| 00:00.02 | ChannelZ | They context for the polycom is probably not right |
| 00:00.09 | sirsquishy | something with the polycom for sure, moved my Softphone to use the Poly's Extension and it works perfectly. |
| 00:00.32 | ChannelZ | either that or the poly has its own internal dialplan and it's failing at the device level |
| 00:00.49 | sirsquishy | I wiped it before setting it up for this |
| 00:00.58 | sirsquishy | so if there is something im not seeing,. |
| 00:01.03 | ChannelZ | the console should say if the call is even hitting Asterisk, and if so why it's failing |
| 00:01.30 | sirsquishy | its hitting asterisk |
| 00:01.42 | sirsquishy | i can see it via SIP logs on asterisk |
| 00:01.56 | ChannelZ | oh yeah you said 'all circuits busy' sorry |
| 00:02.08 | ChannelZ | in any event turn verbose up to 2 or 3 and see what it's doing |
| 00:03.39 | sirsquishy | yea, dont have much else of a choice |
| 00:04.52 | ChannelZ | Well it's a pretty good choice... |
| 00:05.17 | sirsquishy | yea, just hate dealing with loggon on the server for device issues :-) |
| 00:06.09 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 00:07.35 | ChannelZ | ? |
| 00:09.07 | ChannelZ | as I said I think it's probably the wrong context, or the phone is adding some digits onto the dial... but that's the best I can do without seeing anything |
| 00:09.12 | ChannelZ | heading home. BBL. |
| 00:17.01 | [TK]D-Fender | We don't know it's a device issue |
| 00:17.15 | [TK]D-Fender | We don't see anything to actually comment on. |
| 00:17.24 | [TK]D-Fender | You should probably change that. |
| 00:17.40 | [TK]D-Fender | PASTEBIN is your friend |
| 00:27.49 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 00:27.49 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:37.12 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 00:47.58 | Shishir | Hello, I have a remote Asterisk IAX2 trunked with a local Asterisk. Remote asterisk can call local extensions succesfully. But, Local asterisk upload bandwidth and packet is 2 times higher than the download bandwidth |
| 00:49.02 | Shishir | Any advice will be helpful |
| 00:55.11 | hardwire | ? |
| 00:56.06 | Shishir | upload bandwidth consumption is double than the download bandwidth consumption in the local asterisk |
| 00:56.22 | Shishir | calls originated from the remote asterisk to local asterisk |
| 00:56.27 | Shishir | using g729 codec only |
| 01:12.44 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.167.62.231) |
| 01:26.58 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:42.27 | *** join/#asterisk dgeary2 (~david@42.241.74.64) |
| 01:45.06 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 01:47.53 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 01:57.12 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 02:00.30 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
| 02:07.44 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 02:08.16 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.38) |
| 02:17.24 | *** join/#asterisk lvlinux (~n1gg@c-50-147-64-9.hsd1.tn.comcast.net) |
| 02:30.59 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 02:33.27 | *** join/#asterisk radic (~radic@dslb-094-216-253-221.pools.arcor-ip.net) |
| 02:58.18 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 03:02.23 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 03:05.13 | SeRi | damn cc is getting hit hard with som "sophisticated" SDDoS |
| 03:08.53 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 03:08.56 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 03:21.50 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 03:34.15 | ChannelZ | who? |
| 03:36.32 | carrar | Call Centric |
| 03:36.54 | carrar | https://twitter.com/Callcentric |
| 03:37.27 | carrar | They've been under some DoS attack for a 20+ hours |
| 03:37.29 | SeRi | Sorry about that... was away playing with the new nas |
| 03:37.33 | SeRi | carrar: yeap |
| 03:37.43 | SeRi | even to their new redirected server |
| 03:39.16 | SeRi | just changed to srv.callcentric.com and that one worked |
