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| 00:37.03 | killown | what is better TDM400P or AEX410? |
| 00:37.26 | WIMPy | Going digital. |
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| 00:46.44 | killown | ? |
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| 01:27.38 | zemmali-voip | hi guys, please i need help to Installing FreePBX on Ubuntu 12.04 |
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| 01:33.33 | WIMPy | Wrong channel. Try #freepbx |
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| 01:35.44 | zemmali-voip | sorry WIMPy |
| 01:37.26 | WIMPy | NP, just telling you where to get help. |
| 01:37.58 | zemmali-voip | thanks for your help |
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| 02:29.10 | KNERD | How do I diabled the BUILD NATIVE in the MENUSELECT_CFLAGS category? |
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| 02:58.30 | poseidon | I'm going to be using asterisk ami to find out basic information for a call center (ie who is on a call, how long, if they put the person on hold, etc.) |
| 02:59.24 | poseidon | any suggestions for where to start with learning how to get this information. I have found various pages, but I'm still losted in someo f the terminology. ie ParkedCalls, I'm not sure want constitutes a "parked" call. |
| 03:05.41 | KNERD | ~book |
| 03:05.41 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 03:07.56 | KNERD | ~book @poseidon |
| 03:08.05 | KNERD | ~book@poseidon |
| 03:08.17 | poseidon | ~book poseidon |
| 03:08.19 | ChannelZ | A parked call is a call that has been put on hold in an Asterisk parking lot |
| 03:08.55 | ChannelZ | A parking lot being somewhere you transfer a caller, and they get assigned a sequential extension where they can be picked up by someone else. |
| 03:10.19 | *** join/#asterisk SlicerDicer (~SlicerDic@pdpc/supporter/monthlygold/slicerdicer) |
| 03:11.01 | SlicerDicer | if I have _800........ how do I prepend 1 to it? So I can make it be 1800foobard ultimate result |
| 03:11.27 | ChannelZ | like when you dial you want to add the 1? |
| 03:11.47 | SlicerDicer | I want it to add after |
| 03:12.06 | ChannelZ | exten => _800XXXXXXX,1,Dial(SIP/something/1${EXTEN}) |
| 03:12.39 | SlicerDicer | rgr thanks |
| 03:13.24 | ChannelZ | adapt as necessary |
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| 03:38.39 | poseidon | ~ami |
| 03:38.40 | infobot | AMI is the Asterisk Manager Interface, a way to control an Asterisk server (and retrieve information) via a TCP/IP socket. More information is available at http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html and http://voip-info.org/wiki/view/Asterisk+manager+API |
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| 03:44.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:21.45 | andross | _800NXXXXXX is a more precise match fwiw |
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| 06:25.17 | igcewieling | you would be suprized how many people dial 8000 XXX XXXX |
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| 06:32.53 | kaldemar | .n |
| 06:33.32 | carrar | .y |
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| 06:50.02 | drmessano | .x |
| 06:50.15 | drmessano | OH NO, ALL THE CALLS |
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| 07:34.09 | schmidts | good morning |
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| 08:16.03 | seik0 | Hi, guys. I need some advice. We use asterisk very close to databases and use ODBC to connect to many databases (10 at the moment), some of which are a bit remote. If we lost connection to even one of those DB, then asterisk odbc engine hangs until connected or failed, not allowing to do any PBX action |
| 08:18.26 | seik0 | Tried to use CURL to connect to applications connected to DB instead of immediate access to DB, but failed, because CURL is some kind not parallel, serial. Bug is published on bug-tracker |
| 08:21.35 | seik0 | And one more thing: as I remember, ODBC queries are serial too int asterisk, m? |
| 08:22.29 | seik0 | Seems everybody is sleeping now |
| 08:22.40 | seik0 | so, i'll be back =) |
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| 08:35.42 | wdoekes | seik0: queries and curl requests are only serial if called from the same thread |
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| 08:36.41 | wdoekes | however, odbc queries do use mutexes since recently, so they can block each other, unless you use pooled (more than one) connections |
| 08:37.46 | wdoekes | but.. incoming calls are generally handled by a single thread.. so if your query locks there, it would be "serial", as you say |
| 08:38.36 | seik0 | wdoekes, wait a sec, i'll look for an issue |
| 08:40.21 | seik0 | wdoekes, does this make sense: https://issues.asterisk.org/jira/browse/ASTERISK-18708 , or, possibly, i've lost something? |
| 08:40.22 | LieutPants | [ASTERISK-18708] [Status: Open] [Assigned: dcabot] func_curl hangs channel under load - https://issues.asterisk.org/jira/browse/ASTERISK-18708 |
| 08:44.48 | wdoekes | I'm probably wrong at 9:37.. every dialplan should get its own thread |
| 08:45.10 | seik0 | so did i think ) |
| 08:45.42 | wdoekes | in which case you'd be looking at locking issues in func_curl.. that should be easy enough to reproduce |
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| 08:49.55 | wdoekes | seik0: looking at your patch: I wonder why global_curl_info is locked during curl_easy_setopt |
| 08:50.22 | wdoekes | just moving that up should fix your problem |
| 08:52.12 | seik0 | it's not my issue ) |
| 08:52.16 | seik0 | not my patch |
| 08:52.19 | seik0 | but i' |
| 08:52.27 | seik0 | but i've tried it |
| 08:52.36 | wdoekes | does it do what you expect? |
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| 08:55.06 | seik0 | it's not working at all, crashing asterisk, needs deeper insight to make it work |
| 08:56.34 | wdoekes | could you try what happens if you run everything from a local channel? |
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| 08:57.02 | wdoekes | i.e. from the incoming extension, do Dial(Local/s@the_real_context) |
| 08:57.16 | wdoekes | and run the curl stuff from [the_real_context] s,1 |
| 08:57.34 | seik0 | with patched func_curl? |
| 08:57.39 | seik0 | or as is |
| 08:57.40 | seik0 | ? |
| 08:57.42 | wdoekes | as-is |
| 08:57.53 | wdoekes | if it crashes your stuff, then forget that patch |
| 08:59.20 | seik0 | ok, i need some time, need to prepare some long-running func on app-side, others are faster |
| 08:59.49 | seik0 | crashes, you mean? blocking? |
| 09:00.18 | seik0 | or crashing at all, with lighting and fire? |
| 09:00.45 | wdoekes | crashes != blocks |
| 09:00.51 | seik0 | ok =) |
| 09:00.55 | wdoekes | things were already blocking for you, right? |
| 09:01.00 | seik0 | right |
| 09:01.27 | seik0 | but, why should it crash with unpatched func_curl? |
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| 09:05.32 | wdoekes | never mind that. are you setting CURLOPTs somewhere else than in during the normal call run? |
| 09:06.35 | wdoekes | I'm assuming you have a 5 line dialplan right now, trimmed down to show exactly the problem and nothing more? |
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| 09:08.27 | seik0 | i'm useing CURLOPT everytime before I make CURL |
| 09:09.15 | seik0 | this is because withing the same run I need to make CURL-request with different access rights |
| 09:09.40 | seik0 | of course, this may be eliminate due to bug |
| 09:10.27 | seik0 | at least, may be tried to |
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| 09:12.21 | seik0 | now i have to use two different versions of asterisk (1.8.* and 1.4.*) and one of them does not accept http://user:pass@... authentification in curl request |
| 09:13.30 | wdoekes | I have no idea what you're saying |
| 09:13.46 | wdoekes | try what this does: http://pastebin.com/dqB7ezbz |
| 09:20.23 | seik0 | i'll try |
| 09:20.46 | seik0 | thanks |
| 09:21.39 | seik0 | to be clear, where else i cat set CURLOPT except during noraml call run? |
| 09:21.44 | seik0 | *can |
| 09:22.04 | wdoekes | the possibility exists, and that's when globals start getting used, so you don't want that |
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| 09:24.20 | seik0 | i can set CURLOPT(userpwd) in [globals] section? Or what do you meen? |
