were the reason for occasional problems in call reception |
| 23:06.57 | fiesch | those would randomly apply to all calls passing through the PBX |
| 23:07.11 | WIMPy | Sure. I'd like to know what kind of pattern you're finding. |
| 23:08.01 | fiesch | there are specific lines, like the one i am at right now, which will always produce the error, i have only one had a call passing thorugh fine out of 200 |
| 23:08.04 | WIMPy | And have you compared that pattern with the other box with the HFC card? |
| 23:08.13 | WIMPy | Is that box also running Asterisk? |
| 23:08.40 | fiesch | while on most other lines (like my cell for instance) i haven't been able to reproduce the error even once |
| 23:09.02 | gordonjcp | woo and indeed yay |
| 23:09.04 | fiesch | the other box is running swyx on windows, the error is not reproducable on that box |
| 23:09.20 | gordonjcp | incoming and outgoing calls, to and from my SIP phones in here and to a real landline |
| 23:09.37 | fiesch | gordonjcp: yay ;) |
| 23:09.57 | gordonjcp | oh, okay, spoke too soon ;-) |
| 23:10.27 | WIMPy | Ok, have you tried calling from other MSNs of that line and/or to other DDIs? |
| 23:10.49 | fiesch | WIMPy: yes, same pattern each time |
| 23:11.02 | *** join/#asterisk troyt (~troyt@PariahZero.broker.freenet6.net) |
| 23:11.06 | WIMPy | fishy |
| 23:11.21 | fiesch | very ^^ |
| 23:11.27 | WIMPy | Anything special about that line? |
| 23:12.23 | WIMPy | The part of another box not having the issue still smells like a very low level issue to me. |
| 23:12.29 | fiesch | when i use a analog fax machine on that line i can hear cracking noises on the line, i assume that there is a slightly loose wire in the net connection somewhere |
| 23:12.34 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
| 23:12.44 | WIMPy | But depending on caller is quite interesting. |
| 23:12.51 | fiesch | but this isn't the only line with that issue |
| 23:12.54 | WIMPy | But I came across such things earlier. |
| 23:13.26 | fiesch | i had to pull the system from production use because customers were complaining after falling through to my "catchall" extension |
| 23:14.06 | WIMPy | And a garbeled called party number is the only issue you see? |
| 23:14.27 | fiesch | yes, the only oddity |
| 23:14.44 | WIMPy | Extremely strange. |
| 23:14.53 | fiesch | and always missig some of the trailing digits |
| 23:15.13 | fiesch | like it would consider the transmission of the called party number over too soon |
| 23:15.33 | WIMPy | Ah. ok. |
| 23:15.56 | WIMPy | Does your overlap receive work? |
| 23:16.15 | fiesch | you caught me in a "hm?" moment |
| 23:17.20 | fiesch | i was - up until now - in the very beneficial position not having to get into the guts uf things with asterisk so that does not ring a bell with me |
| 23:17.23 | fiesch | *of |
| 23:18.15 | WIMPy | Does it fail for all calls from standard phones? |
| 23:18.41 | fiesch | not all, i was able to get one through from this line today |
| 23:18.42 | WIMPy | POTS/ISDN on a regular phone line without VOIP or whatnot. |
| 23:19.25 | WIMPy | Do you have immediate enabled in chan_dahdi? |
| 23:19.27 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 23:19.30 | fiesch | the testline is a analog phone over a ISDN line (with a simple pbx) |
| 23:20.21 | *** join/#asterisk Russ (~russ@206.29.182.152) |
| 23:20.27 | WIMPy | Looks like it comes down to a simple configuration issue. |
| 23:20.35 | fiesch | lemme look up on this, i use freepbx on this box and i never find my configs on bash when i need to |
| 23:21.06 | WIMPy | Sounds interesting. |
| 23:21.18 | WIMPy | Maybe it will break it again if you find out how to fix it. |
| 23:21.49 | fiesch | nah, when you put your additions in the right places this is a suprisingly sound system for a web interface |
| 23:22.08 | WIMPy | Yes, but that's not an addition. |
| 23:22.22 | fiesch | http://pastebin.com/YLTDVS0N |
| 23:22.35 | WIMPy | This is something it should have set up correctly. |
| 23:23.18 | WIMPy | Try to add "immediate=no". |
| 23:23.34 | fiesch | for completeness' sake the system.conf http://pastebin.com/0sD2jk18 |
| 23:23.42 | WIMPy | And you probably don;t want to set the "dialplan"s to national. |
| 23:23.53 | *** join/#asterisk mindCrime (~chatzilla@24.106.207.82) |
| 23:24.23 | fiesch | the dialplans are a thing for themselves as i had a hard time getting lync and asterisk getting to likeeach other with prefixes |
| 23:24.30 | WIMPy | Oh, it's only the 1st span. |
| 23:24.43 | fiesch | yes, the second one is the active one |
