| 00:03.39 | raden | whats the difference between SIP and LOCAL ? like whena call is being placed > |
| 00:03.56 | raden | how can i make asterisk use a different service to dial outbound ? |
| 00:03.57 | [TK]D-Fender | SIP is SIP. |
| 00:04.01 | [TK]D-Fender | Local is DIALPLAN |
| 00:04.24 | Sidrov | anyone here ? |
| 00:04.27 | Sidrov | is there a bug in AGICommand_record -> BEEP ? |
| 00:04.29 | raden | SIP is SIP ? hmmmm |
| 00:04.38 | Sidrov | i can't get out the damm BEEP |
| 00:05.44 | [TK]D-Fender | SidSo far I've watched you ignore advice on your syntax for the regular dialplan call to it repeatedly |
| 00:05.59 | [TK]D-Fender | Sidand perhaps you should SHOW US in full detail what you're doing. |
| 00:06.01 | [TK]D-Fender | ~pb |
| 00:06.02 | 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. |
| 00:06.08 | [TK]D-Fender | Sidrov: ^^^^ |
| 00:17.29 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 00:30.30 | ChannelZ | raden: SIP is a channel using the SIP protocol. LOCAL is a virtual channel which feeds back into the dialplan. |
| 00:30.43 | raden | ok |
| 00:32.36 | raden | there anything I need to do special if im runnning this over a radio bridge ? |
| 00:32.46 | raden | keep getting sip retransmit errors |
| 00:34.34 | ChannelZ | Well that can be anything.. |
| 00:35.08 | ChannelZ | Either packets aren't making it out of your system/firewall, aren't making it to the other end, or both of those are working and the remote end's responses aren't making it back to you |
| 00:35.14 | WIMPy | Get some heavy tools to remove anythign in the way. |
| 00:35.36 | raden | you can be on a call about 30 seconds before it drops |
| 00:35.54 | ChannelZ | Lousy connection I guess |
| 00:36.07 | ChannelZ | does it have super high or erratic latency? |
| 00:36.08 | raden | connection solid |
| 00:36.11 | raden | -39 |
| 00:36.21 | raden | [Aug 17 19:53:41] WARNING[25406]: chan_sip.c:3912 retrans_pkt: Maximum retries exceeded on transmission 35f438ae-3e00b86b@192.168.2.51 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. |
| 00:36.22 | raden | [Aug 17 19:53:41] WARNING[25406]: chan_sip.c:3939 retrans_pkt: Hanging up call 35f438ae-3e00b86b@192.168.2.51 - no reply to our critical packet (see doc/sip-retransmit.txt). |
| 00:36.26 | raden | thats what i get everytime |
| 00:36.53 | ChannelZ | on a LAN even |
| 00:37.09 | ChannelZ | The device is perhaps wonk and not paying attention to said critical SIP messages |
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| 00:42.52 | raden | well its kinda a lan |
| 00:44.02 | ChannelZ | VPN? |
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| 00:47.47 | raden | no |
| 00:47.52 | raden | try WISP with 70 clients |
| 00:48.04 | raden | asterisk box sits inside |
| 00:48.08 | raden | so all the clients are nated |
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| 00:48.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 00:49.30 | Sidrov | is there a bug in AGICommand_record -> BEEP ? |
| 00:49.31 | Sidrov | i can't get out the damm BEEP |
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| 04:02.34 | SeRi | any body around? |
| 04:03.12 | carrar | I'm your huckleberry |
| 04:04.07 | ChannelZ | let's get it on |
| 04:04.32 | carrar | OPPAN GANGNAM STYLE !!!!!!!!! |
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| 04:27.57 | Micc | does anyone here have any experience with mitel phones in sip mode? I can't seem to find a way to transfer using the programmable keys. I want to make a blf/xfer key or at least be able to xfer then press the speed dial key. |
| 04:29.14 | carrar | use a Mitel 3300 controll!! |
| 04:29.16 | carrar | controller |
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| 04:29.55 | Micc | whats that? the phones are 5330's |
| 04:30.16 | Assid | wassup |
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| 05:07.35 | ChannelZ | they don't have a transfer button that works with SIP? |
| 05:26.32 | ChannelZ | I should switch all my MOH over to Portal Music for Testing |
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| 13:13.30 | retentiveboy | Anybody here manage to get a Digium D40 to get VLAN config from LLDP? Know what specifically it's looking for? |
