| 00:08.59 | nny | how would [2-9]xxx|[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[1-9]xxx prevent a polycom phone from dialing 10 digit numbers? Am i missing somehing? |
| 00:09.13 | nny | (I am trying to add 4 digit extensions) |
| 00:10.58 | carrar | Whats the first pattern you are looking for? |
| 00:11.24 | justdave | 2-9 followed by 3 more digits and it'll stop because that's first (guessing) |
| 00:17.22 | nny | carrar I want to add 4 digit dialing to the default dialplan |
| 00:17.34 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 00:17.41 | nny | i have tried it other ways |
| 00:17.58 | nny | [2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxx|[1-9]xx |
| 00:19.57 | carrar | Might study how digit maps work first |
| 00:22.57 | carrar | Check out the admin guide from their web site |
| 00:23.56 | nny | yeah I think the phone is prioritizing my strings, I may have it fixed |
| 00:24.29 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 00:24.56 | *** join/#asterisk x86 (~x86@i.am.leet.org) |
| 00:25.30 | x86 | anyone been having problems with Google Voice and Asterisk 1.8.0? |
| 00:26.16 | x86 | It was working fine for both inbound and outbound calling for about a month, now when I try to make a call, it says "Remote peer reported an error, trying to establish the call anyway" |
| 00:26.31 | x86 | and it reports ringing, but it doesn't terminate the call... other end never rings |
| 00:26.54 | WIMPy | That's a known issue. |
| 00:27.08 | x86 | is there a fix? |
| 00:27.34 | WIMPy | Not sure. You could check issues.asterisk.org. |
| 00:27.49 | WIMPy | But I think it's still open. |
| 00:27.52 | x86 | weak |
| 00:28.11 | x86 | I mean, it was working fine... did something change on Google's side? |
| 00:28.23 | x86 | are they deliberately trying to block Asterisk or something insane like that? |
| 00:28.34 | WIMPy | yes |
| 00:28.42 | WIMPy | no idea |
| 00:31.44 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 00:43.18 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 00:44.57 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 00:46.38 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 00:47.20 | *** join/#asterisk lal00 (~eduardo@201.151.40.230) |
| 00:47.52 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:54.35 | mr_steve | x86: I was just looking at that issue, someone cooked up a patch, but I haven't tested it yet. |
| 00:54.41 | mr_steve | x86: https://issues.asterisk.org/view.php?id=18412 |
| 00:57.43 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 00:57.55 | x86 | thanks |
| 01:00.16 | x86 | mr_steve: you think that patch made it into 1.8.1? |
| 01:02.44 | mr_steve | x86: Nah, that patch was just posted a few hours ago |
| 01:02.59 | x86 | damn |
| 01:03.04 | mr_steve | indeed. |
| 01:03.25 | x86 | wonder if the patch will apply clean to 1.8.1, or if I'll have to run some SVN version of Asterisk or something |
| 01:05.24 | x86 | seems I can't get to the patch in links... it send the http request, but then nothing |
| 01:07.48 | x86 | w00t, it did apply mostly clean... a little fuzz but not much :) |
| 01:20.42 | *** join/#asterisk m_tadeu (~quassel@89.180.197.82) |
| 01:27.50 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 01:40.27 | mr_steve | Ah, success. I just built 1.8.0 for Ubuntu with that patch, and I can dial out again. yay |
| 01:44.17 | blee | Has anyone had issues with voicemail (that I am saving into mysql) reappearing as old? |
| 01:44.19 | blee | IE |
| 01:44.32 | blee | new message, I delete it, a day later, it reappears as an old message |
| 01:44.57 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 01:45.18 | blee | I honestly dont know how its possible, but i have a client who is complaining about it |
| 01:46.12 | blee | I have a suspicion that maybe shes just listening to voicemails and not deleting them |
| 01:46.14 | blee | but |
| 01:46.28 | blee | really im not sure |
| 01:51.01 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 01:52.36 | shmaltz | blee, what is the problem? |
| 01:52.53 | blee | I just have a client who says she deletes messages and they reappear as old a day later |
| 01:53.25 | shmaltz | ask her what button deletes messages |
| 01:53.33 | blee | haha |
| 01:53.37 | blee | 6? |
| 01:53.39 | blee | :P |
| 01:53.44 | shmaltz | blee I'm serious |
| 01:53.54 | shmaltz | you'll be surprised how stupid users are |
| 01:54.02 | blee | i sat in front of her |
| 01:54.04 | blee | and watched her do it |
| 01:54.06 | blee | but |
| 01:54.10 | blee | i just dont see |
| 01:54.13 | blee | how voicemails reappear |
| 01:54.18 | shmaltz | I once traveled 80 miles because none of the computers work |
| 01:54.26 | blee | so i bet shes just retarded |
| 01:54.28 | pabelanger | mr_steve: Ya, dvossel just looked at the issue and fixed it today :) |
| 01:54.38 | pabelanger | should make people happy again |
| 01:54.39 | shmaltz | blee, they dont when you watched she listened to the prompts and pressed 9 |
| 01:55.21 | shmaltz | when I got there I plugged them in and showed the lady that they were not plugged in, she said to me: I didn't wait for you to show up so you could make fun of me |
| 01:56.28 | blee | haha |
| 01:56.34 | blee | ifi was a fly on the wall.... |
| 01:58.41 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
| 01:59.11 | blee | Yeah im convinced shes just retarded..... |
| 01:59.16 | blee | how do i say this politely to her? |
| 01:59.17 | blee | heh |
| 02:00.49 | WIMPy | Just tell her that phones are still a relatively new invention and that these things can happen from time to time. |
| 02:01.30 | shmaltz | blee, tell her to get a rotary phone |
| 02:01.31 | *** join/#asterisk d-tech (~d-dtech@72.245.233.107) |
| 02:02.52 | blee | ll |
| 02:02.55 | blee | lol |
| 02:03.47 | joeyjones | blee: you have control of the voicemail server? |
| 02:03.54 | blee | yes everything |
| 02:04.04 | blee | Saved to our mysql cluster |
| 02:04.12 | joeyjones | then write a cronjob script to delete her voicemails everyday :p |
| 02:04.18 | blee | rofl |
| 02:04.26 | joeyjones | she can't complaing then |
| 02:04.43 | blee | shes the same lady that complained that she had to delete all her messages since she just listened to them in her email |
| 02:04.50 | shmaltz | joeyjones, those type of ppl ALWAYS have what to complain about |
| 02:05.03 | blee | shes like |
| 02:05.10 | blee | "it takes me 15 minutes to delete all my messages" |
| 02:05.11 | blee | uhh |
| 02:05.16 | blee | 7...6....7...6? |
| 02:05.17 | joeyjones | blee: set it to delete when is sends to email |
| 02:05.24 | joeyjones | *when it |
| 02:05.27 | blee | we just turned off her email notifications |
| 02:05.29 | blee | heh |
| 02:05.36 | shmaltz | I like this one: |
| 02:05.38 | shmaltz | pbx*CLI> show uptime |
| 02:05.40 | shmaltz | System uptime: 1 year, 33 weeks, 1 day, 2 hours, 26 minutes, 32 seconds |
| 02:05.41 | shmaltz | pbx*CLI> |
| 02:06.15 | blee | not bad |
| 02:06.59 | joeyjones | my vps has 300 days of uptime |
| 02:07.11 | blee | i restart our boxes every week :P |
| 02:07.26 | shmaltz | this one is a short one, I posted last week one on the list that was on for 611 days |
| 02:07.32 | shmaltz | blee, why? |
| 02:07.34 | joeyjones | it likelt went down when their billing system Fed me and decided i owed 6 months of bills + opverages, and they disconnected me |
| 02:08.04 | joeyjones | *likely |
| 02:08.08 | joeyjones | *overages |
| 02:08.19 | joeyjones | well, late fees not overages |
| 02:09.37 | joeyjones | i always look for unmetered servers... |
| 02:09.54 | joeyjones | i had a buddy that raked up $2k in overages ina month for a $150/month server |
| 02:09.59 | joeyjones | *in a |
| 02:10.02 | joeyjones | *racked |
| 02:10.18 | shmaltz | ~sleep |
| 02:10.18 | infobot | well, sleep is overrated, and a poor substitute for caffeine. |
| 02:10.27 | shmaltz | ~sex |
| 02:10.27 | infobot | [~sex] updatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; emerge --oneshot condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; emerge -C condom; make clean; sleep; |
| 02:12.51 | shmaltz | ~google asterisk |
| 02:25.25 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 02:47.07 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 03:03.11 | plut0 | anyone had success with fax detect on 1.8 with sip? |
| 03:05.45 | DeHackEd | I have it on 1.6... |
| 03:11.00 | *** join/#asterisk coppice (~chatzilla@m121-203-200-174.smartone-vodafone.com) |
| 03:26.18 | plut0 | DeHackEd: doesn't seem to work on 1.8 |
| 03:48.12 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 03:50.05 | *** join/#asterisk Siblor (~putthibon@58.136.24.13) |
| 03:56.26 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 04:21.28 | *** join/#asterisk luckman212_phone (~luckman21@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 04:24.30 | luckman212_phone | guys can I ask a g722 question? |
| 04:26.58 | luckman212_phone | er, more of a general codec preference question really |