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| 09:25.39 | seik0 | or... in fact, i'm really using something like Set(CURLOPT(userpwd)=${GLOBVAR1}:${GLOBVAR2}) |
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| 09:43.03 | Curs0r | Hello all. Hopefully simple predicament I find myself in here. I grabbed asterisk-gui from svn, made everything fine, configure checked config, everything looks perfect. Asterisk is responding on localhost:8088 but it isn't serving anything up at /gui/static/config/index.html |
| 09:43.25 | WIMPy | Try #asterisk-gui |
| 09:43.35 | Curs0r | d'oh! right you are |
| 09:43.57 | Curs0r | heh, that looks like a lively channel :) |
| 09:44.49 | WIMPy | I guess #freepbx is the only gui channel with at least a little activity. |
| 09:44.50 | Curs0r | All links in the topic are 404 haha |
| 09:45.42 | Curs0r | I think I'll just abandon the crusade. Obviously just going to chase my tail for days on that one |
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| 09:52.25 | mirela666 | hello, is there any reason why would cdr_addaptive_odbc.conf configuration be killing *, so that it won't get back up |
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| 09:54.17 | AlHafoudh | hi all |
| 09:54.46 | AlHafoudh | what SIP softphone do you use for testing SIP connections and trunks? without "fancy" interface, just a lot of trace and debug |
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| 10:01.56 | wdoekes | mirela666: you're not specific on what happens |
| 10:02.06 | wdoekes | s/on/about |
| 10:02.17 | slicknick5181 | Hello, I have run into a problem with sound on the console chan_oss.c:489 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 10:02.45 | slicknick5181 | In fact there is no file of that name but how do I point the console at the correct device |
| 10:03.08 | WIMPy | What is the correct device? |
| 10:03.23 | WIMPy | That looks as if you don;t have OSS with a working device. |
| 10:03.53 | wdoekes | slicknick5181: ; Set the device to use for I/O |
| 10:03.53 | wdoekes | <PROTECTED> |
| 10:03.55 | slicknick5181 | Thats the other thing I don't know where to find my sound information, I'm on xubuntu 10.04 |
| 10:04.22 | slicknick5181 | wdoekes, But where do I point that |
| 10:04.23 | WIMPy | Maybe you want chan_alsa instead? |
| 10:04.45 | slicknick5181 | WIMPy, I do use ALSA but idk how to change it from dsp |
| 10:04.49 | slicknick5181 | i mean oss |
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| 10:06.22 | mirela666 | wdoekes: I have TrixBox + Web-MeetMe and when I use cdr_Addaptive: |
| 10:06.23 | mirela666 | [wmm] |
| 10:06.23 | mirela666 | connection=meetme ;Note that this matches res_odbc.conf |
| 10:06.23 | mirela666 | table=cdr |
| 10:06.23 | mirela666 | Asterisk dies with 127 code |
| 10:06.43 | mirela666 | wdoekes: when I comment that everything works ok |
| 10:07.08 | WIMPy | slicknick5181: As I said: Use chan_alsa instead of chan_oss. |
| 10:08.04 | slicknick5181 | WIMPy, How do I switch over to chan_alsa I found both files |
| 10:08.23 | WIMPy | modules.conf |
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| 10:09.18 | mirela666 | wdoekes: ofcours last line in log is : |
| 10:09.18 | mirela666 | [Mar 22 10:50:24] VERBOSE[4992] logger.c: == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': [Mar 22 10:50:24] VERBOSE[4992] logger.c: == Found |
| 10:10.17 | wdoekes | mirela666: fire up asterisk from gdb: |
| 10:10.22 | wdoekes | # sudo gdb |
| 10:10.28 | wdoekes | (gdb) run -c |
| 10:10.35 | wdoekes | er |
| 10:10.41 | wdoekes | $ sudo gdb `which asterisk` |
| 10:10.43 | wdoekes | ... run -c |
| 10:11.38 | mirela666 | wdoekes: no command gdb on that server, how will that help me, to trace error better? |
| 10:12.27 | wdoekes | if it crashes, you'll see why |
| 10:12.42 | mirela666 | wdoekes: thx :) |
| 10:13.25 | wdoekes | (I suspect something in your generic odbc config though.. missing driver or something) |
| 10:13.38 | wdoekes | (did you test connecting to your db with isql?) |
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| 10:16.55 | mirela666 | no |
| 10:18.14 | mirela666 | isql: relocation error: /usr/lib/libmyodbc3.so: symbol strmov, version libmysqlclient_15 not defined in file libmysqlclient.so.15 with link time reference |
| 10:18.30 | mirela666 | i guess it is odbc issue |
| 10:19.33 | slicknick5181 | I now appear to have lost my console all together |
| 10:19.43 | slicknick5181 | even help shows no console commands |
| 10:19.57 | slicknick5181 | and I cant find console in the modules to load |
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| 10:29.59 | srp_ | hi, I have installed asterisk-11.2.1 on ubuntu-32bit with ARA (mysql database via odbc). I am using Ekiga as my SIP client for making test calls. The problem is that ekiga (on ubuntu64bit system) gives a "Could not register (Transport error)" whenever I try to register to my Asterisk setup... where as everything works fine if ekiga is running on ubuntu32bit machine.. any ideas if this is a problem with Asterisk or Ekiga... (PS: Ekiga works fine for Ekiga.ne |
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| 10:56.33 | Ice_Strike | Morning |
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| 11:23.25 | seik0 | wdoekes: ok, I tried call on local channel with original func_curl and with your patch. dialplan is simple CURLOPT(...) and CURL(...), where CURL just waits 30 seconds. With original func_curl CURL calls stacks in time (so +30 seconds every next call to wait until previous is done). With patched there were no any waits and calls were completed simulateniously |
| 11:24.15 | seik0 | Shortly, patch works |
| 11:24.21 | wdoekes | seik0: and I assume you get the same results if you remove the local channel |
| 11:24.53 | seik0 | it's harder to test on live channels with many calls |
| 11:25.02 | seik0 | it's midday here |
| 11:25.24 | seik0 | but yes, i think it will work |
| 11:25.54 | seik0 | am i right, this patch already in asterisk code? |
| 11:26.11 | wdoekes | I'm missing a verb |
| 11:27.37 | seik0 | em |
| 11:27.41 | seik0 | =) |
| 11:27.56 | seik0 | what do you mean? |
| 11:28.27 | wdoekes | are you asking whether that patch is committed? it's not |
| 11:31.44 | seik0 | yes, asking that. It's patch you've just made? Just unlock global list a bit earlier, so why it's not commited in issue with "Major" severity? Should be a reason |
| 11:33.19 | wdoekes | seik0: describe your findings on the bug report please |
| 11:33.40 | brian98 | hi guys |
| 11:33.50 | wdoekes | I cannot answer off hand what the unlock does there that late |
| 11:34.10 | brian98 | If I am currently using a free fax for asterisk licence do I need to purchase 1 to get 2 channels or 2 ? |
| 11:36.28 | seik0 | wdoekes: ok, i'll do it |
| 11:36.36 | srp_ | Hello, Assuming I have two registered users 'A' and 'B' with Asterisk 11.2.1, how do I make user 'A' call user 'B' with the url sip:B@<ap addr> rather than calling him on an extension configured in extensions.conf ? |
| 11:37.00 | *** join/#asterisk iondream (~iondream@host-64-47-124-106.masergy.com) |
| 11:37.03 | srp_ | 'A' and 'B' are SIP users .. |
| 11:41.18 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.152) |
| 11:41.20 | WIMPy | srp_: You can only call extensions. But you cann make extensions A and B, off ourse. |
| 11:45.18 | srp_ | WIMPy: i'm using ARA (mysql) for loading sip users from db.. i need something like ekiga.. create an username and then we can make calls like username@ekiga.net.. how do i accomplish this in asterisk? |
| 11:46.09 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 11:46.13 | WIMPy | As I said: Create extensions. |
| 11:46.23 | WIMPy | Noone says they have to be numerical. |
| 11:48.39 | srp_ | WIMPy: instead of adding each extension in extension table for each user, can i match using a regex? like.. for all alphabetical extensions do this.. something like that.. |
| 11:49.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:49.30 | WIMPy | You can use patterns, yes. But if you want one letter extensions you need to be carefull, as there are special extensions in that area. |
| 11:52.18 | *** join/#asterisk kuruption (kuruption@laffs.at.the.lol.lawlz.lulz.liberlawls.com) |
