| 00:00.19 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 00:00.57 | ChannelZ | grinder13: these don't make complete sense; From Server A you are registering as 1001 on 192.168.1.2 - which server is that IP? Because I don't see 1001 on Server B |
| 00:03.42 | ChannelZ | also it would help to see console output from one of the failed calls |
| 00:03.43 | grinder13 | yes, need to clarify that. Server A is 192.168.1.2 and Server B is 192.168.2.2 |
| 00:04.06 | ChannelZ | ok so you are registering to yourself, but at this point that's unrelated |
| 00:05.07 | grinder13 | so my confs look ok? |
| 00:06.17 | ChannelZ | well, I think, without knowing what you're dialing and from where |
| 00:08.14 | grinder13 | for example 1000@ServerA calls 2000@Server B (and generally 1XXX@ServerA<--->2XXX@ServerB) |
| 00:10.26 | ChannelZ | re: show verbose console output because something doesn't add up |
| 00:11.17 | ChannelZ | but I'm at work and need to leave so I'll be MIA for a bit, someone else might be pick up |
| 00:12.26 | grinder13 | well ok, thanx anyway. I 'll post again tomorrow, because I have to live as well and close the lab |
| 00:12.40 | grinder13 | have to leave :p |
| 00:14.04 | grinder13 | just a question: What about the pedantic option. I am not sure I understand what that is about by looking at the sip.conf it seems that it solved the problem for some people |
| 00:14.27 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
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| 00:22.01 | kuku | I'm having an issue with two SIP calls getting glued together, even though I have canreinvite=no. http://pastebin.ca/1897574 |
| 00:22.30 | *** join/#asterisk imox1234 (~imox1234@p4FC5C4B3.dip0.t-ipconnect.de) |
| 00:23.32 | russellb | what is the problem? |
| 00:24.08 | carrar | People keep answer the phone!!! |
| 00:24.12 | carrar | answering |
| 00:24.19 | russellb | jerks |
| 00:24.25 | carrar | yeah |
| 00:24.56 | kuku | russellb: The problem is that when the calls get joined, there is no more audio |
| 00:25.23 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 00:26.32 | sawgood | If the user's on an Asterisk box (using SIP extensions) claim calls keep dropping after 5 minutes or less (and nothing like this shows in the switch logs), what would be a good Asterisk CLI process to try and capture this? |
| 00:26.42 | sawgood | Or, should Wireshark be used instead? |
| 00:29.18 | carrar | You have wireshark installed on a asterisk box? |
| 00:29.56 | carrar | or using tcpdump on the asterisk box and using wireshark to view that capture? |
| 00:30.06 | joobie | kuku, sounds like you have a problem with RTP |
| 00:30.10 | carrar | (on another machine) :) |
| 00:30.22 | carrar | ~sipnat |
| 00:30.23 | infobot | rumour has it, sipnat is 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 |
| 00:30.44 | joobie | kuku, can you make other types of calls on your setup? Is it just this type of call playing up? |
| 00:30.44 | sawgood | I have both Wireshark (GUI and non GUI) and tcpdump installed on the Asterisk box |
| 00:31.31 | kuku | joobie: no issues dialing in and talking to a sip device, or other way around. Only when I dial into asterisk, and have asterisk forward the call (righ away, in the dialplan ) to an external number |
| 00:32.15 | *** part/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 00:35.46 | p3nguin_ | People keep answering the phone... THAT IS TERRIBLE! :D |
| 00:38.09 | kuku | joobie: I will try to adjust rtp settings, but I'm worried that it reinvites the calls. |
| 00:40.47 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
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| 00:46.07 | joobie | sawgood, sip debug |
| 00:47.05 | joobie | kuku, sip debug to the rescue |
| 00:47.10 | joobie | kuku, you can see if it's sending the invite |
| 00:50.29 | kuku | ok |
| 00:50.58 | emora | How does Asterisk select the appropriate codec to use for each call? |
| 00:52.00 | joobie | emora, you can specify a preference in the config |
| 00:52.03 | Tim_Toady | it doesnt, its just relies on the allowed codecs you have specified |
| 00:54.00 | joobie | which is how asterisk can select the appropriate codec |