| 04:34.28 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 04:34.31 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 04:39.21 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 04:50.04 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rclijsktvemeaypq) |
| 04:56.28 | *** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net) |
| 05:22.01 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 05:22.04 | *** part/#asterisk plut0 (~cory@cpe-74-76-182-29.nycap.res.rr.com) |
| 05:32.44 | sawgood | If I have an include statement inside of a context, will the include statement be 'processed' before any of the exten => entries? |
| 05:40.47 | *** join/#asterisk Leppers (~Leppers@69-161-26-211.static.acsalaska.net) |
| 05:48.29 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 05:53.37 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 05:56.48 | *** join/#asterisk the_5th_wheel (~edd@webster.cybertek.co.za) |
| 05:57.11 | the_5th_wheel | Morning all. I have a ss7 link Im trying to get working, and im getting Received CIC=17 for unequipped circuit (typ=RSC), link 'l1'. |
| 05:57.16 | the_5th_wheel | can anyone assist? |
| 05:58.15 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 05:58.22 | *** join/#asterisk Denial- (Denial@drgi.co.uk) |
| 05:59.50 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 05:59.50 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 05:59.50 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 05:59.51 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 05:59.51 | *** join/#asterisk zigzapgabe (~gabe@c-71-57-35-169.hsd1.il.comcast.net) |
| 06:00.26 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-ejhhflhuhqgndkoj) |
| 06:07.06 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 06:14.57 | *** part/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 06:24.15 | *** join/#asterisk sshock (~sshock@sshock.net) |
| 06:28.11 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 06:59.10 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:59.30 | ChannelZ | sawgood: yes, it's not really linear in that fashion |
| 07:02.37 | sawgood | oh ok |
| 07:03.08 | sawgood | I have an include => above all the exten => entries, but the context is not using it |
| 07:06.58 | ChannelZ | it should be |
| 07:07.13 | ChannelZ | unless there's maybe a bug with it being the first thing in a context perhaps |
| 07:10.36 | *** join/#asterisk SeTTleR (~bernd@p5DDECFAB.dip.t-dialin.net) |
| 07:13.43 | kaldemar | sawgood: short abswer to your original question: no. |
| 07:13.54 | kaldemar | s/abswer/answer/ |
| 07:13.56 | sawgood | ok ... |
| 07:14.37 | sawgood | how do I make a context 'leave out' to another context ... and then come back to the original context |
| 07:14.52 | kaldemar | in a context, first all extensions in most accurate match order, then includes in the configuration order. |
| 07:15.04 | p3nguin | You don't. I think extension matching and includes are covered in the book. |
| 07:15.15 | sawgood | ok |
| 07:15.29 | kaldemar | sawgood: applications Gosub and Return. |
| 07:15.48 | kaldemar | but an extension needs to match first. |
| 07:16.17 | kaldemar | sounds like you want an ubmrella context that just includes others. |
| 07:16.59 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 07:17.45 | mercutioviz | quick question - where on disk are the voicemail prompts stored - the ones that the user records? ("Hi this is Jim, plz leave me a message") |
| 07:18.37 | sawgood | Basically, I am looking for a way to have any outgoing call to an (800) number to go to a single context (that way I do not have to include all the 800 mappings in each context) |
| 07:21.01 | kaldemar | what do you have now and what are you expecting it to do? |
| 07:22.07 | sawgood | I have the 800 exten = > going to a different provider then the rest of the calls |
| 07:22.53 | sawgood | I would like any outgoing 800 call to go out via the exten => settings in a separate context |
| 07:23.53 | *** join/#asterisk Gwayne (~Gwayne@opensuse/member/gwayne) |
| 07:25.02 | *** join/#asterisk jsjc (~jsjc@32.Red-81-35-21.dynamicIP.rima-tde.net) |
| 07:25.21 | kaldemar | sawgood: make a context that includes two or more other contexts. one that has the 800 extension and one or more that take care of the rest. |
| 07:25.43 | *** part/#asterisk Gwayne (~Gwayne@opensuse/member/gwayne) |
| 07:26.10 | sawgood | So, in this new context (with two includes) ... there should be NO exten => statements? |
| 07:27.18 | jsjc | I been reading an error that seems quite common to some people with "BAD" sip providers. https://issues.asterisk.org/view.php?id=8091 and I wonder if this Regfailuretimeout = 60 is actually a valid parameter in sip.conf? |
| 07:27.49 | p3nguin | You could simply define an exten for _800NXXXXXX and be done with it. |
| 07:27.59 | kaldemar | sawgood: if you don't want anything to match before the includes. |
| 07:28.28 | p3nguin | The best match wins, so that will beat out a _NXXNXXXXXX |
| 07:28.34 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 07:28.36 | jsjc | yesterday I was trying to explain this isse here but my irc client died for some weird reason... |
| 07:28.40 | kaldemar | jsjc: no, it's not a valid parameter. |
| 07:29.15 | sawgood | kaldemar: very good point (have only includes) no exten => |
| 07:29.19 | sawgood | thanks I'll try that right now |
| 07:29.57 | p3nguin | My way would be less work, but whatever. |
| 07:30.52 | jsjc | kaldemar as mentioned yesterday the internet connection between my asterisk and sip provider gives some issues some times, and when drops out I try to recconect a few times then the sip provider returns me with Forbidden - wrong password on authentication for REGISTER... and asterisk stops trying to reconnect. If I reload sip starts working no problems. Is there anyway I could tell my asterisk |
| 07:30.52 | jsjc | to do not give up trying to register?? |
| 07:31.36 | mercutioviz | FWIW I found my files |
| 07:32.10 | kaldemar | jsjc: registertimeout parameter defines something like that, however. |
| 07:32.54 | kaldemar | jsjc: don't know how it behaves after a forbidden, though. |
| 07:33.33 | jsjc | after a forbidden it just stops... |
| 07:33.50 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 07:33.54 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 07:33.58 | kaldemar | jsjc: try registerattempts=0 if your version has the parameter, it should make asterisk try to register until the other end accepts. |
| 07:34.10 | spenguin[work] | hey ChannelZ |
| 07:35.01 | jsjc | kaldemar will try it again, but I had it like that with no change against forbidden (I will try again tho) |
| 07:35.03 | kaldemar | hm. the default seems to be 0 so unless you have something else defined, that won't help. |
| 07:35.59 | sawgood | kaldemar: what 'if' a match can be made in more than one context inside the include => contexts? |
| 07:37.12 | kaldemar | sawgood: includes are processed in the order they are in a context. the first matching extension will take the call. |
| 07:37.26 | sawgood | kaldemar: perfect! |
| 07:49.12 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 07:52.10 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 07:53.55 | the_5th_wheel | can anyone help with an ss7 issue? I have my link as INSERVICE, but the channels are RESET PENDING, and Im getting this error l4isup.c:4803 l4isup_event: Received CIC=25 for unequipped circuit (typ=RSC), link 'l1' |
| 07:55.38 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:55.57 | sawgood | kaldemar: seems to work as you described (have a context with ONLY include statements) ... and the includes are processed in order ... |
| 07:56.23 | sawgood | Now, is there anyway to continue this pattern IF I have a need for a exten => statement in the context? |
| 07:56.36 | sawgood | I only need one exten => statement (to set caller ID information) |
| 07:57.24 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:00.13 | sawgood | Or, is there a way to set caller id with the Set(CALLERID(all) function and not tie this to exten => Set(CALLERID(all) |
| 08:01.01 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:01.23 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:01.25 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:06.52 | sawgood | I would like to a way to set caller ID in the context (and then dump out to the include statements) |
| 08:07.56 | *** join/#asterisk Tim_Toady (~moi@79.103.38.73.dsl.dyn.forthnet.gr) |
| 08:10.40 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 08:10.57 | kaldemar | sawgood: hint: _X. and app Goto or do it multiple times. |
| 08:12.41 | sawgood | not sure what you mean when you say, "or do it multiple times" |
| 08:13.58 | sawgood | exten => _X.,1,Set(CALLERID(all)=company name <5555555555> |
| 08:14.09 | sawgood | If this line is set, the includes will not work |