| 11:52.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.37) |
| 11:55.07 | srp_ | WIMPy: but there is one problem.. say i want the users registered to my asterisk (lets call XYZ) to call some sip user from other service providers (say ABC)... now if a sip user of XYZ calls a sip user of ABC, the call ends abruptly with no answer.. why is this? any ways to solve this? |
| 11:56.04 | WIMPy | Because they don't accept your call? |
| 11:56.35 | WIMPy | Most ITSPs will only accept calls from authenticated users. |
| 11:57.25 | srp_ | WIMPy: nope.. for example, take this address (sip:9991429732@sip.tropo.com) mentioned on Tropo's homepage.. it works if i call with an ekiga account but not from a sip account from my asteisk setup... |
| 11:58.10 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 11:58.20 | WIMPy | Do you have somethin in your dialplan to match and route URIs? |
| 11:58.29 | srp_ | WIMPy: the link where that sip address is mentioned is this, https://www.tropo.com/ |
| 11:58.55 | srp_ | WIMPy: but that would make it specific right? |
| 11:59.37 | WIMPy | You could make a matchall and check for a "@" in the extension. |
| 12:00.36 | WIMPy | Some phones will be able to do it themselves again if you create an empty account. |
| 12:02.07 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 12:05.45 | srp_ | WIMPy: for that tropo address, when i call that address, does that hit my asterisk dialplan? because I tested it and its not matching for matchall at all!... |
| 12:07.30 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:11.51 | WIMPy | Depends on your phone. |
| 12:19.58 | *** join/#asterisk benasse (~Thunderbi@proformatique1-gw-std.alionis.net) |
| 12:22.09 | seik0 | I compile patched version of module (saying, func_curl.so), unloads old one from running instance (module unload func_curl.so), replace old file with new one, make "module load func_curl.so" and asterisk crashes. |
| 12:23.00 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:32.27 | *** join/#asterisk madhatt3r (~madhatt3r@62.117.203.84.dyn.user.ono.com) |
| 12:32.32 | madhatt3r | hello all |
| 12:32.53 | madhatt3r | i need to do a follow me across trunks but its not working |
| 12:33.38 | madhatt3r | list is as follows: 101 (trunk A), 201 (trunk B), 202 (trunk B) |
| 12:33.43 | madhatt3r | never reaches trunk B |
| 12:36.02 | madhatt3r | any tips? |
| 12:36.35 | *** join/#asterisk davlefouAMD (~david@41.225.34.79) |
| 12:37.01 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 12:38.32 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 12:42.51 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 12:43.43 | *** join/#asterisk iondream (~iondream@host-64-47-124-106.masergy.com) |
| 12:44.22 | [TK]D-Fender | madhatt3r, Show us what you're doing along with the actual call. |
| 12:44.23 | [TK]D-Fender | ~pb |
| 12:44.23 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 12:44.29 | [TK]D-Fender | ^^^ |
| 12:53.38 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:54.15 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 12:55.11 | *** join/#asterisk serafie (~erin@nat/digium/x-mdwbojvsypnnaiho) |
| 13:00.00 | *** join/#asterisk kresp0 (~kresp0@81.61.25.130.dyn.user.ono.com) |
| 13:01.02 | madhatt3r | http://pastebin.com/g00VCt0u |
| 13:01.04 | madhatt3r | trunk A |
| 13:01.28 | madhatt3r | http://pastebin.com/4rAPKvXM |
| 13:01.30 | madhatt3r | trunk B |
| 13:02.08 | madhatt3r | basically incoming call goes to IVR in A, then option 1 is selected (corresponds to extension in B) |
| 13:02.30 | madhatt3r | extension in B has a follow me that points to an extension in A |
| 13:04.09 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 13:06.20 | *** join/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 13:07.35 | *** join/#asterisk taylorbyte2013 (~cyberninj@120.156.35.244) |
| 13:09.41 | [TK]D-Fender | madhatt3r, -- dialparties.agi: Filtered ARG3: 201-102-103 <- that's the hunt list |
| 13:09.52 | madhatt3r | yes |
| 13:10.17 | madhatt3r | 201 is server B |
| 13:10.21 | [TK]D-Fender | madhatt3r, it tried 201 |
| 13:10.23 | madhatt3r | 102 103 are in A |
| 13:10.27 | [TK]D-Fender | faile dafter 10s |
| 13:10.27 | madhatt3r | yes |
| 13:10.33 | madhatt3r | correct |
| 13:10.52 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 13:11.21 | [TK]D-Fender | <PROTECTED> |
| 13:11.25 | madhatt3r | why would it say no hunt members? |
| 13:11.28 | [TK]D-Fender | And determined nothing left to dial |
| 13:11.37 | madhatt3r | maybe its the strategy |
| 13:11.53 | madhatt3r | first available i have it set |
| 13:12.00 | madhatt3r | which is kinda wrong now that i think |
| 13:13.00 | [TK]D-Fender | I think it's wrong you think you can include internal extensions from another server like that |
| 13:13.02 | [TK]D-Fender | ^ |
| 13:13.19 | madhatt3r | ok# |
| 13:13.26 | madhatt3r | i tried with misc application and destination |
| 13:13.35 | [TK]D-Fender | Show us your follow-me |
| 13:13.38 | madhatt3r | but didnt work so i went this other way |
| 13:13.49 | madhatt3r | ..im using elastix |
| 13:13.50 | madhatt3r | 201 |
| 13:13.50 | madhatt3r | 102 |
| 13:13.50 | madhatt3r | 103 |
| 13:13.57 | [TK]D-Fender | madhatt3r, this is a server B problem |
| 13:14.02 | madhatt3r | ok |
| 13:14.03 | [TK]D-Fender | SCREEN SHOT <------- |
| 13:14.13 | [TK]D-Fender | show us what you're dialing on that side |
| 13:15.23 | madhatt3r | http://postimg.org/image/e0avbgtwx/ |
| 13:15.59 | madhatt3r | i had it set to firstavailable, now just changed it to hunt, I also tried with |
| 13:16.08 | madhatt3r | # at the end of 102 and 103 |
| 13:16.11 | [TK]D-Fender | you told server B to just hunt 2 extensions that aren't real extensions on that server |
| 13:16.13 | [TK]D-Fender | ^ |
| 13:16.25 | madhatt3r | understood |
| 13:16.52 | [TK]D-Fender | if you want to use the TRUNK DIAL method to reach them you have to add a character to their entry |
| 13:17.00 | [TK]D-Fender | read the tool-tip for this |
| 13:17.17 | madhatt3r | ok |
| 13:19.23 | *** join/#asterisk gusto (~gusto@ppp-62-216-206-179.dynamic.mnet-online.de) |
| 13:22.27 | *** part/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:23.15 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 13:24.19 | *** join/#asterisk FUF (fuf@evilgeni.us) |
| 13:24.23 | *** part/#asterisk FUF (fuf@evilgeni.us) |
| 13:28.49 | madhatt3r | i dont know where to find this tool tip |
| 13:29.27 | [TK]D-Fender | madhatt3r, float your cursor over the underlined field name.... |
| 13:30.46 | madhatt3r | which field name |
| 13:30.51 | madhatt3r | trunk dial? |
| 13:30.58 | madhatt3r | follow-me list? |
| 13:31.00 | [TK]D-Fender | <[TK]D-Fender> if you want to use the TRUNK DIAL method to reach them you have to add a character to their entry |
| 13:31.52 | madhatt3r | ok |
| 13:31.53 | madhatt3r | working now |
| 13:32.18 | madhatt3r | had tried that one before but had the wrong strategy |
| 13:32.22 | madhatt3r | niceOne |
| 13:32.25 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 13:32.31 | madhatt3r | thanksALotta |
| 13:32.32 | eXcAliBuR | hello, how do do |
| 13:32.53 | [TK]D-Fender | eXcAliBuR, Try a toilet |
| 13:33.26 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
| 13:33.30 | eXcAliBuR | :P |
| 13:34.11 | eXcAliBuR | i'ma play with SLA today |
| 13:34.12 | eXcAliBuR | :) |
| 13:35.30 | [TK]D-Fender | ~wglwat |
| 13:35.30 | infobot | hmm... wglwat is well, good luck with all that |
| 13:35.32 | [TK]D-Fender | :) |
| 13:38.14 | *** join/#asterisk skirmisha (~vk@46.47.82.189) |
| 13:38.19 | skirmisha | hi guys |
| 13:38.55 | eXcAliBuR | ~ty |
| 13:38.56 | infobot | i heard ty is thank you, or cobb -- a baseball player |
| 13:38.57 | skirmisha | how can i check whether device state is working properly? On some extension BLF is not working and i am trying to understand why |
| 13:39.10 | eXcAliBuR | i learn quickly |
| 13:39.11 | eXcAliBuR | :) |
| 13:39.12 | skirmisha | i have tigase server where all asterisk servers connect to |
| 13:39.26 | skirmisha | i can see the servers in buddy list |
| 13:39.30 | [TK]D-Fender | eXcAliBuR, What are you looking to use it with? |
| 13:39.39 | skirmisha | any ideas where to check |