| 23:24.57 | WIMPy | Unless you know what you're doing, stick to "unknown". |
| 23:25.16 | fiesch | but i basically really do horrible thing to the numbering schemes in the course of callrouting between systems |
| 23:25.23 | WIMPy | national is a rather nasty default. |
| 23:26.08 | WIMPy | Yes, set everything to 'unknown' and try again. |
| 23:27.52 | fiesch | hm added the immediate line, had a good old fashioned asterisk restart, same behaviour |
| 23:28.26 | fiesch | Tried 2 times, the first time it clipped one digit, the second one 2 digits |
| 23:29.54 | WIMPy | Ok, it must be the dialplan then. |
| 23:30.13 | WIMPy | But that's definitely something you need to got to #freepbx for. |
| 23:30.40 | fiesch | hm sure? I mean the intense span debug shows the number coming in clipped way before the dialplan is touched |
| 23:31.08 | WIMPy | That's perfectely normal. |
| 23:31.28 | WIMPy | Additional digits will be sent wen the caller types them into their phone. |
| 23:32.16 | WIMPy | You will see the absence of "sending complete" in the setup message. |
| 23:33.37 | fiesch | hm really weird |
| 23:33.49 | gordonjcp | this is crazy, it's cheaper to call a mobile in Canada with sipgate than it is to call one in the UK, *from* the UK |
| 23:33.52 | WIMPy | No |
| 23:33.57 | fiesch | i assumed they would be sent as a block from the analog phone through the small isdn pbx |
| 23:34.11 | WIMPy | No. Why should they? |
| 23:34.46 | fiesch | but ok that gives me something to read up on ;) thank you very much for pulling me out of my dead end |
| 23:34.49 | WIMPy | You'd need a timeout for that. Like most SIP ATAs do. |
| 23:35.13 | WIMPy | ... and which is just plain horrible. |
| 23:35.32 | gordonjcp | SIP dialplans |
| 23:35.35 | gordonjcp | *fun* |
| 23:36.08 | WIMPy | gordonjcp: That only works if you only want to be able to call a pre-known set of numbers. |
| 23:36.11 | gordonjcp | I have to say, I haven't a bloody clue how chan_sccp is supposed to work, but it works extremely well ;-) |
| 23:36.14 | fiesch | all i know is that i will _very_ much enjoy that beer when this card is finally doing what it's supposed to do one fine day |
| 23:36.52 | *** join/#asterisk filo1234 (~filo2@unaffiliated/filo1234) |
| 23:37.18 | WIMPy | I have seen an installation with a PRI and FreePBX, but TBH i have no idea if it worked correctly. |
| 23:37.27 | *** join/#asterisk NDT (nunya@cpe-72-226-104-247.nycap.res.rr.com) |
| 23:37.47 | WIMPy | But without, it shouldn't be hard to do. |
| 23:37.49 | gordonjcp | the way you define lines for sccp seems to fly in the face of decoupling phone IDs and extension IDs |
| 23:37.50 | fiesch | I have done several BRI ones with freepby, but this is the first one on PRIO with the dahdi addin for freepbx |
| 23:37.51 | filo1234 | hi all, sorry there is an italian asterisk channel? |
| 23:38.40 | WIMPy | You could have the same issue on BRIs if they are with DDI. |
| 23:39.26 | fiesch | they actually are but i didn't hav an issue there, still running smoothly |
| 23:39.37 | WIMPy | And you will definitely have an issue if you connect phones. |
| 23:39.59 | WIMPy | Shorter? |
| 23:40.11 | WIMPy | Maybe it's just a timeout thing. |
| 23:40.13 | [TK]D-Fender | [18:33]gordonjcpthis is crazy, it's cheaper to call a mobile in Canada with sipgate than it is to call one in the UK, *from* the UK <-- North America doesn't have a rate to call "mobiles" like so much of Europe does at all. |
| 23:40.20 | fiesch | hm.. lemme check up on that |
| 23:41.15 | fiesch | yep, one digit shorter |
| 23:41.25 | filo1234 | well..I have installed asterisk on an Ubuntu 10.04 server, I have downloaded italian sounds but how can I set asterisk to use italian sounds, like Playback(hello-world) ? |
| 23:42.03 | WIMPy | filo1234: Set CHANNEL(language) |
| 23:42.19 | WIMPy | On some channeltypes you can set it per peer. |
| 23:42.20 | *** join/#asterisk billy_ran_away (~billy_ran@173-167-194-14-ip-static.hfc.comcastbusiness.net) |
| 23:42.44 | filo1234 | WIMPy: in wich file? or you mean from cli? |
| 23:43.06 | fiesch | filo1234: this is for inside a dialplan |
| 23:43.11 | WIMPy | In your dialplan. |
| 23:43.55 | fiesch | WIMPy: |
| 23:44.04 | billy_ran_away | Hi, I'm trying to use a Asterisk 10 almost as a Jabber server so I can support multi video conferencing… I've installed and configured Asterisk 1.8 for home user, so just one user, one dial plan to (Google Talk), so I'm familiar with Asterisk, but does anyone have any tips for my goal? |
| 23:44.05 | fiesch | WIMPy: you actually saved me here |
| 23:44.08 | filo1234 | WIMPy: sorry can I have an example? |
