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| 02:29.35 | dmz | howdy y'all, quick question; i want to make a dialplan that will receive a call, execute an agi script to determine if some function is ready (say quering status of bill payment or whatever); the agi script determins that the request will take 5 min to process; i'd like to park the call into something where they get onhold music until time limit is up then drop them back into the agi to check status. i know i could have agi park them & then spool itself & u |
| 02:29.36 | dmz | npark but any way of parkign w/a time limit & exit context within asterisk driectly? |
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| 03:49.33 | [TK]D-Fender | dmz: dial a local channel with m & M(). MoH while calling, time delay built it, and what you'll dial will actuall check constantly for that exit condition. You will never "answer", but will simply choose when to abort. |
| 03:51.07 | *** join/#asterisk techwerkz (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 03:53.05 | techwerkz | I have a context in musiconhold.conf for setting up silence in instances moh is called. To do this I have an application=/dev/null ... this worked fine up until 1.8.3.x and later ... now I get res_musiconhold.c:659 monmp3thread: poll() failed: Interrupted system call flooding the CLI. I remove that context and all is well. |
| 03:53.20 | techwerkz | Any other suggestions? |
| 03:53.21 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:58.29 | ectospasm | techwerkz: copy the seconds of silence into your MOH directory, and be sure that a break of silence plays between each MOH file. |
| 03:58.36 | ectospasm | just a guess, though. |
| 04:00.21 | dmz | [TK]D-Fender interesting; and just to be clear what is m& M()?? |
| 04:00.26 | dmz | goes to google |
| 04:00.42 | ectospasm | dmz: you can always do "core show application Dial" |
| 04:00.52 | ectospasm | Asterisk is pretty well self documenting in that regard. |
| 04:01.43 | dmz | hmm you know i've never gotten info from core that way; learn something new every day :) |
| 04:04.09 | ectospasm | dmz: same thing works for dialplan functions. |
| 04:07.00 | *** part/#asterisk Bullmoose (~Bullmoose@71-33-2-206.bois.qwest.net) |
| 04:11.39 | dmz | [TK]D-Fender; so using a macro, from what you said i could have the channel have say a 30 second timeout and have the macro keep testign it's check for 30 sec and if it succeeds have call "GOTO" the right place; if not i can loop back & ask if they want to keep holding or hangup and loop through again |
| 04:12.37 | [TK]D-Fender | dmz: Sorry, M() not needed... |
| 04:12.59 | [TK]D-Fender | dmz: the dialed local channel does everything you need while keeping it in MoH until it decides to let go. |
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| 04:13.38 | dmz | ok i'll dig into that some |
| 04:13.50 | dmz | i may bounce more off y'all in a bit :) gotta get my agi checks done 1st |
| 04:14.09 | dmz | darn perl isn't returning hash variables properly :( |
| 04:31.58 | [TK]D-Fender | checkout time... later all |
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| 05:56.07 | schmidts | good morning |
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| 06:01.31 | qakhan | i have TDM800p in my asterisk. when someone hangup call from his ext call doesnt hangup and line get busy, but when caller hangup call then call get hangup |
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| 07:17.03 | sawgood | How would I write up something in extensions.conf to do this: _1NXXNXXXXXX,1,Dial(SIP/ITSP/${EXTEN}) ..... but prior to this 1 priority (I would like a lookup done from a .txt file to determine if a specific NPA was selected (and if so) ... do something else? |
| 07:18.02 | sawgood | I have 65 NPA (area codes) that if there is a destination for, I need the dialplan to use a backup vendor |
| 07:20.30 | *** join/#asterisk _omer (~omer@116.71.182.250) |
| 07:21.16 | _omer | hello, Is Queue_Log file format is different in Asterisk 1.8 ? currently I am using Asterisk 1.6 |
| 07:32.31 | ChannelZ | sawgood: easiest way is probably an AGI |
| 07:32.47 | ChannelZ | or use AstDB |
| 07:33.03 | ChannelZ | easier if you're not really a scripter I guess |
| 07:36.32 | sawgood | thank you |
| 07:37.09 | kenshin | The fast busy thing
come to think of it I have the exact reeverse problem |
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| 07:39.03 | kenshin | if I call (from my cell to sip phone\0 and hung up I get the fast busy on sip phone (DIAL exists with non zero code) and if sip phone calls and I hangup Dial exits normally and I can manage the fast busy signal (via a macro) |
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| 07:39.43 | kenshin | in scenario one the whole dial plan gots boom (nothing get executed after Dial exits with non zero status |
| 07:40.15 | ChannelZ | Is this through an ITSP? |
| 07:40.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:b52d:b288:200c:da01) |
| 07:41.31 | kenshin | my set up (Providers ) -> asterisk -> ATA (an st780wl the damn thing is forcing me to use asterisk
it has 2 phone lines but only one provider ) |
| 07:41.48 | kenshin | (had I known I would have stayed away from it) |
| 07:42.40 | kenshin | My asterisk box is a kind of the one ring to rule them all . |
| 07:43.07 | ChannelZ | But is the ATA even involved? You said cell phone/sip phone |
| 07:43.28 | kenshin | cell phone is cell phone
sip phone is ATA |
| 07:43.40 | ChannelZ | so it's an analog phone on the ATA |
| 07:43.45 | kenshin | yep |
| 07:44.03 | kenshin | I;m suspecting the at a here
but I may be doing something wrong too |
| 07:44.14 | ChannelZ | And when you hang up on the cell (incoming via ITSP) the hangup doesn't get detected, but if you hang up the analog phone it does? |
| 07:44.23 | _omer | hello, Is Queue_Log file format is different in Asterisk 1.8 ? currently I am using Asterisk 1.6 |
| 07:46.21 | kenshin | if I call (from cell) and answer analog phone and hang up cell hang dial exits with non zero status dial plan drops there
(so I get annoying fast busy signal on analog phone) |
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| 07:46.50 | ChannelZ | well that's coming from your ATA |
| 07:47.06 | kenshin | that's what I thought. |
| 07:47.07 | ChannelZ | And seems normal |
| 07:47.18 | ChannelZ | (How else are you supposed to know the person hung up on you?) |
| 07:47.37 | kenshin | usually for incoming calls
best way to