| 08:18.04 | kaldemar | sawgood: if the call matches an extension, it will not match to any other extension in another context unless you make it go to that extension. |
| 08:18.23 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es) |
| 08:18.42 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:18.59 | sawgood | So, is there a way inside a context to 'set' CALLERID(all) without using the exten => command? |
| 08:19.20 | kaldemar | sawgood: do the caller id change in every context, i.e. multiple times, or change it somewhere else and then go to the correct context and extension. |
| 08:20.27 | kaldemar | sawgood: no, there is no way to set a caller id without an extension. a dialplan is nothing but extensions, contexts are just a way to structure them. you can't do anything to a call without extensions. |
| 08:21.17 | sawgood | So, I need to have a 2nd entry in the context which does something like this: exten => _X.,2,Goto (unknown variables) |
| 08:21.22 | joeyjones | sawgood: you could always put it as an included portion of other extensions |
| 08:21.46 | sawgood | oh good point |
| 08:22.06 | joeyjones | or a macro |
| 08:22.18 | *** join/#asterisk dimm (~miniadmin@unaffiliated/dimm) |
| 08:22.35 | joeyjones | but that would only be worth it if there's more than a couple line in common |
| 08:23.26 | kaldemar | sawgood: sure, for example Goto({IF(["${EXTEN:0:3}" = "800"]?800-context:other-context)},${EXTEN},1) or something like that. |
| 08:23.53 | sawgood | ty |
| 08:24.04 | kaldemar | that has a missing $ before {IF |
| 08:29.31 | kaldemar | and another $ before the expression (first [). |
| 08:29.40 | sawgood | working on it now |
| 08:35.02 | sawgood | exten => _X.,2,Goto(name of new context) |
| 08:35.09 | sawgood | is that a valid Goto statement? |
| 08:36.06 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:37.20 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 08:37.54 | kaldemar | sawgood: no. see core show application Goto. |
| 08:38.20 | kaldemar | and it is preferred to use n instead of 2 as the priority. |
| 08:54.11 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 08:54.15 | sawgood | kaldemar: I now have my calls coming into one context, setting the CALLERID, and then a Goto command sending things off to another context, but once it 'arrives' at the new context, it errors out |
| 08:54.28 | sawgood | mentioning this ... "sent to invalid extension in context" |
| 08:54.31 | kaldemar | pastebin... |
| 08:54.54 | kaldemar | show the config and a failing call. |
| 09:04.43 | *** join/#asterisk minsa (~minsa@c-24-5-121-157.hsd1.ca.comcast.net) |
| 09:09.55 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:12.03 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:12.09 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 09:12.47 | *** join/#asterisk fofware (~fabian@host91.190-136-209.telecom.net.ar) |
| 09:15.31 | Montys | sawgood: I Suggest that you explore the posibility to use GotoIf intead the Goto statement that you posted on this chat |
| 09:15.39 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:16.37 | Montys | I would make you dialplan more easier to understand |
| 09:16.45 | ChannelZ | I've been away so perhaps this was answered, but I'm wondering why not just do the 800 match by the extension instead of IFing anything |
| 09:17.44 | kaldemar | Montys: or goto with an IF function.. |
| 09:18.19 | Montys | ChannelZ: Well... I have the same case, I've been away so I not sure what is the current configuration, I gave an answer accoording to the few lines that I read |
| 09:18.24 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 09:18.28 | Montys | kaldemar: gotoif |
| 09:19.10 | kaldemar | Montys: is this a start for a yes/no competition? :) |
| 09:19.51 | Montys | kaldemar: let me create an example for you, this will take me a couple minutes |
| 09:21.26 | kaldemar | GotoIf($["${EXTEN:0:3}" = "800"]?800-ctx,${EXTEN},1:other-ctx,${EXTEN},1) vs. Goto(${IF($["${EXTEN:0:3}" = "800"]?800-ctx:other-ctx)},${EXTEN},1) |
| 09:22.01 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 09:22.23 | kaldemar | Montys: i think i know what i'm talking about. :P i just think that the second one is prettier. |
| 09:23.48 | Montys | kaldemar: http://pastebin.com/7DdPx96s |
| 09:25.46 | Montys | kaldemar: Pretty much depends how you want to do it. In my personal opinion I like to separate things so It can be easy to read, if you prefer to have everything on a single line, it's ok too |
| 09:28.28 | kaldemar | it's pretty much a matter of how used you are to looking at those brace hells. |
| 09:31.10 | *** join/#asterisk bjornts (~BTS@2001:700:200:10:c62c:3ff:fe16:a044) |
| 09:31.15 | *** join/#asterisk datalay (~datalay@217.131.39.154) |
| 09:31.22 | Montys | kaldemar: what is the reason that you use the goto? |
| 09:32.22 | Montys | (I just want to understand more your dialplan, I think that there is a possibility that I could do it easier) |
| 09:32.38 | kaldemar | Montys: not having to repeat ${EXTEN},1 |
| 09:33.41 | Montys | ? |
| 09:34.12 | Montys | are you dialing out or receiving a call when asterisk uses this context |
| 09:34.13 | Montys | ? |
| 09:34.14 | kaldemar | take a look at my earlier example. |
| 09:35.36 | Montys | I think ChannelZ was right |
| 09:35.48 | Montys | it better to use extention matching |
| 09:35.58 | kaldemar | read back, i was just giving sawgood examples. |
| 09:36.13 | datalay | i have analog phone santral, we have internal extensions, i want to configure a remote office for reaching our internal extensions with analog phone, |
| 09:36.21 | Montys | yeah but that does not work for all the 800 numbers |
| 09:36.23 | kaldemar | by extension matching, you need to write the same line twice. |
| 09:36.28 | Montys | you need to take into consideartion |
| 09:36.43 | Montys | that the 800 numbers are: 800, 811 822 833 844 855 866 877 |
| 09:36.50 | datalay | can you give keyword about my problem |
| 09:36.56 | datalay | what i need |
| 09:37.11 | Montys | so you will need to check for easy possible contamination |
| 09:37.42 | kaldemar | Montys: that was an example, he got what he asked for. |
| 09:37.44 | Montys | s/easy/every |
| 09:37.58 | Montys | lol |
| 09:37.59 | Montys | ok |
| 09:38.31 | kaldemar | if someone says 800, i interpret it as 800, not a bunch of other numbers. :) |
| 09:39.04 | Montys | I see your point lol :P |
| 09:39.07 | kaldemar | datalay: what kind of connectivity do the sites have? |
| 09:39.44 | datalay | i have adsl internet connection |
| 09:39.53 | datalay | and i have pstn, analog line |
| 09:40.34 | datalay | two sides have adsl connection / pstn line |
| 09:40.39 | kaldemar | i suppose you want to connect them over ip? |
| 09:41.03 | datalay | goal is |
| 09:41.15 | datalay | if i get calls from other office with analog calls |
| 09:41.21 | datalay | i dont know what i need |
| 09:41.37 | datalay | i can configure and use asterisk |
| 09:41.54 | datalay | if i get calls from other office with analog phones |
| 09:42.17 | datalay | other office should work like an internal extension |
| 09:42.50 | ChannelZ | Well that's just a matter of constructing your dialplan |
| 09:42.52 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 09:43.01 | Montys | datalay: Two Asterisk server + Two analogs Card + A good contection between those site + knowledge |
| 09:43.07 | kaldemar | ~thebook |
| 09:43.07 | infobot | thebook is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 09:43.18 | Montys | and that one ^^ :P |
| 09:43.27 | kaldemar | the book has examples on connecting two systems with SIP or IAX2. |
| 09:44.57 | datalay | what do you think about this product: http://www.alpagut.com/default.aspx?pageID=18&PID=1074 |
| 09:45.06 | *** join/#asterisk m_tadeu (~quassel@89.180.197.82) |
| 09:46.17 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 09:46.56 | kaldemar | datalay: it should work well with asterisk. |
| 09:51.21 | datalay | Montys, internal phone network is not using asterisk.. it s hardware product of karel |
| 09:51.42 | datalay | asterisk can talk with karel's extensions? |
| 09:54.17 | Montys | datalay: I going to assume that the Karel is the brand, and the phone is analog (and dont use any weird and/or property signaling) |
| 09:54.49 | Montys | .... so... Yes asterisk should be able to talk to the phone |
| 09:55.09 | Montys | datalay: Can you connect the phone directly to the PSTN? |
| 09:56.00 | kaldemar | datalay: do you have a karel PABX or just phones? |
| 10:02.03 | datalay | yes i cant |
| 10:02.17 | datalay | yes i can connect the phone directly to the PSTN |
| 10:02.33 | datalay | i have karel PBX |
| 10:05.00 | datalay | Montys, kaldemar i have internal extension. i dont want to connect directly to pstn line, |
| 10:05.03 | kaldemar | what kind of interfaces does the karel PBX have? it needs to have something to connect to an asterisk box. |
| 10:05.17 | datalay | i think it has IP/trunk |
| 10:05.27 | kaldemar | what protocol? |
| 10:05.39 | datalay | iam not sure about that i should ask |
