| 00:00.02 | drudge` | the readme.txt for asterisk 1.8.9.3 is telling me to check out doc/AST.pdf for secuirty info - im not finding AST.pdf or AST.txt anywhere |
| 00:00.05 | ChannelZ | well their they are blocking the return SIP traffic or something else is going on elsewhere in your network setup that those packets aren't making it back to you |
| 00:00.23 | sawgood | cool ... I appreciate your help and time! |
| 00:00.24 | drudge` | was it deprecated for teh asterisk-admin-guide.pdf thats in the doc folder? |
| 00:01.19 | ChannelZ | drudge`: maybe, it used to be in the doc folder |
| 00:02.03 | ChannelZ | sawgood: actually, what is your IP? |
| 00:02.52 | sawgood | which side, sir? |
| 00:02.57 | ChannelZ | yours |
| 00:03.15 | sawgood | well, I'm remote at 173.13.158.28 I belive |
| 00:03.23 | *** join/#asterisk cyborg-one (1000@212-178-1-144.broadband.tenet.odessa.ua) |
| 00:03.28 | volga629 | <ChannelZ>: I got prompt fixed just spent some money for proper recording and all working nice |
| 00:03.30 | ChannelZ | Their debug says it's coming from a 24.x.x.x IP but your registering using some 65.x.x.x IP |
| 00:03.45 | sawgood | right 24 = client 65 = the ITSP |
| 00:03.54 | ChannelZ | oh ok |
| 00:03.58 | ChannelZ | duh |
| 00:04.20 | sawgood | I'm setting up host=dyndns.org address now |
| 00:04.25 | ChannelZ | so then what's this 173 IP? |
| 00:04.38 | sawgood | I'm SSH into the equipment from 173.x.x.x |
| 00:05.00 | ChannelZ | your Asterisk box is the 24.* ? |
| 00:05.07 | sawgood | yes |
| 00:05.13 | sawgood | Asterisk at the ITSP too |
| 00:05.16 | ChannelZ | ok don't care about the 173 one then |
| 00:05.26 | sawgood | I figured so |
| 00:05.32 | ChannelZ | (your ITSP lets you SSH to them?) |
| 00:05.52 | sawgood | yes ... in this case ... |
| 00:06.55 | ChannelZ | Do you have externip or anything in your sip.conf? |
| 00:07.12 | drudge` | i'll re-dl the sourcetarball from asterisk.org, maybe it'll be there |
| 00:08.20 | ChannelZ | it's in my 1.8.7.1 dir anyway |
| 00:08.36 | sawgood | I think the ISP is blocking port 5060 on their OEM equipment ... |
| 00:09.00 | sawgood | even with a host=static IP (calls cannot arrive to the PBX) |
| 00:09.14 | sawgood | I cannot register a softphone either |
| 00:09.24 | sawgood | other port forwarding works 100% |
| 00:09.32 | drudge` | weird, that didnt have the AST.pdf or AST.txt in it either |
| 00:09.58 | ChannelZ | yeah I just downloaded .9.3 and it aint there |
| 00:10.43 | drudge` | maybe they just re-named it to that Admin guide, is that new? or was that in the others as well |
| 00:11.00 | ChannelZ | I just looked at both, yes that's what they did |
| 00:11.11 | drudge` | sweet, thanks ChannelZ |
| 00:11.11 | ChannelZ | Asterisk-Admin-Guide.pdf is what AST.pdf was |
| 00:11.43 | drudge` | this one client is using a@h with 1.2 and freepnbx 2.6, if you cna beleive it |
| 00:11.51 | drudge` | they wonder why they have "odd/weird/random issues" |
| 00:11.52 | ChannelZ | although it's quite a few less pages. hmm. |
| 00:12.10 | ChannelZ | Wow, 1.2.. kickin it oldschool |
| 00:12.13 | ChannelZ | Guess if it ain't broke.. |
| 00:12.42 | drudge` | well, all the freepbx upgrading they did - while leaving asterisk un-upgraded - caused random things to not work correctly |
| 00:13.33 | ChannelZ | well even when you're running the right versions FreePBX doesn't always work correctly |
| 00:13.37 | drudge` | Friday i showed them freepbx with that new GUI and they instantly froze up and their butts puckered up, lol was funny |
| 00:14.26 | drudge` | i assured them i wouldnt use that GUI since they are used to the freepbx 2.7 interface |
| 00:15.37 | *** join/#asterisk adeel (~adeel@72.53.72.133) |
| 00:16.04 | ChannelZ | sawgood: you cannot register a softphone from inside or outside the network? |
| 00:16.16 | ChannelZ | I assume outside |
| 00:16.52 | sawgood | outside |
| 00:16.56 | ChannelZ | They're probably blocking 5060 or doing something for their own voice service offering, can you use another port? |
| 00:17.05 | sawgood | neat question ... |
| 00:17.05 | volga629 | ChannelZ: Thank you for help and time. All working right now I will need do more staff like radio on hold, but it will be tomorrow |
| 00:17.23 | sawgood | can I put a port= (inside of sip.conf) |
| 00:17.55 | ChannelZ | volga629: sure have fun |
| 00:18.52 | volga629 | again thank you |
| 00:20.11 | *** join/#asterisk cyborg-one (1000@31.31.97.128) |
| 00:22.53 | ChannelZ | well you can put a port on the end of your externaddr but then that will break everything else connecting to you unless you change them |
| 00:23.09 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:23.17 | sawgood | I'm doing that now for testing |
| 00:24.45 | sawgood | its weird how SSH and other port forwarded ports work but not 5060-5069 and/or 7001 now (which is what I changed things too) |
| 00:26.40 | sbrath | ChannelZ: I gave up on the AudioCodes, I'm sending it back. |
| 00:27.11 | ChannelZ | well the way to really test it for sure is to shut down asterisk completely and use something like tcpcat to connect on those ports (udp) and see if they're really not making it to you |
| 00:27.35 | sbrath | is there anyway to force an include in a context in the dialplan to have higher preferece than s ? |
| 00:27.38 | sawgood | very good point |
| 00:27.39 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 00:27.55 | ChannelZ | but it sounds like you have a router in front of your linux box, and a modem in front of the router, so it could be getting chewed up anywhere |
| 00:28.14 | ChannelZ | not tcpcat.. netcat I meant |
| 00:28.22 | sawgood | ty |
| 00:28.51 | ChannelZ | sbrath: huh? |
| 00:29.07 | sawgood | I'm going to speak to Motorola soon aboubt the cable modem/gateway appliance |
| 00:29.21 | sawgood | maybe there is some SIP ALG engine running which is no fun! |
| 00:29.40 | ChannelZ | well you don't have a static IP at the moment, right? |
| 00:29.50 | sawgood | dyndns.org until tomorrow |
| 00:29.55 | ChannelZ | so your modem is doing NAT to your side of the network |
| 00:30.19 | sawgood | no nat from the modem (live IP) ... to a DD-WRT router (as DHCP between the two) |
| 00:30.30 | sbrath | I'm trying to get a routine to run in [macro-exten-vm] at the beginning of the call. |
| 00:30.40 | sbrath | It's on a freepbx setup :( |
| 00:31.10 | ChannelZ | so your modem is bridging |
| 00:31.22 | sawgood | yes, I hope so |
| 00:31.32 | sawgood | other port fowarding works ... |
| 00:31.54 | sawgood | I can access * through the Motorola ... then DD-WRT ... port forwarding to Linux/* |
| 00:32.28 | ChannelZ | well I'd try a quick netcat test on both sides then |
| 00:32.35 | sawgood | In progress now ... |
| 00:41.42 | ChannelZ | whistles the Jeopardy theme |
| 00:44.17 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 00:56.40 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 01:20.16 | p3nguin | sbrath: s isn't a preference, it's an extension. And macros use extension s. |
| 01:20.31 | p3nguin | And we don't know jack about FreePBX in this non-FreePBX channel. |
| 01:49.45 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
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| 03:36.09 | ChrisInSydney | I have just wasted 2 hours chasing a vm to email issue with Elastix and Exchange |
| 03:36.21 | ChrisInSydney | arghhhhh |
| 03:37.14 | ChrisInSydney | sbrath: Just been reading. Try on a basic Asterisk system. build a realy simple sip.conf and extensions.conf to test |
| 03:37.39 | ChrisInSydney | working in a FreePBX environment is just confusing |
| 03:38.53 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
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| 03:59.50 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 04:07.23 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
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| 04:47.40 | *** join/#asterisk Dovid (d508795a@gateway/web/freenode/ip.213.8.121.90) |
| 04:47.55 | Dovid | is there any stats of how many download of asterisk per country? |
| 04:51.39 | *** join/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 04:51.43 | mattwj2002 | hi all |
| 04:52.55 | *** join/#asterisk ipengineer (~ipenginee@cpe-66-25-39-35.tx.res.rr.com) |
| 04:53.01 | Dovid | hi |
| 04:53.14 | mattwj2002 | hi Dovid |
| 04:54.35 | mattwj2002 | maybe you could answer a questions for me that no one even acknowledged before...is there a way to search for what providers has what DIDs available? |
| 04:54.36 | Dovid | hi |
| 04:54.56 | Dovid | mattwj2002: depends in wha country |
| 04:55.00 | mattwj2002 | US |
| 04:55.34 | mattwj2002 | I am looking for a DID search engine of sorts |
| 04:55.41 | Dovid | not really |
| 04:55.52 | Dovid | in the US there is tnid.org/ |
| 04:56.00 | Dovid | where you can see who owns a sepcific number |
| 04:56.44 | mattwj2002 | I don't even care about a specific number |
| 04:56.51 | Dovid | localcallingguide.com will show you who was assigned what but who actually has the number is anyones guess. there is no guide to who owns what. if you want you can do a LNP dip on every number and see who owns what |
| 04:56.54 | mattwj2002 | just specific cities |
| 04:57.10 | Dovid | what are you looking to do? is there a specific DID that you want? |
| 04:58.25 | p3nguin | What area code and exchange are you looking to get? |