answer then dial .. or just dial ? |
| 07:48.09 | kenshin | the annoying part is that after that the dial plan stops
|
| 07:48.24 | ChannelZ | Yes because the channel terminated. |
| 07:48.38 | kenshin | the other scenario
it continues and I have same =>n, Macro (do-hangup) |
| 07:49.22 | kenshin | where I play congestion (I think, the not so fast not aggressive busy tone ) |
| 07:49.53 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:50.49 | ChannelZ | Well from the dialplan's point of view, on an incoming call the dialplan is executing on the calling channel, not the called channel. |
| 07:51.24 | kenshin | is there a way around it ? |
| 07:51.39 | ChannelZ | The called channel (your phone) is simply bridged to the calling channel, there is no dialplan involved. |
| 07:52.49 | ChannelZ | hmm |
| 07:53.07 | kenshin | (intercept incoming channel create an other (from art to at a) then connect both (keeping the second one alive |
| 07:53.39 | kenshin | I tried answer -> dial (ext) didn;t work |
| 07:53.45 | kenshin | (got the same result) |
| 07:54.09 | kenshin | (the diff the timer on my cell phone start right away -- I don;t see dialing) |
| 07:55.37 | ChannelZ | again, when you Dial someone else, there is no dialplan executing from THEIR point of view. It's a function. |
| 07:55.59 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 07:57.09 | ChannelZ | What version of asterisk? |
| 07:58.24 | kenshin | 10.3.1 |
| 07:58.57 | ChannelZ | You might try the 'e' option of Dial, which will make the 'h' extension execute for the called channel |
| 07:59.20 | kenshin | (this seems the ATA's fault, it detects that the other end has hang up and kills the channel |
| 07:59.28 | ChannelZ | Although it actually does it for both channels, so you might get some errors on the console in regards to the channel that hung up |
| 07:59.36 | ChannelZ | No, Asterisk hangs up the channel. |
| 08:00.08 | ChannelZ | Your ITSP detects the hangup, tells Asterisk, and Asterisk terminates the bridge between it and the ATA you dialed. |
| 08:00.19 | kenshin | Mmm, I tried with the g option but not the e |
| 08:00.45 | ChannelZ | The ATA is reacting by giving you a fast busy on the handset because it doesn't know what else to do/thats what it's configured to do/that's the only thing it can do |
| 08:01.12 | kenshin | (checked .. seems no way to configure this on the ATA ) |
| 08:01.15 | ChannelZ | the 'g' option continues the dialplan on the callING channel if the callED channel hangs up |
| 08:01.57 | kenshin | I'll try the 'e' |
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| 08:08.03 | kenshin | F seems a sound choice too |
| 08:08.05 | kenshin | no? |
| 08:08.34 | ChannelZ | Yes |
| 08:10.05 | ChannelZ | off to bed, have fun |
| 08:11.38 | kenshin | thank ChannelZ |
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| 09:34.11 | skirmisha | guys |
| 09:34.28 | skirmisha | why do i get this msg for t38 when such msg is send |
| 09:34.40 | skirmisha | chan_sip.c: Strange... The other side of the bridge don't have udptl struct |
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| 09:45.18 | qakhan | i have TDM800p it doesnt hangup calls |
| 09:45.21 | qakhan | please help |
| 09:48.50 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:48.52 | Ice_Strike | Hello |
| 09:49.11 | Ice_Strike | Why this happen |
| 09:49.13 | Ice_Strike | [May 2 10:48:57] NOTICE[1513]: chan_sip.c:12707 sip_reg_timeout: -- Registration for '1358xx@london.voip.ms' timed out, trying again (Attempt #41) |
| 09:54.49 | *** part/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:56.01 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:56.34 | jacc0 | because the registration timeout timer ended? and the asterisk is re-registering |
| 09:57.24 | Ice_Strike | how to fix it? |
| 09:57.30 | jacc0 | 'trying again (Attempt #41)' could mean that london.voip.ms is offline or is no longer accepting registrations |
| 09:57.43 | Ice_Strike | I have tried different voip.ms server |
| 09:57.47 | Ice_Strike | same issue |
| 09:58.04 | jacc0 | are they pingable from the asterisk server |
| 09:58.08 | jacc0 | ? |
| 09:58.18 | Ice_Strike | sip show peers show its connected? |
| 09:58.20 | jacc0 | ping london.voip.ms -t 30 |
| 09:58.33 | Ice_Strike | voipms/1358xx 174.34.146.1xx N 5060 Unmonitored |
| 09:59.09 | Ice_Strike | yes ping work |
| 10:00.07 | jacc0 | it does not show it is connected |
| 10:00.30 | jacc0 | it shows it has been registerd somehere in the past |
| 10:00.35 | Ice_Strike | ah |
| 10:01.00 | Ice_Strike | any other solution? |
| 10:01.20 | jacc0 | so are you sure this works: ping london.voip.ms -t 30 |
| 10:01.41 | Ice_Strike | 4 bytes from 78.129.153.xx: icmp_seq=1 ttl=52 time=15.9 ms |
| 10:01.41 | Ice_Strike | 64 bytes from 78.129.153.xx: icmp_seq=2 ttl=52 time=18.5 ms |
| 10:01.43 | Ice_Strike | yes |
| 10:01.53 | jacc0 | or did you do : ping 174.34.146.1xx -t 30 |
| 10:02.07 | jacc0 | and what about the ip adres |
| 10:02.12 | jacc0 | why are they different? |
| 10:02.13 | Ice_Strike | let me try that |
| 10:02.27 | jacc0 | 78.129.153.xx != 174.34.146.1xx |
| 10:02.28 | Ice_Strike | i have no idea? |
| 10:02.43 | jacc0 | My guess is you are having DNS problems |
| 10:03.03 | Ice_Strike | ping google.com work though |
| 10:03.23 | jacc0 | some DNS cache might not have been updated |
| 10:03.43 | Ice_Strike | hmmm |
| 10:03.50 | jacc0 | if IP address of london.voip.ms is changes you might need to flush DNS |
| 10:04.00 | Ice_Strike | how? |
| 10:05.00 | jacc0 | service nscd restart |
| 10:05.19 | jacc0 | and restarting asterisk might help |
| 10:05.52 | Ice_Strike | service nscd restart not exist |
| 10:06.21 | jacc0 | '/etc/init.d/nscd restart |
| 10:06.27 | jacc0 | without the quote |
| 10:06.35 | Ice_Strike | -bash: /etc/init.d/nscd: No such file or directory |
| 10:06.42 | jacc0 | what OS? |
| 10:06.43 | qakhan | i have TDM800p it doesnt hangup calls |
| 10:06.44 | qakhan | please help |
| 10:06.50 | Ice_Strike | CentOS |
| 10:06.59 | jacc0 | did you try google? |
| 10:07.39 | jacc0 | http://lmgtfy.com/?q=flush+DNS+centos |
| 10:08.35 | Ice_Strike | lol |
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| 10:19.11 | Ice_Strike | jacc0 It look like its working now, thank you |
| 10:20.37 | qakhan | anyone know why doesnt TDM800p hangup calls |
| 10:25.41 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 10:32.53 | Ice_Strike | Why my hardware phone wont connect to my asterisk server? |