| 00:54.21 | emora | joobie: I know I can specify a format. But the case is that I can specify several. In reality, if I dont Aasterisk will be forced to transcode. |
| 00:55.06 | emora | If I allow G722 and alaw, what criteria does it use to select the appropriate codec |
| 00:55.38 | kuku | joobie: there is an Invite |
| 00:55.39 | emora | Actually, I will have g722, g729, alaw and ulaw |
| 00:56.14 | kuku | joobie: I do have insecure=port,invite |
| 00:56.48 | joobie | emora, in sip.conf, allow= is in order of preference |
| 00:57.27 | joobie | just do disallow=all in your global and then allow=codec1,codec2, etc in your user section |
| 00:58.13 | emora | Precedence is established exclusively by the order of allow= directives in sip.conf ? |
| 00:58.48 | joobie | kuku, my knowledge for your problem is limited - but if you are seeing the calls bridge from the control channel |
| 00:58.56 | joobie | then it will just be rtp that is falling over |
| 00:59.07 | joobie | in which case look at sip debug and see what rtp ports are being used |
| 00:59.18 | joobie | and make sure there's no firewall / routing issues in the way |
| 00:59.53 | joobie | emora, nod, so long as you disallow=all in global |
| 00:59.56 | emora | I dissallow all then allow the codecs I want my endpoint to support. But what happens when my endpoint supports wideband (g722) and a call comes in with alaw? My endpoint supports alaw too. Will asterisk opt for alaw or will it transcode? |
| 01:01.36 | joobie | emora, im not sure.. but i'd guess that if you specify allow=g722,g729,alaw and a call came in only supporting alaw.. it would use alaw |
| 01:02.03 | joobie | i dont think it would transcode - otherwise it would negate the disallow=all allow=specific_codecs_only options |
| 01:03.41 | emora | My question is because I'm looking to understand what criteria Asterisk uses to select a codec. On the one hand, I want the best call quality possible when wideband is available but I want to avoid transcoding when a narrowband call comes in. |
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| 01:04.09 | joobie | dood |
| 01:04.12 | joobie | disallow=all |
| 01:04.18 | joobie | means dont accept any codec |
| 01:04.31 | joobie | allow=specific_codec, means allow only these codecs, in the order specified |
| 01:05.07 | joobie | so if you want to support wideband when it's avaialble as your "first priority", throw it in as the first allow option |
| 01:05.14 | joobie | then put your other codecs after that |
| 01:05.23 | emora | joobie: dont get me wrong, but you are stating the obvious |
| 01:05.23 | joobie | then bob's your uncle |
| 01:05.52 | joobie | emora, no shit.. i don't understand what you mean then |
| 01:06.26 | joobie | if you have an endpoint connected to asterisk with ulaw |
| 01:06.40 | joobie | and a call comes in connected to asterisk via alaw |
| 01:06.49 | joobie | of course it will transcode? |
| 01:06.52 | emora | what will asterisk do when an alaw call comes in on a channel that supports alaw and then bridges that call to bridge it to a endpoint that supports g729 and alaw? |
| 01:07.04 | joobie | ahh |
| 01:07.11 | joobie | it will transcode |
| 01:07.23 | joobie | im guessing |
| 01:07.28 | joobie | just test it bro |
| 01:07.34 | joobie | and feed back the response |
| 01:08.29 | kuku | joobie: but I don't want to reInvite... I want to keep the call |
| 01:09.16 | kuku | joobie: plus the rtp port is just 5060: Via: SIP/2.0/UDP 192.168.1.11:5060;branch=z9hG4bK58e3d50e;rport=5060;received=1 |
| 01:09.53 | joobie | kuku, send us your dialplan for that portion so i can see what ur trying to do |
| 01:13.16 | sawgood | Why is it so 'hard' to find an ITSP offering DID numbers in Alaska? |
| 01:13.39 | kuku | joobie: the dialplan ? Call comes in, and its a one liner, dial(sip/XXXXXX@voicepulse-primary) |
| 01:16.07 | kuku | joobie: I also made the astersik box DMZ |
| 01:16.32 | p3nguin_ | That's part of the problem, I'm sure. |
| 01:17.02 | p3nguin_ | Why do all these people think they need to put their asterisk systems in a DMZ when they don't even understand what a DMZ is? |
| 01:21.30 | joobie | werd |
| 01:21.40 | joobie | kuku, what are the firewall rules on your DMZ |
| 01:22.25 | kuku | its all open |
| 01:25.06 | kuku | I made some changes |