| 04:59.06 | mattwj2002 | 715-403 |
| 05:00.18 | mattwj2002 | oh I see level 3 has one of those exchanges |
| 05:00.37 | mattwj2002 | does level 3 have byod? |
| 05:01.22 | Dovid | level3 may have been assigned the range but numbers in that range can actually be with anyone |
| 05:01.34 | Dovid | Level wont deal with you <= 20,000 a month |
| 05:01.42 | mattwj2002 | :( |
| 05:01.46 | Dovid | maybe you can try some one that sells them |
| 05:01.50 | Dovid | like myphonecompany.com |
| 05:02.01 | Dovid | i know they used level3 when i worked with them years ago |
| 05:04.29 | mattwj2002 | okay cool |
| 05:04.31 | ipengineer | you can go with bandwidth.com they resell Level 3. We use them to get L3 numbers |
| 05:04.44 | mattwj2002 | nice |
| 05:05.09 | mattwj2002 | I appreciate your help guys....I was getting kind of mad at his channel |
| 05:05.22 | mattwj2002 | I was not even getting acknowledged :( |
| 05:05.27 | mattwj2002 | *this |
| 05:05.36 | p3nguin | ~ask |
| 05:05.36 | 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. |
| 05:05.45 | p3nguin | Note the last part. |
| 05:05.55 | mattwj2002 | :P |
| 05:06.03 | mattwj2002 | or against our will |
| 05:06.04 | mattwj2002 | :P |
| 05:07.19 | beardy | "Don't be demanding." is also a general rule. |
| 05:07.29 | p3nguin | Some ITSPs will say that numbers are carrier order only, and you basically have to have them put in an order with the carrier to even know if it's available. |
| 05:08.17 | Dovid | if Level3 owns the number then google level3 reseller or something like that |
| 05:08.26 | p3nguin | I looked in the list at VoIP.ms and they don't have that exchange listed. They don't even have any Ladysmith numbers at all listed, assuming there are other exchanges for that area. |
| 05:08.34 | Dovid | and as beardy says don't demand. people here don't like that |
| 05:09.29 | p3nguin | If you want quicker answers, you should ask during regular USA business hours. That's when there's the most activity here. |
| 05:09.48 | ipengineer | We have a carrier account through bandwidth. I just looked up that rate center for you they service both ELS and LI numbers to that rate center |
| 05:09.49 | Dovid | p3nguin: where r u located? |
| 05:09.59 | p3nguin | IL, USA |
| 05:10.10 | Dovid | isnt it 11:00 PM out there? |
| 05:10.16 | Dovid | on the weekend |
| 05:10.33 | p3nguin | Yes. 23.10, Sunday night. |
| 05:10.33 | Tyrael1 | p3nguin... where in IL, near Chicago by chance? |
| 05:10.43 | Tyrael1 | (from the burbs here) |
| 05:10.44 | p3nguin | southern part |
| 05:10.47 | Dovid | wow. an american that uses 24 hour time. impressive |
| 05:11.31 | mattwj2002 | :P |
| 05:11.45 | Dovid | here i thought i was the only one. i got in to the habbit when i worked in EMS. |
| 05:14.35 | ipengineer | Does anyone on here know when asterisk SCF version 1 is supposed to be released? |
| 05:15.15 | Dovid | never even had time to pay with it. from what i have read it will replace the need for Opensips for load balancing and redundancy? |
| 05:16.28 | ipengineer | Yes and create a seamless API for multiple servers. Each server can function as an individual component of asterisk (i.e. voicemail server, registrar server, etc) |
| 05:16.46 | Dovid | finally the evolution of asterisk has caught up |
| 05:17.08 | Dovid | no dumbed down docs on scf like there is for asterisk |
| 05:17.29 | ipengineer | Yes.. I am ready to start playing with it. I believe they have a beta version but no good documentation |
| 05:17.45 | Dovid | there is a basic wiki but no "real examples" |
| 05:18.05 | Dovid | i am not good with docs. better with examples. monkey see monkey do |
| 05:18.11 | ipengineer | I have seen a few examples but the wiki is terribly put together. it needs to be cleaned up |
| 05:18.23 | ipengineer | yep Im the same way.. |
| 05:18.28 | Dovid | anything on voip-info? |
| 05:19.03 | ipengineer | I haven't looked. Nothing comes up on google and I would think they would be one of the top results if so |
| 05:20.13 | Dovid | the average **asterisk expert** knows how to use a gui and click, click put in username 100 and password 100. sooo sad |
| 05:27.56 | mattwj2002 | Dovid: I have asterisk running on an openwrt router! :D |
| 05:28.18 | Dovid | mattwj2002: Your an exception to the rule |
| 05:28.36 | mattwj2002 | I am a hobbyist not a corporation though |
| 05:28.46 | mattwj2002 | thanks but I am no expert |
| 05:29.43 | mattwj2002 | I understand your point though |
| 05:31.20 | mattwj2002 | I think there is a push to get more retards into IT too....I hear commericals about get trained in IT on the radio all the time....they want to lower the cost of the profession |
| 05:31.30 | mattwj2002 | *getting |
| 05:35.10 | *** part/#asterisk mattwj2002 (~matt@wikisource/pdpc.active.mattwj2002) |
| 05:36.05 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 05:45.26 | [TK]D-Fender | Anyone who runs * off some premade GUI doesn't necessarily have any experiece with * at all... |
| 05:51.33 | p3nguin | SysAdmins are the same way. Everyone is looking for someone who can use, or install and use, webmin. |
| 05:51.42 | *** join/#asterisk odin_ (~Odin@93-97-168-38.zone5.bethere.co.uk) |
| 05:51.45 | p3nguin | It's lame. |
| 05:52.58 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 05:53.01 | p3nguin | You want me to work on your stuff, AND you want me to use webmin or freepbx to control it? I don't know how to use that shit because I don't have the *need* to use that shit. I actually know how to configure things. |
| 05:53.23 | learath | boring :) |
| 05:53.35 | learath | webguis are sexy! |
| 05:55.21 | p3nguin | Maybe in the same way manly lesbians are sexy. *shudder* |
| 05:58.09 | learath | is there any way to manually toggle mwi status from the CLI? |
| 06:03.12 | ChannelZ | you could use 'sip notify' to re-inform a peer probably, if that's what you're asking |
| 06:03.43 | learath | hmm that'll just blindly send a sip notify? |
| 06:04.05 | ChannelZ | no, read the usage |
| 06:04.22 | ChannelZ | I'm not entirely sure what you mean by "toggle mwi status" anyway |
| 06:04.40 | learath | I want to force message waiting indicator status as a test |
| 06:05.14 | learath | to see if I have the phone configured correctly |
| 06:05.40 | ChannelZ | see sip_notify.conf, make a copy of the example clear-mwi but set it to fake number of messages.. use 'sip notify' on the peer to send the fake message |
| 06:06.42 | learath | perfect |
| 06:06.57 | learath | do I have to reload anything, or is sip_notify.conf reread when requested |
| 06:07.53 | ChannelZ | Seems to be parsed on demand here anyway |
| 06:08.07 | learath | hmm |
| 06:08.40 | ChannelZ | or not. |
| 06:08.53 | ChannelZ | it's read when chan_sip is loaded |
| 06:09.00 | ChannelZ | so a 'sip reload' would be in order. |
| 06:09.42 | learath | yep, appears to be the case |
| 06:10.04 | learath | so set Messages-Waiting: yes and Content=>Voice-Message: 0/1 (0/1) or something like that? |
| 06:11.49 | ChannelZ | do 1/0 (0/0) |
| 06:12.13 | *** join/#asterisk ipengineer (~ipenginee@cpe-66-25-39-35.tx.res.rr.com) |
| 06:12.23 | learath | ok thanks |
| 06:13.43 | ChannelZ | new/old (new-urgent/old-urgent) |
| 06:13.55 | learath | ahhh ok |
| 06:14.05 | learath | I was guessing the other way around |
| 06:14.12 | ChannelZ | though it probably doesn't even matter, the device will probably light the light with any value |
| 06:14.35 | ChannelZ | or should anyway |
| 06:15.11 | learath | nice theory, sadly it's a Cisco, so it's Special |
| 06:15.14 | ChannelZ | guess it depends on what it thinks about old messages, even if they are 'urgent' |
| 06:15.47 | *** join/#asterisk ipengineer (~ipenginee@cpe-66-25-39-35.tx.res.rr.com) |
| 06:22.17 | *** join/#asterisk ayrjola (~ayrjola@89.18.236.11) |
| 06:43.19 | *** join/#asterisk damex|laptop (~damex@84.42.42.22) |
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| 06:45.22 | damex|laptop | hello, i was trying using asterisk for a while and started trying elastix now. after some testing i was found out that everything working except situation "asterisk not detecting situation, whan call ended by calling side (incoming calls) and continuing to ring till IVR menu" can anyone recommend me something to check? i can paste any configs out there. |
| 06:46.17 | damex|laptop | asterisk -rvvv shows that it continuing to play ivr menu recorded voice and then following timeout extension. |
| 06:58.49 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:59.20 | *** join/#asterisk roham (~ali@31.184.187.2) |
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| 07:44.09 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 07:44.23 | mirelab | hello :) |
| 07:44.30 | *** join/#asterisk robl^_ (~robl^@pdpc/supporter/active/robl) |
| 07:58.55 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 07:59.31 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 07:59.44 | mirelab | has anyone worked with HASH function? |
| 07:59.49 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:01.08 | ChannelZ | newp |
| 08:02.40 | mirelab | I'm wondering how to read next row when I Set(HASH(test)=<SQL query>) from that test |
| 08:02.54 | mirelab | documentation is weak :| |
| 08:04.08 | kaldemar | concentrace on the SQl query. does it give you more than one row? |