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| 11:13.54 | qakhan | i m using TDM800P when i hangup call there is dead air for 1 min and then call get disconnected |
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| 11:34.23 | jacc0 | yw |
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| 13:11.49 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 13:11.49 | *** topic/#asterisk is #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.1 (2012/04/23), 1.8.11.1 (2012/04/23), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 13:12.40 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:15.56 | bobb_WU | good morning! anyone knowledgeable about DTMF issues? |
| 13:20.53 | WIMPy | ~ask |
| 13:20.54 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:23.41 | bobb_WU | yeah well i put a bunch of stuff in the window yesterday and no one was around to ask :P |
| 13:24.11 | pabelanger | ~patience |
| 13:24.12 | infobot | from memory, patience is a Godly attribute, or the solution for most things. |
| 13:24.13 | bobb_WU | i'm having trouble between a mitel 3300 and our asterisk voicemails. DTMF has stopped working |
| 13:24.28 | jacc0 | :) |
| 13:24.35 | bobb_WU | it works from my asterisk phones, but anything from the mitel side is not recognized |
| 13:24.35 | jacc0 | I might be able to help |
| 13:24.47 | jacc0 | asterisk phones? |
| 13:24.50 | bobb_WU | this happened after updating to 1.8, and some extensive problems in mitel |
| 13:25.07 | WIMPy | And now to the information: How is that thing connected to Asterisk? |
| 13:25.07 | bobb_WU | well its an analog connected to digium card on my asterisk sandbox |
| 13:25.18 | jacc0 | okay |
| 13:25.23 | bobb_WU | i'm doing that webinar today, the asterisk phones look cool so far |
| 13:25.24 | pabelanger | how are you passing DTMF? |
| 13:25.31 | bobb_WU | mitel is rfc2833 |
| 13:25.50 | bobb_WU | i looked at my sip peer defs yesterday and had no value in dtmfmode |
| 13:25.51 | pabelanger | what version is your 3300? |
| 13:26.13 | pabelanger | I've never had any issue with asterisk and mitel 3300 w/ DTMF |
| 13:26.21 | bobb_WU | version of what? i know our database is outdated |
| 13:26.29 | jacc0 | you could use 'tshark -R rtpevent' to show all DTMF send by rfc2833 (on the first network interface) |
| 13:27.11 | pabelanger | the firmware on your 3300, what version? I wanted to see if you are running something newer then me |
| 13:27.20 | bobb_WU | no we are definitely not |
| 13:27.31 | bobb_WU | our sx-2000s are keeping us from updating |
| 13:27.40 | pabelanger | k |
| 13:27.47 | bobb_WU | i'm pulling up the site |
| 13:28.00 | jacc0 | @bobb_WU: look in sip.conf under [general] ; if the sip peer account does not have dtmfmode set then asterisk will use the general setting |
| 13:28.44 | jacc0 | but to see if the mitel is even sending rfc2833 your could use : 'tshark -R rtpevent' |
| 13:29.17 | jacc0 | if the mitel is not sending any rtpevent; then your problem is on the mitel side |
| 13:29.17 | pabelanger | Also, where is the DTMF issue. You said when your phone dials into asterisk? |
| 13:29.24 | bobb_WU | isn't the general setting rfc2833? and i'm doing realtime sip- changed dtmfmode to rfc2833 and auto with no change |
| 13:29.37 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:30.12 | jacc0 | isn't the general setting rfc2833? no, the general setting is set in sip.conf under [general] |
| 13:30.12 | pabelanger | bobb_WU: Are you coming across the 3300, into asterisk to check voicemail? Then DTMF does not work? |
| 13:30.24 | bobb_WU | tshark- is that a packet capture? its not installed on my system |
| 13:30.38 | jacc0 | it is textmode wireshark |
| 13:30.40 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 13:30.48 | bobb_WU | the route is 3300 -> asterisk relay -> asterisk voicemail |
| 13:30.51 | jacc0 | I couldn't do without it |
| 13:30.56 | jacc0 | install it |
| 13:31.12 | pabelanger | bobb_WU: Right, but if I dial into asterisk without using the 3300 does DTMF work? |
| 13:31.20 | bobb_WU | yes it does |
| 13:31.36 | bobb_WU | i call asterisk -> asterisk voicemail and everything works just fine |
| 13:31.43 | pabelanger | And how are you terminating to the 3300? analog, T1/PRI? |
| 13:32.09 | jacc0 | What about licenses? You will need one for every DSP -> that is 2 licenses for 2-way speech |
| 13:32.14 | bobb_WU | PRI i'm thinking |
| 13:32.43 | pabelanger | how is the PRI passing DTMF? Usually out-of-band |
| 13:32.56 | pabelanger | regardless, I doubt this is an asterisk issue and more a 3300 issue. |
| 13:33.20 | WIMPy | Err, I'd call OOB on PRI very unusual. And AFAIK no go with dahdi. |
| 13:33.20 | bobb_WU | i think it is out of band because i see a similar connection from both sides when doing sip debugging |
| 13:33.52 | jacc0 | bobb_WU: just install tshark and see if the mitel is sending any rtpevent |
| 13:33.58 | jacc0 | if not your problem is in the mitel |
| 13:34.03 | bobb_WU | ok firmware is 4.0 SP3 |
| 13:34.37 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:34.40 | bobb_WU | we have had this working, something changed on the system |
| 13:35.25 | bobb_WU | there have been 4 or 5 venders on site due to a 3300 failure over the past month |
| 13:35.27 | *** join/#asterisk tomwish (~Tom@97-117-164-105.phnx.qwest.net) |
| 13:36.56 | pabelanger | sip ssp set level 3 |
| 13:36.56 | jacc0 | that sounds familiar |
| 13:37.00 | pabelanger | sip ssp set storage c |
| 13:37.06 | pabelanger | on the 3300 to enable sip debugs |
| 13:37.10 | pabelanger | check the logs and see what is going on |
| 13:37.15 | pabelanger | remember to disable it when done |
| 13:38.17 | jacc0 | if the mitel hangs a lot you should lower registration interval and implement at least 1 second pauze between every call passed to the mitel |
| 13:38.27 | bobb_WU | how to disable it? |
| 13:38.42 | bobb_WU | it hasn't been hanging, mostly resiliency issues |
| 13:39.09 | pabelanger | sip ssp set level 0, I think |
| 13:39.23 | pabelanger | been a while since I did troubleshooting with a 3300 |
| 13:41.55 | jacc0 | GET TSHARK OR WIRESHARK RUNNING!! filter for `rtpevent` to make an analysis independent of the state of the network interfaces of both mitel and asterisk |
| 13:43.24 | jacc0 | I've seen mitel logs and asterisk cli output with full debug info that did not show any trafic while there realy was trafic (becasue the deamon wasn't bind to any port) |
| 13:43.59 | bobb_WU | ok i will do that |
| 13:44.11 | bobb_WU | i have tcpdump installing. |