| 01:25.13 | carrar | sawgood, cause there's only like a couple providers in AK and they make it pricy |
| 01:25.24 | kuku | Now I get Auto fallthrough, channel 'SIP/Epd66Nvn97-b7909ad0' status is 'NOANSWER' |
| 01:25.32 | carrar | ACS, GCI & AT&T |
| 01:25.43 | carrar | maybe more now |
| 01:26.05 | carrar | ACS being the incumbent |
| 01:26.14 | joobie | how is it a dmz if its all open? |
| 01:26.23 | carrar | haha joobie |
| 01:26.30 | p3nguin_ | Why it is so hard to forward port 5060 for SIP and a range (often 10000-20000) for RTP, rather than inappropriately assigning a DMZ. |
| 01:26.32 | carrar | I wasn't gonna say anything on that |
| 01:26.55 | kuku | well, this dmz assigns 1-to-1 |
| 01:27.28 | kuku | p3nguin_: I did that. In case the comcast router was doing it inproperly, and added dmz. |
| 01:27.56 | kuku | p3nguin_: the comcast router allows only 5000 ports, so I did from 10000-14000, I also updated rtp.conf to reflect those changes. |
| 01:28.11 | kuku | I also stopped/started astersik ( not reload ) |
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| 01:45.04 | kuku | ok. Somehow it started working... |
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| 02:09.11 | p3nguin_ | majik |
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| 04:07.41 | ChannelZ | Clap your hands |
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| 04:30.06 | joako | In the dialplan how can I get a variable number of digits input? |
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| 05:37.08 | DogBoy | joako, http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-5 |
| 05:37.18 | DogBoy | "dialplan basics" |
| 05:37.48 | joako | I ended up doing: exten => getdigit,1,Background(silence/5); exten => 0,1,Set(OURNUMBER=${OURNUMBER}${EXTEN}); exten => 0,2,Goto(getdigit,1); exten => #,1,Set(CALLERID(number)=${OURNUMBER}); exten => #,2,Goto(voicemenu-custom-10,s,1) |
| 05:38.04 | joako | Need to pass callerid between ivrs on pstn link that won't pass correct callerid |
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| 07:18.23 | raden | I have a PAP2T connecting to asterisk over the internet !!! not LAN !!! I can call from the PAP2t everything is fine audio both ways if i call the pap2t from any phone I can hear person on pap2t but they cannot here me |
| 07:20.20 | raden | oh jesus now its working WTF |
| 07:21.32 | raden | It works yea |
| 07:21.36 | raden | just dont know why |
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| 07:48.24 | mattwj2002 | hi all |
| 07:57.02 | *** join/#asterisk aceio (c2cbd7fe@gateway/web/freenode/ip.194.203.215.254) |
| 07:57.34 | aceio | hello |
| 07:58.28 | aceio | iam trying too prevent international call how too i create it |
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| 08:08.34 | ChannelZ | aceio: change your dialplan so it's not possible to dial whatever international codes |
| 08:14.40 | aceio | ChannelZ: can you show me an example plz |
| 08:17.25 | aceio | ChannelZ: this how i allow outbound calls exten => _X.,1,Dial(DAHDI/g1/${EXTEN},${DIAL_TOUT},HrS(600)TWK) |
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| 08:31.03 | FILLVAIO3 | Hi guys |
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| 08:35.35 | E-bola | If i can play audio out through my soundcard through both ALSA and OSS with various players, why cant i get asterisk to use either of them through the console? |
| 08:35.54 | E-bola | i have both modules compiles, but neither of them are able to produce any audio output over my speakers :( |
| 08:36.39 | E-bola | this is newest asterisk 1.6.2.9 |
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| 09:02.14 | ChannelZ | aceio: you basically have a complete wildcard, people can dial anything they want. Make the pattern more specific. I don't know what normal local/long distance numbers look like to you compared to international numbers |
| 09:02.22 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 09:04.27 | ChannelZ | Here in the US for instance we generall need an 'exit code' of 011 to dial another country... so we don't allow 0 to be the first digit since no numbers in the US ever start with 0 |
| 09:06.18 | aceio | ChannelZ: okay gotcha |