| 08:06.17 | kaldemar | if it does, the SQL function should give key-value pairs. they can be output in the dialplan with ${HASH(test,key)}. |
| 08:06.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:09.00 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 08:09.22 | ChannelZ | or you can just say "screw this noise" and write an AGI script in something else |
| 08:11.04 | mirelab | kaldemar: yes more rows |
| 08:11.10 | kaldemar | "mode" explains the multirow functionality in func_odbc.conf.sample |
| 08:11.58 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:13.58 | mirelab | kaldemar: http://pastebin.com/AfM10UG0 |
| 08:14.25 | mirelab | kaldemar: ok i'll read it now :) thx |
| 08:18.01 | *** join/#asterisk Dovid (~Dovid@office.mypbxmanager.net) |
| 08:19.06 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 08:19.08 | schmidts | good morning |
| 08:19.41 | mirelab | good morning :) |
| 08:19.54 | ChannelZ | Hi ho, hi ho |
| 08:20.52 | schmidts | wow its monday and you have such a good mood? what happend? |
| 08:20.53 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 08:20.55 | Dovid | morning |
| 08:21.12 | mirelab | always with good mood :P |
| 08:21.16 | *** join/#asterisk jzaw (~jzaw@macbook.dzki.co.uk) |
| 08:21.22 | Dovid | cause after monday comes tuesday, followed by wednesday then thursday and then FRIIIIIIIIDAY |
| 08:22.20 | schmidts | Dovid +1 |
| 08:23.52 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 08:23.55 | v0lZy | hello |
| 08:24.10 | v0lZy | Again, I turn to you in need of some advice |
| 08:24.15 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 08:24.44 | v0lZy | I have to define an outgoing pattern for phone calls |
| 08:25.04 | v0lZy | I successfully configured it to some degree so that I make internal calls directly... and for external calls, I press 0 |
| 08:25.32 | v0lZy | problem is, I have to send my ITSP <countrycode><number> |
| 08:25.40 | v0lZy | so... for example |
| 08:26.06 | v0lZy | I press 0 to dial out, then 44 for UK then area code then number |
| 08:26.16 | *** join/#asterisk wonderworld (~ww@dsdf-4db53135.pool.mediaWays.net) |
| 08:26.17 | v0lZy | but when i call within my own country |
| 08:26.28 | v0lZy | operator also expects me to put my own country code in |
| 08:26.47 | v0lZy | so.. 0, 386 for my country, then the rest |
| 08:27.04 | v0lZy | thing is, my users are used to dialing 01, 02, 03, 04 etc for different area codes |
| 08:27.05 | v0lZy | then number |
| 08:27.15 | v0lZy | so if my user dials for example |
| 08:27.42 | v0lZy | 02<phone number> i have to inject 386 in between |
| 08:27.55 | v0lZy | so 0|+386<whatever else> ? |
| 08:28.31 | ChannelZ | You know of ${EXTEN} yes? |
| 08:30.12 | ChannelZ | I'm bad at international dialing so I don't know what all these codes mean |
| 08:30.30 | v0lZy | its probably an askozia thing |
| 08:30.52 | ChannelZ | so say someone dialed 025555555 you need it to really dial 0386025555555 ? |
| 08:31.08 | v0lZy | mm... let me be a bit more specific |
| 08:31.14 | ChannelZ | please |
| 08:31.22 | v0lZy | theres a difference between what they dial, and what I really need to dial, yes |
| 08:31.29 | ChannelZ | sure, typical. |
| 08:31.36 | v0lZy | my users press 0 to initate calling outside my PBX |
| 08:31.59 | v0lZy | my provider requires that i always give a FULL number... like even if im calling within my own country, i have to provide my country code. |
| 08:33.09 | ChannelZ | exten => _0XXXXXXX,1,Dial(SIP/whatever/0386${EXTEN:1}) would take 05555555 that the user dialed and dial 03865555555 |
| 08:33.11 | v0lZy | the convention here is that we dont do that on normal lines so people are used to 2 things. First... when dialing within your city, you do not enter your area code, just the phone number u want to dial. second: when u dial to another city but within your country, you do a 0<areacode><phone number> |
| 08:33.38 | ChannelZ | you can apply the technique accordingly |
| 08:33.54 | v0lZy | then they are used that for outgoing calls (beyond our country), they do a 00<country code><area code><phone number> |
| 08:34.06 | v0lZy | now what I want to do is this |
| 08:34.14 | v0lZy | I want my user to have to press 0 to initiate outgoing calls |
| 08:34.16 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:34.26 | kaldemar | v0lZy: why? |
| 08:34.33 | ChannelZ | steering digits |
| 08:34.46 | v0lZy | im trying to mimic what we have |
| 08:34.53 | v0lZy | on our current ISDN |
| 08:35.02 | v0lZy | (my users are rather dumb) |
| 08:35.06 | v0lZy | and unadaptable |
| 08:35.20 | v0lZy | so... |
| 08:35.23 | mzb | so much easier to ignore the leading 0 method and work from local up to full international, then customising digits per outbound |
| 08:35.33 | ChannelZ | I've sort of answered whatever your question is |
| 08:35.50 | v0lZy | Well... in an ordinary scenario, if my user wanted to dial to a different city |
| 08:35.52 | v0lZy | they'd dial |
| 08:35.56 | kaldemar | drop the need to enter a 0 to dial out. then match 00 in the beginning for international calls and 0 to another area. otherwise, use the current area. |
| 08:36.09 | ChannelZ | ${EXTEN} represents the extension matched by the dialplan. You can use ${EXTEN:number} to skip over digits at the head, pull out whatever you want, and add any other digits you want |
| 08:36.15 | v0lZy | 02<phone number> where 2 is the area code and 0 is the national consesus for initiating calls external to your local area |
| 08:36.21 | v0lZy | so they will be dialing |
| 08:36.31 | v0lZy | 02 123 45 67 |
| 08:36.43 | v0lZy | or 03 123 45 67 |
| 08:36.52 | v0lZy | and if they want to call a mobile number |
| 08:37.01 | v0lZy | they go 040 123 456 for example |
| 08:37.10 | v0lZy | but if they dial outside the country |
| 08:37.13 | mzb | kaldemar: that sounds like a contradiction ;) |
| 08:37.23 | mirelab | v0lZy: yes as Kaldemar said read about asterisk patterns |
| 08:38.01 | mirelab | http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
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| 08:38.03 | kaldemar | mzb: not if you use patterns properly. |
| 08:38.08 | mzb | agreed |
| 08:38.18 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 08:38.27 | mzb | but as it reads I can see how it would be difficult to understand |
| 08:38.42 | kaldemar | 00Z vs. 0Z |
| 08:38.52 | v0lZy | if they dial outside the coutnry they want to do it wiht 00 <country code> <area code without the leading 0> <phoen number> |
| 08:39.13 | v0lZy | i have an aditional 0 in this system |
| 08:39.24 | v0lZy | AND ... my reception desk has an extension 00 |
| 08:39.28 | v0lZy | hence... my dilema |
| 08:40.02 | mzb | cat /dev/reprogramme > /dev/v0lZy |
| 08:40.11 | kaldemar | i don't see a dilemma there. |
| 08:40.12 | mirelab | Does anyone know if * can trigger an event when Peer is unreachable |
| 08:40.19 | mirelab | [Feb 23 19:05:25] NOTICE[4068] chan_sip.c: Peer 'DLGW05' is now UNREACHABLE! Last qualify: 41 |
| 08:40.29 | v0lZy | 00 > dial reception, 002 123 45 67 > dial to area code 2, 000385 45 678 890 > dial to croatian |
| 08:40.39 | v0lZy | mzb, lol |
| 08:40.45 | mirelab | so when that happens to trigger event that will send me sms or mail...etc |
| 08:41.09 | jacc0 | @mirelab: add the code you want to execute in chan_sip.c after line 4048 |
| 08:41.10 | v0lZy | this is how they want to dial |
| 08:41.16 | v0lZy | but my provider always accepts only |
| 08:41.25 | v0lZy | <country code><area code without leading 0><phone number |
| 08:41.44 | v0lZy | so if a user dials 0, then 02 , i have to send this as 3862<remainder> |
| 08:41.44 | ChannelZ | ${EXTEN:1} |
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| 08:41.54 | v0lZy | if they dial |
| 08:41.54 | mzb | ^^ note the :1 |
| 08:41.55 | mirelab | jacc0: uf i\m not yet ready to open the hood of * |
| 08:41.57 | ChannelZ | like I said 5 minutes ago |
| 08:42.06 | v0lZy | 000385 i have to send this as 385<reminder> |
| 08:42.10 | mzb | v0lZy: the ':1' will strip the leading 0 |
| 08:42.12 | v0lZy | remain* |
| 08:42.27 | mirelab | jacc0: and better not compile it with my knowledge of c++ "P |
| 08:42.28 | v0lZy | is there any other way to define patterens mzb? |
| 08:42.31 | v0lZy | like i am using askoziapbx |
| 08:42.35 | v0lZy | and it users a system with |
| 08:42.42 | v0lZy | !.|+NZX |
| 08:42.56 | v0lZy | its based on asterisk but has a gui. |
| 08:43.06 | ChannelZ | groans |
| 08:43.18 | ChannelZ | goodnight |
| 08:43.33 | mirelab | ChannelZ: night |
| 08:43.40 | mzb | listen to what the guys are telling you ... there's only a very small gap between your understanding and what you're being told |
| 08:44.22 | mzb | <ChannelZ> exten => _0XXXXXXX,1,Dial(SIP/whatever/0386${EXTEN:1}) would take 05555555 that the user dialed and dial 03865555555 |
| 08:44.31 | v0lZy | night ChannelZ |
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| 08:45.03 | v0lZy | mzb: but do i have to define multiple matching patterens' |
| 08:45.28 | v0lZy | i guess my dilema is is this a match in the form of first encountered digit.. or match of first encountered sequence of digits? |
| 08:45.29 | mzb | combination of leading 0 and length of the EXTEN should get you there |
| 08:45.48 | mzb | then it's all just order of logic |