| 13:44.22 | jacc0 | that will work 2 |
| 13:44.44 | jacc0 | not sure if you can filter out like in wireshark |
| 13:44.47 | bobb_WU | do you know how to segregate the RTP traffic with it? otherwise i'll plug it into our analyzer |
| 13:45.20 | bobb_WU | i'm going to run to the server room and figure out my ip address of the remote listener i built |
| 13:45.35 | jacc0 | I use tshark with 'tshark -R rtpevent' (that will only show dtmf send by rfc2833) |
| 13:46.10 | bobb_WU | is there another name? like a package name for gentoo maybe |
| 13:46.45 | bobb_WU | "portage tshark" in google only returns wireshark results |
| 13:47.58 | bobb_WU | wait a sec, is tshark the CLI cmd for wireshark? |
| 13:48.22 | bobb_WU | running to the server room ( have to drive ) so i will be back in a few |
| 13:48.26 | bobb_WU | thanks for all the help so far! |
| 13:50.00 | jacc0 | no tshark is not a cli command |
| 13:57.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:57.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:58.18 | jacc0 | sorry, i did not read it right |
| 13:58.35 | jacc0 | <PROTECTED> |
| 13:59.42 | leifmadsen | aye |
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| 14:14.09 | bobb_WU | alrighty got two seperate captures going |
| 14:14.14 | bobb_WU | making some test calls |
| 14:14.18 | jacc0 | :) |
| 14:17.25 | bobb_WU | pabelanger: where do i look for those specific logs (SSP) |
| 14:19.04 | bobb_WU | i see lots of dtmf events from the asterisk box |
| 14:19.24 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:21.17 | p3nguin | wtf |
| 14:21.19 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:21.43 | p3nguin | -- Registration for '7381410e0@sipgate.com' timed out, trying again (Attempt #1363) |
| 14:21.44 | bobb_WU | yep tried three different mitel phones around the office and none show up with DTMF |
| 14:21.57 | p3nguin | That's a lot of attempts. I guess sipgate died. |
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| 14:28.48 | Katty | lfjwoeifjalsdjfoawienfalskjdf0q2w93ur0aqwsflkasfjlaksqueeeeeeeeeeeeeee |
| 14:29.21 | din3sh | Katty:gone crazy with debugging |
| 14:29.45 | anonymouz666 | how can I read the input from "asterisk -rv" in the shell? |
| 14:29.57 | anonymouz666 | i mean |
| 14:29.59 | anonymouz666 | the output |
| 14:30.14 | anonymouz666 | and use the output as input to a "grep" |
| 14:30.55 | leifmadsen | anonymouz666: pipe |
| 14:31.04 | leifmadsen | anonymouz666: asterisk -rvvvv | grep "foobar" |
| 14:31.54 | anonymouz666 | oh thanks. very easy one. I was trying something different. :P |
| 14:32.03 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:11:70d1:f1d6:1083:8388) |
| 14:32.45 | catphish | i'm having trouble with "Set(CALLINFO=Call-Info: ;answer-after=0)" and realtime |
| 14:32.50 | Katty | well, gone crazy for sure. |
| 14:33.08 | catphish | it seems to evaluate to: SIPAddHeader("SIP/atech-charlie-0000075a", "answer-after=0") |
| 14:34.28 | catphish | the same things happen even if add any number of backslashes before the ; |
| 14:35.40 | [TK]D-Fender | catphish, Show us that actual whole picture, not pieces we can't corroborate. |
| 14:35.42 | [TK]D-Fender | ~pb |
| 14:35.43 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:35.44 | [TK]D-Fender | ^^^ |
| 14:36.59 | catphish | no, that's the whole issue |
| 14:37.19 | jacc0 | bobb_WU: so you see no rtpevent comming from the mitel? then your problem is not in asterisk |
| 14:37.23 | catphish | the same thing is documented here: http://lists.digium.com/pipermail/asterisk-users/2007-September/196183.html |
| 14:37.35 | *** join/#asterisk tomwish (~Tom@97-117-164-105.phnx.qwest.net) |
| 14:38.26 | catphish | [TK]D-Fender: i'd be interested to know what you think is missing from my report though that is relevent |
| 14:38.54 | elred_ | anyone job-seeking ? |
| 14:39.00 | [TK]D-Fender | catphish, No, I do not see that actual line this is parsed out from. Nor a real complete call. Also, * 1.2 is completely unsuported at this point It is EOL |
| 14:39.01 | elred_ | http://www.andrexen.com/wp-content/uploads/2012/04/HR-SEARCH-ANDREXEN-FR-SIP-ENGINEER-201204.pdf |
| 14:39.32 | Katty | job seeking? nah |
| 14:39.33 | catphish | [TK]D-Fender: my issue is in 1.8.7, the report i linked to simply said it worked correctly in 1.2 |
| 14:39.37 | Katty | i could use a little less, tbh |
| 14:39.44 | jacc0 | elred_: maybe |
| 14:39.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:40.13 | [TK]D-Fender | catphish, Show complete dialplan, complete CLI with SIP debug. Not just bits and pieces like this and bringing up things that are not the actual problem. |
| 14:41.53 | Katty | qwellery |
| 14:42.41 | catphish | [TK]D-Fender: unfortunately there's no way i can dump my whole dialplan, and the sip debug isn't necessary as the problem shows in the log before any data is transmitted, but i'll put what's relevent in a paste |
| 14:44.12 | [TK]D-Fender | catphish, Also we're at 1.8.11.1, you are 4 full releases behind. The very first thing anyone will ask you to do is confim on the latest |
| 14:44.43 | catphish | can probably do that |
| 14:45.08 | [TK]D-Fender | catphish, You show 2 different halves between what you set and what we see. my trust factor shrinks exponentially seeing that kind of mismatch |
| 14:45.21 | catphish | i hoped a relevent developers would be able to spot the problem |
| 14:45.33 | catphish | that mismatch *is* the problem i'm asking about |
| 14:45.38 | [TK]D-Fender | catphish, You haven't even really shown us the basics. |
| 14:46.09 | catphish | [TK]D-Fender: http://paste.codebasehq.com/pastes/prxn8a6p57c5jz7hd8 |
| 14:46.13 | [TK]D-Fender | catphish, \ <- ESCPE your slashes |
| 14:46.46 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 14:46.56 | catphish | the same occurs when escaping the semi-colon with one, 2 or 4 backslashes :( |
| 14:47.15 | *** join/#asterisk Samuel_CTX (~svanderp@62-213-207-59.colo.housingcenter.be) |
| 14:47.38 | [TK]D-Fender | catphish, Well upgrade first then provide new backup |
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| 14:50.11 | bobb_WU | will you confirm for me that tshark -R rtpevent is showing me RFC2833 packets? |
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| 14:57.14 | Samuel_CTX | has anyone experience with Xorcom Astribank 2 with EEPROM 2.1? |
| 15:01.42 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 15:02.34 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 15:03.34 | bobb_WU | jacc0: is there a major difference between RTP EVENT 60 and 62? |