| 09:06.20 | ChannelZ | I think from the UK international always starts with 00 so you'd construct your pattern accordingly |
| 09:06.54 | aceio | ChannelZ: that is correct |
| 09:06.55 | ChannelZ | anyway you can use various pattern characters for the extension to match only certain digits |
| 09:07.07 | *** join/#asterisk emora (~emora@213.37.33.105.dyn.user.ono.com) |
| 09:08.07 | aceio | so once i get the pattern right, how to i drop the call |
| 09:08.21 | ChannelZ | Well the idea is to match the stuff you DO want to allow, not the stuff you don't. |
| 09:08.49 | ChannelZ | But otherwise you can just 'Hangup' |
| 09:08.55 | ChannelZ | Or maybe play back an error message first |
| 09:08.58 | ChannelZ | but anyway |
| 09:11.45 | aceio | okay thank you.. |
| 09:12.02 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 09:12.55 | ChannelZ | the Z character might help you, which means only digits 1-9, and N which is 2-9 |
| 09:14.00 | ChannelZ | so _0NX. would be 'a 0 followed by 2-9 followed by any numbers of 0-9' |
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| 09:18.59 | aceio | cool i willl try that |
| 09:19.25 | ChannelZ | well that wasn't necessarily a useful pattern for you, just an example. So construct accordingly :) |
| 09:19.31 | ChannelZ | off to bed for me |
| 09:20.13 | aceio | ChannelZ: sure i shall let you know how it go's |
| 09:22.57 | FILLVAIO3 | Guys, is there anybody know software for collect and manage call statistics? |
| 09:24.00 | *** join/#asterisk nickzxcv (nick@schmalenberger.us) |
| 09:24.22 | nickzxcv | does anybody know about compiling dahdi on sparc64? |
| 09:24.37 | nickzxcv | i just tried to compile the freebsd port and I got ===> dahdi-2.3.0rc2 is only for i386 amd64, while you are running sparc64. |
| 09:36.42 | emora | How should I interpret output of core show translation: alaw & ulaw -> g729 encoding is at 12000 microseconds. ?? |
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| 10:14.29 | DelphiWorld | nickzxcv: ask here about dahdi :D :P xD :-;) |
| 10:14.37 | nickzxcv | lol |
| 10:14.58 | DelphiWorld | russellb: you are new for me;) |
| 10:17.20 | nickzxcv | so i'm compiling dahdi on freebsd on sparc64, and apparently BIAS is already defined in machine/frame.h and its redefined in drivers/dahdi/dahdi-base.c |
| 10:17.51 | nickzxcv | i'm trying to figure if its a redefinition of the same thing, or different things with the same name |
| 10:18.21 | *** part/#asterisk DelphiWorld (~Delphi@196.20.95.121) |
| 10:23.14 | nickzxcv | from the comment /* define the add-in bias for 16 bit samples */ i think its a namespace collision, so i'm renaming it DAHDIBIAS :-/ |
| 10:25.12 | nickzxcv | ah, unknown opcode :( |
| 10:25.25 | nickzxcv | i guess dahdi really isn't portable to sparc64 :( |
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| 10:51.49 | nickzxcv | it looks like the assember parts of dahdi, are its versions of test_and_set_bit and test_and_clear_bit |
| 10:53.03 | nickzxcv | i suppose linux has a sparc64 implementation of those and bsd doesn't have them at all, and the author of the bsd port of dahdi only ported the i386 version? |
| 10:59.26 | nickzxcv | interesting, in linux for sparc i think it doesn't even use assembler to do the function |
| 10:59.53 | nickzxcv | but its still sparc specific i guess, because the point of it is to test the cpu about this |
| 11:01.32 | nickzxcv | there should still be a more portable way to do the test maybe though? |
| 11:08.27 | nickzxcv | this sounds like a good solution http://www.mail-archive.com/freebsd-hackers@freebsd.org/msg25708.html ? |
| 11:14.29 | *** join/#asterisk mva (mva@desktop.mva.name) |
| 11:14.37 | mva | hi there! |
| 11:15.06 | mva | can somebody tell me, how i can fix this: |
| 11:15.12 | mva | == Registered translator 'lintog722' from format slin to g722, cost 999 |
| 11:15.12 | mva | [Jul 10 17:47:49] WARNING[14801]: translate.c:654 __ast_register_translator: plc_samples 160 format f |
| 11:15.15 | mva | ? |
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| 11:24.55 | nickzxcv | apparently the portable way to do test_and_set_bit and test_and_clear_bit on freebsd, is to use atomic_cmpset to emulate them? |