| 08:45.58 | v0lZy | Can i have a patteren that matches 0[1-9] and a patteren that matches 00[1-9] and a patteren that matches 000[1-9] ? |
| 08:46.44 | v0lZy | I think im slowly getting it |
| 08:46.46 | v0lZy | hold on, let me try |
| 08:47.15 | mzb | when I get confused about something like this, I find it helpful to write or type a sample dialplan ... and then diagnose all the sample numbers I can think of until the dialplan is right |
| 08:48.18 | Dovid | mzb: As much as we just want you to listen to what we have to say, for myself i learn by example as well. same reason i never got in to SCF yet |
| 08:48.44 | mzb | you want me to listen? |
| 08:50.08 | mzb | no idea what you're talking about Dovid |
| 08:50.16 | mzb | good luck v0lZy, writing it all down will get you there eventually |
| 08:52.20 | v0lZy | ill try |
| 08:52.23 | v0lZy | thanks mzb |
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| 08:54.49 | Dovid | oops. i mized things up |
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| 09:57.22 | rjvvliet | Anyone how can help explain what could cause an T2_TIMEOUT with T.38 Fax receive using res_fax_spandsp |
| 09:58.37 | itsurkg | hello all, newbie here. I just install dahdi 2.5, asterisk 10 and asterisk-gui 2.1 on centos. i have a xorcom usb device with 8 fxo ports. Well i can see 8 ports to configure while executing dahdi_cfg -vv command but the problem is it says "no channel type registered for DAHDI" when i make outgoing call. Any help would be appreciate. |
| 10:01.36 | rjvvliet | itsurkg: not that familair with dahdi but, check the CLI for the DAHDI Channels available, just type "dadhi" than press TAB |
| 10:02.21 | rjvvliet | itsurkg: it looks like you only configured the dahdi drivers, not the channels for asterisk. |
| 10:03.04 | kaldemar | itsurkg: you don't have chan_dahdi in asterisk. |
| 10:03.31 | kaldemar | itsurkg: how did you install asterisk? |
| 10:04.11 | itsurkg | kaldemar, i compiled the source |
| 10:04.34 | rjvvliet | itsurkg: in what order did you compile? |
| 10:04.43 | itsurkg | i can't type dahdi in asterisk cli no command |
| 10:04.58 | itsurkg | first i compiled dahdi asterisk and asterisk-gui |
| 10:06.03 | kaldemar | itsurkg: see in "make menuselect" that you have chan_dahdi selected under "Channel Drivers". if not, select it and recompile. |
| 10:06.23 | rjvvliet | itsurkg: the no command means that chand_dhadi is not loaded, do the compile could be cone wrong as kaldemar already sugested. |
| 10:06.40 | kaldemar | itsurkg: first see what "module load chan_dahdi.so" gives you in CLI. |
| 10:07.16 | rjvvliet | itsurkg: look at the asterisk logs and be sure the compile of chan_dahdi when ok, check the menuselect if it has been detected. |
| 10:07.28 | kaldemar | itsurkg: you might already have the module but it won't load due to a config error. |
| 10:08.10 | rjvvliet | kaldemar: i think ill step aside ;-) you know mutch more of DAHDI. |
| 10:08.37 | itsurkg | Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: cannot open shared object file: No such file or directory |
| 10:11.52 | kaldemar | ok, that confirms you don't have it installed. proceed to make menuselect, enable the module and recompile. |
| 10:13.12 | itsurkg | kaldemar, thanks, i am on it. |
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| 10:21.47 | itsurkg | kaldemar, it shows XXX on chan_dahdi and i can't select. Any thing i am missing? |
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| 10:23.15 | pavlix | hello folks, is asterisk capable of prepaid credit billing? |
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| 10:24.06 | WIMPy | itsurkg: Did you configure after installing dahdi? |
| 10:25.35 | itsurkg | WIMPy, yes..i have already configured dahdi |
| 10:26.15 | itsurkg | do i have to remove all before re-compiling |
| 10:29.31 | kaldemar | itsurkg: no, you need to run asterisk's configure script after installing dahdi. |
| 10:30.55 | kaldemar | XXX indicates that you don't have or asterisk does not know about dependencies for a module. configure makes it search libraries or headers that it needs for modules. |
| 10:32.37 | itsurkg | kaldemar, got it. I can see chan_dahdi now it is selected by default |
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| 11:41.17 | pavlix | is asterisk capable of prepaid credit billing? |
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| 11:43.35 | kaldemar | pavlix: sure. but not out of the box. |
| 11:44.14 | pavlix | kaldemar: it was rather hard to google more information, I got more spam than ham |
| 11:44.57 | pavlix | kaldemar: is the backend billing part built in or is it some external plugin? |
| 11:45.16 | Dovid | pavlix: Asterisk can do what ever you set it to do. |
| 11:45.32 | Dovid | asterisk is like a pen and paper. you can draw what ever you want |
| 11:47.00 | pavlix | Dovid: so is C :) |
| 11:47.25 | kaldemar | pavlix: there is no built-in billing. asterisk prints call data records for calls, you must use something else to parse them and handle billing. |
| 11:48.21 | kaldemar | pavlix: you also need to handle balances in dialplan if you want them prepaid. |
| 11:48.32 | pavlix | kaldemar: ok but how would I stop running calls then< |
| 11:50.48 | kaldemar | pavlix: many ways to do it, which all depend on what the system is like. |
| 11:51.57 | kaldemar | pavlix: if only one call is allowed at a time for a certain customer, you can for example give a limit for the call length with option L() of app Dial. otherwise you'd need some nasty polling or similar. |
| 11:53.14 | pavlix | yep, the latter is what I'm afraid of |
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| 11:53.20 | kaldemar | some solutions here, don't know if they're any good: http://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications |
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| 11:55.29 | pavlix | thanks |
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| 12:07.13 | davidduffy | we been using a2billing and it works ok |
| 12:07.39 | davidduffy | quite a robust billing platform |
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| 12:47.45 | rjvvliet | Anyone who can help explain what could cause a T2_TIMEOUT with T.38 Fax receive using res_fax_spandsp and possibly how to solve it, also happens with res_fax_digium. |
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| 13:09.24 | zkn | Hi, how to interpret the following: chan_sip.c:19829 handle_response_invite: just did sched_add waitid(1011820) for sip_reinvite_retry for dialog QFib4qT9EIWAN2oSo4GYc4mLfNxPoyvu in handle_response_invite |
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| 13:21.06 | rjvvliet | Sorry to ask again:Anyone who can help explain what could cause a T2_TIMEOUT with T.38 Fax receive using res_fax_spandsp and possible solution,also happens with res_fax_digium. |
| 13:28.18 | coppice | rjvvliet: its usually no response from the far end. it could be you got some responses, but later communication hiccupped or stopped. |
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| 13:30.38 | rjvvliet | coppice: thans, it only happens with one sender,while is using the sam path through the TSP. must say, seems happens less often using spandsp then when i was using ffa not sure why.. |
| 13:31.30 | coppice | spandsp is more robust in some ways if you have quirky communication. if you have no communication at all it really doesn't matter which you use :-) |
| 13:31.31 | rjvvliet | coppice: i now have a wireshark running whole day, hoping to capture the problem, any spandsp debugging i can enable. |
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| 13:31.55 | v0lZy | hello guys |
| 13:32.03 | v0lZy | I have one more problem (hopefully, the last one) |
| 13:32.13 | v0lZy | I've configured how to send the numbers to my ITSP |
| 13:32.24 | v0lZy | It works as it should when i am dialing |
| 13:32.31 | v0lZy | but i get their number displayed when they callme |
| 13:32.37 | v0lZy | and if i hit redial, im missing the outgoing zero |
| 13:32.49 | v0lZy | was wondering if anyone has a solution to that? (its probably a phone thing?) |
| 13:33.55 | rjvvliet | coppice: Yep thats true, still searching where the connection is failing, the TSP tells me the do not generate the T.38 but their trunk provider does using Alcatel devices, the just pass it along using asterisk 1.4 Custom build. |
| 13:34.24 | kaldemar | v0lZy: add a 0 to CALLERID(num) for incoming calls. |
| 13:34.56 | kaldemar | v0lZy: that's one reason to drop the extra zero when dialing out. |
| 13:35.20 | v0lZy | hm... |
| 13:35.40 | coppice | rjvvliet: I haven't done much debugging against alcatels. I don't know if that means they aren't used much, or they just don't give much trouble |
| 13:36.11 | v0lZy | indeed i have that option kaldemar , thanks |
| 13:36.13 | v0lZy | lets see if it works |
| 13:36.41 | coppice | rjvvliet: if you turn on the debug logging spandsp gives quite a detailed log of what's going on. you might not want it on all the time, as it can really fill the logs on a busy system |
| 13:36.57 | rjvvliet | coppice: mmm, i'll wait and see what the capture brings. |
| 13:37.33 | rjvvliet | coppice: What command is spandspp using for the debugging, asterisk is set to verb=4 Deb=4 but fax set debug does not seem to work. |
| 13:38.16 | coppice | rjvvliet: I'm not sure how it is controlled these days, but there is a separate selection for FAX debug level |