| 15:04.38 | pabelanger | Samuel_CTX: try tzafrir_laptop |
| 15:05.15 | nny | I would like to have a context to dial two DAHDI channels and challenge/response the lines for answering the originating channel. (To accept the call, press 1). Does anyone have a basic framework or some app suggestions on how to do this? |
| 15:07.01 | *** join/#asterisk jm_s2s (~jm_work_s@static-72-64-129-133.tampfl.fios.verizon.net) |
| 15:07.55 | Gugge | nny: followme ? |
| 15:08.13 | jm_s2s | has anyone patched 1.4.x with qualify5.backport-to-1.4.patch? |
| 15:09.08 | catphish | i'm guessing my bug lies in this loop :) for (stringp = row[i], chunk = strsep(&stringp, ";"); chunk; chunk = strsep(&stringp, ";")) { |
| 15:09.11 | Samuel_CTX | pabelanger: thanks, I already PMed him. But got no responce yet. (and I kinda in a hurry since I should get our new 2 Astribanks wormking this today, and well, all the other Astribanks I always installed had EEPROM 1.1 or less. Now the 2 new ones are with EEPROM 2.1 and I'm unable to get the FPGA loaded. |
| 15:09.41 | Samuel_CTX | (Even with Dahdi Tools 2.6.1) |
| 15:09.58 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:10.07 | pabelanger | your best be is to contact xorcom directly |
| 15:10.15 | pabelanger | since it is their hardware |
| 15:12.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:14.30 | Katty | helllllllllloooooooooooooo nurse. |
| 15:14.31 | Katty | i mean tony. |
| 15:15.31 | nny | Gugge: been meaning to play with that and see how much it handles, derp, thanks |
| 15:19.59 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
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| 15:28.40 | catphish | does anyone know why mysql realtime data is split on semi-colon? |
| 15:30.04 | Qwell | catphish: Because that's the only way you can have multiple values for things like allow/deny |
| 15:30.13 | Qwell | err, allow/disallow |
| 15:30.19 | Katty | HI QWELL |
| 15:30.24 | Qwell | ohai |
| 15:30.30 | Katty | hugs on Qwell |
| 15:30.33 | catphish | Qwell: makes perfect sense, unfortunately there seems to be no way to escape it :( |
| 15:30.34 | Qwell | You were like *point*, right there |
| 15:30.48 | Katty | right.... there? |
| 15:30.57 | Katty | in gatlinburg? |
| 15:31.01 | Qwell | tn |
| 15:31.05 | nny | anyone have a suggestion on a 4 line ATA for some testing? Looking at the linksys spa400 right now |
| 15:31.06 | Katty | yes, yes i was |
| 15:31.13 | Katty | josh was in nashville |
| 15:31.38 | Katty | twisted. |
| 15:32.40 | Katty | so i was only about 2 hours from huntsville at one point |
| 15:32.43 | [TK]D-Fender | nny, by ATA the default assumption is FXS, as "T" = terminal. You need lines, or phones? |
| 15:32.46 | Katty | which would have been oh about...2am |
| 15:32.53 | Qwell | tsk, tsk |
| 15:32.56 | nny | [TK]D-Fender: lines |
| 15:32.58 | Katty | in the wee hours of monday morning |
| 15:33.13 | [TK]D-Fender | nny, AudioCodes MP-114 |
| 15:33.22 | coppice | Katty: I'm only 200ms from Huntsville |
| 15:33.26 | nny | [TK]D-Fender: long story short I have a client with ringback playng on lines and the contact tech needs to be shown that eliminating 90% of the analog chain will eliminate the issue |
| 15:33.40 | nny | (ringback playing on one line when another line is ringing) |
| 15:33.50 | Katty | cape is about...320ish miles |
| 15:33.55 | Katty | from huntsville |
| 15:34.04 | catphish | [TK]D-Fender: my issue still exists in the latest version, i'll patch it |
| 15:34.11 | Katty | i should go visit someday |
| 15:34.15 | catphish | Katty: thanks for the *pointer |
| 15:39.57 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v017-202.mobile.uci.edu) |
| 15:40.43 | urvg4 | hi all,all attempts to register my soft phone shows unauthorized |
| 15:41.04 | urvg4 | I checked the username/password still no joy |
| 15:41.12 | urvg4 | any help pls? |
| 15:43.17 | jm_s2s | catphish: do you know how to patch a file? I am trying to apply a patch to 1.4.26, qualify5.backport-to-1.4.patch |
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| 15:46.27 | nny | wonders why he's never payed with follow me before |
| 15:46.32 | nny | played* |
| 15:52.27 | catphish | jm_s2s: the patch command is uaually your friend, eg: patch < patchfile.diff |
| 15:52.39 | catphish | but im no expert on it, i usually use git or apply matched manually |
| 15:52.46 | catphish | *apply patches |
| 15:53.35 | jm_s2s | I tried patch chan_sip.c < xx.patch but I got some hunk errors |
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| 15:55.57 | catphish | im afraid i don't know, if its not too big, apply it manually |
| 15:56.10 | catphish | if there are actual problems though, you will struggle |
| 15:56.43 | catphish | what's the difference between realtime_mysql and realtime_multi_mysql |
| 15:59.49 | catphish | oh, one returns a whole config and the other just an array of fields |
| 16:00.21 | urvg4 | my soft phone shows unauthorized,have checked and rechecked the username/password |
| 16:00.29 | urvg4 | any help pls? |
| 16:01.08 | catphish | urvg4: please set the asterisk verbosity and paste anything that appears in the log while trying to register |
| 16:01.18 | urvg4 | ok |
| 16:04.06 | catphish | ooh, i don't need to escape semi-colons, i can just use the chunk decoder to my advantage when crafting the data |
| 16:04.27 | urvg4 | have about 8 concurrent calls on,can't pick out the stubborn sip phone |
| 16:06.23 | p3nguin | Is sipgate broken for anyone else today? I've had a couple thousand register attempts time out since around 2:45 AM Eastern. |
| 16:11.13 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:20.58 | catphish | anyone who cares: the solution to my semi-colon problem is "Call-Info: ^3Banswer-after=0" |
| 16:21.15 | catphish | it gets decoded to ; |
| 16:22.33 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-150-21.user.veloxzone.com.br) |
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| 16:24.40 | [TK]D-Fender | catphish, Well.. a hack-like way of making it work around the flaw.... |
| 16:25.04 | [TK]D-Fender | catphish, You seem to have pinpointed the real failure... did you end up submitting a better patch/report on the tracker? |
| 16:26.13 | p3nguin | same => n,SIPAddHeader(Call-Info:\;answer-after=0) <----- this works for me. |
| 16:26.27 | catphish | p3nguin: it doesn't work in mysql |
| 16:26.32 | p3nguin | I didn't read your initial complaint. |
| 16:26.49 | catphish | because in mysql realtime ; is a separator |