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| 12:54.33 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 12:59.19 | grinder13 | hello guys! I have a problem with a "482 loop detected" error. the setup is two servers over a SIP trunk. here are my confs: http://pastebin.com/9Xqqf5Q1 (for ServerA with IP 192.168.1.2) and http://pastebin.com/2tYU1N2Z (for ServerB with IP 192.168.2.2). and here a log when I am trying to call extension 2000@ServerB from extension 1000@ServerA using the "channel originate" command |
| 12:59.41 | grinder13 | http://pastebin.com/KDV8EhtB |
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| 13:10.18 | *** join/#asterisk af_ (~getsmart@78.134.21.122) |
| 13:13.07 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 13:30.09 | af_ | where could i find pinout for a E1 cable? |
| 13:30.22 | af_ | normal and crossover |
| 13:32.34 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:32.50 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:35.02 | WIMPy | 1 Tx+ 2 Tx- 4 Rx+ 5 Rx- |
| 13:37.45 | af_ | WIMPy, that's normal or cross? |
| 13:38.42 | WIMPy | That's the pin-out. For a normal cable any patch cable will do. For X-Over you need to cross 12 and 45. |
| 13:39.32 | af_ | patch cable? mmmh |
| 13:40.18 | af_ | like a bri S cable? any wire in the same position? |
| 13:43.43 | *** join/#asterisk [Outcast] (~anonymous@24-151-142-169.dhcp.oxfr.ma.charter.com) |
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| 13:43.51 | WIMPy | For a BRI you usually only have a 8P4C cable. A normal Cat5+ patch cable will fit both BRI and PRI. |
| 13:54.39 | af_ | WIMPy, you mean I can use ethernet cables? |
| 13:55.08 | WIMPy | yes |
| 14:03.09 | chuckf | loves standards |
| 14:06.07 | *** join/#asterisk hipitihop (~denis@203.132.229.236) |
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| 14:36.45 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 14:38.49 | grinder13 | hello! anybody for this? I have a problem with a "482 loop detected" error. the setup is two servers over a SIP trunk. here are my confs: http://pastebin.com/9Xqqf5Q1 (for ServerA with IP 192.168.1.2) and http://pastebin.com/2tYU1N2Z (for ServerB with IP 192.168.2.2). and here is a log when I am trying to call extension 2000@ServerB from extension 1000@ServerA using the "channel originate" command: http://pastebin.com/KDV8EhtB |
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| 15:00.10 | hipitihop | anyone here familiar with integrating asterisk with mythtv ? in particular I'm tryong to follow these instructions, but the OSD does not show any caller details. http://www.voip-info.org/wiki/view/Asterisk+tips+MythTV+integration |
| 15:00.10 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 15:02.56 | russellb | hipitihop: start by trying to run the mythtvosd command manually, not from asterisk |
| 15:03.05 | russellb | then go back to messing with asterisk once you have it working as you expect |
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| 15:45.16 | grinder13 | anyone for this? It's a problem with a "482 loop detected" error. the setup is two servers over a SIP trunk. here are my confs: http://pastebin.com/9Xqqf5Q1 (for ServerA with IP 192.168.1.2) and http://pastebin.com/2tYU1N2Z (for ServerB with IP 192.168.2.2). and here is a log when I am trying to call extension 2000@ServerB from extension 1000@ServerA using the "channel originate" command: http://pastebin.com/KDV8EhtB |
| 15:47.51 | [TK]D-Fender | grinder13: Line 91 : sending to 192.168.1. |
| 15:48.01 | russellb | grinder13: an invite is begin sent from .1.2 to .1.2 |
| 15:48.03 | [TK]D-Fender | grinder13: Line 91 : sending to 192.168.1.2 |
| 15:48.19 | russellb | so it's telling you "wtf?" |
| 15:48.21 | [TK]D-Fender | grinder13: Line 97 : Contact is 192.168.1.2 |
| 15:48.29 | [TK]D-Fender | grinder13: it IS call itself. |
| 15:49.20 | russellb | and this is Asterisk sending a call to SIP/1000, which is apparently itself |
| 15:49.58 | [TK]D-Fender | Wait..... |
| 15:50.14 | [TK]D-Fender | grinder13: Is SIP/1000 a softphone RUNNING on your * server as well? |
| 15:50.24 | grinder13 | yes, I 've seen what you are saying. but where is the error? |
| 15:50.25 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 15:50.29 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 15:50.30 | grinder13 | Fender, no |