| 03:40.49 | jpsharp | Pity there's no death penalty or at least a public flogging for the basement dwellers who do that kinda crap. |
| 03:41.44 | SeRi | +1 |
| 03:44.59 | jpsharp | Or to paraphrase another friend "Put them in a room alone with a angry rotweiler that has a taste for BALLS". |
| 03:45.30 | carrar | turns out it's some chic doing this |
| 03:45.37 | carrar | dog is out of luck |
| 03:46.23 | SeRi | chic? |
| 03:46.25 | SeRi | lol |
| 03:46.31 | jpsharp | Well, the make sure the dog hasn't been fixed. |
| 03:46.50 | SeRi | LOL |
| 04:08.18 | *** join/#asterisk Kraln (~kraln@69.169.90.239) |
| 04:15.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:29.30 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 04:38.03 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 04:40.05 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 04:42.42 | *** join/#asterisk evilman_home (kvirc@2.92.147.80) |
| 05:15.56 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 05:17.12 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.197) |
| 05:48.09 | *** join/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 05:52.42 | jaxon007_ | Guys, If I call from my mobile to another mobile and if another mobile Disconnect my call then one message get Played like ""Telco name" subscriber you are trying to reach is busy at the moment, Pls stay on line or call later". But if tried to call another mobile from my asterisk with same scenario. I cant hear Busy message and after 5 sec Call gets hung Up. How to solve this? |
| 06:03.15 | *** join/#asterisk jsjc (~Adium@221.Red-83-46-237.dynamicIP.rima-tde.net) |
| 06:05.12 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:23.33 | *** join/#asterisk MorkBork (~mark@unaffiliated/morkbork) |
| 06:24.44 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 06:58.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 07:48.22 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 07:57.52 | jaxon007_ | Guys, If I call from my mobile to another mobile and if another mobile Disconnect my call then one message get Played like ""Telco name" subscriber you are trying to reach is busy at the moment, Pls stay on line or call later". But if tried to call another mobile from my asterisk with same scenario. I cant hear Busy message and after 5 sec Call gets hung Up. How to solve this? |
| 08:21.12 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 08:50.53 | *** join/#asterisk Neptu (~Neptu@c213-89-2-159.bredband.comhem.se) |
| 09:02.43 | *** join/#asterisk echo083 (~adam@80.125.173.151) |
| 09:04.21 | echo083 | hello, i configured asterisk through freepbx it is working great in my local network but when i connect from the internet i receive the error message 401 unauthorized i my sip softphone |
| 09:04.54 | echo083 | do you know if there is a particular action to make it work out of my local network ? |
| 09:07.15 | echo083 | noone ? |
| 09:37.22 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 09:37.52 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 09:49.11 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 09:54.34 | WIMPy | jaxon007_: Don't use SIP. It's not laid out to do so. |
| 09:55.42 | WIMPy | (and neither is Asterisk) |
| 10:43.03 | *** join/#asterisk youjelly (~bwahahaha@39.47.250.217) |
| 10:44.06 | jaxon007_ | WIMPy: can i use IAX? |
| 10:49.37 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:50.52 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 10:51.22 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 10:51.52 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 11:07.46 | trumee | does linksys new ATAs SPA122 have working support of srtp with *? |
| 11:28.58 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul) |
| 11:29.04 | bipul | ##devil |