| 13:39.10 | tzanger | coppice: got a moment? I have an audio question for you |
| 13:39.25 | rjvvliet | coppice: Yep, with the ffa module it wast setting the 'fax set debig on' of using the 'd' option on receivefax but nothing happens, unlus i'am really missing somthing. |
| 13:39.26 | coppice | OK |
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| 13:40.38 | coppice | rjvvliet: I have no idea how much FFA logs, but spandsp is quite detailed. you won't miss the log it produces :-) |
| 13:40.39 | tzanger | http://www.mixdown.ca/dump/rx.wav is a clip from the asterisk demo coming from a PRI card I'm developing... the audio "phases" in and outand I'm wondering if you have heard this kind of error before |
| 13:40.47 | rjvvliet | coppice: Also, i seem to have trouble forcing the fax maxrate to 9600, ffa does not houner it, and spandsp not always, still se the T38MaxRate 14440 comming by.. |
| 13:41.06 | tzanger | it sounds to me like I'm sampling incorrectly and losing bits |
| 13:41.34 | coppice | rjvvliet: T38MaxRate is handled by the module, not spandsp |
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| 13:42.58 | rjvvliet | coppice: a, oke, still finding out where the separation is in those modules,thought that part was de tech modules part. |
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| 13:44.23 | coppice | well, T38MaxRate is SDP stuff, so I assume it is handled in the common module |
| 13:44.25 | rjvvliet | coppice: THANKS..... just remembered something, there is a logger fax destination and spandsp seems to be using this |
| 13:44.48 | tzanger | the funny part is that if I emit a constant pattern the remote card sees it just fine with hexdump -C. patlooptest sees errors though. I'm getting the logic analyzer out today to see more clearly but was wondering if you'd heard that kind of audio error before |
| 13:44.49 | rjvvliet | coppice: whereas. ffa did not and logged to the debug channel |
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| 13:47.08 | rjvvliet | coppice:Thanks, I'll take all suggestions and restart examaning the logs. |
| 13:47.32 | rjvvliet | coppice: the fax log is 75M so got somthing to read ;-) |
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| 16:35.47 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 16:35.47 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.1.3 (2012/02/23), 1.8.9.3 (2012/02/23), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 16:37.12 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:38.26 | ollii | what do i have to activate to announce date and time of a recorded voicemail message which is retrieved via voicemailmain (1.8.9 and 1.4.42) ? |
| 16:39.30 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 16:39.52 | leifmadsen | ollii: pretty sure that was just enabled by default... |
| 16:40.24 | *** join/#asterisk wonderworld (~ww@dsdf-4db51711.pool.mediaWays.net) |
| 16:40.39 | cusco | leifmadsen: would you by any chance know if there is a way, on calling the Queue() app, disable the periodic_announce, or set its frequency ? |
| 16:41.41 | [TK]D-Fender | ollii, well documented in the sample config for voicemail.conf |
| 16:41.42 | leifmadsen | cusco: not off the top of my head, probably need to set frequency to zero or something |
| 16:41.45 | leifmadsen | check the documentation |
| 16:41.50 | [TK]D-Fender | ollii, its a box parameter |
| 16:41.52 | cusco | is there a var, that I can set? I tried the announceoverride option but I can't find info about that flag |
| 16:42.07 | cusco | Im reading https://wiki.asterisk.org/wiki/display/AST/Application_Queue |
| 16:42.18 | cusco | announceoverride doesn't say much |
| 16:42.34 | ollii | leifmadsen: yeah...but no it isnt...it tells me the callers cid: http://pastebin.com/N2ukcjTW |
| 16:43.02 | leifmadsen | shrugs |
| 16:43.17 | leifmadsen | cusco: check documentation via asteriskdocs.org as well |
| 16:43.17 | ollii | [TK]D-Fender: i'll check that...thanks |
| 16:43.18 | [TK]D-Fender | "message envelope" |
| 16:43.38 | p3nguin | blizzow: How did you get the one that you have currently installed? |
| 16:43.49 | ollii | yeah ok...the german translation of envelope makes no sense for me in this context ;) |
| 16:43.52 | ollii | i try that |
| 16:45.07 | *** join/#asterisk F|shie (~chatzilla@182.177.32.55) |
| 16:47.04 | blizzow | p3nguin: I believe it was compiled from source. (I inherited this system) |
| 16:47.24 | p3nguin | Then you'll have to build the correct (new) version in the same manner. |
| 16:47.53 | ollii | according to the sample voicemail conf, envelope should be on by default...i tried setting it manually to yes/no but no change at all |
| 16:48.02 | cusco | also in http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id288932.html there is no explanation on the flag announceoverride |
| 16:48.06 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.168) |
| 16:48.48 | b0ot | Is there anything that would allow asterisk call managers to communicate with one another and update each others extensions (trunks etc) when a remote call manager addes new extensions |
| 16:48.57 | b0ot | I thought it was called dhadi or something |
| 16:48.59 | p3nguin | ollii: After you edit voicemail.conf, you have to reload the voicemail module. |
| 16:49.08 | ollii | yeah i did that |
| 16:50.57 | [TK]D-Fender | b0ot, what is a "call manager"? "update extensions" , add extensions? HUH? |
| 16:52.13 | p3nguin | Sounds like human concept rather than something pertaining to asterisk. |
| 16:52.16 | b0ot | So if I have extensions 3001, 3002, 3003 on my local asterisk box and a remote box adds an extension 4001 my box would get a "trunk" where if I dialed 4001 it would know to send the call to the remote call manager |
| 16:52.46 | b0ot | I thought there was some sort of distributed system that could run on top of asterisk that would automatically update and sync information |
| 16:53.09 | [TK]D-Fender | b0ot, No, nothing of the sort |
| 16:53.27 | [TK]D-Fender | b0ot, This is a bunch of dumb text files, not some integrated system for a specific appliaction. |
| 16:53.58 | [TK]D-Fender | application* |
| 16:54.41 | b0ot | :( |
| 16:55.01 | b0ot | There is no way to have asterisk box's communicatae with one another in any automatic fashion? |
| 16:55.18 | b0ot | maybe it was called DUNDi |
| 16:55.20 | b0ot | or something |
| 16:55.50 | [TK]D-Fender | b0ot, What are you expecting? You haven't defined how "extensions" are even being added... or what you're usage of the term even means |
| 16:56.47 | [TK]D-Fender | b0ot, And for DUNDi, you must already specify the "trunk" to use. It just uses it as a search path for valid dialplan extensions. |
| 16:57.37 | [TK]D-Fender | DUNDi does not "advertise". When queried (by a call) it will make match if there is one. |
| 16:57.50 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 17:02.16 | blizzow | Is there a quick way to see the voicemail status (enabled vs. disabled) of all my extensions? |
| 17:03.14 | *** join/#asterisk myyrdin (~textual@74-93-34-252-Sarasota.hfc.comcastbusiness.net) |
| 17:03.57 | [TK]D-Fender | blizzow, "extensions" don't have voicemail |
| 17:04.34 | p3nguin | Book, anyone? |
| 17:04.37 | p3nguin | ~book |
| 17:04.37 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 17:51.36 | catphish | when receiving a fax with receivefax, do i need to do subsequent processing in extension h? |
| 17:52.02 | Qwell | Why would you? |
| 17:53.13 | rjvvliet | catphish: only i you would like to further process the received TIFF file, like emailing |
| 17:53.32 | catphish | that is exactly what i want to do |
| 17:53.44 | rjvvliet | catphish: take a look at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf |
| 17:53.44 | Qwell | Why would that have to be done in h? |
| 17:53.50 | catphish | right now i have an email script in the same extension after receivefax, but it often doesn't execute |
| 17:54.20 | catphish | i guess execution stops after receivefax because the remote hangs up at that point |
| 17:54.22 | rjvvliet | Qwell: because the fax processing continues in the h extension when the receive is successfull |
| 17:54.55 | catphish | unfortunately i have fax extensions in the same context as others |
| 17:55.02 | rjvvliet | catphish: the FFA quide has een perfect example in Receive and send faxing. |
| 17:55.35 | rjvvliet | catphish: best is to use e sepparete context. |
| 17:56.03 | catphish | yeah, that makes sense, will just be really complicated to set up in my environment :) |
| 17:56.07 | rjvvliet | catphish: i have an contaxt named [app-rx-fax] and use a goto after setting some VARs |
| 17:57.27 | rjvvliet | catphish: Complicated? when you use the normal extension so set some Vars and then use goto(app-rx-fax,s,1) |
| 17:58.03 | catphish | sure, that's fine, just means i need to create the app-rx-fax context |
| 17:58.12 | rjvvliet | catphish: is what i do, i just set the email adres on the real fax exten did. |
| 17:58.16 | catphish | not a huge job i hope |
| 17:58.30 | p3nguin | qwell: The call will die after the fax has been received, so most people deal with it in h. |
| 17:58.47 | rjvvliet | catphish: Yep, you have to do some work. hey come the softwrae is free ;-) |
| 17:58.57 | b0ot | Does cisco have anything like dundi? |