| 15:50.48 | [TK]D-Fender | grinder13: What is it? |
| 15:50.50 | grinder13 | SIP/1000 is registered by ServerA |
| 15:51.09 | grinder13 | check my confs. you 'll see exactly what I 'm doing |
| 15:51.20 | Trixboxer | grinder13, if its a trunk dont use it as extension over ServerA |
| 15:51.46 | [TK]D-Fender | Trixboxer: that makes no sense... |
| 15:51.48 | russellb | grinder13: you're having the server register with itself? |
| 15:51.53 | russellb | that doesn't make any sense |
| 15:52.20 | Trixboxer | "SIP/1000 is registered by ServerA" and used as extension on the same server |
| 15:52.26 | grinder13 | russell, I have it like that because I want to generate calls from ServerA->ServerB |
| 15:53.12 | russellb | ok, well, the whole SIP/1000 thing needs to be ripped out of the picture, it's not necessary and it is what is causing this problem |
| 15:53.18 | [TK]D-Fender | grinder13: SIP/100 is SereverA. You are having * call itself. |
| 15:53.26 | [TK]D-Fender | 1000* |
| 15:53.42 | [TK]D-Fender | grinder13: Why is ServerA registering to ITSELF? |
| 15:54.10 | russellb | change SIP/1000 to something that is a peer defition for serverb |
| 15:54.15 | [TK]D-Fender | grinder13: And what is the point of Having A call itself to call out to B? |
| 15:54.33 | [TK]D-Fender | russellb: I think he should be using a LOCAL channel on A to call out to B. |
| 15:54.44 | [TK]D-Fender | russellb: He just seems to shove SIP in every place he can. |
| 15:54.55 | russellb | well, maybe ... point is, SIP/1000 is bogus |
| 15:55.03 | [TK]D-Fender | russellb: Indeed.. |
| 15:56.10 | grinder13 | I 've tried with local as well. the thing is that I 'm getting these errors with other calls as well, eg from 1001@ServerA->1002@ServerA or 1001@ServerA->2000@ServerB |
| 15:56.17 | grinder13 | 1001, 1002 are softphones |
| 15:56.33 | grinder13 | i 'll post the logs from home to see. |
| 15:56.43 | grinder13 | i have to leave the lab now :( |
| 15:56.50 | [TK]D-Fender | grinder13: grinder13 If 1000 is a softphone why is ASTERISK REGISTERING AS IT <---- |
| 15:56.52 | russellb | before you go any further, you need to convince yourself of why SIP/1000 is wrong |
| 15:57.15 | russellb | it's pretty important for understanding |
| 15:57.15 | grinder13 | Fender, 1000 is NOT a softphone |
| 15:57.33 | [TK]D-Fender | grinder13: <grinder13>1001, 1002 are softphones <-------------- |
| 15:57.38 | [TK]D-Fender | oops |
| 15:57.38 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 15:57.42 | [TK]D-Fender | digit fail |
| 15:57.47 | russellb | [TK]D-Fender: you lose |
| 15:58.07 | grinder13 | as I said I just want to generate calls from 1000@ServerA -> 2000@ServerB and do traffic measurements |
| 15:58.11 | [TK]D-Fender | grinder13: Still 1000 IS that server itself . Why are you having * call itself? |
| 15:58.30 | Trixboxer | grinder13, exchange the register => statement on both the servers |
| 15:58.31 | grinder13 | but I am NOT calling 1000 from 1000 |
| 15:58.39 | grinder13 | thats what I am saying |
| 15:58.57 | [TK]D-Fender | grinder13: You are |
| 15:59.00 | russellb | dude, I'm telling you, Asterisk is not doing anything wrong here. It is doing what you tell it to. trust me. :-) |
| 15:59.01 | grinder13 | I am doing a "channel originate SIP/1000 extension 2000" |
| 15:59.13 | russellb | that statement tells Asterisk as step 1, to call SIP/1000 |
| 15:59.31 | russellb | before extension 2000 is even brought into the picture, step 1 is to call SIP/1000 |
| 15:59.34 | russellb | that's what you see in your log |
| 15:59.42 | grinder13 | hmm, ok |
| 15:59.48 | [TK]D-Fender | grinder13: http://pastebin.com/9Xqqf5Q1 <--- line 4. It ir registering to ITSELF. Line 15. we can SEE where ServerB is ... |
| 16:00.04 | [TK]D-Fender | grinder13: B is 2.2 |
| 16:00.11 | [TK]D-Fender | grinder13: So A is point to ITSELF |
| 16:00.17 | grinder13 | I 'll post the logs from the other calls from home as I said |
| 16:00.31 | grinder13 | security guy wants me out of the lab |
| 16:00.35 | russellb | I don't need to see any more logs :-) |
| 16:00.48 | russellb | tell the security guy to STFU |
| 16:01.09 | russellb | tell him you have work to do and for him to go eat a doughnut |
| 16:01.38 | grinder13 | i told him so, but hell :( |