| 11:38.09 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 12:02.42 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 12:04.28 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 12:07.57 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:09.37 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:14.56 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 12:30.51 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 12:33.20 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 12:35.53 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 12:37.34 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 12:45.37 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 12:47.39 | SeRi | CC still having issues... |
| 12:51.49 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:54.51 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 12:55.49 | *** join/#asterisk echo083 (~adam@gateway/tor-sasl/echo083) |
| 12:55.55 | echo083 | hello |
| 12:56.08 | echo083 | what is the freepbx dedicated channel please ? |
| 13:04.20 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 13:09.59 | LittleFool | Hello, i want to give a customer a extension and 10 numbers. Is it somehow possible to restrict the CID of this extension to only allow those 10 numbers? |
| 13:11.16 | kaldemar | echo083: #freepbx |
| 13:11.39 | echo083 | yes found it ... just the channel was long to load sorry |
| 13:11.50 | echo083 | thanks anyway |
| 13:14.14 | *** join/#asterisk n3hxs (~n3hxs-5@TOROON63-3096776761.sdsl.bell.ca) |
| 13:16.21 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 13:16.21 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:25.06 | *** part/#asterisk jaxon007_ (~jay@123.252.144.92) |
| 13:37.39 | *** join/#asterisk Shishir (~Shishir@114.130.38.246) |
| 13:39.45 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:40.06 | Shishir | Hello, I have a remote Asterisk IAX2 trunked with a Local Asterisk. Local asterisk registers itself with the Remote asterisk in trunk mode both ends. Remote asterisk can call local extensions succesfully. However, it is notices that Local asterisk is consuming double upload bandwidth than the download bandwidth. Local asterisk is behind NAT, however Remote asterisk is not. |
| 13:41.03 | Shishir | Looking for some advice, why the local asterisk may consume double the amount of upload bandwidth while download bandwidth is OK as per the g729 standard |
| 13:45.49 | *** join/#asterisk wonderworld (~ww@dsdf-4db528d7.pool.mediaWays.net) |
| 14:05.16 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 14:17.07 | WIMPy | LittleFool: What channel? But the answer is most probably that you have to check in your dialplan. |
| 14:17.42 | LittleFool | WIMPy: all SIP |
| 14:18.31 | WIMPy | Like I said then. |
| 14:18.37 | LittleFool | ok |
| 14:27.50 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 14:29.58 | *** join/#asterisk chasingSol (~ao@41.232.124.124) |
| 14:40.45 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 14:42.50 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 14:42.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:49.22 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 14:49.32 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 14:56.49 | *** join/#asterisk robl^ (~robl^@pdpc/supporter/active/robl) |
| 15:03.43 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-77-106-250.mia.bellsouth.net) |
| 15:05.02 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 15:06.31 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 15:08.03 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 15:13.44 | echo083 | WIMPy, my soft phone need a delay to enable authentication when i start it it says "unauthorized" i wait few seconds and reconnect and all is fine |
| 15:13.57 | echo083 | WIMPy, any idea of a asterisk parameter to change or try ? |
| 15:18.45 | *** join/#asterisk puzzled (~patrick@5ED2AFC0.cm-7-3c.dynamic.ziggo.nl) |
| 15:22.37 | echo083 | sorry i have to leave have a good day |
| 15:30.52 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 15:34.08 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 15:35.26 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:36.41 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 15:37.10 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 15:37.56 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-88.red.bezeqint.net) |