| 10:06.10 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:07.01 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:07.07 | datalay | it s : http://www.karel.com.tr/karel/product_detail.jsp?id=74 DS200M |
| 10:07.09 | kaldemar | if it supports standard SIP or H.323, you should be able to connect it to asterisk. actually, if the both sites have such a PBX, you don't even need asterisk in between if connectivity is all you need. |
| 10:08.28 | *** join/#asterisk binbash_ (~peter@ip4da5c213.direct-adsl.nl) |
| 10:09.05 | datalay | other side doesnt have anything |
| 10:09.12 | datalay | it s new office |
| 10:09.21 | datalay | we didnt buy anything yet |
| 10:09.54 | *** join/#asterisk anebi (~anebi@91.207.191.17) |
| 10:10.27 | kaldemar | my turkish is a bit rusty. however that page only mentions VoIP, which gives no clues. |
| 10:11.50 | datalay | Linksys SPA3102 1 Port FXO 1 Port FXS VoIP Gateway can be a solution for me? |
| 10:11.56 | *** part/#asterisk anebi (~anebi@91.207.191.17) |
| 10:11.57 | datalay | because it s cheap, |
| 10:12.17 | spenguin[work] | hrm I cant express how much I hate cisco for the SPA3102 |
| 10:12.30 | spenguin[work] | still awiting on them for an update |
| 10:12.58 | spenguin[work] | well itll work fine if you arent in Asia or such and dont need callerid |
| 10:13.30 | coppice | and if you don't mind echo, or a high failure rate, or quirky T.38, or..... |
| 10:14.17 | datalay | dont i need a sip server?, or can i configure it point to point |
| 10:20.04 | *** join/#asterisk glam (~glam@113.88.224.170) |
| 10:42.26 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 10:44.56 | *** join/#asterisk Orbixx (~Orbixx@213.229.88.67) |
| 10:57.59 | *** join/#asterisk dimm (~guest@unaffiliated/dimm) |
| 10:58.01 | *** join/#asterisk ccesario_ (~ccesario@187.75.139.188) |
| 11:04.51 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 11:05.20 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:05.31 | AdvoWork | ive got 1 isdn line coming in, which then gets split into 6 channels, which our trixbox system running on asterisk uses. How can i tell if all the 6 channels are running ok, any ideas please? |
| 11:09.00 | *** join/#asterisk ningia (~gain@mail.ufficyo.com) |
| 11:14.08 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 11:26.51 | *** join/#asterisk tecnico (~tecnico@pdpc/supporter/active/tecnico) |
| 11:35.51 | *** join/#asterisk Marini (~Marini@80.90.80.78) |
| 11:35.55 | Marini | hello everybody |
| 11:36.00 | Marini | i have a serious problem |
| 11:36.06 | Marini | after some time my asterisk is telling |
| 11:36.15 | Marini | chan_sip.c:6102 add_content: Out of space, can't add anymore |
| 11:36.27 | Marini | and i cannot do calls, i have to restart asterisk manually |
| 11:36.35 | Marini | there is space on hard drive |
| 11:36.50 | Marini | there are plenty of those messages , just starting suddenly |
| 11:37.01 | Marini | did somebody have come accross this? |
| 11:38.42 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 11:39.46 | Marini | hello |
| 11:40.17 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:40.22 | Marini | asterisk svn 1.4, VN-branch-1.4-r274579 |
| 11:43.44 | Marini | hello |
| 11:43.49 | Marini | anybody here? |
| 11:44.31 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:07.27 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 12:10.08 | *** join/#asterisk ukine_work (~ukine@152-144.97-97.tampabay.res.rr.com) |
| 12:11.13 | Marini | hello anybody here, i have a problem |
| 12:25.26 | Marini | hello |
| 12:29.35 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 12:34.28 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 12:56.48 | *** join/#asterisk LorenzBarth (~Miranda@62.216.165.71) |
| 12:57.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:57.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:00.01 | *** join/#asterisk Hatrix76 (~Hatrix@164.Red-80-36-218.staticIP.rima-tde.net) |
| 13:02.15 | Hatrix76 | Hi, I just installed AstManProxy but for some reasons it will not work correctly. I can do the same without problems with the manager interface directly, with astmanproxy if I, e.g. send a Login record, I just get the name of the proxy and the version back and the socket closes. I am connected to asterisk (netstat shows it and the debug output from astmanproxy confirms that the connection is established). but I it just plain do |
| 13:02.16 | Hatrix76 | es not work, any ideas? |
| 13:04.41 | eduzimrs | I guess it's not possible to use Asterisk BLF function for |
| 13:04.42 | eduzimrs | queues... Can someone confirm that? I'm looking for that type of |
| 13:04.42 | eduzimrs | function with calling queues. I have Grandstream Yealink T26P phones. |
| 13:04.42 | eduzimrs | <PROTECTED> |
| 13:05.10 | eduzimrs | Ops Yealink SIP T26P |
| 13:06.23 | WIMPy | Hatrix76: I have no idea if it supports (different) users at all, but it is my understanding that there has been no need to use it any more for quite some time. |
| 13:17.10 | leifmadsen | Hatrix76: well AstManProxy is quite old and unmaintained afaik |
| 13:17.32 | leifmadsen | ya, the manager interface contains security now I believe |
| 13:17.44 | leifmadsen | russellb would know better, he wrote that chapter :) |
| 13:18.25 | Siblor | Hello, anybody know how to originate call and auto-answer when target phone accept the call? |
| 13:20.02 | russellb | i know nothing |
| 13:20.20 | Siblor | ;) |
| 13:21.16 | leifmadsen | Siblor: depends on the phone, but with Polycom's you can set a SIP_HEADER() to make it auto-answer |
| 13:22.19 | leifmadsen | like here: http://ofps.oreilly.com/titles/9780596517342/ch10.html#AdditionalConfig_id257169 |
| 13:22.30 | russellb | bam |
| 13:22.32 | leifmadsen | and I guess it's SIPAddHeader() you use |
| 13:25.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:27.01 | Siblor | leifmadsen: Thank you, I will look at this one. |
| 13:27.19 | Siblor | Asterisk support in my country said I could call to application instead of exten. He said agi should be able to solve this problem. |
| 13:28.54 | Siblor | What do you think? |
| 13:29.00 | kaldemar | Siblor: what is the answering part when the target accepts the call? |
| 13:29.56 | kaldemar | i.e. what is the overall scenario? you originate a call to a phone and connect it to what? |
| 13:31.50 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:34.06 | *** join/#asterisk jazzy-J (~jazzy-J@92.249.107.65) |
| 13:34.42 | Siblor | When target accepts call, Asterisk will auto-answer and start record the conversation between target phone and current dummy channel. |
| 13:36.15 | kaldemar | not much of a conversation... you can set app Record as the other end of the origination if that's all you want. |
| 13:37.17 | leifmadsen | sounds like a complicated scenario for simple functionality :) But sure, you could just using Originate() (or originate a call in some method such as call files, AMI, etc.) and upon answer connect that person to the Record() application |
| 13:37.52 | leifmadsen | Siblor: your wording made it sound like you wanted paging, but in fact that is not at all what you want :) |
| 13:40.44 | Siblor | The scenario is we don't have a physical phone connect to extension. Does it possible? // Do you means create application 'Record' and call to this application instead of exten? |
| 13:41.47 | kaldemar | you don't have to create the Record application, it already exists. :) and yes that is possible and very easy. and no need to complicate it with AGI. |
| 13:42.48 | kaldemar | you aren't trying to spy on someone with an auto-answering phone are you? |
| 13:43.33 | blee | Anyone know a g729 capable softphone for linux? |
| 13:44.59 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:45.11 | Siblor | This is internal PBX at my office. ;) |
| 13:45.45 | *** join/#asterisk reber (~reber@212-198-99-56.rev.numericable.fr) |
| 14:01.28 | *** join/#asterisk toxictrash (~toxictras@62.87.237.123) |
| 14:04.30 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.151.106) |
| 14:07.50 | *** join/#asterisk titter (~Justin@c-98-208-153-116.hsd1.fl.comcast.net) |
| 14:08.06 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 14:16.16 | *** part/#asterisk jpmcallister (~jpmcallis@200.242.28.231) |
| 14:17.19 | Katty | gooooooooooooooooooooooooood morning!!!! |
| 14:17.25 | Katty | IT"S FRIDAY! |
| 14:23.45 | phix | Katty: no it is Saturday |
| 14:23.59 | Katty | even better! YAH! |
| 14:24.04 | Katty | dances with phix |
| 14:25.03 | phix | :D |
| 14:25.13 | phix | does a lil jig then gets over it |
| 14:25.33 | Katty | pfft |
| 14:25.35 | Katty | it's friday! |
| 14:25.38 | phix | What's new $title Katty? |
| 14:25.39 | Katty | more dancing! less reservedness! |
| 14:25.53 | phix | heh, I need more alcohol for that |
| 14:25.58 | Katty | pfft. |
| 14:26.02 | Katty | you don't need alchohol. |
| 14:26.08 | Katty | you just need an awesome group of friends |