| 17:59.05 | Qwell | b0ot: ask them |
| 17:59.32 | b0ot | Well I was just interested to try to get a better understanding of what Dundi is used for |
| 17:59.44 | catphish | rjvvliet: i know, was just hoping for a workaround on this occasion :) |
| 17:59.55 | catphish | i'll make a context and jump to it, thanks |
| 17:59.57 | rjvvliet | catphish: Sorry ;-) |
| 18:00.27 | WIMPy | b0ot: To look up numbers, just like the name says. Not unlike ENUM. |
| 18:00.31 | Qwell | b0ot: http://ofps.oreilly.com/titles/9780596517342/asterisk-CHP-5.html |
| 18:01.09 | catphish | rjvvliet: i am responsible for http://atechtelecoms.com - cool product, unreliable fax receiving! |
| 18:02.21 | rjvvliet | catphish: looks nice, are u using spandsp or ffa ? |
| 18:02.23 | p3nguin | catphish: http://pastebin.com/6RQV9nEx |
| 18:02.29 | catphish | spandsp |
| 18:02.55 | catphish | thinking about it, it's not a big problem at all, the fax-receive context can be hardcoded and just use some variables |
| 18:03.15 | rjvvliet | p3nguin: thank also gave me some ideas |
| 18:04.17 | rjvvliet | catphish: oke, just fount that res_fax_spandsk does not honour FAXOPT(modem) setting :https://issues.asterisk.org/jira/browse/ASTERISK-16409 |
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| 18:05.00 | catphish | rjvvliet: thanks, right now i'm basically using defaults |
| 18:05.04 | catphish | http://paste.codebasehq.com/pastes/vnpzx7xpkm0hlcvchd |
| 18:06.02 | catphish | but i can move it out to its own context and pick up some variables, thanks p3nguin for the info on the other settings, will try some of that |
| 18:06.33 | rjvvliet | catphish: i'am working on a nice fax handler and also replacing a system() perl script for an AGI script |
| 18:06.37 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
| 18:06.59 | catphish | i don't even use perl, i just shell out to mutt |
| 18:07.33 | catphish | in fact my solution is quite similar to p3nguin's - it just needs some extending to handle failures |
| 18:07.47 | rjvvliet | catphish: a saw that, i wanted to give the user a little more info about the fax,also its converted to PDF. |
| 18:07.49 | catphish | and i like the tiff2pdf |
| 18:08.38 | catphish | the pfd you linked to has lots of useful looking settings |
| 18:08.47 | catphish | but i suspect in most cases the defaults are working |
| 18:09.39 | rjvvliet | catphish: looks like p3nguin does more in the dialplan then me, I started using scripts to soon i see.... |
| 18:09.59 | catphish | i like the dialplan method as my dialplan is very dynamic |
| 18:10.27 | p3nguin | I do everything I can in dial plan since I don't do programming. |
| 18:11.08 | catphish | my dialplan is in mysql, but looks like i'd be better hardcoding the receive-fax context and just jumping to it |
| 18:11.26 | rjvvliet | p3nguin: catphish : as we can see there are always more ways to do the same ;-) |
| 18:11.52 | catphish | indeed, my systems have hundreds of companies sharing one asterisk instance, so it pays to keep things neat |
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| 18:14.44 | rjvvliet | catphish: p3nguin : i Wish you the best, gotta go. |
| 18:15.01 | catphish | you too, thanks for the help |
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| 18:21.37 | Gomex | Guys |
| 18:21.51 | Qwell | Gomex |
| 18:22.09 | Gomex | Is there some service to record a good voice for URA in asterisk? |
| 18:22.14 | Qwell | URA? |
| 18:22.39 | Gomex | Qwell, sorry, in english I think is IVR |
| 18:22.57 | Qwell | http://store.digium.com/productview.php?category_id=8&product_code=8IVRPROMPT |
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| 18:40.39 | gokul | Hello, my asterisk app records the voices and plays back to users, but when I choose wav format for record, the playback fails. whereas gsm works |
| 18:40.52 | gokul | Any way to make it work in wav |
| 18:57.14 | akrohn | wav has to be a specific format gokul. I think it has to be mono with 8000 sample rate |
| 18:57.15 | *** join/#asterisk moy (~moy@216.172.42.74) |
| 18:57.42 | akrohn | gokul, more info: http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk |
| 19:13.25 | gokul | akrohn, found another issue with my context, actually I was specifying as filename.wav in the context |
| 19:19.57 | learath | anyone here use broadvoice and have MWI working? |
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| 19:42.55 | pa | is MP3Player still working? |
| 19:44.51 | Qwell | pa: There's really no reason to use it |
| 19:45.00 | Qwell | in fact, there's no reason to use MP3s at all |
| 19:45.04 | pa | hm why? |
| 19:45.20 | pa | what should one use? |
| 19:45.35 | Qwell | some native format |
| 19:48.34 | pa | well.. ideally yes |
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| 19:54.23 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
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| 19:55.15 | Steel_Reign | hello all |
| 19:55.39 | Steel_Reign | i have question about connecting two asterisk servers with sip |
| 19:56.21 | Steel_Reign | i have the trunk setup like this |
| 19:56.59 | Steel_Reign | host=xx.xx.xx.xx |
| 19:57.01 | Steel_Reign | username=username |
| 19:57.03 | Steel_Reign | secret=secret |
| 19:57.05 | Steel_Reign | type=friend |
| 19:57.07 | Steel_Reign | qualify=yes |
| 19:57.09 | Steel_Reign | context=from-trunk |
| 19:57.20 | WIMPy | ~pb |
| 19:57.20 | 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. |
| 19:57.21 | Steel_Reign | my question is this what do they mean by username? |
| 19:57.54 | Steel_Reign | thanks wimpy |
| 19:59.07 | Steel_Reign | what is this username and where does it come from? |
| 20:01.53 | akrohn | so your phone can log into the server |
| 20:02.13 | akrohn | you can make it whatever you want it to be Steel_Reign |
| 20:03.17 | Steel_Reign | ok so is it just a name added in the trunk or does it have to be an extension? |
| 20:03.49 | akrohn | i didn't read everything you wrote... we have a similar thing going on. |
| 20:03.58 | akrohn | one sec |
| 20:04.03 | Steel_Reign | k |
| 20:04.41 | glaz | It can be anything, and if not specified it will use whatever you put in the [] as the username |
| 20:05.01 | glaz | so username is optional |
| 20:05.32 | akrohn | http://pastebin.com/TDLQu6HC ... you don't need username, as glaz said. |
| 20:05.33 | glaz | my sip trunks only have type, context and host |
| 20:06.01 | *** join/#asterisk Georger (~Georger@79.103.238.193.dsl.dyn.forthnet.gr) |
| 20:06.10 | Georger | hello |
| 20:06.20 | Steel_Reign | ok thanks glaz. i a, just trying to connect two servers with sip and found a bunch of tutorials and they all fail |
| 20:06.39 | Georger | i have centos |
| 20:06.39 | Georger | current config of snmpd was working in past for 1.4 asterisk |
| 20:06.39 | Georger | i installed from repositories asterisk-snmp and enabled the subagent |
| 20:06.39 | Georger | snmp is working in machine but cannot fetch info about asterisk |
| 20:06.48 | Georger | snmpwalk -v 2c -c public 127.0.0.1 ASTERISK-MIB::astVersionString |
| 20:06.48 | Georger | ASTERISK-MIB::astVersionString = No Such Object available on this agent at this OID |
| 20:06.55 | akrohn | ~pb |
| 20:06.55 | 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. |
| 20:07.07 | Georger | ok |
| 20:09.15 | Steel_Reign | lol i got the same message georger |
| 20:09.40 | Steel_Reign | i just installed pastebin on my linux machine now |
| 20:10.12 | Georger | http://justpaste.it/rkn |
| 20:10.19 | Georger | i pasted here |
| 20:10.42 | Georger | asterisk 1.8 and snmp |
| 20:11.49 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 20:13.25 | Georger | i guess outdated mibs or agentx not working? |
| 20:16.28 | Steel_Reign | ok i guess its working kinda because i see the activity on the destination server but i am getting "number not in service" message |
| 20:16.39 | Steel_Reign | anyone know why? |
| 20:19.07 | learath | turn debugging on? |
| 20:19.12 | learath | it is usually pretty clear |
| 20:19.26 | learath | such as "403 F Off" |
| 20:20.32 | blizzow | [TK]D-Fender: I'm a little confused. If "extensions" don't have voicemail, what does? |
| 20:20.37 | learath | anyone use broadvoice? |
| 20:21.17 | WIMPy | You can send extensions to voicemail. |
| 20:21.47 | [TK]D-Fender | blizzow, extensions.conf <- extensions |
| 20:21.55 | learath | I'm trying to get MWI working |
| 20:22.01 | [TK]D-Fender | blizzow, And extension is just some number in your dialplan |
| 20:22.40 | [TK]D-Fender | blizzow, A line in your dialplan may USE the Voicemail app. But there is no inherent existance of any such thing unless you created it |
| 20:23.05 | [TK]D-Fender | Steel_Reign, Guess you'd have to look on that destination server... |
| 20:23.39 | blizzow | [TK]D-Fender: Meaning I just have to manually cross-reference voicemail.conf and extensions.conf to see what numbers exist in both? |
| 20:24.27 | [TK]D-Fender | blizzow, You're stating that in a form that seems to assume tehre is any relationship (1:1, etc) between them |
| 20:24.39 | [TK]D-Fender | blizzow, It is what you want it to be. |
| 20:25.02 | [TK]D-Fender | blizzow, I could have a million lines of dialplan to process my calls and never ever use the voicemail app whatsoever |