| 16:26.54 | catphish | with no escape option |
| 16:27.21 | catphish | i was going to code an escape character for ; |
| 16:27.33 | catphish | but then saw it accepts ascii codes |
| 16:27.41 | catphish | which seemed like a perfectly good solution |
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| 16:30.23 | catphish | i'll raise a bug for discussion anyway |
| 16:30.40 | catphish | but its unlikely that anyopne will care as the workaround is good |
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| 16:39.14 | nny | one last sanity check on my end here, I feel like I am confirming gravity, but here it goes |
| 16:40.18 | nny | I have a system where the user is complaining of ring tones playing on other channels. They answer one call, another comes in and they hear a very clear ring on the line (it over powers the caller on the first call). It is only audible by the callee. |
| 16:40.34 | nny | http://pastebin.com/aghQM1Yy and http://pastebin.com/6bMmFGKt |
| 16:40.41 | nny | are the two calls in tandem, seperated by channel |
| 16:41.16 | nny | my gut screams telco or line issue, I am going to bypass the telco with an ATA (while still retaining the card) but I just want to triple check that I haven't missed something trivial |
| 16:42.19 | nny | The phone (a linksys spa504g) doesn't play ringing on the callee end normally when a second call is sent to it, usually it's a beep stutter and/or a line button flash depending on configuration. |
| 16:42.31 | nny | It's also highly intermittent |
| 16:42.41 | nny | 4 test calls, happens once, same (known) local parameter |
| 16:43.06 | nny | excuse me the phone is a 508g |
| 16:43.22 | nny | thoughts laughter or condolences appreciated |
| 16:45.49 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:53.53 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 16:57.05 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 17:00.09 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 17:08.55 | *** join/#asterisk afink (~afink@wsip-70-184-213-65.om.om.cox.net) |
| 17:12.02 | dmz | AGI->set_music('on' is now my new best friend |
| 17:20.52 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
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| 17:39.43 | nny | resending my query to the channel, please see http://pastebin.com/zZyJ9Mne for details. Help appreciated |
| 17:40.00 | nny | if you've been here for a while, it should be recent in the chat history |
| 17:47.35 | talntid | hmm, that sucks |
| 17:54.23 | *** part/#asterisk ryduh (~ryross@ryduh.com) |
| 18:01.15 | leifmadsen | nny: polycom devices? |
| 18:01.16 | pabelanger | wow, nortel BCM suck |
| 18:08.31 | talntid | BCM400? |
| 18:08.56 | talntid | that brings back memories.... "must have java version 1.5!" |
| 18:09.16 | talntid | "and nothing later than update 13!" |
| 18:09.55 | p3nguin | Finally... sipgate has been restored! |
| 18:11.01 | *** join/#asterisk asterisk-Tester (~ramy@80.79.159.165) |
| 18:12.57 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 18:13.35 | bobb_WU | let's say someone else put some a spy group on an extension and made it where it would record calls to a samba share. where would i look to fix the call recording? |
| 18:14.26 | nny | leifmadsen: cisco508g |
| 18:14.57 | leifmadsen | hmmm |
| 18:15.29 | leifmadsen | I ran into a similar issue, and I ended up "fixing" it by changing the call waiting indication stuff on the polycom to silence |
| 18:19.21 | nny | yeah, I think my initial test is going to be straight VoIP this week to 1/2 the issue to either the analog or phone hardware side... I have 100 phones or so out int he wild, this is the first time I've seen it and it's really inconsistent and the ring isn't the normal incoming call on line 2 beep I am use to. I have at least experienced it first hand to know this isn't a ghost or misconstrued issue... |
| 18:19.45 | nny | it sounded like something siezed the channel and played a ring to it |
| 18:19.54 | nny | i couldn't hear the other end at all during each ring |
| 18:20.02 | nny | then again that could be volume differences |
| 18:32.12 | nny | leifmadsen: thanks for advice, i'll keep isolating each apsect till I nail it down and let you know what I find |
| 18:32.16 | nny | aspect* |
| 18:32.57 | talntid | the whole world is in on the prank. We look for your callerid, then play ringing tones while we speak with you :) |
| 18:41.25 | *** topic/#asterisk by mjordan -> #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.0 (2012/05/02), 1.8.12.0 (2012/05/02), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:49.10 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 18:51.26 | *** join/#asterisk timahvo1 (~rogue@41.90.177.215) |
| 18:51.59 | bobb_WU | anyone else joining the webinar? |
| 18:52.30 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:53.49 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 19:02.32 | sawgood | Where do I set the FROM address at in Asterisk, so all outgoing emails with voicemail attachements have the same FROM address (SMTP relay host) |
| 19:02.44 | *** join/#asterisk TheMan (~garry@wsip-24-234-165-140.lv.lv.cox.net) |
| 19:04.38 | pabelanger | sawgood: voicemail.conf |
| 19:06.49 | Qwell | sawgood: Asking a question in multiple channels is very rude. |
| 19:09.20 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
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| 19:19.23 | *** part/#asterisk talntid (~t@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 19:20.21 | sawgood | I understand, but I think this is a sendmail and asterisk thing now |
| 19:21.36 | sawgood | Thing is, I can make outbound emails go to @gmail.com, but any other domain the SMART HOST is not delivering the messages |
| 19:21.57 | sawgood | SMART HOST provider says I need to be sending from an official email address to get to non @gmail.com addresses |
| 19:22.01 | Qwell | #sendmail |
| 19:38.26 | p3nguin | Or forget about sendmail and use something a little more sane. |
| 19:41.49 | p3nguin | s/sane/sensible/ |
| 19:50.18 | *** part/#asterisk asterisk-Tester (~ramy@80.79.159.165) |
| 19:51.03 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 19:51.06 | SaRSAeOL | would this be the place to ask a question about asteriskNOW |
| 19:51.33 | Qwell | SaRSAeOL: Depends on your question, but #asterisknow would probably be most appropriate. |
| 19:51.43 | SaRSAeOL | awesome thanks Qwell |
| 19:55.18 | *** join/#asterisk eepr0m (~r0ute@2001:1af8:feff::ff0:bce:cad0) |