| 14:26.19 | phix | perhaps |
| 14:26.34 | phix | Will you be my friend Katty? :) |
| 14:26.44 | Katty | ^_______^ |
| 14:26.45 | Katty | of course! |
| 14:26.48 | phix | yay |
| 14:27.28 | phix | I will call you everyday! |
| 14:27.34 | Katty | :< |
| 14:27.34 | phix | gets his stalk on |
| 14:27.54 | Katty | i'm just gonna go stand over here now, kthxbai -----------------> |
| 14:28.01 | phix | heh |
| 14:28.07 | spenguin[work] | hello Katty |
| 14:28.12 | Katty | goooood morning! |
| 14:28.16 | Katty | hugs spenguin[work] |
| 14:28.38 | Hatrix76 | Katty, my dear bird lover |
| 14:28.40 | phix | kbai Katty, time for sleep now |
| 14:28.45 | Katty | phix: ninite. |
| 14:29.28 | phix | I have reached my quota for freaking out ppl today |
| 14:29.43 | phix | tomorrow is another day though |
| 14:29.49 | Hatrix76 | phix, a real freaker never reaches any quota! |
| 14:30.03 | Katty | it's only freaky the first time |
| 14:30.09 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:30.24 | Hatrix76 | goes to #debian-woman for some one on one action ... |
| 14:30.33 | dmz | howdy y'all; any recommendations for linux softphones? i'm trying linphone but it's very grany; even when using ulaw / alaw |
| 14:30.45 | phix | Hatrix76: I am only a social / occiational freaker, I don't have all of the skill set yet, but i am working on it |
| 14:30.45 | Katty | i like zoiper. |
| 14:30.51 | phix | any way, *gone* |
| 14:30.56 | spenguin[work] | i like snowball guns |
| 14:31.00 | phix | Hatrix76: I am there :P |
| 14:33.53 | *** join/#asterisk iulhk (iulhk@117.102.32.128) |
| 14:37.27 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 14:41.25 | Hatrix76 | hi, regarding my above question about the astmanproxy, we had very bad experiences with asterisk 1.4.11 - 1.4.25/27 regarding the AMI, it locked the system up (we use it for Click2Call, internal stuff, we have 150 agents in our callcenter). So the less connections I have the better, therefore I would like to use the astmanproxy ... is there anything else I could use? we have asterisk 1.4.37 now ... |
| 14:43.09 | *** join/#asterisk patrick^ (~patrick_@2001:470:b0ea:1:219:21ff:fe4e:f5de) |
| 14:43.13 | leifmadsen | I don't believe anything superceded astmanproxy other than newer versions of Asterisk such as 1.6.2 and 1.8 |
| 14:47.22 | tuxx- | yay, asterisk 1.8.1 |
| 14:49.33 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 14:50.30 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:56.08 | *** join/#asterisk jsjc (~jsjc@143.Red-83-42-187.dynamicIP.rima-tde.net) |
| 15:02.46 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:02.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:06.01 | Hatrix76 | leifmadsen, ... ok, thank you for your help ... |
| 15:06.31 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 15:08.14 | leifmadsen | tuxx-: and 1.8.2-rc1 will be out either later today or early next week |
| 15:08.39 | leifmadsen | tuxx-: has a couple of fixes in the 1.8 branch right now, such as Gtalk fixes and a crash with CDR and call parking |
| 15:08.53 | russellb | Google Voice outbound works again \o/ |
| 15:08.53 | tuxx- | nice |
| 15:08.57 | tuxx- | :-D |
| 15:08.58 | leifmadsen | \o/ |
| 15:09.02 | tuxx- | gj guise. |
| 15:10.03 | Nivex | I may have to upgrade to that, since I route my GV through my old Gizmo account, which is still down :( |
| 15:12.19 | *** join/#asterisk Tim_Toady (~moi@79.103.38.73.dsl.dyn.forthnet.gr) |
| 15:15.31 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:18.58 | *** join/#asterisk ccesario (~ccesario@187.75.139.188) |
| 15:30.45 | *** join/#asterisk coppice (~chatzilla@210.17.157.96) |
| 15:35.17 | *** join/#asterisk ctaloi (~Adium@nat-66-218-1-160.usadatanet.com) |
| 15:35.45 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:35.56 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 15:36.50 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:38.31 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 15:44.24 | *** join/#asterisk Frem (~jamesgeck@64.207.3.161) |
| 15:45.35 | Marini | hello everybody |
| 15:45.35 | Marini | <Marini> <Marini> i have a serious problem |
| 15:45.35 | Marini | <Marini> <Marini> after some time my asterisk is telling |
| 15:45.35 | Marini | <Marini> <Marini> chan_sip.c:6102 add_content: Out of space, can't add anymore |
| 15:45.35 | Marini | <Marini> <Marini> and i cannot do calls, i have to restart asterisk manually |
| 15:45.36 | Marini | <Marini> <Marini> there is space on hard drive |
| 15:45.36 | Marini | <Marini> <Marini> there are plenty of those messages , just starting suddenly |
| 15:45.37 | Marini | <Marini> <Marini> did somebody have come accross this? |
| 15:45.57 | seanbright | that's super annoying |
| 15:46.02 | seanbright | and i know annoying |
| 15:46.05 | *** part/#asterisk ctaloi (~Adium@nat-66-218-1-160.usadatanet.com) |
| 15:48.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:49.00 | telnettech | this is not an asterisk question but I need some communityfeedback.....Looking at purchasing an edgemarc gateway. what are some pros and cons of the 4550 and 5300 series models |
| 15:49.45 | seanbright | Marini: open an issue on https://issues.asterisk.org/ |
| 16:00.50 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:04.36 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 16:08.25 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:11.09 | phix | There's old friends and new friends and even a bear! |
| 16:16.01 | telnettech | anybody with some feedback? |
| 16:17.31 | *** join/#asterisk Evet (~Evet@unaffiliated/evet) |
| 16:18.43 | *** join/#asterisk Sorcier_FXK (~sorcierfx@unaffiliated/sorcierfxk) |
| 16:21.15 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 16:22.47 | timholum | does anyone know if there is a way to set a context wide variable? |
| 16:23.18 | seanbright | a global? |
| 16:25.31 | timholum | seanbright: not exactly, I would like to have everyone in one context's extentions start with 100 and all the one's in a different one start with 101 and that if anyone is in 101 if they dial 200 it dials 101200 and if people from 100 dial 200 it dials 100200 ext..., and I would like to only change one line when I copy the extention file |
| 16:27.00 | seanbright | hmmm, no, i'm not sure then. |
| 16:27.09 | WIMPy | Set the variable in a match-all in that context and then jump to the next. |
| 16:27.26 | WIMPy | Or set global variables with the name of the context in them. |
| 16:27.27 | leifmadsen | [100] |
| 16:27.30 | leifmadsen | include => dialing |
| 16:27.33 | leifmadsen | [101] |
| 16:27.37 | leifmadsen | include => dialing |
| 16:27.40 | leifmadsen | [dialing] |
| 16:27.56 | leifmadsen | exten => _XXX,1,Set(ForDialing=${CONTEXT}${EXTEN}) |
| 16:28.00 | leifmadsen | that may or may not work... |
| 16:28.22 | seanbright | interesting |
| 16:28.34 | leifmadsen | it might if you do: exten => _XXX,1,Set(Prefix=${CONTEXT}) |
| 16:28.37 | leifmadsen | then in [dialing] |
| 16:28.51 | leifmadsen | exten => _XXX,n,Set(ForDialing=${Prefix}${EXTEN}) |
| 16:28.55 | leifmadsen | you'll have to test |
| 16:28.56 | thehar | orly |
| 16:29.12 | leifmadsen | I'm not sure I'd approach it that way, but ya, that |
| 16:29.18 | thehar | excellent |
| 16:29.42 | seanbright | i use setvar=TENANT=XXX in sip.conf |
| 16:29.57 | leifmadsen | that could work too, and probably work better |
| 16:30.21 | timholum | that can be set on a per phone basis, correct? |
| 16:30.34 | seanbright | yes |
| 16:30.48 | timholum | I think that is what I will try, thanks |
| 16:31.17 | *** join/#asterisk russellb_ (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:31.17 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 16:31.34 | *** join/#asterisk DJClean (~djclean@unaffiliated/djclean) |
| 16:31.47 | *** join/#asterisk russellb_ (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:31.47 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 16:37.42 | *** part/#asterisk russellb_ (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:41.41 | *** join/#asterisk ukine_work (~ukine@152-144.97-97.tampabay.res.rr.com) |
| 16:42.42 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 16:43.33 | *** join/#asterisk sol326 (~studyofli@c-71-56-135-99.hsd1.wa.comcast.net) |
| 16:44.55 | sol326 | ~itsplist-us |
| 16:44.55 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:45.07 | gandhijee | ls |
| 16:45.39 | *** join/#asterisk ukine_work (~ukine@152-144.97-97.tampabay.res.rr.com) |
| 16:48.07 | *** join/#asterisk oDesk (~f@77.31.6.14) |
| 16:48.07 | oDesk | hello all |
| 16:49.04 | oDesk | i've decoded my DTMF in SaudiArabia and find that it starts by letter D and ends by C, i can't find this rule into /src/asterisk/main/callerid.c |
| 16:49.19 | oDesk | how i may add this rule |
| 16:49.51 | *** join/#asterisk jazzy-J (~jazzy-J@92.249.107.65) |
| 16:50.39 | oDesk | i mean how to add this condition |
| 16:50.55 | oDesk | the current conditions are as following paste |
| 16:51.01 | oDesk | !paste |