| 20:30.34 | blizzow | [TK]D-Fender: I think I understand. I have to have something in my extensions.conf that triggers the voicemail application. I could make that trigger any myriad of things. Once the voicemail application is triggered, it reads through voicemail.conf and does a voicemail dance for whatever number the extensions.conf is configured to pass to it |
| 20:30.53 | [TK]D-Fender | blizzow, Correct |
| 20:32.05 | learath | [TK]D-Fender: is there a way to in the general section, set "each extension will get it's own extension as a mailbox, by default"? |
| 20:32.19 | learath | IE just set mailbox=$myextension or whatever? |
| 20:32.24 | [TK]D-Fender | learath, I don't think you have been following at all here... |
| 20:32.30 | learath | quite possible :) |
| 20:32.48 | [TK]D-Fender | extensions = extensions.conf . voicemail = voicemail.conf. NO RELATIONSHIP |
| 20:33.05 | [TK]D-Fender | learath, There is no "default" |
| 20:39.41 | ChrisInSydney | learath: Hi, all seems confusing ? Its a bit like Funkadelic, George Clinton, Bootsy Collins, Parlaiment, those guys. Sometimes you cant realy explain it in english, but if you experience,m then it makes perfect sense |
| 20:40.19 | learath | that sentance made my brain hurt :( |
| 20:42.28 | ChrisInSydney | Not as much as extensions, dial plans, voicemail and sip endpoints by the sounds of things |
| 20:42.35 | ChrisInSydney | ;-) |
| 20:42.41 | ChrisInSydney | we have all been there |
| 20:42.48 | learath | ok that makes more sense. yeah |
| 20:42.55 | learath | I've got dialing and extensions working |
| 20:43.11 | learath | though broadvoice is all kinds of fucked up, just trying to get MWI working, which they in theory support |
| 20:44.48 | *** join/#asterisk nny (~Scott@174.107.223.14) |
| 20:46.00 | nny | hi, kill me. I am dealing with a sip provider and they are diagnosing DTMF. I have tcpdumped the interface and shown the DTMF is leaving ala RFC2833 (packet based). Their latest suggestion is to remove dtmfmode=auto from [general] but retain it in the per definition. my gut says this is pointless and I need to escalate the ticket. thoughts? |
| 20:46.13 | nny | peer definition* |
| 20:47.16 | ChrisInSydney | learath: So they have Voicemail and MWI, and you are trying to get their VM to trigger an MWI on a handset registered to you rbox ?? |
| 20:47.21 | ChrisInSydney | your box |
| 20:48.32 | learath | yep. |
| 20:49.01 | learath | in theory I should just set unsolicited_mailbox=$mailboxnumber |
| 20:49.11 | learath | (I put my mailbox number in there) |
| 20:49.18 | learath | but that's not working for whatever reason |
| 20:49.40 | learath | also, because they are dumb, insecure=invite is required |
| 20:49.54 | akrohn | nooooooooo =/ |
| 20:49.54 | ChrisInSydney | Never donr that, but from what I have read it is available |
| 20:50.15 | learath | sadly they document it as "insecure=very" which no longer works |
| 20:50.23 | ChrisInSydney | learath: Most of the ITSPs we use here need insecure invites |
| 20:50.23 | *** part/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 20:50.36 | learath | interesting. seems like a bad idea to me. |
| 20:50.37 | nny | lerath afaik is insecure=port,invite |
| 20:50.41 | nny | learath: |
| 20:50.49 | ChrisInSydney | nny: beat me to it |
| 20:50.51 | learath | nny: yes, but port does not actually seem to be required |
| 20:51.01 | learath | it works fine with just invite, YMMV |
| 20:51.24 | nny | learath: my provider does the same thing, conincidentally they aren't that great at service and complain when anything isn't per their limited instructions |
| 20:51.50 | learath | nny: yeah. broadvoice's docs are flat out wrong, based on asterisk 1.0 or something |
| 20:52.27 | learath | and of course I bought a cisco phone, which adds it's own layer of fun :) |
| 20:52.30 | learath | (9951) |
| 20:52.33 | nny | learath: i am diagnosing DTMF and mine jsut told me to remove dtmfmode=auto from [general], when I have dumped the interface sent them screenshots and proven it's not anything pre pbx. They're kind of stupid |
| 20:53.17 | nny | the biggest problem is they ignore evrything i say and take some extra bit of info as a possible solution, I erroneously showed them my sip.conf and now they are playing scientist on my client's dime |
| 20:53.50 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:54.34 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 20:55.07 | learath | just write a script to iterate through every possible sip.conf :) |
| 20:55.25 | nny | ha |
| 20:55.38 | ChrisInSydney | nny: You using RFC or SIP INFO ? |
| 20:56.07 | nny | ChrisInSydney: Looks like RFC from the packet capture |
| 20:56.09 | ChrisInSydney | learath: You usually do Cisco ?? |
| 20:56.20 | nny | ChrisInSydney: the thing is, I have a dump of the interface between them and the client |
| 20:56.22 | learath | nah, I just like their voip phones |
| 20:56.24 | nny | and it shows clearly the DTMF |
| 20:56.27 | learath | but the 9951 was probably a mistake |
| 20:56.30 | ChrisInSydney | nny: Have you tried SIP INFO ? |
| 20:56.34 | learath | I probably should have gone with the 7961 or whatever |
| 20:56.34 | nny | however they keep jumping back to the pbx/phones, like they ignore it |
| 20:56.54 | nny | ChrisInSydney: their suggestion is dtmf=auto and let SDP sort out the method, I can't argue with it, it's upstream of the pbx |
| 20:57.17 | ChrisInSydney | learath: I have a bunch pf SPA525G2 with lockups on an IPFX system. Fortunately not my problem. |
| 20:57.19 | nny | ChrisInSydney: http://i.imgur.com/ATPzp.png |
| 20:57.36 | ChrisInSydney | nny: Try to force a preference |
| 20:57.54 | ChrisInSydney | nny: I did have issues with Snom 870s and DTMF in earlier firmware |
| 20:58.02 | ChrisInSydney | seems OK now |
| 20:58.06 | nny | ChrisInSydney: check the link |
| 20:58.14 | ChrisInSydney | trying to |
| 20:58.20 | nny | ChrisInSydney: that's the network interface of the PBX -> Provider |
| 20:58.32 | ChrisInSydney | damn YChat |
| 20:58.43 | nny | ChrisInSydney: which means the dtmf packet is leaving the network and supposedly hitting their gateway |
| 20:59.06 | ChrisInSydney | brb |
| 20:59.26 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:59.37 | ChrisInSydney | back |
| 21:01.01 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:01.08 | ChrisInSydney | try agian |
| 21:01.21 | *** join/#asterisk wonderworld (~ww@dsdf-4db5f3fc.pool.mediaWays.net) |
| 21:02.26 | ChrisInSydney | nny; Thats cool, what app is that ?? |
| 21:02.54 | nny | fml |
| 21:02.55 | nny | Asterisk has known issues when setting the dtmfmode under general as well as the trunk. Can you please try this and let us know of your results? |
| 21:02.59 | nny | what?! |
| 21:03.03 | nny | god damn these companies |
| 21:03.17 | nny | ChrisInSydney: wireshark with tcpdump set to output to a file that it can interpret |
| 21:03.35 | ChrisInSydney | cool |
| 21:03.40 | ChrisInSydney | I must upgrade |
| 21:03.43 | ChrisInSydney | ;-) |
| 21:04.07 | learath | wireshark is pretty awesome |
| 21:04.23 | ChrisInSydney | true |
| 21:04.54 | nny | tcpdump -i eth0 -s 65535 -w dtmftest -> save file to pc, open in wireshark, tools telephony calls |
| 21:06.51 | paulc | I discovered that the other day - awesome to play back the audio.. and got me onto the "we transmit 'nothing' rather than 'silcence' whilst recording" - pointed me in the right direction to solev my problem, and now we have happy happy users |
| 21:06.57 | paulc | s/solev/solve/ |
| 21:07.03 | Georger | anyone for the snmp issue? |
| 21:07.04 | paulc | doffs hat to infobot |
| 21:11.00 | learath | what was your problem/solution? |
| 21:13.31 | *** join/#asterisk catphish (~charlie@2001:9d8:2005:12::3) |
| 21:14.12 | catphish | if i use Goto(context, s, 1) then the call ends, will context,h,1 be executed? |
| 21:14.21 | catphish | or the hangup from the original context |
| 21:14.43 | ChrisInSydney | Guys: Have to run |
| 21:14.47 | ChrisInSydney | good luck |
| 21:15.32 | [TK]D-Fender | catphish, it hangs up where you are |
| 21:16.10 | catphish | i think that's what i want :) |
| 21:17.29 | catphish | i've forgotten what an extensions.conf looks like, too much time with the luxury of realtime |
| 21:25.54 | *** join/#asterisk Henchman21 (~rakata@208.102.127.220) |
| 21:26.15 | Henchman21 | sweet just got my spa-3000 in the mail and got it hooked up dialing asterisk and picking up the phone woohoo |
| 21:26.28 | Henchman21 | http://66.172.12.214/ |
| 21:27.25 | *** join/#asterisk iamgalen (~galen@173.164.41.185) |
| 21:28.08 | [TK]D-Fender | checkout time, later all |
| 21:29.12 | iamgalen | Has anyone even come across diagnostic messages from the Digium transcoder cards? Specifically the Wildcard TCE400P. My net searches have come up with zero results. |
| 21:30.16 | jgowdy | I'm trying to change the log level for the console |
| 21:30.24 | jgowdy | help says "logger set level DEBUG" |
| 21:30.32 | jgowdy | I get no such command |
| 21:30.38 | akrohn | core set verbose 10 |
| 21:30.41 | jgowdy | thanks |
| 21:30.45 | akrohn | core set debug 10 |
| 21:30.49 | akrohn | np |
| 21:31.07 | akrohn | that will also change the logging levels in your log files |
| 21:31.47 | akrohn | for a more permanent solution, modify logger.conf |
| 21:33.12 | learath | so, does anyone know how to manually toggle MWI flags inside asterisk? |
| 21:33.21 | learath | IE, I'd like to force MWI on (and off) for extensions |