| 13:39.52 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:39.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:39.54 | eXcAliBuR | I want to share my analog lines so people can see when they are in use |
| 13:39.56 | [TK]D-Fender | skirmisha, "core show hints" |
| 13:40.04 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:29e7:b75f:6641:5a1a) |
| 13:40.10 | [TK]D-Fender | eXcAliBuR, On what phones? |
| 13:40.23 | eXcAliBuR | best phone in the world, Digium |
| 13:40.27 | eXcAliBuR | :) |
| 13:40.36 | eXcAliBuR | do I get points for saying that |
| 13:40.46 | [TK]D-Fender | With somebody possibly... |
| 13:42.05 | [TK]D-Fender | eXcAliBuR, You could just use basic hints on the channels without going through th rest of the "SLA" process |
| 13:42.12 | skirmisha | i have show hints |
| 13:42.30 | skirmisha | it shows everything, but when ext dial it does not update the status |
| 13:42.48 | skirmisha | basically can you tell me what watchers means in above output? |
| 13:45.17 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 13:47.20 | eXcAliBuR | will just sit here idle, revving occasionaly |
| 13:48.28 | jmetro | vrooom |
| 13:49.11 | srp_ | I am planning to buy an analog FXO/FXS card.. but when I looked up the vendor sites I see additional prices for modules. What are these modules and do i require them? My requirement is an analog TDM 2 port FXO card... |
| 13:49.45 | *** part/#asterisk gonewage (~gonewage@smtp.rwrlaw.com) |
| 13:50.50 | [TK]D-Fender | srp_, the cards are MODULAR and support different kinds of ports. The card itself is the backing but you still need to buy the specific modules for the combination YOU want |
| 13:51.23 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 13:52.32 | *** join/#asterisk _Corey_ (~corey@173.161.229.46) |
| 13:55.41 | srp_ | [TK]D-Fender: so the card is just like a thing to plug in modules is it? |
| 13:56.11 | [TK]D-Fender | yes |
| 13:56.21 | srp_ | [TK]D-Fender: so for 2 port fxo, i need two fxo modules? |
| 13:56.27 | [TK]D-Fender | typically |
| 13:56.42 | [TK]D-Fender | the high-densitycards can use 4-port modules |
| 13:57.37 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:29e7:b75f:6641:5a1a) |
| 13:57.49 | srp_ | [TK]D-Fender: I looked up this thing (http://www.x100p.com/products/FXO.php) which is the cheapest of the lot. does it have modules or just the card? how do i find out? |
| 13:59.06 | [TK]D-Fender | just the card |
| 13:59.34 | [TK]D-Fender | And X100's are flakey as far as callerid reliability, etc go |
| 13:59.43 | _Corey_ | oooh, they come with a hologram so you know they're genuine though.... ;) |
| 13:59.44 | [TK]D-Fender | Remeber that ... |
| 13:59.49 | [TK]D-Fender | ~ygwypf |
| 13:59.49 | infobot | ygwypf is, like, You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 14:00.14 | srp_ | [TK]D-Fender: :) so, i need to buy a module seperately to work with this right? :-| |
| 14:00.31 | [TK]D-Fender | srp_, no, that card is an INDEPENDENT 1-port card |
| 14:00.36 | [TK]D-Fender | non-modular |
| 14:01.15 | srp_ | [TK]D-Fender: and one more thing on my mind.. it says it supports zaptel.. i wonder if i'll be able to use it with asterisk-11.2.1 which has dahdi... |
| 14:02.09 | [TK]D-Fender | yes |
| 14:02.27 | [TK]D-Fender | But still the X100-type cards are the bottom of the food chain |
| 14:03.07 | srp_ | [TK]D-Fender: ya.. i understand.. this is only for experimentation.. so i'm choosing the cheapest one.. :) |
| 14:03.35 | [TK]D-Fender | srp_, What is your end goal? |
| 14:04.20 | srp_ | [TK]D-Fender: I'm working on a academic project based on asterisk.. so want to use it for that.. for prototyping and basic usage... |
| 14:04.46 | [TK]D-Fender | Does it require it being a PCI card? |
| 14:05.11 | [TK]D-Fender | What are the parameters of the project and what do you think you'd be doing outside of this specific project? |
| 14:05.50 | srp_ | [TK]D-Fender: PCI.. ah.. no.. I thought this was the default thing.. just plug it to the PCI port and configure is what i thought.. |
| 14:06.19 | [TK]D-Fender | srp_, Ok, answer the rest... |
| 14:06.58 | srp_ | [TK]D-Fender: parameters are that, i'm supposed to build a pbx with asterisk and freepbx and add / delete / configure extensions... have some basic ivr and voicemail options for the user... |
| 14:07.28 | srp_ | [TK]D-Fender: and using ARA with mysql |
| 14:08.45 | *** join/#asterisk evil-man (~evil-man@insider-mail.icf.org.ru) |
| 14:09.34 | *** join/#asterisk serafie1 (~erin@24.214.158.242) |
| 14:10.06 | srp_ | [TK]D-Fender: I hope this card will help me do all the above.. right? |
| 14:10.09 | [TK]D-Fender | srp_, Linksys SPA-3102 <- |
| 14:10.20 | [TK]D-Fender | that'll give you 1 FXO, and 1 FXS on the device |
| 14:10.35 | srp_ | [TK]D-Fender: checking it.. |
| 14:10.37 | [TK]D-Fender | and you won't need DAHDI. It will use SIP to talk to * |
| 14:11.22 | [TK]D-Fender | srp_, also, FreePBX does not use ARA. Those ideas are not compatible |
| 14:11.40 | [TK]D-Fender | And is a terrible test of "learning something" |
| 14:13.28 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 14:13.42 | jeffspeff | anybody here use opennms? |
| 14:13.56 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 14:14.02 | srp_ | [TK]D-Fender: but I need to connect my asterisk setup to a PSTN telephone line finally.. to make outbound calls from sip phones to normal telephones.. I think its not possible in that Linksys device right? |
| 14:16.07 | srp_ | [TK]D-Fender: oh.. that linksys device has a FXO.. so i can connect it to telephone lines.. :) |
| 14:18.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.37) |
| 14:18.53 | [TK]D-Fender | srp_, one PHONE, one LINE |
| 14:20.39 | srp_ | [TK]D-Fender: but if i have a PRI line (instead of a normal telephone line), then I can have more than one continuous active channels with just one FXO port right? |
| 14:21.03 | [TK]D-Fender | srp_, correct |
| 14:21.16 | [TK]D-Fender | srp_, but normal people don't have those. The cost is considerable |
| 14:21.39 | [TK]D-Fender | srp_, the service as well as the hardware |
| 14:22.01 | *** join/#asterisk Rhomber (~david@60-240-245-17.static.tpgi.com.au) |
| 14:25.13 | *** join/#asterisk FUF (~fuf@shoreline.phuph.org) |
| 14:26.12 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 14:34.33 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 14:38.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:40.44 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:41.36 | eXcAliBuR | i feel slightly un smart :( |
| 14:41.58 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:42.29 | srp_ | [TK]D-Fender: I checked out the entire specs of that Linksys device.. its amazing.. I have a doubt.. why do more people prefer the cards (digium/sangoma/etc.) instead of such Linksys voice gateway that you just showed me.. |
| 14:46.35 | [TK]D-Fender | srp_, this device isn't great either but it has 2 ports, doesn't require DAHDI (more installation options), doesn't have to physically install in the server, etc. I still wouldn't runa company on it for FXO, but it's a question of fitting the need |
| 14:46.51 | [TK]D-Fender | srp_, if I wanted a real card for a bunch of lines I'd go for a quality card |
| 14:48.24 | srp_ | [TK]D-Fender: sounds reasonable :) Thanks for the help... you made my day :D |
| 14:53.23 | [TK]D-Fender | srp_, You're welcome |
| 14:53.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:53.58 | [TK]D-Fender | srp_, side-benefit : the linksys box is commidty gear you'll probably get faster & easier in regular distribution channels |
| 14:54.47 | *** join/#asterisk chatran (~chatran_@186.212.75.244) |
| 14:54.50 | chatran | hello |
| 14:55.07 | chatran | i have an problem |
| 14:55.17 | chatran | with pap2t |
| 14:55.23 | chatran | behind adsl modem |
| 14:55.48 | chatran | when i call i can listen them and them do not listen me ? |
| 14:56.01 | chatran | its firewall of modem ? |
| 14:56.03 | jeev | ?? nat |
| 14:56.05 | jeev | heh |
| 14:56.05 | kaldemar | ~sipnat |
| 14:56.05 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 14:56.24 | *** join/#asterisk aliban (~aliban@217.130.151.218) |
| 14:56.27 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:56.27 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:56.31 | igcewieling | Does anyone know what this might mean? chan_sip.c:20897 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog '085d0db44e8e04a50cd659ed06bdf8a7@209.220.119.17:5060'. Giving up. |