| 16:01.56 | russellb | oh well |
| 16:02.03 | [TK]D-Fender | grinder13: here are my confs: http://pastebin.com/9Xqqf5Q1 (for ServerA with IP 192.168.1.2) <------------then why is there a REGISTER in this pastebin pointing to ITSELF |
| 16:02.12 | [TK]D-Fender | grinder13: WAKE UP |
| 16:02.18 | grinder13 | talk to you soon |
| 16:02.22 | russellb | [TK]D-Fender: CALM DOWN |
| 16:02.23 | russellb | :-p |
| 16:02.24 | Trixboxer | grinder13, exchange the 'register => ' statement on both the servers |
| 16:02.36 | Trixboxer | :) |
| 16:03.13 | [TK]D-Fender | o/ |
| 16:03.41 | russellb | welcome to Saturday, folks |
| 16:03.50 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 16:03.51 | [TK]D-Fender | russellb: Where the crazies run free... |
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| 16:18.04 | xheliox | yawns |
| 16:19.26 | p3nguin_ | shoves in a stale twinkie |
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| 17:20.54 | *** part/#asterisk bn-7bc (bjarne@raskerepc.holmedal.net) |
| 17:21.18 | grinder13 | hello again! here to sort out the 482 issue i was talking earlier |
| 17:24.26 | [TK]D-Fender | [12:02]<[TK]D-Fender>grinder13: here are my confs: http://pastebin.com/9Xqqf5Q1 (for ServerA with IP 192.168.1.2) <------------then why is there a REGISTER in this pastebin pointing to ITSELF |
| 17:24.57 | [TK]D-Fender | grinder13: pastebin is for server whose IP is 192.168.1.2. It has a register for 1000 that points to its OWN IP. |
| 17:25.06 | [TK]D-Fender | 'griIs this now entirely clear? |
| 17:25.10 | [TK]D-Fender | grinder13: Is this now entirely clear? |
| 17:25.30 | [TK]D-Fender | grinder13: WHY is it pointed to itself? |
| 17:26.09 | *** join/#asterisk bn-7bc (bjarne@raskerepc.holmedal.net) |
| 17:26.37 | [TK]D-Fender | \o/ |
| 17:27.11 | *** join/#asterisk grinder13 (~grinder@cpc2-sgyl2-0-0-cust1286.sgyl.cable.virginmedia.com) |
| 17:30.21 | [TK]D-Fender | grinder13: So did you get that before your connection flaked out? |
| 17:31.08 | grinder13 | well, ok. i was studuying what you were saying earlier and I think that i got it |
| 17:31.55 | grinder13 | so the question no is: how can I generate calls from ServerA to ServerB? maybe with the "console dial" command? |
| 17:32.22 | [TK]D-Fender | grinder13: You both adeed a useless layer to this.. and then broke it. |
| 17:32.28 | grinder13 | no softphones involved |
| 17:32.36 | [TK]D-Fender | grinder13: what do you want o have on each side of the call? |
| 17:33.15 | grinder13 | I want to A to call B and B to answer by playing a pre-recorded message |
| 17:33.32 | grinder13 | and then do performance measurements |
| 17:34.03 | grinder13 | nothing hard for the experienced Asterisk user |
| 17:34.54 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 17:36.31 | grinder13 | and repeat the experiment with the IAX protocol |
| 17:36.31 | [TK]D-Fender | grinder13: originate local/extentodialotherserver@dialplancontextthathasit extension extenwithplaybackstuffetc@context |
| 17:37.24 | grinder13 | ok, i c |
| 17:37.49 | grinder13 | what about the "console dial" command? |
| 17:41.03 | *** join/#asterisk blaines (~blaines@wsip-98-174-253-36.ph.ph.cox.net) |
| 17:41.37 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 17:41.45 | [sr] | hi WIMPy |
| 17:41.52 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 17:41.57 | [sr] | WIMPy: my kernel modules mISDN crash when i disconnect a call :S |
| 17:51.32 | [TK]D-Fender | grinder13: No |
| 17:51.43 | [TK]D-Fender | grinder13: Console dial is to use the CLI as a softphone |
| 17:51.58 | [TK]D-Fender | grinder13: You want to send an automated recording. |
| 17:52.36 | [TK]D-Fender | grinder13: You could remove 1 side of the local dialplan by doing : |
| 17:53.01 | [TK]D-Fender | grinder13: originate SIP/peerforserverB/number extension extenwithplaybackstuffetc@context |
| 17:54.46 | grinder13 | if "console dial" works as a softphone from the CLI then isn't that want exactly I want to do? generate a number of calls with the "console dial" cmd from ServerA to ServerB and ServerB answer by playing the pre-recorded message? |
| 17:55.06 | grinder13 | also I didnt get the second part of what you just said |