| 15:42.49 | *** join/#asterisk mobile_gordita (~Robert@ip70-188-50-186.rn.hr.cox.net) |
| 15:43.27 | *** join/#asterisk Greek-Boy (~Greek-Boy@197.152.130.133) |
| 15:43.37 | Greek-Boy | hello. |
| 15:43.47 | Greek-Boy | What is the best way to deal with a client behind NAT? |
| 15:44.19 | Greek-Boy | he has set his firewall to port forward back to his internal IP address where he is running his soft-phone but there is still one-way audio. |
| 16:05.57 | [TK]D-Fender | The same as always. |
| 16:06.07 | [TK]D-Fender | *'s settings for this havent really changed. |
| 16:06.18 | [TK]D-Fender | So show us what you've set up for your server side and thier peer |
| 16:06.20 | [TK]D-Fender | ~pb |
| 16:06.21 | 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. |
| 16:08.40 | *** join/#asterisk Nemus (~Nemus@67.161.251.38) |
| 16:13.52 | Shishir | [TK]D-Fender: hi |
| 16:23.37 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 16:45.14 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 16:46.15 | *** join/#asterisk drupin (~drupin@unaffiliated/drupin) |
| 16:46.43 | *** part/#asterisk drupin (~drupin@unaffiliated/drupin) |
| 16:53.44 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 16:56.37 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:57.30 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:01.35 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 17:07.11 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 17:12.59 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 17:42.41 | *** join/#asterisk corretico (~luis@190.211.93.11) |
| 17:46.19 | *** join/#asterisk ibm2 (~ibm@unaffiliated/ibm2) |
| 17:46.58 | ibm2 | afternoon anyone got a suggestion for a web front end for asterisk that is free? |
| 17:48.04 | WIMPy | There are quite a few, but we don't support frontends here. |
| 17:49.08 | ibm2 | no worries, not looking for support for them, just suggestions |
| 17:49.42 | WIMPy | I haven't fond a comparison or even a complete list so far. |
| 17:49.55 | ibm2 | gotcha |
| 17:50.21 | *** part/#asterisk ibm2 (~ibm@unaffiliated/ibm2) |
| 18:05.45 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 18:18.16 | *** join/#asterisk mobile_gordita (~Robert@66-87-94-253.pools.spcsdns.net) |
| 18:28.13 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 19:15.30 | *** join/#asterisk mokmeister (~mokmeiste@109.78.118.61) |
| 19:34.31 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:42.56 | *** join/#asterisk nightrid3r (~kvirc@d54C6B801.access.telenet.be) |
| 19:43.52 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 19:45.47 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 19:51.52 | *** join/#asterisk echo083 (~adam@gateway/tor-sasl/echo083) |
| 19:51.55 | echo083 | hello |
| 19:56.02 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 19:56.02 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 19:56.02 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 19:56.02 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 19:56.02 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 19:56.18 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:56.18 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:56.18 | *** join/#asterisk hurdman (~ygcheny@bender.r0b0t.fr) |
| 19:56.18 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 19:56.18 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 19:56.18 | *** join/#asterisk DigiDaz (~quassel@ks305112.kimsufi.com) |
| 19:57.09 | *** join/#asterisk salviadud (~rkdkjd@187-163-110-206.static.axtel.net) |
| 19:57.25 | salviadud | Does anyone know how to check for polarity problems |
| 19:57.35 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 19:57.39 | salviadud | my box cannot make a simple switch after receiving a call from DAHDI |
| 19:57.44 | salviadud | on a particular line |
| 19:57.49 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:58.33 | *** join/#asterisk mokmeister (~mokmeiste@109.78.118.61) |