| 14:56.34 | chatran | ok |
| 14:59.24 | Greenlight | igcewieling: I see those from time to time. Asterisk sometimes goes a little crazy with it's SIP messages (from looking at a capture), esepcially it allowing REINVITES |
| 14:59.37 | Greenlight | *if |
| 15:00.30 | Greenlight | If you do a packet capture, and then lookup the call in the capture, you'll probably see the craziness I speak of |
| 15:01.37 | igcewieling | ok then, how to I stop Asterisk from going crazy? 8-) |
| 15:01.58 | Greenlight | Do you allow directmedia? |
| 15:02.09 | igcewieling | Honestly, we've had no end of problems after upgrading to Asterisk 1.8 |
| 15:02.12 | igcewieling | Greenlight: yes. |
| 15:02.37 | igcewieling | because none of our endpoints are behind NAT and CPU usage is significant if we disable reinvites |
| 15:02.40 | Curs0r | Despite my strong self-advice not to do so, I stayed up all night and whipped asterisk-gui into submission :) The deed is done |
| 15:05.08 | [TK]D-Fender | Curs0r, Add our strong advice to your own :) |
| 15:05.54 | Curs0r | [TK]D-Fender, it was one of thos personal vendetta, can't let the software win type things |
| 15:06.41 | eXcAliBuR | VROOM |
| 15:07.09 | [TK]D-Fender | Curs0r, Even when it works... you still lose :) |
| 15:07.43 | Curs0r | I had fun, and it works. Seems like a win to me. |
| 15:09.25 | [TK]D-Fender | Curs0r, That's just how insidious it is! You get more entrenched and then you NEVER get out! |
| 15:09.59 | Curs0r | Indeed, asterisk is at once the most confusing and joyfully esoteric thing I've played with in server-land in a long time |
| 15:10.36 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 15:11.02 | eXcAliBuR | thats a big word |
| 15:11.11 | Curs0r | in? |
| 15:14.08 | Greenlight | igcewieling: Disabling directmedia sorted it for us. I'm guessing there are some bugs in the SIP handling otherwise. |
| 15:14.54 | Greenlight | igcewieling: In terms of CPU load, it did increase but it's nothing crazy |
| 15:15.17 | igcewieling | Greenlight: how many calls do you handle at the same time? |
| 15:16.14 | Greenlight | 300-700 |
| 15:16.44 | Greenlight | We've even ran that through a VPS before, no issues |
| 15:17.01 | Greenlight | Currently, though, it runs on an ESXi guest |
| 15:18.26 | Greenlight | You'll want to ensure the codecs etc all match, and it'll do p2p bridging, the packets dont even hit asterisk core |
| 15:19.56 | igcewieling | we are very careful about codecs |
| 15:21.07 | igcewieling | 700 calls with media going through Asterisk on one server?? |
| 15:23.10 | Greenlight | Peak, yea. Generally we're sitting in the 400-500 region |
| 15:24.14 | Greenlight | If it's working correctly, you should see it mention p2p bridging in the logs (debug?) |
| 15:25.15 | Greenlight | I was actually only going to carry the media temporarily as a fix to the types of errors you saw, until I could sort more permenantly, but since it worked so well we've stuck with it |
| 15:25.41 | igcewieling | <PROTECTED> |
| 15:26.24 | Greenlight | I think local bridging covers both core and p2p bridging , from the last time I looked at that bit of code |
| 15:27.04 | igcewieling | would be nice if the message indicated which it did. |
| 15:27.12 | Greenlight | Indeed |
| 15:27.49 | Greenlight | iirc there's a massive "if" block in the code, which I refer to as the "stars aligning", if all that's okay, then it'll p2p bridge |
| 15:28.17 | igcewieling | we expect to have to disable directmedia when we get rid of our RTP proxy anyway (because of carrier routing policies) but I'd still like it to work until then |
| 15:29.14 | igcewieling | we are putting a 3rd server into our Asterisk cluster -- I'll feel better when we have it deployed |
| 15:29.17 | Greenlight | Enable debug and see if you can spot anything; we definetly had it display in the CLI somehow |
| 15:30.22 | Greenlight | What sort of CPU load are you seeing, and how many calls ? |
| 15:31.26 | igcewieling | I don't quite understand CPU load in a multi-CPU enviroment, but Asterisk takes about 30% of CPU at around 100 calls. |
| 15:31.37 | igcewieling | 16 CPU box |
| 15:32.05 | Greenlight | Yea, that sounds okay. 30% is nothing. That with directmedia disabled ? |
| 15:32.35 | Greenlight | You're "maxxed out" load is 1600% |
| 15:32.55 | Greenlight | Well, unless you're hyper-threaded |
| 15:35.26 | jmetro | 1600%... huh |
| 15:36.33 | Greenlight | 100% is only 1 core |
| 15:36.40 | ccherrett | can anyone point me to an example extentions.conf file using a sangoma card? or does sangoma even matter once the card is configured and working, does it just use something like g0? |
| 15:36.45 | Greenlight | Presuming you're going by CPU% in top |
| 15:37.46 | Greenlight | For example I've an 8 core box and asterisk process on there sits around 170-250% |
| 15:37.50 | eXcAliBuR | my voicemail gets to the part" the person at extension ### is unavailable" then it hangs up |
| 15:38.29 | eXcAliBuR | and i gets things like this: [Mar 22 11:41:49] NOTICE[1421]: chan_sip.c:25975 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 255 |
| 15:38.47 | eXcAliBuR | but i added it in my voicemail.conf |
| 15:39.01 | ccherrett | when I call the box I see: [Mar 22 09:33:50] WARNING[4702][C-0000000e]: pbx.c:6390 __ast_pbx_run: Channel 'DAHDI/4-1' sent to invalid extension but no invalid handler: context,exten,priority=from-zaptel,s,1 |
| 15:39.14 | ccherrett | I can use the CLI to place a call |
| 15:39.23 | ccherrett | to an external phone line |
| 15:39.54 | ccherrett | so I am done to configuring my card in asterisk |
| 15:40.03 | ccherrett | just don't know where to start |
| 15:43.57 | jmetro | attach something and dial it? |
| 15:44.18 | kaldemar | ccherrett: the fact that you use a sangoma card is irrelevant. your warning already tells you that you should have exten s in context from-zaptel. |
| 15:44.38 | eXcAliBuR | http://pastebin.mozilla.org/2236619 can someone have a looksee at that and let me know whats wrong? |
| 15:44.49 | jeffspeff | anybody here use opennms? i'm having some troubles getting it to work with asterisk. |
| 15:44.57 | ghost75 | someone has experience with bluetooth pstn gateway ? |
| 15:46.51 | ghost75 | chan_mobile |
| 15:47.47 | [TK]D-Fender | eXcAliBuR, We don't know what you expect or why... so nothing looks blatantly wrong there to us |
| 15:48.27 | [TK]D-Fender | eXcAliBuR, For your Vm question... devices aren't supposed to subscribe for it... you're supposed to define the box in your peer |
| 15:51.23 | *** join/#asterisk Xserver (~Xserver@180.188.255.11) |
| 15:51.42 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 15:53.33 | eXcAliBuR | i can't leave a message |
| 15:53.38 | eXcAliBuR | it hangs up on me |
| 15:53.42 | eXcAliBuR | >:( |
| 15:55.11 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 15:57.25 | Xserver | any suggestion for a billing package for asterisk ? |
| 15:57.30 | Xserver | astpp or vbilling ? |
| 15:57.41 | Xserver | or anything similar ? |
| 16:02.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:06.53 | *** join/#asterisk F41L (~herp@unaffiliated/f41l) |
| 16:08.51 | F41L | Hey all, question for ye... who do you use as a SIP trunk provider? I am working to move over to voip, and having a hard time finding a sip trunk provider that seems good quality and for a decent price. I was wondering if you all had any insight you'd be willing to share. |
| 16:09.19 | carrar | ~itsp |
| 16:09.19 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 16:09.23 | F41L | My first choice -looks- like it would be broadvox. |
| 16:09.43 | carrar | Where are you at? |
| 16:09.46 | F41L | California |
| 16:09.57 | carrar | ~itsplist-us |
| 16:09.57 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:10.18 | F41L | I looked at broadvox, but was also interested in didlogic, but they.. being a telephony company, don't have a telephone number to speak with a sales rep? |
| 16:10.52 | carrar | What kind of call volume? |
| 16:11.15 | F41L | also, carrar, I was looking for opinions on them moreso, not just a listy-list :3 But I appreciate the response. |
| 16:11.25 | F41L | at most 4 concurrent calls, and 1 fax |