| 23:44.10 | fiesch | it's overlapdial=yes |
| 23:44.42 | WIMPy | Oh, I thought that was only used in NT mode. |
| 23:44.57 | WIMPy | Well, I guess I'm not that much in to dahdi. |
| 23:44.59 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-mgbyvmaezcntsztu) |
| 23:44.59 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
| 23:45.37 | fiesch | now i see the behaviour you mentioned, the call passes in clipped but gets passed to the dialplan as the full number (after a short wait period) |
| 23:46.11 | WIMPy | Ok, beer goes to Flensburg ;-) |
| 23:46.24 | fiesch | I'll send it up from Munich |
| 23:46.27 | [TK]D-Fender | filo1234: "core show function CHANNEL" |
| 23:47.33 | WIMPy | Sorry to Digium :-) |
| 23:47.45 | pdtpatrick1 | Question .. how's video on asterisk? i know there's configurations in sip.conf that mention video support. Has anyone experience in this area .. what did you use to make it work? if there's a wiki or guide somewhere, please link . Thanks |
| 23:47.52 | fiesch | no i'll actually open one up right now.. I was pulling my hair for nights on end on this |
| 23:48.20 | fiesch | WIMPy: thank you very much! |
| 23:48.21 | [TK]D-Fender | pdtpatrick1: Enable the codecs for your peers. Set vidiosupport=yes in [general]. The end. |
| 23:48.40 | WIMPy | Doesn't look too well for OpenVoxes support, I guess. |
| 23:48.57 | [TK]D-Fender | pdtpatrick1: It'll negotiate with everything else. No transcoding, no "mixing" for conferencing. app_conference supports "follow the speaker" for video, but not "mixing" |
| 23:48.58 | billy_ran_away | Did anyone see my message? |
| 23:49.08 | billy_ran_away | Not sure I had identified yet... |
| 23:49.34 | WIMPy | billy_ran_away: yes |
| 23:49.47 | [TK]D-Fender | billy_ran_away: My comment to pdtpatrick1 largely applies to you as well |
| 23:49.49 | billy_ran_away | WIMPy: I guess I meant the big long one... |
| 23:49.52 | pdtpatrick1 | [TK]D-Fender, interesting. Do u know of any devices it prefers or as long as the device has video (example: softphone on laptop) it should work ? |
| 23:50.03 | fiesch | WIMPy: honestly - you are not in a good position if you need support from openvox directly - especially not if you're in Europe as they have normal "CN" working hours |
| 23:50.15 | billy_ran_away | WIMPy: |
| 23:50.16 | [TK]D-Fender | pdtpatrick1: * doesn't care about devices, just protocols. |
| 23:50.18 | fiesch | and for the professional side of things... you're correct |
| 23:50.23 | billy_ran_away | I'm hoping to use Blink as my clients |
| 23:50.26 | [TK]D-Fender | pdtpatrick1: * passes video through, that's all |
| 23:50.31 | billy_ran_away | It's a soft phone client, icanblink.com |
| 23:51.04 | WIMPy | billy_ran_away: A 6 line one. |
| 23:51.41 | WIMPy | fiesch: Don't trust north americans on anything telephony related. Especially not ISDN. |
| 23:51.51 | filo1234 | [TK]D-Fender: Set(CHANNEL(language)=it) is right? |
| 23:52.59 | filo1234 | or without SET? because syntax says CHANNEL(item) |
| 23:53.33 | [TK]D-Fender | filo1234: Good start.... that is how to set it in the dialplan in specific places. You can also set languages on devices in sip.conf, etc via "language=it" for example so all calls from that device use it as the starting language for the channel |
| 23:53.37 | WIMPy | filo1234: Exactely the way you wrote it. |
| 23:53.58 | fiesch | WIMPy: I'll try to keep that in mind.. (I'm on a web client.. i actually don't have a clue what commands this thing supports) |
| 23:54.37 | [TK]D-Fender | filo1234: Use of that function is most beneficial for things like IVR's where you enter a language choice and want it to pull the right recordings without making extra copies and duplicating code, etc |
| 23:56.01 | *** join/#asterisk chigambamukoko (~chatzilla@fl-76-3-18-120.dhcp.embarqhsd.net) |
| 23:56.24 | filo1234 | [TK]D-Fender: ok thnks a lot |
| 23:56.33 | chigambamukoko | I have asterisk and a2billing, everything installed, just little trouble with the call routing, any takers? quick buck anyone? |
| 23:57.20 | WIMPy | chigambamukoko: Now I have to think about Married With Children. |
| 23:58.00 | chigambamukoko | Comone WIMPY don't be a WUSS |
| 23:58.27 | chigambamukoko | that kinda sounded fun |
| 23:58.50 | pdtpatrick1 | Question .. does anyone have a softphone or IM client they'll recommend that can integrate the phone and jabber? im currently running asterisk server and jabber separately (openfire as jabber server). |
| 23:59.35 | chigambamukoko | anyway, WIMPy, I think this is should be a walk in the park for you |