| 16:51.04 | oDesk | !pastie |
| 16:51.17 | WIMPy | ~pb |
| 16:51.17 | 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.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. |
| 16:51.48 | WIMPy | Hmm. Wasnt the UAE the first to have 100% iSDN coverage in the world? Now this... |
| 16:52.36 | WIMPy | hopes he doesn't get sued by Apple for pressing Shift a little too late. |
| 16:52.57 | oDesk | http://pastebin.com/U8ZxR7j2\ |
| 16:55.19 | oDesk | anyone ? |
| 16:57.42 | *** join/#asterisk ccesario (~ccesario@189-29-50-127-ac.cpe.vivax.com.br) |
| 17:00.29 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.137.107) |
| 17:07.54 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 17:19.21 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 17:21.48 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:22.35 | gandhijee | hey, would it be alot of work to get a channel driver from 1.6.1 to work in 1.6.2? |
| 17:24.56 | JerJer | heh? |
| 17:25.46 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 17:25.58 | JerJer | gandhijee: i hope you men 1.8 -> 1.6 |
| 17:26.07 | JerJer | mean * |
| 17:26.11 | JerJer | damn a key |
| 17:40.06 | *** join/#asterisk pithen (~pithen@mail.graphlogic.com) |
| 17:41.20 | pithen | My asterisk console is showing thousands msgs like "Registration from 'callcentric' ... failed - No matching peer found".. I've removed all references to callcentric from my sip.conf but cant stop these messages |
| 17:41.26 | pithen | any ideas where they're coming from? |
| 17:41.30 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 17:44.05 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:44.08 | leifmadsen | pithen: means that end point is attempting to register. The only way to stop it is to block that IP address. |
| 17:44.12 | gandhijee | no i mean 1.6.2 |
| 17:44.25 | gandhijee | i have a driver for a funky device that works in 1.6.1 |
| 17:44.38 | leifmadsen | gandhijee: there wouldn't be that much difference -- I suggest you try it. |
| 17:44.41 | leifmadsen | then you'll know |
| 17:44.45 | gandhijee | i did |
| 17:44.57 | gandhijee | thats why i am asking, i get some errors about not beable to read the tone |
| 17:44.57 | leifmadsen | we can't tell you how hard it'll be as we have no knowledge of the channel driver |
| 17:48.15 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 17:50.34 | pithen | @leifmadsen: thank god for iptables, it stopped... |
| 17:50.41 | leifmadsen | yep :) |
| 17:50.53 | leifmadsen | probably would have been automatic if you were using fail2ban :) |
| 17:51.28 | pithen | i may just do that...not sure why callcentric would have suddenly started flooding me like that |
| 17:52.27 | oDesk | my DTMF callerid starts by letter D and ends by C, i can't find condition into /src/asterisk/main/callerid.c how i would add it |
| 17:52.29 | oDesk | http://pastebin.com/U8ZxR7j2 |
| 17:52.39 | russellb | ~fail2ban |
| 17:52.39 | infobot | somebody said fail2ban was a program to ban people using iptables based on information in logs: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk. For information on setting up fail2ban with Asterisk, see the Security chapter of Asterisk: The Definitive Guide - http://ofps.oreilly.com/titles/9780596517342/ch25.html |
| 17:52.46 | leifmadsen | pithen: it may not actually be callcentric -- could just be someone trying to authenticate as callcentric |
| 17:52.59 | leifmadsen | russellb: +1 |
| 17:57.38 | oDesk | oh sorry seems it includes the D and C |
| 18:00.03 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 18:00.53 | *** join/#asterisk fofware (~fabian@host14.190-137-126.telecom.net.ar) |
| 18:01.02 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 18:02.25 | cmnky | shoecaca ... i seek the great white pstn |
| 18:03.46 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:06.44 | Jasnejac | ita all about the last mile |
| 18:06.50 | Jasnejac | its* |
| 18:07.19 | Jasnejac | blackpool has full wifi coverage |
| 18:07.56 | cmnky | "remote exploit in the wild against the Exim mail agent." |
| 18:09.13 | cmnky | just a heads up for exim users |
| 18:12.42 | thehar | link? |
| 18:12.54 | thehar | oh it's on /. |
| 18:12.55 | thehar | lol |
| 18:12.56 | cmnky | slashdot.org |
| 18:13.12 | *** join/#asterisk Tim_Toady (~moi@79.103.38.73) |
| 18:13.46 | p3nguin | aych tee tee pee colon slash slash slashdot dot org? |
| 18:14.34 | cmnky | squishy splashy nee nee nee |
| 18:14.38 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.140.238) |
| 18:15.13 | cmnky | waits for the news article about how "hackers on IRC are now speaking in code" |
| 18:18.47 | *** join/#asterisk Evet (~Evet@unaffiliated/evet) |
| 18:24.03 | Nugget | "The hacker group known as 'anonymous'..." |
| 18:24.06 | Nugget | those guys are everywhere! |
| 18:25.11 | cmnky | well, they are finally fixing BGP ... maybe that fix DDoS as well |
| 18:25.26 | cmnky | er..they'll |
| 18:26.37 | cmnky | you can detect a DDoS on the servers being attacked ... and filter it, but your incomming bwidth is still clogged, but outgoing is clear |
| 18:27.01 | cmnky | a UDP protocol to send an information packet to your ISPs "security advisory" server could be sent out |
| 18:27.25 | cmnky | ISPs would in turn then share the info, and the attackers could get shut down in much faster time than now |
| 18:28.30 | cmnky | that is ... if someone writes the RFC ;) |
| 18:31.55 | oDesk | module load app_cdr.so failed, how i can load this module ? |
| 18:33.27 | oDesk | asterisk/full #=> Module 'app_cdr.so' already exists. |
| 18:39.06 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 18:43.53 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 18:43.59 | angryuser | Hello |
| 18:44.06 | timholum | is there any simple way to have a user be able to change there voicemail pin code? ( preferably without asterisk realtime )? |
| 18:44.45 | angryuser | Can someoen tell me if B2BUS reply 401 Unatorized instead of 403 Authorization required is SIP stadart possible ? |
| 18:45.25 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:49.50 | angryuser | ok found it, it is normal |
| 18:51.00 | *** part/#asterisk jazzy-J (~jazzy-J@92.249.107.65) |
| 18:52.53 | *** join/#asterisk marl_scot1 (~marl_scot@cpc1-cowc4-0-0-cust381.14-2.cable.virginmedia.com) |
| 18:53.08 | marl_scot1 | !seen whimpy |
| 18:58.42 | *** part/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 19:14.49 | *** join/#asterisk rrb3942 (~rbullock@208.34.96.186) |
| 19:18.51 | *** join/#asterisk Scorpio2007 (~Scorpio20@jose-tc.ctc.biz) |
| 19:22.53 | Scorpio2007 | exten => 551,1,Set(_PICKUPMARK=5512) |
| 19:22.53 | Scorpio2007 | exten => 551,2,Dial(SIP/SomeSipPhone&SIP/SomeOtherIpPhone) |
| 19:22.54 | Scorpio2007 | exten => *22,1,Pickup(5512) |
| 19:22.58 | Scorpio2007 | not working any help ? |
| 19:23.14 | Scorpio2007 | would like to ring 2 phones when an extension is dialed and i would like to hit *22 and pick up that call |
| 19:25.26 | *** join/#asterisk marl_scot1 (~matt@cpc1-cowc4-0-0-cust381.14-2.cable.virginmedia.com) |
| 19:25.33 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 19:30.48 | marl_scot1 | anyone use dahdi? |
| 19:31.41 | marl_scot1 | have it installed with a 4port beronet bri card, and dmesg is showing it as TE mode. but i need it in NT mode (it is connected to my telco), anyone able to point me to the correct setting to switch it over? |
| 19:33.38 | p3nguin | Dial(SIP/SomeSipPhone&SIP/SomeOtherIpPhone) is certainly the way to call two phones. |
| 19:36.08 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 19:36.57 | Docfxit | Hi, |
| 19:37.28 | marl_scot1 | hi Docfxit |
| 19:37.47 | Docfxit | What is the UDP timeout number in Asterisk? |
| 19:38.19 | p3nguin | Let's start with: What is a UDP timeout number? |
| 19:39.45 | leifmadsen | russellb: 580!!! |
| 19:39.48 | russellb | \o/ |
| 19:41.29 | Docfxit | p3nguin I have a softphone running behind a SonicWall. When a call connects after about 15seconds the called phone hangs up. The softphone doesn't hang up and waits. I'm wondering if the Sonicwall is dropping the UDP packets too soon. |
| 19:42.18 | Docfxit | I have a sip debug if anyone would like to take a look at it. |
| 19:44.39 | Docfxit | In the packet capture I see a number of packets for 5060 and a number of packets for 8000 |
| 19:45.50 | Docfxit | I am forwarding 5060-5080 and 8000-8020 |
| 19:50.56 | p3nguin | What's on 8000-8020? |
| 19:53.25 | p3nguin | Maybe it was just a random range you thought would be fun to forward? |
| 19:54.00 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 19:54.02 | leifmadsen | How important is DUNDi documentation to people? (i.e. should we include it in the third edition?) |
| 19:54.32 | cmnky | include anything that helps people understand how to run, configure, secure, and p0wn, asterisk ;) |
| 19:54.48 | p3nguin | i think if you've got it you should include it. |
| 19:54.51 | cmnky | when it gets too big ... split it into 2 books ;) |