| 16:11.41 | carrar | So you want to ensure your itsp supports t38 |
| 16:12.42 | carrar | I don't have any expirence with the above formentioned ITSP's |
| 16:13.03 | F41L | well, who -do- you have experience with? :) |
| 16:13.13 | carrar | I'm a itsp :) |
| 16:13.25 | carrar | in seattle |
| 16:13.52 | F41L | heh :_ |
| 16:13.54 | F41L | :D |
| 16:17.33 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 16:20.32 | Xserver | check with geils |
| 16:20.37 | Xserver | they offer unlimited sip trunks |
| 16:21.08 | Xserver | www.geils.com |
| 16:21.21 | aliban | ~itsplists-eu |
| 16:21.25 | F41L | Do you have any experience with their call quality and reliability, by chance? |
| 16:21.45 | *** join/#asterisk awilliams (mistik1@unaffiliated/mistik1) |
| 16:22.01 | Xserver | Nope ... unfortunately they do not serve my requirement of dialer traffic .. |
| 16:22.31 | igcewieling | Dialer? |
| 16:22.36 | igcewieling | gets the garlic and holy water |
| 16:22.40 | F41L | robocalls |
| 16:23.03 | jmetro | i dont think garlic water would taste very good. |
| 16:23.07 | F41L | lol |
| 16:23.13 | F41L | they have garlic ice cream |
| 16:23.18 | F41L | that tastes fine. |
| 16:23.22 | igcewieling | jmetro: they are for the eyes of the telemarketers |
| 16:23.23 | jmetro | noty |
| 16:23.28 | Xserver | Yep ... 60% of my clients are using auto-dialers |
| 16:23.33 | aliban | talking about sip provideres |
| 16:23.45 | igcewieling | Wow, that is an amazing concentration of PURE EVIL. |
| 16:24.10 | jmetro | imagine , only one company has to be affected to get rid of all those telemarketers |
| 16:24.31 | aliban | any advice on a sip provider on UK? |
| 16:24.34 | Xserver | lolz.... |
| 16:24.34 | igcewieling | jmetro: *nod* |
| 16:26.24 | Xserver | seriously .... dialer traffic is indeed evil |
| 16:29.40 | Xserver | anyone had any luck with vbilling or astpp ? |
| 16:30.27 | Xserver | or similar applications ? |
| 16:31.36 | *** join/#asterisk kesselklopfer79 (~kesselklo@astaro.starface.de) |
| 16:32.06 | igcewieling | we import the wholesaler CDRs directly into our billing system so we don't worry about billing from Asterisk CDRs |
| 16:32.27 | ghost75 | when using nat=no, allowguest=no and no ports are forwarded on router, is it still possible that asterisk can be "hacked" ? |
| 16:32.46 | malcolmd | um..sure? |
| 16:33.05 | igcewieling | ghost75: all in alwaysauthreject=yes |
| 16:33.36 | ghost75 | how the system can be reached from outside? |
| 16:34.38 | ghost75 | alwaysauthreject is sending always same error i think |
| 16:36.58 | igcewieling | ghost75: what exactly are you trying to do? |
| 16:37.16 | igcewieling | if you simply want to prevent access, then use iptables |
| 16:39.26 | *** join/#asterisk benasse (~Thunderbi@proformatique1-gw-std.alionis.net) |
| 16:39.57 | ghost75 | i want to know if this would be hackable |
| 16:40.08 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:40.18 | igcewieling | ghost75: there is no way to know about bugs discovered in the future |
| 16:40.36 | ghost75 | more like the usual kiddie hackers |
| 16:42.11 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 16:49.44 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:49.54 | *** join/#asterisk mattsl (~user@c-50-142-241-34.hsd1.tn.comcast.net) |
| 16:57.22 | *** join/#asterisk KNERD (~KNERD@24.175.249.177) |
| 16:57.24 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:57.24 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:00.49 | chatran | hi |
| 17:01.27 | chatran | i did put my pap2t behind an modem adsl... i put it on dmz host but still people not listen me, but i listen them. |
| 17:01.53 | chatran | my asterisk have no firewall its ip on internet directly |
| 17:02.24 | chatran | any ideia what is happen? |
| 17:02.40 | igcewieling | chatran: "its ip on internet directly" and "i put it on dmz host" are incompatible statements. |
| 17:02.46 | Greenlight | Indeed |
| 17:02.56 | igcewieling | chatran: make sure you have directmedia=no |
| 17:02.58 | [TK]D-Fender | Champi, You should not DMZ your client side and you did not configfure your peers right on * for them to work from behind NAT |
| 17:03.19 | [TK]D-Fender | igcewieling, Server is public. PAP is behind NAT |
| 17:03.23 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:03.26 | [TK]D-Fender | (and DMZ'd) |
| 17:03.31 | saint_ | hi alllll ! |
| 17:04.07 | igcewieling | It is a little known fact that "DMZ" stands for "Destroy My callZ" |
| 17:04.45 | navaismo | indeed |
| 17:04.54 | *** join/#asterisk edong23 (~quassel@mptc-dhcp-50-220.mptelco.com) |
| 17:05.51 | chatran | igcewieling asterisk have internet ip no firewall... pap2t is behind modem adsl |
| 17:06.13 | chatran | i did try put port redirect too.. still no work |
| 17:06.21 | igcewieling | chatran: you have one of the easiest and most common setups. |
| 17:06.32 | igcewieling | chatran: it won't work right if you forward ports. |
| 17:06.46 | igcewieling | so stop forwarding ports, stop putting the PAP in the DMZ |
| 17:06.48 | chatran | yes !!! but not work !!! and im dont know why... maybe becose protocol |
| 17:06.59 | KNERD | Is mpg123 needed anymore for since asterisk 11? |
| 17:07.03 | igcewieling | chatran: do you have directmedia=no? |
| 17:07.14 | chatran | igcewieling let me see |
| 17:07.25 | chatran | on asterisk ? |
| 17:07.29 | chatran | sip.conf ? |
| 17:07.29 | igcewieling | chatran: the fact it doesn't work likely has NOTHING WHATSOEVER to do with port forwwarding |
| 17:07.33 | igcewieling | chatran: correct |
| 17:07.35 | chatran | sec |
| 17:07.56 | malcolmd | KNERD: mpg123 is needed if you're playing back mp3 files for moh |
| 17:08.07 | KNERD | groovy, and thanks |
| 17:08.11 | malcolmd | np |
| 17:08.40 | [TK]D-Fender | chatran, Fix your peers. you peer settings are wrong for the phones. |
| 17:09.43 | chatran | can i past here ? |
| 17:09.49 | Greenlight | ~pb |
| 17:09.49 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:10.14 | chatran | is 4 lines :( |
| 17:10.51 | chatran | http://pastebin.com/0JWUTr5Q |
| 17:10.54 | chatran | igcewieling |
| 17:10.58 | chatran | look my config |
| 17:11.02 | chatran | on sip.conf |
| 17:11.17 | chatran | i did put what you say, still no work |
| 17:11.24 | igcewieling | that is a template |
| 17:11.31 | Greenlight | What's in [general] ? |
| 17:12.02 | igcewieling | I bet he is using port and bindport or bindip |
| 17:12.12 | Greenlight | agrees |
| 17:12.20 | chatran | in general ? |
| 17:12.21 | chatran | sec |
| 17:12.32 | Greenlight | Or has a route that tries to be "SIP aware" |
| 17:12.34 | Greenlight | *router |
| 17:12.57 | chatran | http://pastebin.com/wYDxWnb8 |
| 17:13.00 | Greenlight | They should all be burned, horrid devices those SIP ALG routers |
| 17:13.19 | chatran | Greenlight :) pap2t ? |
| 17:13.51 | saint_ | can someone recommend a good sip wireless phone ? |
| 17:14.09 | [TK]D-Fender | chatran, that is only HALF of your peer config and you are missing bits |
| 17:14.13 | igcewieling | saint_: Kirk wireless from Polycom |
| 17:14.34 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 17:14.41 | igcewieling | or if you are a cheap SOB and ATA with a standard cordless phone |
| 17:14.47 | igcewieling | s/and/an |
| 17:15.24 | chatran | [TK]D-Fender half ? missing what ? |
| 17:15.33 | Greenlight | chatran: Not sure what the default if not specifcied is, but you should at least define "nat=force_rport,comedia" and "externip=" |
| 17:15.44 | saint_ | igcewieling: nah, I'm looking for a full sip solution.. |
| 17:16.56 | chatran | Greenlight ok i put |
| 17:16.59 | igcewieling | Greenlight: his server is not begind nat, so externip is not needed |
| 17:17.01 | chatran | lets test |
| 17:17.21 | chatran | i think my problem is the modem |
| 17:17.26 | igcewieling | chatran: have you read ANYTHING on line about Asteisk and NAT? Doesn't look like it. If you had you would put nat=yes in your config |
| 17:17.29 | chatran | becose i have same config on other modem |
| 17:17.34 | [TK]D-Fender | chatran, ... |
| 17:17.39 | igcewieling | chatran: you can't know that until you actually have a correct config |
| 17:17.41 | Greenlight | nat=yes is depreached iirc |
| 17:17.45 | [TK]D-Fender | chatran, You showed us only a TEMPLATE that holds HALF of your settings. |