| 21:33.26 | learath | hopefully via cli commands |
| 21:33.53 | akrohn | i think i'm wrong. but. i want to say it is phone-specific =/ |
| 21:34.18 | akrohn | or something equally annoying |
| 21:34.20 | learath | MWI? |
| 21:34.44 | akrohn | maybe? |
| 21:34.54 | jgowdy | From what I heard, MWI is phone specific |
| 21:35.11 | learath | ok |
| 21:35.23 | learath | but asterisk has to keep track of it somehow right? |
| 21:37.29 | akrohn | closest to it: "just use externnotify= and a script that touches the msg<num>.txt file in the user's vm directory" |
| 21:38.22 | akrohn | here's a good (but old) article: http://asterisk.mdaniel.net/2006/12/06/mwi-notification/ |
| 21:38.31 | learath | ok thanks |
| 21:43.49 | nny | so consensus, what do you guys do for SIP based providers? Seems like this one always has had DTMF issues ( I suspect upstream from them, their usual response is they change the route/vendor upstream.) |
| 21:43.49 | nny | I constantly have to suggest a provider to our client, and after 6 years I have yet to find one that doesn't suffer from some variance of this issue |
| 21:44.29 | nny | clients*. I refer tons of traffic to them and it seems i always regret it |
| 21:44.52 | akrohn | VoIP Innovations rocks, imo |
| 21:45.02 | akrohn | but i think they are reseller only |
| 21:45.39 | nny | akrohn i'd be ok with reseller if they had a billing portal and service that didn't require diagnosing dtmf heh |
| 21:45.59 | akrohn | they have a portal and really awesome tech support / CS |
| 21:46.16 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 21:46.21 | nny | seems they do have a portal |
| 21:46.56 | nny | interesting. I have been meaning to look into that as an alternative method to just suggesting providers. It's risky, but the revenue seems appropriate for the risk if you have enough clients |
| 21:47.00 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 21:47.16 | nny | besides, i end up doing 75% of the diagostic right now as it is anyways :\ |
| 21:47.55 | akrohn | the base price you mean? we ported a bunch of our toll-free DIDs over there in order to use up our monthly allotment |
| 21:53.07 | pigpen | so, I as asked by a user if polycom has a phone that has "more line buttons" so they can "monitor" everything (lines, extensions, hints) on a single phone. (Small office, stupid request) |
| 21:53.22 | pigpen | Naturally, they didn't like the cost of the side "car" for their polycoms?. |
| 21:54.16 | pigpen | So does anyone have any recommendations of desktop (windows I am afraid) "viewers" of sort, so they can watch sip "in use" type devices. (yes, they are using sip gateway as well) |
| 21:54.52 | _Corey_ | pigpen: If they can wait a few weeks, Digium's D70 phone might do the job pretty cost-effectively |
| 21:55.22 | pigpen | you know, I did mention Digium's phones?.to that may be the best choice. I guess in the mean time i need to get one to play with. |
| 21:57.37 | pigpen | _Corey_, so does the button light up on the side list? |
| 21:58.00 | pigpen | oh, shit that is a "display" |
| 21:59.27 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 22:00.31 | _Corey_ | pigpen: Hmm, I haven't configured the sidecar stuff yet actually... ;) I'll let you know tomorrow |
| 22:00.56 | pigpen | very cool?pls let me know. |
| 22:01.10 | pigpen | I have a customer that will move to them 100% if all is good. |
| 22:01.17 | pigpen | 5 per store, 380 stores. |
| 22:01.32 | pigpen | dam they look like a polycom. |
| 22:01.39 | learath | bada-bing |
| 22:01.40 | pigpen | tks bty. |
| 22:01.51 | learath | ok, so the cisco phone does mwi with asterisk fine |
| 22:01.55 | learath | that's beautiful |
| 22:01.56 | _Corey_ | yeah, they're nice... no worries. if you need a reseller, let me know! ;) |
| 22:02.17 | pigpen | yeah, I am a reseller, setup with a good distributor. |
| 22:02.32 | pigpen | I am sure if they sell the other digum stuff they will the phones too. |
| 22:02.41 | woleium | hola peeps :-) |
| 22:03.23 | woleium | is anyone aware of a good resource for diagnosing problems with asterisk, after the fact? |
| 22:04.27 | woleium | our phones started saying "unavailable" whenever we tried to make calls about an hour ago. A reboot of the server has restored functionality, and I can't seem to find anything relevant in the logs. |
| 22:08.21 | *** join/#asterisk jsjc (~Adium@161.Red-83-45-143.dynamicIP.rima-tde.net) |
| 22:15.45 | *** join/#asterisk greenwolf (cc98c95b@gateway/web/freenode/ip.204.152.201.91) |
| 22:17.18 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 22:24.03 | blizzow | woleium: /var/log/asterisk/full would be a good place to start. |
| 22:25.18 | woleium | thanks blizzow, i have been reviewing that file, but there aren't any exceptions that I can see |
| 22:26.46 | blizzow | you may want to check if there is anything in the system logs as well. See if something was hampering the asterisk process. |
| 22:27.05 | woleium | sure, checking now |
| 22:27.54 | woleium | one thing i did check before i rebooted was a 'sip set debug peed [my_extension]' and tried to make a call |
| 22:28.05 | woleium | there was no output |
| 22:28.20 | woleium | which suggests that the server was unavailable to the phone |
| 22:28.42 | woleium | however a sip show peers showed 16ms ping :-/ |
| 22:29.08 | woleium | i guess that's cached though... |
| 22:29.59 | *** join/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 22:30.16 | woleium | here is the relevant bit of the 'full' log up to the reboot, if anyone fancies lending a trained eye :-) http://pastebin.com/hzTvaUC3 |
| 22:30.54 | Steel_Reign | question. if i am trying to connect to asterisk server together and make the trunk do i also need to make the outbound routes for them? |
| 22:36.07 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:37.10 | Steel_Reign | hello D-Fender |
| 22:37.30 | Steel_Reign | got a question for you have been so helpful in the past |
| 22:37.40 | Steel_Reign | if i am trying to connect to asterisk server together and make the trunk do i also need to make the outbound routes for them? |
| 22:38.23 | *** join/#asterisk adeel (~adeel@72.53.72.133) |
| 22:39.32 | hackeron | when I start asterisk I see [Feb 27 22:38:25] WARNING[3883]: loader.c:393 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref -- any ideas what is causing it? |
| 22:40.34 | [TK]D-Fender | Steel_Reign: How else do you expect your system to know what to pass between them? |
| 22:41.18 | Steel_Reign | no idea buddy just trying to learn all this stuff. Thank you once again. |
| 22:43.26 | nny | odd |
| 22:43.47 | nny | so I am testing dtmf, i have a tcpdump like here http://i.imgur.com/ATPzp.png |
| 22:44.32 | nny | the other end wasn't responding, does a tcpdump of the interface used to communicate with the SIP provider prove that at the very least the issue is post pbx? |
| 22:45.54 | nny | dtmf logs show that the pbx was receiving (ex: DTMF end '1' received on SIP/109-00000002, duration 120 ms) and the dump was showing the packets in stream. |
| 22:46.10 | nny | now.. shit it works.. and I don't know why |
| 22:46.20 | nny | but every tother test prior and nothing. |
| 22:46.58 | nny | but testing the same provider/number on another pbx worked. I updated to the latest 1.6.2.X on the test system, but I did this last time the issue arose and nothing changed. |
| 22:48.07 | *** join/#asterisk jpsharp (jsharp@ohno.mrbill.net) |
| 22:48.44 | nny | so 1.) there's a bug between 1.6.2.6 and 1.6.2.22 that was dtmf related that wouldn't show up in normal tests 2.) the provider changed something or 3.) magic |
| 22:51.46 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 22:55.05 | nny | ANNNND https://issues.asterisk.org/view.php?id=18189 |
| 22:55.06 | nny | kill me |
| 22:55.15 | nny | SSRC is different in my dump |
| 22:55.47 | nny | it's since been fixed, looks like i have yet another thing to search for when reporting DTMF issues |
| 22:56.26 | ChannelZ | I've been having all kinds of dtmf problems lately |
| 22:57.40 | *** part/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 23:02.55 | nny | ChannelZ: well to asterisk's credit this is an older issue, although why they would change the SSRC value (not RFC compliant at all) is odd and beyond me |
| 23:03.03 | nny | jesus that was a bitch to spot |
| 23:03.26 | nny | i kept jumping between the working and non working one.. my brain was going Einhorn.. Finkle.. Einhorn.. Finkle... |
| 23:04.13 | nny | now i feel bad for badmouthing the provider, although you'd think they'd have a way to see packets being ignored on their gateway |
| 23:04.27 | nny | oh nothing wrong here.. except all these dtmf packets being rejected |
| 23:07.23 | ChannelZ | Mine have just been choppy and some remote IVRs (my bank) throw up on them all the time. I think it must be my ITSP but I need to narrow it down better, maybe it's my bandwidth. Need to get some packet prioritization happening. |
| 23:29.41 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 23:32.07 | nny | ChannelZ: hmm i'd think with dtmf the choppy part would only be an issue if you were using inband pre provider/upstream |
| 23:40.55 | *** join/#asterisk twodogs (~twodogs@hendra.biohazard.seattle.wa.us) |
| 23:48.22 | *** join/#asterisk cyborg-one (1000@31.31.97.128) |