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| 13:34.24 | stasiu7 | Hi guys. Do you know where I can find the config file with path to the .wav file that gets played after reaching vm? Right now it's set to greet.wav and I want to change it for specific extensions. Thanks! |
| 13:42.22 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 13:42.24 | Assid | hi |
| 13:43.42 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:46.35 | Assid | http://pastebin.com/5Z7hniTp |
| 13:46.51 | Assid | i am able to receive acall and forward it to the other dahdi port.. |
| 13:47.44 | Assid | however, im stuck on trying to make an outgoing call. |
| 13:50.34 | Assid | WIMPy: you around by any chance? |
| 13:55.41 | carrar | isn't "forward it to the other dahdi port" in and of it's self "making an outgoing call" ? |
| 13:56.32 | Assid | well.. its its coming in from the vendor and out INTO the NEC device |
| 13:56.52 | Assid | i am now trying to reverse the process.. where the NEC pbx sends me the call and i send it to the vendor |
| 13:56.59 | Assid | ifi i call from sip device.. it works fine.. |
| 13:57.12 | carrar | DO you have your dadhi ports assigned to a group? |
| 13:57.39 | Assid | also im supposed to have the NEC port show red when its connected.. thats not happening |
| 13:57.59 | Assid | err.. yeah.. whatever the default configured |
| 13:58.26 | carrar | just Dial(DAHDI/g1/2061112222) |
| 13:59.07 | Assid | yeah if i do it statically.. it works.. but if the call comes from the NEC .. its only taking 1 digit at a time |
| 13:59.15 | carrar | or exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) |
| 13:59.32 | Assid | ive set it up currently to send EVERYTHING down the vendors pipe |
| 14:00.11 | carrar | What do you mean 1 digit at a time |
| 14:00.24 | carrar | doesn't make sense |
| 14:00.35 | Assid | like if its 9124512345 .. its trying to call extension 9 only |
| 14:01.28 | carrar | so change the config on the nec |
| 14:01.37 | carrar | to not do 9 |
| 14:01.45 | Assid | no no.. the mobile numebrs are 9 |
| 14:01.55 | Assid | its sending 1 digit at a time man.. like an old analogue phone |
| 14:02.09 | carrar | Sounds like you need to make changes on the NEC |
| 14:02.17 | carrar | to make it dial better |
| 14:02.38 | Assid | if i connect the same thing directly to the vendor.. it works |
| 14:03.31 | carrar | Are these PRI's? |
| 14:04.32 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:07.41 | Assid | yeah |
| 14:07.46 | Assid | okay i managed to get it to worl |
| 14:07.52 | Assid | im about 90% done |
| 14:09.16 | Assid | hmm i need to give a dialtone.. so i guess its time to go DISA.. so i can accept numbers |
| 14:10.23 | carrar | just change the NEC |
| 14:10.32 | carrar | to not do the dial 9 first for dialtone |
| 14:12.24 | carrar | that doesn't sounds like a PRI |
| 14:12.51 | Assid | the NEC ghets the pri on 0 |
| 14:13.01 | carrar | what does that mean |
| 14:13.13 | Assid | NEC <-> asterisk <-> vendor |
| 14:13.28 | carrar | WHat is vendor |
| 14:13.32 | Assid | all the extensions are connected to the NEC at the moment |
| 14:13.44 | Assid | carrar: the PRI vendor / telephone co. |
| 14:14.12 | carrar | So just putting ASterisk inline of between your NEC PBX and telco |
| 14:14.15 | carrar | thats simple stuff |
| 14:14.41 | carrar | Why would your telco require 9 first to dial out? |
| 14:15.24 | carrar | some PBX's require EXTENSIOSN to dial 9 first to dial out |
| 14:15.29 | carrar | but not the telco |
| 14:15.52 | Assid | it doesnt |
| 14:15.57 | Assid | my cell number begins with 0 |
| 14:15.59 | Assid | err.. 9 |
| 14:16.29 | carrar | Are you in the USA? |
| 14:16.37 | Assid | no.. india |
| 14:16.46 | carrar | so 9123123123 |
| 14:16.56 | carrar | one long string of numbers |
| 14:17.08 | Assid | http://www.voip-info.org/wiki/view/Asterisk+cmd+DISA -- im trying to do this: exten => s,1,Dial(Srx/tgExternalPtP//${CALLERIDNUM}) ; example for Sirrix cards my line is : exten => s,n,Dial(DAHDI/i1//${CALLERIDNUM}) |
| 14:17.14 | carrar | and the NEC is not sending that? |