| 17:17.47 | igcewieling | Greenlight: I don't care. |
| 17:18.03 | saint_ | any other recommendation than polycom ..? |
| 17:18.12 | igcewieling | Greenlight: especially since nat=yes may be partially broken in Asterisk 11 |
| 17:18.17 | chatran | [TK]D-Fender its not a template :) is my config and it works well behind other modems |
| 17:18.23 | chatran | sec |
| 17:18.26 | [TK]D-Fender | saint_, Ok, what doesn't Polycom do for you? |
| 17:18.29 | chatran | i will test again |
| 17:18.37 | Greenlight | igcewieling: nat=yes should not be used from what the CLI groans at you if you use it |
| 17:18.57 | [TK]D-Fender | chatran, Half the options are NOT there |
| 17:19.02 | saint_ | [TK]D-Fender: nothing.. i just like to have more than one option .. |
| 17:19.03 | [TK]D-Fender | chatran, because as-is it is broken |
| 17:19.04 | chatran | the best is dmz or port forward ? |
| 17:19.17 | Greenlight | chatran: Neither |
| 17:19.22 | Greenlight | You shouldn't need either |
| 17:19.23 | KNERD | Where are the docs on the CONFIGURE file so I can set the correct CPU? I am just not seeing them. |
| 17:19.25 | chatran | nat = port forward |
| 17:19.31 | Greenlight | No |
| 17:19.40 | igcewieling | chatran: NO! NAT != PORT FORWARD |
| 17:19.44 | chatran | than how i redirect on modem ? |
| 17:19.50 | Greenlight | sighs |
| 17:19.57 | igcewieling | chatran: when you port forward you prevent Asterisk from fixing up the packets so they work with NAT |
| 17:20.00 | chatran | i have port trigger too |
| 17:20.08 | Greenlight | turn all that crap off |
| 17:20.12 | [TK]D-Fender | chatran, No forwarding at all of any kind |
| 17:20.16 | chatran | ok |
| 17:20.21 | Greenlight | Then, as we've been saying, fix your settings. |
| 17:20.23 | [TK]D-Fender | chatran, And show us your COMPLETE peer settings for both of them |
| 17:20.27 | igcewieling | chatran: the outbound packet from the endpoint to asterisk will open a port forward in NAT just like it does with HTTP and every other protocol you use. |
| 17:20.36 | chatran | how i will came out to internet if not using nat ? |
| 17:20.40 | saint_ | igcewieling: what kirk model of the polycom 3040, 4020, 4040, 5020, 5040 |
| 17:20.45 | Greenlight | You WILL use NAT |
| 17:20.51 | igcewieling | chatran: same way a web browser gets to the internet |
| 17:20.54 | Greenlight | We never said not to use NAT |
| 17:20.58 | igcewieling | saint_: no idea. |
| 17:21.16 | igcewieling | saint_: check the prices and proceed from there. |
| 17:21.27 | chatran | ok its nat ... on modem is: redirect ports |
| 17:21.33 | Greenlight | No |
| 17:22.06 | chatran | on modem have this option |
| 17:22.07 | Greenlight | Port redirection should not be neccissary |
| 17:22.13 | igcewieling | I give up. chatran I wish you the best of luck, but until you can start following instructions it is unlikely you'll get this working. |
| 17:22.20 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 17:22.32 | chatran | ig got mad |
| 17:22.49 | Greenlight | Yea, cause you're not listening |
| 17:22.56 | chatran | im trying |
| 17:23.10 | chatran | is 2/3 people writing at same time |
| 17:23.39 | chatran | so, on modem i dont need port redirect ( nat / DNAT ) to inside my lan ? |
| 17:23.49 | chatran | just nat to out ? |
| 17:24.03 | Greenlight | Yea |
| 17:24.12 | chatran | ok, removed |
| 17:24.15 | Greenlight | No port redirection, no DMZ. Just normal settings. |
| 17:24.17 | saint_ | when i look for kirk, i end up on this page http://spectralink.polycom.com .. their web site sucks |
| 17:24.27 | Greenlight | If you can access the internet, then that should work. |
| 17:24.36 | chatran | ok, lets test |
| 17:24.42 | Greenlight | Now, go fix your settings on Asterisk |
| 17:24.49 | Greenlight | *then* you can test |
| 17:26.29 | eXcAliBuR | i'm back |
| 17:26.36 | eXcAliBuR | i had sushi for lunch |
| 17:26.37 | eXcAliBuR | :] |
| 17:26.41 | eXcAliBuR | is happy |
| 17:26.50 | chatran | eXcAliBuR :(~) |
| 17:27.00 | chatran | Greenlight is working normal |
| 17:27.04 | chatran | on other places |
| 17:27.19 | Greenlight | chatran: What do you mean ? |
| 17:27.33 | chatran | i have SAME config behind other modem |
| 17:27.38 | eXcAliBuR | now for my voicemail problem, it says the person at extension... bit then hangs up |
| 17:27.47 | Greenlight | chatran: Then, it's your modem |
| 17:27.49 | chatran | its modem or pap2t the problem but... |
| 17:27.52 | eXcAliBuR | where to start |
| 17:28.11 | chatran | its an tp-link / td 8810 |
| 17:29.55 | eXcAliBuR | autofallthrough = yes/no ... right now it's at yes... is that normal? |
| 17:33.22 | eXcAliBuR | >:( |
| 17:33.28 | eXcAliBuR | now it just times out |
| 17:35.54 | chatran | Greenlight |
| 17:35.56 | chatran | [Mar 22 14:34:10] WARNING[18991]: channel.c:5128 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) |
| 17:36.01 | eXcAliBuR | redlight |
| 17:36.07 | eXcAliBuR | o.O |
| 17:36.17 | chatran | eXcAliBuR i dont know |
| 17:36.57 | Greenlight | You're using g729 at one side and alaw at the other, and don't have the ability to transcode (you need a license) |
| 17:36.59 | [TK]D-Fender | chatran, codec failure |
| 17:37.10 | saint_ | Is there a way to use the Dial command with the p option (privacy) to make a phone ring 10 sec, play a message to the caller , then ring the phone again ? |
| 17:37.17 | chatran | [TK]D-Fender its not failure becose is WORKING |
| 17:37.24 | saint_ | when I do that, if I pick up at the first time and hang up, the phone rings right away again |
| 17:37.26 | chatran | but still give this message |
| 17:38.20 | *** join/#asterisk jkroon (~jkroon@kerberos.uls.co.za) |
| 17:38.50 | chatran | i think this is becose pap2t is trying all modules |
| 17:38.51 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 17:40.37 | chatran | im upgrading the asterisk, on other asterisk have g729 with no license working |
| 17:40.44 | chatran | i dont know how |
| 17:40.47 | chatran | but have :) |
| 17:42.49 | chatran | Greenlight and this: 14:41:10:287 ERROR: No Open LogicalChannels - Failed FindLogicalChannelByChannelNo (incoming, ooh323c_3 |
| 17:42.51 | saint_ | [TK]D-Fender: so beside teh $450 sip polycom, do you have any other brand you would recommend, slightly cheaper ? |
| 17:42.52 | chatran | ? |
| 17:44.41 | jmetro | saint_ why not Dial the number for 10 sec, background the sound file, then dial again |
| 17:46.31 | saint_ | jmetro: that is what i do. my issue is that if i pickup the first 10 sec, chose 2 in the privacy (to send to voicemail), then hangup , the phone rings again right away |
| 17:47.50 | jmetro | hm.. show us the console output |
| 17:48.12 | saint_ | gimme a sec |
| 17:48.18 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
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| 17:53.30 | saint_ | jmetro: you don t see anything interesting in the console, you still want it ? You see the Dial, the Background, and the Dial again. |
| 17:53.55 | saint_ | which is what it is somehow supposed to do |
| 17:53.56 | jmetro | when you press 2 on the privacy menu, it doesnt do anything? |
| 17:54.19 | saint_ | instead of sendine me to voicemail, it plays the Background , then goes to the next action which is the Dial |
| 17:54.28 | saint_ | if I remove the background and dial , then it goes to voicemail |
| 17:54.49 | jmetro | and do you have same = > n,voicemail(ext) after the background and dial? |
| 17:55.20 | saint_ | hold on, let me past my extendion.conf |
| 17:55.44 | jmetro | put the console output in there too |
| 17:55.52 | *** join/#asterisk areski (~areski@80.174.128.6.dyn.user.ono.com) |
| 17:56.46 | kresp0 | chatran, please read this: http://www.catb.org/esr/faqs/smart-questions.html |
| 17:57.00 | saint_ | jmetro: http://pastebin.com/V7aya1jN |
| 17:57.12 | kresp0 | i guess that it will help you more than any other reading right now |
| 17:57.12 | saint_ | jmetro: and I think i just understood by looking again at the source what is going on |
| 17:57.28 | saint_ | I think when you press 2 to send to voicemail, you have a DIALSTATUS that comes back, and depending on that, you can do whatever you want. |
| 17:57.57 | saint_ | I am going to add a Message to print the dial status just before the background, and see if I can make it work this way |
| 17:58.18 | jmetro | right |