| 19:55.21 | eepr0m | hi guys |
| 19:55.46 | eepr0m | can anyone tell me a good softphone app, freeware =) |
| 19:56.56 | sawgood | Use CounterPath and/or Zoiper and you'll be ok |
| 19:57.13 | sawgood | It is amazing the things Zoiper can do for testing (G.729 and faxing) |
| 19:57.42 | leifmadsen | eepr0m: Zoiper or Jitsi |
| 20:02.14 | anonymouz666 | what kills the counterpath software is the damn dnd button |
| 20:06.15 | *** join/#asterisk damg (~damg@pD952B4B2.dip.t-dialin.net) |
| 20:07.24 | sawgood | DND is a feature you have to manually change/setup with CounterPath |
| 20:11.01 | damg | sip trace tells me that my (*) capabilities are 0x0, although I've tried various stuff including allow=all in sip.conf . the translation table is filled with times and the log reports at least ulaw and alaw as being loaded ( format_pcm.so, codec_ulaw.so ) |
| 20:11.15 | damg | is there anything I'm missing? |
| 20:12.34 | p3nguin | disallow=all, then on the next line allow=ulaw |
| 20:13.09 | SaRSAeOL | hugs Qwell |
| 20:13.12 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 20:13.19 | Qwell | SaRSAeOL: Bribes accepted. |
| 20:13.51 | SaRSAeOL | Qwell: bribes are BEFORE you solve my problem… nothing in it for me now ;) |
| 20:13.52 | *** join/#asterisk jsjc (~Adium@213.37.155.15) |
| 20:14.05 | SaRSAeOL | Qwell: next time i promise |
| 20:14.07 | Qwell | I never was very good at the payment thing. |
| 20:14.25 | SaRSAeOL | lol |
| 20:14.42 | Qwell | He'll be back. They always come back. |
| 20:14.52 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 20:14.58 | Qwell | I told you so. |
| 20:15.15 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-178-9.phlapa.fios.verizon.net) |
| 20:15.56 | p3nguin | Now this is really bizarre: I have a phone with a MAC address on the tag starting with 0013, so I have everything configured for the MAC address starting with 0013, but the phone didn't get the correct IP address. I checked the DHCP leases, and it shows that particular device has a hardware address starting with 02:13. HOW THE HELL IS THAT POSSIBLE?! |
| 20:16.33 | damg | i believe I've got a run-away problem with a context. I clashed allow=all into every section there and now it accepts codecs. |
| 20:16.42 | damg | p3nguin, thank you for the hint |
| 20:17.09 | p3nguin | It used to have a MAC address matching what is on the tag. How can it just change? |
| 20:17.30 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 20:17.33 | p3nguin | damg: Allow the codecs you need to use, not all codecs. |
| 20:17.38 | Qwell | p3nguin: The rest matches what you expect? |
| 20:17.42 | p3nguin | yes |
| 20:17.46 | Qwell | weird |
| 20:17.51 | p3nguin | Only the second char has changed. |
| 20:18.35 | p3nguin | I have a static dhcp entry in two dhcp servers for the original MAC address, and I know for a fact it used to obtain the reserved IP address. |
| 20:19.00 | *** join/#asterisk timahvo1 (~rogue@41.90.253.37) |
| 20:19.04 | p3nguin | Is it possible for an ethernet bridge device to mangle a MAC address of a connected device? |
| 20:19.39 | Qwell | maybe |
| 20:19.59 | p3nguin | I just can't imagine how the first byte of the MAC could change from 00 to 02. |
| 20:20.31 | p3nguin | I'll test the bridge theory. |
| 20:23.04 | p3nguin | OMG |
| 20:24.02 | p3nguin | I changed the phone to another bridge device and now it got the IP address I had expected it to have in the first place. |
| 20:24.17 | *** join/#asterisk timahvo1 (~rogue@41.90.253.37) |
| 20:25.13 | p3nguin | Now to change it back and see if it breaks again. |
| 20:25.31 | Qwell | are these wireless devices? |
| 20:27.14 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 20:29.06 | p3nguin | Yes, there's one wired segment on each side of a wireless link. The bridge joins the two wired segments. |
| 20:29.24 | Qwell | fun |
| 20:29.59 | p3nguin | I have never seen anything like this before. |
| 20:30.11 | p3nguin | I switched it back to the other bridge and the MAC is mangled again. |
| 20:30.25 | Qwell | must be a feature of the bridge |
| 20:30.50 | damg | p3nguin, I'll try to nail it down later, thank you for the hint. hopefully I'm the only pebkac in my experiment. |
| 20:31.32 | eepr0m | leifmadsen: thanks! |
| 20:31.38 | *** part/#asterisk TheMan (~garry@wsip-24-234-165-140.lv.lv.cox.net) |
| 20:32.22 | p3nguin | This is really messing with me. |
| 20:33.08 | p3nguin | Now I have to find out if I can duplicate the IP address for two MACs in the dhcp reserve. |
| 20:33.39 | bobb_WU | anyone still around from my questions about Mitel and asterisk problems with DTMF? i found the solution |
| 20:34.36 | pabelanger | which was? |
| 20:34.43 | bobb_WU | 'Force sending SDP in initial Invite message' needed to be set to 'Yes' |
| 20:34.52 | bobb_WU | that single option was the problem |
| 20:35.43 | bobb_WU | that was the setting on the SIP Peer Profile |
| 20:36.58 | *** join/#asterisk carloimperia (~carloimpe@151.16.161.72) |
| 20:39.58 | damg | is dancing around his table. |
| 20:41.55 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:52.05 | p3nguin | Okay, so I tested another device with this goofy bridge. |
| 20:52.32 | SaRSAeOL | luck? |
| 20:52.52 | p3nguin | The first byte of the device's MAC gets changed from 58 to 5a. |
| 20:52.57 | Qwell | lol |
| 20:53.01 | Qwell | that is amazing |
| 20:53.10 | p3nguin | It's a first for me. |
| 20:55.45 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
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| 21:04.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:06.44 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-daxiedyzzahfzttt) |
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| 21:13.36 | kenshin | hello |
| 21:14.22 | eepr0m | iehlow |
| 21:14.30 | kenshin | is there a predefined variable in the context of 'extension.con' that is called $INCOMING ? (where can I find the list of all the predefined variables ?) |
| 21:14.48 | Qwell | There are ${EXTEN} and ${CONTEXT} |
| 21:16.49 | kenshin | I'm trying to read an ext.conf I found in some forum
it speaks of ${INCOMING} but I see no where where it is defined .. |
| 21:19.14 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 21:20.27 | *** part/#asterisk eepr0m (~r0ute@2001:1af8:feff::ff0:bce:cad0) |
| 21:21.17 | ChannelZ | it might be implied |
| 21:21.45 | ChannelZ | Or just not there in the piece they showed. |
| 21:22.54 | ChannelZ | In any event it's not a 'standard' channel variable if that's what you're asking |