| 20:01.08 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 20:01.08 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 20:01.08 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 20:01.08 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.174) |
| 20:01.08 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 20:02.34 | *** join/#asterisk angryuser_laptop (~angryuser@2a02-8422-1230-bb00-8868-3b94-ee42-ba4b.rev.sfr.net) |
| 20:04.12 | *** part/#asterisk sirsquishy (~sirsquish@unaffiliated/sirsquishy) |
| 20:05.28 | WIMPy | salviadud: Is there a link or are you asking to unrelated questions there? |
| 20:06.28 | WIMPy | s/to/two/ |
| 20:06.44 | *** join/#asterisk odenkos (~odenkos@unaffiliated/odenkos) |
| 20:08.26 | salviadud | let me explain |
| 20:08.57 | salviadud | The phone company is giving several pstn lines for incoming calls |
| 20:09.04 | salviadud | 4 lines |
| 20:09.26 | salviadud | 3 out of 4 work on a 8 port card |
| 20:10.08 | salviadud | Some expert guy says that the line that doesn't work has a polarity issue |
| 20:10.30 | salviadud | I want to know how to change the polarity on my card |
| 20:10.44 | salviadud | if there is a quick way to do it |
| 20:10.50 | salviadud | maybe modify a config file... |
| 20:11.29 | *** join/#asterisk af_ (~getsmart@78.134.61.252) |
| 20:12.27 | jpsharp | Flip the polarity by making a new cable that flips the middle pair. |
| 20:13.26 | WIMPy | Ok, but tha simple switch thing is for the opposite direction. |
| 20:15.45 | salviadud | if my rj 11 has only 2 copper wires, would switching burn my card? |
| 20:16.11 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 20:16.34 | jpsharp | no, not unless you've got serious wiring issues on your line already. |
| 20:16.48 | WIMPy | It can ony affect things like disconnect supervisuion. |
| 20:17.30 | WIMPy | Just to make calls, polarity doesn't matter. |
| 20:17.33 | jpsharp | Yes. A line will "work", as in ring and answer calls, even if the pair is flipped. |
| 20:18.20 | salviadud | I'm gonna flip the pair, but I won't be able to do so until monday |
| 20:18.33 | salviadud | And I'll be back with a WIN, thanx guys |
| 20:20.26 | *** join/#asterisk aross42 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 20:23.04 | nny | anyone have a suggestion for playing activating a feature code from AMI on a specific channel? |
| 20:38.00 | nny | I figure I need to do get BRIDGEPEER from the channel I want to send it from and then REDIRECT the BRIDGEPEER to my feature's context but not sure how from AMI |
| 20:38.15 | nny | I don't think dbget would grab it |
| 20:38.49 | nny | ahh |
| 20:38.51 | nny | GetVar |
| 20:38.57 | nny | think I can work with this |
| 20:40.48 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 20:46.17 | *** join/#asterisk mobile_gordita (~Robert@66-87-95-173.pools.spcsdns.net) |
| 20:55.05 | *** join/#asterisk bytemaster (~qweq@host81-150-217-167.in-addr.btopenworld.com) |
| 21:09.41 | *** join/#asterisk vlad_starkov (~vlad_star@77.41.88.210) |
| 21:12.37 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 21:14.43 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 21:16.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:17.23 | hesco1 | is there some guide to understanding sip debug output someone might point me to? |
| 21:17.48 | WIMPy | The SIP RFC. |
| 21:18.15 | hesco1 | thanks, looking that up now |
| 21:20.08 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 21:21.37 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 21:21.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 21:21.56 | hesco1 | wow, nice comprehensive answer. I guess a short primer is more what I'm looking for, heading to google I guess. |
| 21:30.28 | volga629 | Hello WIMPy, Error message where DTMF issue on snom 370 remember ? |
| 21:30.38 | volga629 | access to vm |
| 21:31.25 | volga629 | This need set on snom 370 DTMF via SIP INFO: on, this is reported as bug and will be fixed in next releases |
| 21:32.24 | volga629 | right now asterisk understand all DTMF values from snom phone |
| 21:33.21 | volga629 | and might be it related also to codec g729 |