| 19:54.51 | leifmadsen | cmnky: that is not the answer I was looking for :) DUNDi is not *required* to use Asterisk :) |
| 19:55.11 | leifmadsen | p3nguin: well the thing is it's out of date, so I'm curious if I should really spend my time updating it, or whether I should work on something else. |
| 19:55.16 | leifmadsen | January 1st is the due date |
| 19:55.23 | Docfxit | p3nguin RTP ports are 8000-8020 |
| 19:55.39 | cmnky | fails to read minds ... again {sigh} |
| 19:55.42 | p3nguin | That's a pretty weird range for RTP. |
| 19:55.47 | marl_scot1 | leifmadsen, as someone who is struggling to get dahdi working at this moment, YES INCLUDE the docs :) |
| 19:55.57 | leifmadsen | marl_scot1: DUNDi, not DAHDI |
| 19:56.09 | marl_scot1 | lol, sorry, miss read :( |
| 19:56.40 | cmnky | leifmadsen, if its a fringe topic and you gotta prioritize ... include whatever the most people / installations will benefit from |
| 19:56.50 | leifmadsen | *facepalm* |
| 19:56.53 | leifmadsen | that's what I'm asking :) |
| 19:57.04 | leifmadsen | I'm just curious how important DUNDi is to people |
| 19:57.06 | Docfxit | p3nguin That's what the softphone states. |
| 19:57.15 | leifmadsen | you can see how far we are and what is being included thus far at http://ofps.oreilly.com |
| 19:57.16 | cmnky | i dont even know what DUNDi is ;) |
| 19:57.21 | leifmadsen | well there ya go :) |
| 19:57.21 | marl_scot1 | could someone answer a quick one, that always confuses me? i have an * box with ISDN card, connected to my telco (BT), should the card be in NT or TE mode? |
| 19:57.47 | cmnky | perhaps if i could read about it, in a book somewhere maybe, id know ;) |
| 19:58.05 | p3nguin | docfxit: Maybe that's why it drops. If you don't have Asterisk configured for those ports (most people don't use those ports for RTP), there won't be any media and the SIP channel should close down. |
| 19:58.17 | Docfxit | p3nguin That's what is in my rtp.conf |
| 19:58.45 | p3nguin | Well that's weird, too, but at least it's consistent. :/ |
| 19:59.30 | Docfxit | p3nguin I have run this softphone on the Same LAN as Asterisk and it doesn't hang up. |
| 19:59.46 | *** join/#asterisk asilva (~structz@gandalf.ai.unesp.br) |
| 19:59.53 | asilva | Hi all |
| 20:00.02 | asilva | I'm experiencing something from the underworld |
| 20:00.11 | asilva | from nothing my asterisk dials to everyone and nothing shows on consoleeeee |
| 20:00.30 | asilva | everyone = sip users |
| 20:00.45 | Docfxit | p3nguin The problem happens when it's remote through a SonicWall TZ 210 wireless Nite |
| 20:01.35 | leifmadsen | russellb: 588! |
| 20:02.37 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:02.38 | Docfxit | p3nguin maybe you could tell what the problem is if you see the sip debug? http://pastebin.com/jhZsSC3K |
| 20:03.50 | p3nguin | Maybe, if you hadn't modified it before pasting. |
| 20:07.56 | Docfxit | p3nguin sorry. I typed it in. This is a paste - http://pastebin.com/jhZsSC3K |
| 20:10.25 | p3nguin | Same thing. A modified debug is something I am not interested in. |
| 20:11.54 | *** part/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 20:12.36 | Docfxit | p3nguin How can I hide the phone number and IP addresses to protect the innocent? |
| 20:13.51 | p3nguin | I guess you can use a private pastebin, that way only people here can see it. |
| 20:15.26 | atan | I see Google Voice has a price sheet up... can one use their Google Voice account to route tons of calls? Or is the ToS setup to allow only use by one user? =S I figure so long as someone is paying they should not care... but?? =D |
| 20:16.39 | Docfxit | p3nguin I will reproduce it and post back |
| 20:18.11 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:20.28 | *** join/#asterisk freddyk (~freddyk@host253-83-dynamic.3-79-r.retail.telecomitalia.it) |
| 20:23.40 | timholum | does asterisk config files basicly read them from top to bottom? so if I had exten => 200,1,Dial(Someone) and further down had exten => 2XX,1,Dial(adifferentperson) it would dial Someone? correct? |
| 20:24.34 | atan | I just switched to 1.8 and my command is not being called after a recorded call. exten =>_NXXNXXXXXX,n,Set(MONITOR_EXEC="/usr/share/php/cfrecordedcall.php", ^{TOUCH_MONITOR_OUTPUT}) |
| 20:24.46 | atan | Did this change in 1.8 or something? =) 1.4 called this script without any issue |
| 20:26.22 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.146.131) |
| 20:27.28 | *** join/#asterisk brezular (~brezular@adsl-dyn244.78-98-185.t-com.sk) |
| 20:29.53 | timholum | atan, I do know a few things have changed from 1.4 to 1.8, but I dont know if that is one of them |
| 20:30.07 | timholum | it looks like you have already change the | to , in it |
| 20:30.17 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 20:30.57 | timholum | I am in the proccess as well of setting up a 1.8 system, when all I have used in the past is 1.4 :) |
| 20:32.25 | atan | ...and my mysql CDR is all foobar'd |
| 20:32.28 | atan | =| =) |
| 20:32.57 | timholum | mysql CDR installed just fine for me on 1.8 |
| 20:33.37 | atan | That could be the issue: I didn't install it. |
| 20:33.40 | atan | My bad! =D |
| 20:33.54 | atan | I just grabbed source, configured, make, make install. Now I'm where I am now. |
| 20:33.56 | Docfxit | p3nguin This is the sip debug http://pastebin.com/pm4XQ0xy |
| 20:34.02 | timholum | make menuconfig |
| 20:34.33 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:35.14 | timholum | under Add-ons cdr_mysql |
| 20:35.36 | timholum | I beleve you need to have the mysql client and maybe mysql-client devel package's installed |
| 20:35.48 | atan | Got those. Good there. |
| 20:35.49 | timholum | if you have not installed them yet you will need to ./configure again |
| 20:36.31 | atan | timholum, so from the top, I'm in /usr/src/asterisk-1.8-stable/1.8/cdr now. What do I do? |
| 20:36.59 | timholum | you should just need to be in /usr/src/asterisk-1.8-stable/ |
| 20:37.10 | atan | Okay. In there I have a folder 1.8 with the files in it. |
| 20:37.11 | timholum | make menuconfig |
| 20:37.34 | timholum | this should bring up a menu of optoins |
| 20:37.36 | atan | ...WHAT IS THIS! |
| 20:37.48 | atan | On this is neat. A menu. Sweetness. |
| 20:37.54 | timholum | :) |
| 20:38.17 | atan | So in here I can't do anything with app_mysql. It's XXX |
| 20:38.31 | timholum | you must not have the mysqlclient installed |
| 20:38.50 | timholum | what os is it? |
| 20:39.07 | atan | Debian |
| 20:39.24 | atan | <PROTECTED> |
| 20:39.24 | atan | <PROTECTED> |
| 20:39.33 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 20:39.42 | atan | Let me try to apt-get that I suppose |
| 20:39.49 | timholum | I dont know the exact packages for debian, mine is centos, but install mysql-server mysql-connector-odbc mysql-devel mysql |
| 20:39.56 | timholum | then rerun ./configure |
| 20:40.08 | *** join/#asterisk mercestes (~mercestes@c-98-195-181-19.hsd1.tx.comcast.net) |
| 20:40.20 | atan | I don't want the MySQL server on this box. It connects to a remote server. |
| 20:40.32 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 20:40.34 | timholum | that should not be a requirement |
| 20:40.49 | timholum | I just basicly listed off everything on my system that had to do w/mysql |
| 20:41.02 | timholum | I beleve the mysql mysql-devel and mysql-odbc are the big ones |
| 20:41.21 | timholum | one sec, ill check on my ubuntu box to see what the correct package names are for debian |
| 20:41.29 | atan | http://packages.debian.org/search?keywords=mysql&searchon=names&suite=stable§ion=all just looking here now |
| 20:41.35 | atan | mysql-client might work |
| 20:41.59 | timholum | ibeleve you need libmyodbc |
| 20:42.01 | timholum | as well |
| 20:43.12 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 20:43.13 | timholum | and mysql-client |
| 20:44.45 | atan | While I wait for this to configure, do you know if the database table setup is still the same? |
| 20:45.15 | timholum | mine was |
| 20:45.30 | atan | I just installed those packs via apt-get then configured, and make menuconifig and still x'd out |
| 20:45.56 | timholum | hmm, try mysql-devel as well |
| 20:46.11 | atan | http://packages.debian.org/lenny/asterisk-mysql says http://packages.debian.org/lenny/asterisk-mysql |
| 20:46.13 | atan | err |
| 20:46.17 | atan | libmysqlclient15off |
| 20:47.05 | timholum | :) that might be what it is :) |
| 20:47.34 | atan | I don't see a mysql-devel here http://packages.debian.org/search?keywords=mysql&searchon=names&suite=stable§ion=all =\ |
| 20:49.28 | carrar | people still use mysql? :) |
| 20:49.36 | thehar | mysql > carrar |
| 20:49.47 | thehar | ;] |
| 20:49.49 | carrar | postgres > mysql |
| 20:51.04 | thehar | nah |
| 20:51.06 | thehar | disagree |
| 20:51.41 | carrar | everyone is entitled to their own wrong view point :) |
| 20:51.50 | carrar | heh |
| 20:51.53 | *** join/#asterisk ningia (~gain@host143-85-dynamic.11-79-r.retail.telecomitalia.it) |