| 21:23.10 | ChannelZ | See the asterisk.pdf doc |
| 21:23.22 | *** join/#asterisk s[x] (~sx]@ppp59-167-157-96.static.internode.on.net) |
| 21:23.40 | ChannelZ | forgets the exact name now |
| 21:24.15 | kenshin | That was my question, thanks ChannelZ |
| 21:24.36 | kenshin | it seems that the file I'm reading was generated by AMP (of freepbx) |
| 21:26.52 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 21:27.59 | ChannelZ | Maybe I'm losing my mind but I thought there also used to be a 'channel-variables.txt' or somesuch type readme but I don't see it in the source now. Oh well. |
| 21:29.52 | p3nguin | There's no telling what kinds of variables FreePBX creates. That's why we don't deal with FreePBX here. |
| 21:30.24 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 21:31.29 | p3nguin | You can check if it is a global var by 'dialplan show variables' on the AstCLI. |
| 21:33.03 | kenshin | I was just stating, freebpx is freepbx :) |
| 21:34.35 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 21:36.42 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 21:45.49 | leifmadsen | ChannelZ: that information is likely on the Asterisk wiki now (wiki.asterisk.org) |
| 21:50.09 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:52.28 | fprior | Hi, I'm testing trunk between 2 * in LAN. If I don't use "fromuser=" option I've receive error "username mismatch, have <x>, digest has <y>" , If I use "fromuser=" option I lost CallerID. Any idea ? |
| 21:53.37 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 21:58.03 | [TK]D-Fender | fprior: "sendrpid=yes", "trustrpid=yes" <- BOTH sides. |
| 22:06.17 | fprior | [TK]D-Fender, thanks. only callerid(num) is received, CallerId(name) is "device" , is normal ? |
| 22:07.02 | [TK]D-Fender | Go prove what each side started with... also you should NOT have CID set in the peers |
| 22:09.15 | *** join/#asterisk s[x] (~sx]@eth589.qld.adsl.internode.on.net) |
| 22:13.10 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 22:18.55 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 22:25.08 | fprior | [TK]D-Fender, yes, callerid was missing in conf. Last doubt: is possible avoid fromuser option ? I've test insecure=port but no change. |
| 22:26.34 | p3nguin | Don't set a caller id. Don't set fromuser. Do set sendrpid and trustrpid. |
| 22:26.49 | [TK]D-Fender | fromuser shouldn't mess with the name |
| 22:26.53 | [TK]D-Fender | Show us the call |
| 22:27.02 | p3nguin | http://pastebin.com/Ag7tknm2 |
| 22:27.09 | [TK]D-Fender | and configs |
| 22:31.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:37.52 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-mrnyxvqeokyskswf) |
| 22:41.49 | *** join/#asterisk mhaddog (~mhaddog@z65-50-118-232.ips.direcpath.com) |
| 22:43.01 | fprior | http://pastebin.com/kW3fQbbN [note:if I add "fromuser" option call start successful] |
| 22:48.01 | *** join/#asterisk pavlz (~pavlz@unaffiliated/pavlz) |
| 22:48.44 | pavlz | hi |
| 22:48.58 | pavlz | one query |
| 22:49.39 | pavlz | it is possible to use asterisk + batman for untrusted mobile phones ? |
| 22:50.34 | pavlz | i tink that could be interestin' to see how the two ones work |
| 23:03.19 | *** join/#asterisk tunaru (bc1b3da7@gateway/web/freenode/ip.188.27.61.167) |
| 23:05.01 | tunaru | anyone has experience with GoIP gateways? The provider doesn't want to tell me the credentials for telnet to gain access to the running Linux instance. How to find the credentials? |
| 23:06.08 | pavlz | very strange |
| 23:06.36 | pavlz | i use telnet without any problem, and i got tiscali |
| 23:20.15 | *** join/#asterisk arnotixe (~arnotixe@190.131.183.143) |
| 23:21.20 | arnotixe | hi I have set up manager.conf and in "manager show settings" in CLI, I DO get Manager (AMI): Yes and TCP Bindaddress: 127.0.0.1:5038, however even after restarting asterisk, netstat -l shows no listening service on port 5038. Do I need configure somewhere else too, to use manager? |
| 23:22.36 | pabelanger | arnotixe, pb your manager.conf |
| 23:23.38 | arnotixe | pabelanger, http://pastebin.com/kXmUE91k |
| 23:24.29 | pabelanger | arnotixe, look right |
| 23:25.28 | arnotixe | yep but netstat -l shows nothing on that 5038... |
| 23:25.34 | p3nguin | manager show settings |
| 23:25.55 | p3nguin | after you've restarted it. |
| 23:26.02 | p3nguin | What does it report? |
| 23:26.26 | pabelanger | arnotixe, trying telnet 127.0.0.1 5038 |
| 23:26.27 | Nugget | telnet is eeeeeeevil! |
| 23:27.56 | p3nguin | Are you checking netstat with root? What does "lsof -i tcp:5038" show you? |
| 23:29.01 | *** join/#asterisk tekzilla (~jon@hmbg-5f765b27.pool.mediaWays.net) |
| 23:29.01 | arnotixe | pabelanger, just times out. |
| 23:29.05 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:29.11 | arnotixe | p3nguin, with root yes |
| 23:29.35 | p3nguin | You skipped my first question. |
| 23:29.40 | arnotixe | nothing on lsof |
| 23:30.01 | p3nguin | It is not listening if both netstat and lsof say it isn't listening. |
| 23:30.25 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:30.47 | *** join/#asterisk mazpe (~lesterm@li315-205.members.linode.com) |
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| 23:31.27 | arnotixe | manager show settings http://pastebin.com/6ALpgnmc |
| 23:32.23 | arnotixe | Strange. I'm away for a while, but any suggestions... |
| 23:35.02 | *** join/#asterisk angryuser_laptop (~angryuser@2a01:e34:ee09:8850:7dc5:d08d:add7:ee2c) |
| 23:36.37 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 23:45.54 | *** join/#asterisk mazpe (~mazpe@74.212.157.66) |
| 23:47.34 | mazpe | I'm trying to upgrade my polycom phones 550 to to the latest sip firmware and bootrom. I already downloaded the split file 4.0.2 RevB Split, which BootROM do i use 4.3.1? |
| 23:55.14 | pavlz | i don't know polycome phones 550 |
| 23:56.09 | pavlz | about mobiles phones you must to asks to GNUtoo-desktop which is on #lp-it and works on Replicant http://replicant.us |
| 23:56.43 | pavlz | anyway, it possible to do batman + asterisk |
| 23:57.13 | pabelanger | batman can do anything |
| 23:57.29 | pavlz | and this means freedom, p2p network, but before it's necessary to solve the problem of power safe mode |
| 23:58.12 | mazpe | mobiles phones? |
| 23:58.39 | pavlz | power safe mode is a problem for the batteries of mobile phones, they was not thinked to work for long time without shutdown the communication |
| 23:58.55 | pavlz | yes, mobiles phones |
| 23:59.24 | pavlz | bye bye TELCO's |