| 14:17.36 | Assid | well i manahged to do that.. right now.. if i call from my cordless phone.. PREDIAL the number and hit call |
| 14:18.08 | Assid | hence my line above.. where i want to generate a dialtone.. and let it go out through that |
| 14:18.36 | carrar | SO you telco waits for 9 then presents another dialtone? |
| 14:18.52 | Assid | no.. forget the 9 !! its part of thenumber |
| 14:19.02 | Assid | my telco accepts the final number i want to call ONLY |
| 14:20.27 | Assid | okay heres the thing.. you know how you press 0 .. and you get a dialtone ? |
| 14:20.30 | Assid | how do i do that ? |
| 14:20.46 | carrar | lets see some logs |
| 14:21.03 | carrar | You don't do that with PRI's |
| 14:21.12 | carrar | thats between the PBX and HANDSETS |
| 14:21.20 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:21.21 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:22.12 | Assid | correct.. im on the handset.. which sits behind the NEC |
| 14:22.20 | Assid | i press 0.. so i can dial out |
| 14:22.23 | carrar | or do you actually want to call the phone number of 0 to the telco and hten send DTMF after the connection is made? |
| 14:22.23 | Assid | http://pastebin.com/kgm4MukG |
| 14:22.35 | Assid | no.. i want to fake the temp dialtone |
| 14:24.29 | carrar | I don't see any number dia;ed |
| 14:24.32 | carrar | thats just the hangup |
| 14:25.06 | carrar | You haven't matched anything |
| 14:26.54 | Assid | yes cause i dialled nothing.. i just hit 0 |
| 14:27.03 | Assid | i need to generate a dialtone |
| 14:27.16 | Assid | "If all you want is an immediate dialtone you could also dial your gateway technology (Zaptel, Dadhi, Capi, Srx...) with no target specified (see the double // in the example), e.g.:" |
| 14:27.20 | Assid | thats what i want to do |
| 14:27.30 | Assid | i want to generate an immediate dialtone so i can call the DAHDI |
| 14:28.18 | carrar | Well you already know how to do that |
| 14:28.20 | carrar | so do that |
| 14:28.37 | carrar | but I'd ch ange the NEC if I were you |
| 14:28.43 | carrar | so it doesn't do that crap |
| 14:31.21 | Assid | http://pastebin.com/T0dSE4iS |
| 14:31.25 | Assid | there |
| 14:31.28 | Assid | thats what im doing |
| 14:31.30 | Assid | this is workin |
| 14:32.10 | carrar | If you are using PRI's, ware are you using s |
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| 15:16.03 | *** join/#asterisk Sidrov (~sid@AC9EA707.ipt.aol.com) |
| 15:16.13 | Sidrov | Hello all |
| 15:18.15 | Sidrov | AGI RECORD FILE <filename> <format> <escape digits> <timeout> [offset samples] [BEEP] [s=<silence>] => Is there any way to shutoff the BEEP ??? |
| 15:26.42 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:28.45 | [TK]D-Fender | Sidrov: Skip that parameter and fill in all the others |
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| 16:14.06 | Sidrov | [TK]D-Fender -> same thing with or without. Documentation doesn't provide a value which disables it. Should find sourcecode of record command ? |
| 16:14.39 | Sidrov | it's annoying |
| 16:17.59 | [TK]D-Fender | And you have NOT shown me a call with AGI debug that shows you've followed my suggestions. |
| 16:18.00 | [TK]D-Fender | ~pb |
| 16:18.01 | 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. |
| 16:18.02 | [TK]D-Fender | ^^^^ |
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| 16:19.00 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:19.00 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 16:51.12 | Assid | heya |
| 16:51.24 | Assid | so i think i mostly got my pri working |
| 16:52.44 | Sidrov | [TK]D-Fender : |
| 16:52.47 | *** join/#asterisk mazpe (~lesterm@li315-205.members.linode.com) |
| 16:52.58 | Sidrov | AGI Rx << RECORD FILE /tmp/myfile wav "" "2000" "" "1" |
| 16:52.58 | Sidrov | <PROTECTED> |
| 16:53.17 | Sidrov | this is debug output |
| 16:54.18 | ChannelZ | RECORD FILE /tmp/barf ulaw # 10000 s=2 |
| 16:54.23 | ChannelZ | works for me, no tone. |
| 16:54.53 | Sidrov | it can be because of chanspy ? |
| 16:55.21 | ChannelZ | Since when did Chanspy become a part of this? (this is why we've been asking you for complete information for 2 days) |
| 16:56.22 | Sidrov | i thought using chanspy on same channel which is recorded has nothing to do with that |