| 21:33.59 | jpsharp | If you don't have something like SIP INFO or RFC2889 working to transport DTMF, g729 is going to distort the heck of the tones. |
| 21:35.36 | volga629 | true |
| 21:36.12 | volga629 | I am getting this message is possible disable vm for this user ? app_voicemail.c:2414 inboxcount2: Couldn't find mailbox 2006 in context device |
| 21:37.58 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:39.40 | [TK]D-Fender | volga629: What does your PEER say? |
| 21:41.13 | volga629 | sip_additional.conf:mailbox=2006@device |
| 21:43.10 | WIMPy | volga629: A bug in the current FW or where? |
| 21:43.38 | volga629 | it was reported to snom |
| 21:43.45 | [TK]D-Fender | volga629: Well YOU pointed it there. Stop doing things you don't want to do |
| 21:45.46 | volga629 | yes, exactly this mobile extension want disable vm completely for this user. |
| 21:46.15 | [TK]D-Fender | facepalm |
| 21:48.36 | *** part/#asterisk nny (~Scott@174.107.223.14) |
| 22:00.53 | volga629 | no gui setup it working yes just comment out mailbox reload all gui and in test machine freepbx it override when apply settings |
| 22:09.21 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 22:25.12 | *** join/#asterisk dandate2 (~dan@180.190.233.70) |
| 22:27.05 | *** join/#asterisk BlackBishop (~dexter@ipv6.d3xt3r01.tk) |
| 22:37.41 | *** join/#asterisk k610 (4e1df1ba@gateway/web/freenode/ip.78.29.241.186) |
| 22:38.43 | k610 | how can i register my sip clients to sip.mydomain.tld instead of using my server's ip |
| 22:43.29 | *** join/#asterisk j4m3s_ (~j4m3s_@c-71-196-88-32.hsd1.fl.comcast.net) |
| 22:45.47 | [TK]D-Fender | k610: get a domain and point it to your server |
| 22:49.38 | Shishir | Hello, I have a remote Asterisk IAX2 trunked with a Local Asterisk. Local asterisk registers itself with the Remote asterisk in trunk mode both ends. Remote asterisk can call local extensions succesfully. However, it is notices that Local asterisk is consuming double upload bandwidth than the download bandwidth. Local asterisk is behind NAT, however Remote asterisk is not. |
| 22:49.49 | Shishir | looking for some advice |
| 22:50.40 | WIMPy | Shishir: Have you looked at te traffic? |
| 22:51.02 | WIMPy | Have you configured differen packetization? |
| 22:51.26 | Shishir | WIMPy: no, haven't done anything like that |
| 22:52.24 | Shishir | here is my configuration |
| 22:52.25 | Shishir | http://pastebin.com/TCmBYHQh |
| 22:52.37 | [TK]D-Fender | Shishir: You keep asing the same thing over and over. SHOW US this bandwidth variance and proof as to where it's coming from. |
| 22:53.08 | BlackBishop | wdoekes: hi. |
| 22:53.14 | BlackBishop | ping* |
| 22:53.22 | Shishir | [TK]D-Fender: i keep asking at different times because most time this channel no body talks |
| 22:53.32 | Shishir | [TK]D-Fender: sorry if you are disturbed |
| 22:53.45 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-76-42.pppoe.mtu-net.ru) |
| 22:54.05 | [TK]D-Fender | Shishir: And you aren't showing anything meaningful. Configs don't prove what's happening. You are not thinking on this at all. You want to pinpoint a problem show us actual proof. |
| 22:54.10 | Shishir | [TK]D-Fender: could you please point me how can I show you the bandwidth variance? showing channels again? |
| 22:54.35 | [TK]D-Fender | Shishir: You are the claiming there is a variance. YOU have to prove it to US. |
| 22:54.49 | WIMPy | What about a bit of tcpdump/wireshark/whatever? |
| 22:55.08 | [TK]D-Fender | WIMPy: If he can't prove it, how could he even claim it? |
| 22:55.21 | Shishir | I can do tcpdump |
| 22:55.45 | Shishir | [TK]D-Fender: i m monitoring the bandwidth of the local asterisk server from my router. |
| 22:56.08 | [TK]D-Fender | Shishir: I'm not seeing real proof anywhere.... |
| 22:57.15 | Shishir | [TK]D-Fender: what type of real proof u want |
| 22:58.00 | [TK]D-Fender | Shishir: You are claiming * is using a disproportionate amount of bandwidth on upstream vs downstream. PROVE IT. |
| 23:01.01 | *** join/#asterisk unicron (~unicron@the.wrong.domain.name) |
| 23:03.23 | unicron | anyone have a favorite toll free termination provider? |