| 17:59.09 | kresp0 | chatran, about your codec warning: just remove g729 from sip.conf |
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| 18:09.41 | saint_ | jmetro: it won t work |
| 18:09.51 | saint_ | jmetro: because if i do not answer, I have a NOANSWE status |
| 18:09.54 | saint_ | NOANSWER |
| 18:10.07 | saint_ | and if I pick up and want to send to voicemail, I have the same status |
| 18:10.24 | saint_ | my goal was that if I did not pickup up, it would try again |
| 18:11.55 | jmetro | hm |
| 18:11.59 | jmetro | so it gives noanswer both ways. |
| 18:12.03 | saint_ | yeah |
| 18:12.14 | jmetro | and the console output doesnt say anything about when you pick up? |
| 18:12.22 | saint_ | nah |
| 18:12.35 | saint_ | no biggies |
| 18:12.44 | jmetro | write your own privacy then |
| 18:12.50 | saint_ | i had asterisk call my cell phone and i wanted to break it and try again so it did not go in my cell phone voicemail |
| 18:12.58 | *** join/#asterisk Vince-0 (~Vin@41.183.7.34) |
| 18:13.33 | jmetro | write a confirmation AA that the callee hears |
| 18:14.01 | saint_ | i ll work on that later. gotta go to real work. thanks for the hint. |
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| 18:49.42 | chatran | kresp0 pap2t have some protocols |
| 18:49.54 | chatran | what i do to work ? i put alaw there |
| 18:49.58 | chatran | its ok? |
| 18:57.25 | kresp0 | alaw is ok |
| 18:57.32 | chatran | [Mar 22 15:56:39] WARNING[21134]: channel.c:5128 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) |
| 18:57.33 | chatran | [Mar 22 15:56:39] WARNING[21134]: channel.c:5128 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) |
| 18:57.36 | chatran | kresp0 |
| 18:57.49 | chatran | i have another interface voip |
| 18:58.07 | chatran | and i did not put on sip.conf to allow |
| 18:58.21 | chatran | allow=ulaw |
| 18:58.29 | chatran | is just this on sip.conf |
| 18:58.36 | kresp0 | no |
| 18:59.52 | chatran | ? |
| 19:00.51 | eXcAliBuR | can I have 1 power supply at 120volt and the second one at 240 volt in the same server? |
| 19:02.12 | chatran | is your font bivolt ? |
| 19:02.24 | chatran | normaly have and switch to change the voltage |
| 19:02.49 | chatran | if have switch i think yes |
| 19:03.16 | kresp0 | wrong channel eXcAliBuR. Also: why? and: do you have 240v and 120v power supply on the same building? |
| 19:03.53 | kresp0 | If you do, I guess it could work. But I really doubt that |
| 19:04.00 | malcolmd | yeah, that'd be weird. it's also weird not to have auto-switching power supplies in a server. |
| 19:06.33 | eXcAliBuR | well the power supplys are auto sensing |
| 19:06.51 | eXcAliBuR | we need to change the UPS and want to power crit servers with an extension cord from a wall outlet |
| 19:07.25 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
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| 19:08.46 | *** join/#asterisk crazed1 (themrrober@unaffiliated/themrrobert) |
| 19:08.58 | crazed1 | has Originate been removed from AMI? Whats the way to do it now, i don't see originate the ast 11 command reference |
| 19:09.33 | [TK]D-Fender | crazed1, "channel originate <tab>" |
| 19:09.46 | [TK]D-Fender | crazed1, read the UPGRADE.TXT for the versions you've passed over |
| 19:12.11 | crazed1 | Thx, yea i was just realizing i could do action: command, then what you said |
| 19:12.27 | crazed1 | i will rtfm some more ;) |
| 19:13.37 | crazed1 | can you pass channe lvariables with the channel originate comand? |
| 19:14.36 | *** join/#asterisk kontinuity (~kontinuit@122.178.229.113) |
| 19:23.03 | jmetro | crazed1: i beleive the correct method is to originate to a testing dialplan that will pass the variables for you |
| 19:23.35 | [TK]D-Fender | crazed1, No |
| 19:23.44 | [TK]D-Fender | crazed1, AMI & call-files can |
| 19:28.27 | KNERD | Where are the docs on the CONFIGURE file so I can set the correct CPU flags for the correct type of CPU? I am just not seeing them. typing "./configure --help" only shows the options, not the CPU flags |
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| 20:05.57 | gundy | Anyone here using XO's ESIP near the Denver area? |
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| 20:15.42 | gundy | Never mind all. I got the confirmation I needed. They're down. |
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| 21:27.28 | gtTuna | what are people using for ATAs these days? |
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| 21:38.41 | gtTuna | is the Cisco SPA112 any good? |
| 21:42.37 | kresp0 | gtTuna: I'm using ATAs on cheap setups |
| 21:43.20 | gtTuna | yeah, we don't use many of them...but sometimes we have to throw one in for a low volume fax machine or something |
| 21:43.49 | gtTuna | and we've finally run out of PAP2T in our stock |
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| 21:51.12 | drmessano | I still have some PAP2 v1's I really need to ebay |
| 21:57.22 | Katty | drmessano: inspire me to brave the upsairs |
| 21:57.29 | Katty | drmessano: where the miniture human is. |
| 21:58.48 | drmessano | lol |
| 21:58.59 | drmessano | That sounds like something out of 50 shades |
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| 22:11.49 | KNERD | Since when is Asterisk using PJSIP? |
| 22:13.13 | [TK]D-Fender | SOON |
| 22:15.22 | jmetro | The SPA 112 is pretty good, we use it all the time |
| 22:15.32 | jmetro | terrible web interface like any cisco, but it works. |
| 22:15.51 | KNERD | soon? I am compiling the most current version and I saw some messages about PJSIP in it |
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| 22:44.27 | fabiobik | hello guys, its possible to making a huawei internet modem a gsm gateway? |
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| 23:10.15 | lorsungcu | gundy: ESIP still down for you? |
| 23:10.38 | gundy | lorsungcu: no, we're back up. |
| 23:10.52 | lorsungcu | alright |
| 23:11.00 | lorsungcu | was down in MN around 4PM as well. |
| 23:11.33 | gundy | lorsungcu: how long? |
| 23:11.53 | lorsungcu | not sure, everyone went home, and i dont keep track of it.. |
| 23:11.56 | gundy | This was about 1:00 to 2:00 pm MST. |
| 23:12.00 | lorsungcu | few hours |
| 23:12.05 | gundy | gosh. |
| 23:12.07 | lorsungcu | was down yesterday as well |
| 23:12.13 | lorsungcu | its XO |
| 23:12.21 | gundy | might be the same issue. |
| 23:12.25 | gundy | It's killing us. |
| 23:12.26 | lorsungcu | their SLA is actually for downtime |
| 23:12.34 | lorsungcu | guaranteed to not work 99% of the time |
| 23:12.45 | gundy | How often do you go down? |
| 23:13.09 | lorsungcu | often |
| 23:13.26 | lorsungcu | never gotten proactive alerts from them about it, either |
| 23:13.37 | lorsungcu | always need to call in/open tickets to get them to admit to anything |
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| 23:13.53 | gundy | ugh |
| 23:13.57 | lorsungcu | i'm the PBX vendor; everything is blamed on me until i can get confirmation from XO |
| 23:14.08 | gundy | is not feeling hopeful about the future. |
| 23:14.16 | lorsungcu | did you just sign up? |
| 23:14.40 | lorsungcu | we've got another customer mving to it this month. i am not looking forwrad to it. |
| 23:14.49 | gundy | No, we've been with them on and off. This is for a new client who is *very* sensitive to downtime. |
| 23:15.02 | lorsungcu | ah |
| 23:15.06 | lorsungcu | yeah bad choice, then |
| 23:15.16 | lorsungcu | we have less downtime with flowroute than we do with them |
| 23:15.17 | gundy | lorsungcu: what do you use with them? OpenSIPS, Asterisk, FreeSWITCH? |
| 23:15.23 | lorsungcu | Asterisk |
| 23:15.42 | gundy | Any interop issues? |
| 23:16.00 | lorsungcu | yeah, their caller ID shit is a mess |
| 23:16.19 | lorsungcu | and i do not interop with their support department well at all |
| 23:16.34 | gundy | hehe |
| 23:16.34 | lorsungcu | :p |
| 23:16.49 | gundy | That's the first laugh I've had all day, thanks! |
| 23:19.11 | lorsungcu | we've also got customers using E&M with XO |
| 23:19.19 | lorsungcu | theres almost no support at all for that |
| 23:19.33 | lorsungcu | i go with almost anyone else before using them |
| 23:19.44 | lorsungcu | especially for the money they're asking for |
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