| 16:58.41 | ChannelZ | you are sending "1" for the beep parameter. |
| 16:59.19 | Sidrov | oh yeah.. old version was without s= |
| 16:59.51 | [TK]D-Fender | <PROTECTED> |
| 16:59.57 | [TK]D-Fender | No quotes AT ALL |
| 16:59.57 | sruffell | Assid: what was the issue? |
| 17:00.14 | [TK]D-Fender | When I said to skip that parameter, I mean LEAVE IT OUT ENTIRELY |
| 17:00.29 | Sidrov | ok trying now |
| 17:00.31 | Assid | sruffell: kept hacking away at the config.. between the dahdi config and the dialplan |
| 17:00.35 | Assid | finally got it to twork |
| 17:00.49 | sruffell | umm…that doesn't sound encouraging. It just now works but you don't know why it didn't before? |
| 17:01.31 | sruffell | sends [TK]D-Fender some blood pressure medicine. |
| 17:02.03 | [TK]D-Fender | sruffell: I get very cranky when I get too much blood in my caffeine stream |
| 17:02.12 | sruffell | nods |
| 17:03.22 | Assid | sruffell: various problems.. yday.. pri_cpe / pri_net change.. for one of tyhe ports.. then..callerid .. apparently i had to add the STD dialling code. whcih the NEC didnt used to send.. so it was by luck someone told me i needed it.. |
| 17:04.18 | sruffell | ok…that's more encouraging. |
| 17:04.26 | Assid | sruffell: outgoing calls.. somehow waitexten magically worked (although it didnt yday) ... even tho i had immediate=no.. so i just went through all the configs and kept adding it every possible place |
| 17:05.12 | sruffell | I feel your pain there. I wish there were a way to completely dump all the configs from a running system. So you can see for any given channel what configs are actually in effect. |
| 17:05.22 | sruffell | …and what defaults were used, etc.. |
| 17:05.52 | Assid | sruffell: what got me was the stupid callerid .. it still shows me the 8 digit DID .. not the whole thing including the STD code |
| 17:06.10 | Assid | im actually pre-pending it before i re-send it out to the pri provider |
| 17:10.42 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:15.44 | Assid | sruffell: i dont understand the need for the groups tho .. |
| 17:15.48 | Assid | in dahdi |
| 17:18.59 | ChannelZ | as in group = xx ? |
| 17:22.20 | Assid | yeah in dahdi.. |
| 17:24.34 | ChannelZ | They allow you to group channels so you can dial the group and it will pick the next available free channel.. like Dial(DAHDI/g1/5551212) |
| 17:26.08 | sruffell | what he (she?) said. |
| 17:26.10 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 17:26.17 | Sidrov | [TK]D-Fender ChannelZ thank you for help. IT WORKS ! :) |
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| 17:49.43 | ChannelZ | yay |
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| 19:19.05 | Sidrov | [TK]D-Fender agi RECORD file have end stop silence detector. Is there any way to get recording started after specified silence interval ? |
| 19:19.38 | imox | chan_sip.c:3661 retrans_pkt: Retransmission timeout reached on transmission |
| 19:20.01 | imox | what is this? I don't know what I should do. can someone help me? |
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| 19:50.09 | *** part/#asterisk qmr (~qmr@gamma.qtech.us) |
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| 20:05.39 | kaldemar | imox: see this: |
| 20:05.47 | kaldemar | ~sipnat |
| 20:05.47 | 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. |
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| 20:32.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 21:36.32 | pcs | Q: is anyone here for hire? /msg me if u are, thanks. |
| 21:38.02 | ChannelZ | male escort? |
| 21:38.42 | pcs | hahaha, to help configure an asterisk server. |
| 21:40.15 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 21:40.18 | ChannelZ | oops |
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| 22:54.43 | pcs | calling now. |
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| 22:56.35 | ChannelZ | oh good., |
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| 22:59.21 | danfromuk | Hi, can MixMonitor be paused with PauseMonitor? |
| 23:40.29 | ChannelZ | No, PauseMonitor makes your screen freeze. |
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