| 17:57.24 | [TK]D-Fender | grinder13: You want A's callout to be automated and to play a recording to B. |
| 17:57.35 | [TK]D-Fender | grinder13: This is not a "person" talking |
| 17:58.09 | [TK]D-Fender | grinder13: Actually... this is A calling and sending. if you just want a to call out.. you could potentially use console dial |
| 17:59.05 | grinder13 | Ino, I want A to call and B to answer and play the message back to A |
| 18:05.38 | ChannelZ | yawns and stretches |
| 18:06.10 | ChannelZ | grinder13: so have you removed the bogus 'register' lines in your sip.confs? |
| 18:06.32 | grinder13 | not yet |
| 18:06.42 | ChannelZ | you should |
| 18:06.44 | grinder13 | the security guard kicked me out of the lab |
| 18:07.07 | grinder13 | will do tomorrow |
| 18:07.09 | *** join/#asterisk Tim_Toady (~moi@178.128.16.115.dsl.dyn.forthnet.gr) |
| 18:11.01 | ChannelZ | ssh.... |
| 18:11.14 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 18:12.32 | grinder13 | i can't ssh. the routers/workstations we have at the lab are not connected to the Internet. they are just there for our Cisco stuff, etc |
| 18:15.01 | [TK]D-Fender | grinder13: then "console dial exten@context" and make that exten dial out using server B's peer entry |
| 18:15.15 | [TK]D-Fender | grinder13: You don't even need registrations for anything ehre that I can see |
| 18:18.24 | *** join/#asterisk grinder13 (~grinder@cpc2-sgyl2-0-0-cust1286.sgyl.cable.virginmedia.com) |
| 18:21.42 | [TK]D-Fender | [14:15]<[TK]D-Fender>grinder13: then "console dial exten@context" and make that exten dial out using server B's peer entry |
| 18:21.44 | [TK]D-Fender | [14:15]<[TK]D-Fender>grinder13: You don't even need registrations for anything ehre that I can see |
| 18:23.33 | grinder13 | ok, got it |
| 18:23.42 | grinder13 | thanx :) |
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| 20:29.00 | knctrnl | i am passing a call over to an IAXMODEM through a macro. My DID is coming through on the iaxmodem as "s" from the macro. is there anyway i can have it maintain the original extension? |
| 20:29.00 | knctrnl | exten => 211,1,Macro(Dialfax) |
| 20:29.00 | knctrnl | [macro-Dialfax] |
| 20:29.01 | knctrnl | exten => s,1,Dial(IAX2/faxiax01/${EXTEN}) |
| 20:30.09 | knctrnl | nevermind found it |
| 20:30.14 | knctrnl | MACRO_EXTEN |
| 20:35.40 | eliel | knctrnl: yes, MACRO_EXTEN or passing it as a parameter to the macro |
| 20:50.33 | nightwalk | I know there are several settings that affect it...anyone know off-hand which setting would be *most likely* to cause (dahdi-based) calls to be dropped in the middle? |
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| 21:29.44 | [TK]D-Fender | nightwalk: "callprogress=yes" <- synonymous with "disconnectmycallswhenleastconvenient=yes" |
| 21:31.12 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 21:34.04 | russellb | :-) |
| 21:35.08 | ChannelZ | Wow. http://www.newegg.com/Special/ShellShocker.aspx?cm_sp=ShellShocker-_-11-129-046-_-07102010 |
| 21:35.16 | ChannelZ | They managed to make this thing look like a VCR from the 90s. |
| 21:44.06 | [sr] | nice |
| 21:49.52 | [sr] | i like nostalgic hardware :) |
| 21:50.06 | [sr] | also stuff's like my spectrum 128k with k7's |
| 21:50.16 | [sr] | DataGeneral Servers with DG/UX |
| 21:50.45 | [sr] | oh well... time flies!!! |
| 21:58.08 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 21:58.08 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 21:58.36 | bkruse | Can I haz the asterisk? |
| 21:59.47 | ChannelZ | Yes. Fo free even |
| 22:00.05 | russellb | bkruse: ! |
| 22:00.47 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:00.51 | bkruse | Hey russellb! How are you my friend? |
| 22:01.10 | russellb | good, you? |
| 22:01.17 | bkruse | I am doing pretty good my friend, thank you! |
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| 22:31.05 | mattwj2002 | hi all |
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| 22:56.52 | root52 | Hey All Anyone have experience wiring a sangoma a200 in Australia? I have my 4pin rj11 from the PTSN that has dialtone when plugged into a POTS phone. I plug it into the sangoma FXO port and using the wanpipemon tool see no line voltage voltage on the port. Any trick I missed? |
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