| 20:52.27 | timholum | I prefer mysql just becouse it is what I have used for years :), and from what I hear postgress does not scale as well to multiple servers, but I dont have any experiance with postgress to say that for sure |
| 20:55.00 | timholum | atan: did libmysqlclient15off work? |
| 20:55.13 | atan | No. Still pecking away at the list. On to mysql-client-5.0 to see if it's any better |
| 20:55.27 | atan | did a make clean, still no go |
| 20:56.01 | timholum | hmm, that is dissapointing |
| 20:56.32 | atan | still says Depends on: mysqlclient(E) |
| 20:56.36 | atan | =( pitty |
| 20:57.21 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 20:58.04 | atan | Trying out libmysqlclient-dev for size |
| 20:58.23 | *** join/#asterisk Silicium (~marco@mail.2am.ch) |
| 20:58.25 | Silicium | hi there |
| 20:59.19 | Silicium | i searching for a simple IVR example for 1.6, i'am new at 1.6 and my 1.4 snippets does not longer work |
| 20:59.19 | atan | YEAH BABY! libmysqlclient-dev was it. |
| 20:59.28 | atan | hugs timholum |
| 20:59.45 | timholum | glade to here it worked :) |
| 20:59.54 | Silicium | i just need a simple default sound "press 1 or 2" and then play "you pressed one" or two |
| 21:00.13 | Silicium | google also gives me just 1.4 examples :/ |
| 21:01.15 | timholum | Silicium: 1.4 examples should work for you just change | to , |
| 21:01.48 | timholum | or at least it might work for you :), I dont have any ivr's at this moment to give you direct examples from |
| 21:02.01 | Silicium | i will try, thanks :) |
| 21:04.19 | p3nguin | 1.4 uses commas, too. |
| 21:04.58 | Silicium | i never used the ivr module, but i will try it now |
| 21:05.06 | Silicium | just hacked in extensions |
| 21:05.53 | timholum | p3nguin: did 1.4 use both then, becouse all of my extentions.conf use | |
| 21:06.00 | timholum | between variables |
| 21:07.11 | p3nguin | You could easily get the "press one or two" with BackGround(press&digits/1&or&digits/2). |
| 21:07.39 | atan | So it is setup with mysql I suppose, but t/mp/mysql.sock does not exist for it to use. Crap =D |
| 21:07.52 | p3nguin | If you are using pipe lines in 1.4, you probably brought the file over from 1.2 because I don't have a single pipe line in my extensions.conf on 1.4. |
| 21:07.55 | atan | And nothing is getting inserted to the database =D |
| 21:08.10 | Silicium | i dont want to use any db backend |
| 21:08.20 | Silicium | i junst planned to make it very simple |
| 21:08.29 | Silicium | pres 1 for forward yo my cellphone |
| 21:08.32 | Silicium | two for mailbox |
| 21:08.33 | Silicium | etc. |
| 21:10.52 | Docfxit | p3nguin Were you able to find anything or are you still looking at it? |
| 21:11.59 | p3nguin | When people insist on ruining a debug by trying to hide "secret" information, I typically stop caring and go on about my business. |
| 21:13.13 | Docfxit | p3nguin I posted a second debug for you. |
| 21:13.52 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.109.226.dsl.dyn.forthnet.gr) |
| 21:26.40 | atan | After 'make menuconfig' do you make, or skip to make install? |
| 21:27.03 | thehar | you do make |
| 21:27.12 | Silicium | re |
| 21:42.18 | Nugget | http://pastebin.com/V2QMXHgy <-- at least I've never had to do this |
| 21:47.47 | Silicium | back to the IVR one |
| 21:47.49 | Silicium | http://markmail.org/message/xhqkxm6d5g4cwk7f |
| 21:47.52 | Silicium | i tried that |
| 21:48.03 | Silicium | but it doesnt work because of the MOH |
| 21:58.08 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 22:14.46 | tvc123 | someone have a second to look a a piece of config? The problem that I am having is that when I call over a the sip trunk I am put in the default section of the extensions.conf and not the one I have defined as context in the sip.conf http://pastebin.com/C4VeLjcC |
| 22:16.04 | tvc123 | any hints as to what I need to change (I assume something in my sip.conf |
| 22:20.16 | *** join/#asterisk deeperror (~deeperror@adsl-76-226-157-245.dsl.sfldmi.sbcglobal.net) |
| 22:20.55 | deeperror | How is the random string created/called that is appended to the end of the channel name? |
| 22:21.22 | WIMPy | tvc123: Maybe it doesn't hit your configured friend and comes in as guest? Try to disable guests. |
| 22:22.08 | WIMPy | Well more than maybe :-) |
| 22:22.15 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.154.107) |
| 22:23.08 | tvc123 | ok that makes sense I'll try that |
| 22:23.16 | tvc123 | thanks WIMPy |
| 22:24.09 | *** join/#asterisk poison (~poison@78-21-102-160.access.telenet.be) |
| 22:30.06 | tvc123 | WIMPy: is there anywhere else that could be set besides the sip.conf? |
| 22:30.31 | tvc123 | after removing everything else from my sip.conf it still wants to come through with the default context |
| 22:31.20 | WIMPy | Be sure to have allowguests=no. |
| 22:33.01 | lal00 | is there a way to get the Call-ID sip header from an outgoing call? |
| 22:38.02 | tvc123 | WIMPy: still no luck ... I'll repaste my entire sip.conf one second |
| 22:39.37 | tvc123 | http://pastebin.com/uyct7wJ2 |
| 22:41.09 | tvc123 | lal00: wireshark? |
| 22:42.11 | lal00 | tvc123: I was thinking more along the lines of "how can I store the call-id with the cdr module" :) |
| 22:42.22 | WIMPy | tvc123: Nothing obvious to me. Try to turn up verbose and debug. Mabe that gives a clue. |
| 22:43.08 | tvc123 | WIMPy: thanks I'll keep looking |
| 22:43.28 | tvc123 | I included the sample configs when I built it so maybe it is burried in there somewhere |
| 22:44.08 | WIMPy | Didn't you say "entire sip.conf"? |
| 22:44.37 | WIMPy | The samples contain a lot of suff you probably don't want. |
| 22:48.37 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 22:51.20 | tvc123 | WIMPy: yeah I pulled everything else out of the sip conf just to check and then posted that file |
| 22:51.26 | tvc123 | so that is now my entire sip.conf |
| 22:51.58 | tvc123 | but it may be pulling over from some other file |
| 22:52.23 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:00.55 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 23:01.02 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 23:03.28 | tvc123 | WIMPy: is there a good locate to pull just the basic config files that are needed? |
| 23:06.37 | WIMPy | I don't see any includes in your paste so there shouldn't be any hidden surprises. |
| 23:06.59 | tvc123 | right I was just wondering if it was being set in another file some how |
| 23:07.59 | tvc123 | but it should end up with the context label in the extensions.conf right? |
| 23:08.01 | WIMPy | Some people write very simple configs as simple as possible, othes take the whole sample and check it line by line. |
| 23:08.15 | WIMPy | yes. |
| 23:08.52 | WIMPy | And I don't see how any call could end up in any but the two specified contexts. |
| 23:09.02 | WIMPy | But maybe I'm just too tired. |
| 23:09.26 | tvc123 | k yeah has me confused as well ... but I could be missing something |
| 23:10.57 | *** join/#asterisk caim (~user@unaffiliated/caim) |
| 23:15.42 | tvc123 | is there a debugging option that will show me the route a call is taking through the dialplan i.e. extensions.conf? |
| 23:16.20 | WIMPy | core set verbose 3 |
| 23:16.36 | tvc123 | OK I have core set to 9 |
| 23:16.46 | tvc123 | so that should be giving me what I need |
| 23:17.01 | tvc123 | I'm still baffled at how it is getting to the default context |
| 23:17.03 | WIMPy | Yes. YOu can also add debug. |
| 23:17.55 | tvc123 | ok I guess that makes sense ... although I don't know if that will help me decide why it is going into the default |
| 23:19.08 | WIMPy | Not sure. Might be yu have to use sip debug to get information on what it tries to match. |
| 23:19.42 | tvc123 | yeah I have sip debug on as well |
| 23:19.58 | tvc123 | really I only get that that extension does not exist in default |
| 23:20.07 | tvc123 | if I add it there or include it from there then everything works |
| 23:20.39 | tvc123 | just trying to decide why it goes there in the first place |
| 23:24.12 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-233-75-73.mia.bellsouth.net) |
| 23:32.32 | *** join/#asterisk engrxyz (~cbvvvvv@host81-150-217-173.in-addr.btopenworld.com) |
| 23:32.45 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 23:34.12 | *** join/#asterisk Fruchthoernschen (~Fruchthoe@77.13.159.158) |
| 23:40.48 | *** join/#asterisk mindCrime_ (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 23:42.48 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 23:43.30 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 23:49.27 | *** join/#asterisk UserReg_CL (~COB@pc-45-125-86-200.cm.vtr.net) |
| 23:50.45 | UserReg_CL | hi.. have one ivr and work fine, but have 1 ring and after sound ivr... need not ring ... thank |
| 23:51.46 | *** join/#asterisk ganeshix (~ganeshix@rrcs-24-213-243-188.nys.biz.rr.com) |
| 23:59.25 | *** join/#asterisk hhkahya (~gol@78.188.177.103) |