| 23:05.04 | k610 | [TK]D-Fender: should i set anything special up i in asterisk, I taught setting realm=mydomain.tld in sip.conf would suffice |
| 23:05.31 | [TK]D-Fender | k610: Not necessary |
| 23:05.47 | [TK]D-Fender | k610: Want a domain, get a domain. |
| 23:06.50 | WIMPy | setting realm has nothing to do with it. |
| 23:09.01 | k610 | I guess namecheap.com has some special settings aside of forwaring www and @ to an ip |
| 23:11.16 | Shishir | i have a pcap file from tcpdump |
| 23:18.04 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:19.22 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 23:22.09 | [TK]D-Fender | k610I guess namecheap.com has some special settings aside of forwaring www and @ to an ip <- just saying this doesn't say much for your understanding of DNS.... |
| 23:23.34 | Shishir | http://www.mediafire.com/?21ehpj9959e05sb |
| 23:23.43 | Shishir | here is the output from the tcpdump |
| 23:23.46 | Shishir | hope it helps |
| 23:25.00 | *** join/#asterisk j4m3s (~j4m3s@c-71-196-88-32.hsd1.fl.comcast.net) |
| 23:25.14 | WIMPy | Have you looked at it yourself? |
| 23:25.22 | [TK]D-Fender | Shishir: I'm not seeing you show us prrof of bandwidth consumption. You seem to have enough to make claim and we aren't seeing it. |
| 23:25.55 | j4m3s | do the digium phones support dns srv? |
| 23:26.02 | [TK]D-Fender | WIMPy: Of course he hasn't. He is making a claim and can't show us the basis of it. |
| 23:26.13 | [TK]D-Fender | WIMPy: No clue whatsoever. |
| 23:26.45 | Shishir | [TK]D-Fender: I am not used to tcpdump so I didn't see it. I am used with monitoring from my Mikrotik router. |
| 23:26.58 | Shishir | I can show a screenshot from the router |
| 23:28.12 | [TK]D-Fender | Shishir: Iasked you for proof of consumption about 20 times and only NOW you think about actually showing it? |
| 23:29.46 | Shishir | I didn't understood u wanted a screenshot |
| 23:31.50 | Shishir | http://imageshack.us/photo/my-images/141/iax.jpg/ |
| 23:33.12 | WIMPy | That does not look like it's related to anything VOIP. |
| 23:33.33 | Shishir | this screenshot was taken while there was 4 concurrent g729 calls. |
| 23:33.54 | [TK]D-Fender | Shishir: HTTP. WEB. |
| 23:33.56 | Shishir | yah thats because i used 8080 ports for iax in both asterisk |
| 23:34.32 | Shishir | and even if i do it with some other ports (perhaps 4569) it will be the same, i tested it..so port is not a issue |
| 23:35.30 | [TK]D-Fender | Shishir: Show us the calls. |
| 23:35.41 | WIMPy | Looks like only one side is using trunk mode. |
| 23:36.19 | WIMPy | What version(s) are you using? |
| 23:36.54 | Shishir | Asterisk 1.8.16.0 |
| 23:37.04 | Shishir | WIMPy: both sides have trunk=yes |
| 23:37.22 | WIMPy | Do you have dahdi loaded? |
| 23:38.05 | WIMPy | That verison might still need it. |
| 23:39.02 | Shishir | I have dahdi loaded in the local asterisk |
| 23:39.04 | ChannelZ | IAX trunking used DAHDI? |
| 23:39.11 | [TK]D-Fender | yes |
| 23:39.15 | [TK]D-Fender | Always has. |
| 23:39.22 | ChannelZ | Huh. Didn't know that. |
| 23:39.33 | WIMPy | It was required for timing. |
| 23:39.38 | [TK]D-Fender | It's only documented everywhere since forever.... |
| 23:39.59 | ChannelZ | Makes sense. I've never used trunking, but didn't know that |
| 23:40.39 | WIMPy | I don't know up to what version. |
| 23:43.04 | Shishir | WIMPy: so if there's no dahdi... is it expected for consuming higher upload bandwidth? |
| 23:43.24 | WIMPy | No dahdi, no trunk mode. |
| 23:43.33 | WIMPy | trunk=yes will just be ignored. |
| 23:44.00 | Shishir | WIMPy: I think my installation doesn't have a dahdi |
| 23:44.05 | Shishir | WIMPy: I checked |
| 23:44.48 | WIMPy | Install dahdi or upgrade Asterisk. |
| 23:44.58 | Shishir | WIMPy: upgrade to 10? |
| 23:45.38 | WIMPy | Probably. I don't know when it changed. |
| 23:46.12 | Shishir | ok I'll install dahdi. |
| 23:47.18 | Shishir | WIMPy: dahdi needs to be installed on both sides right? |
| 23:47.33 | WIMPy | yes |
| 23:51.46 | ChannelZ | hmm * is not paying attention to my ~/.asterisk.makeopts |
