| 00:03.04 | *** join/#asterisk pdtpatr1ck_ (~pdtpatric@12.249.4.226) |
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| 00:34.34 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
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| 01:00.40 | riddlebox | is there an option in sip.conf for a sip user to use a different port like say 5090? |
| 01:03.39 | citywok | riddlebox: per user? no. that's a per server setting. |
| 01:05.49 | riddlebox | citywok, I just figured out my problem, I will just use an iax user |
| 01:10.43 | p3nguin | You can set the client port of the SIP peer in each peer definition. port=5061 for example. |
| 01:10.57 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 01:11.10 | p3nguin | That's not the server port, that's the client port. |
| 01:11.27 | p3nguin | You'll have to also change the port in the device itself. |
| 01:14.26 | riddlebox | p3nguin, yeah I figured that, I think if I use iax I can avoid a lot of issues, with what I am trying to accomplish |
| 01:15.39 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 01:43.34 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 01:49.24 | dijib | p3nguin: how have you been? |
| 01:52.06 | woleium | evenin' all :-) |
| 01:52.40 | woleium | Are there any doc's on routing in * |
| 01:52.59 | woleium | I guess it's all done by hand with dial plans? |
| 01:55.09 | dijib | dialplan routing? |
| 01:55.15 | dijib | or whats your specific question? |
| 01:55.37 | woleium | we have (for arguments sake) three sites, all on the same routed network, and all with their own * box. links A-B and B-C have low latency pipes, but C-A is has lag and jitter |
| 01:56.14 | woleium | can I get C-A calls to route via B? |
| 01:57.13 | woleium | I could do it using IP routing (which seems most logical), but I wondered if there was any kind of multi path routing in * |
| 01:57.53 | woleium | simiar to the way exchange uses OSPF |
| 01:57.57 | citywok | asterisk isn't going to do any network layer routing for you |
| 01:58.12 | woleium | no, i understand that :-) |
| 01:58.20 | citywok | if you want to do that with asterisk you would probably need to route the call to C through B and have all 3 asterisk boxes in the call |
| 01:59.12 | citywok | somebody else may know of something I don't, but that's the only solution I can think of. There's no other magic. |
| 02:00.06 | woleium | I guess that's the whole point of VoIP really :-) |
| 02:00.17 | woleium | to be able to control routes using IP |
| 02:01.13 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 02:01.55 | p3nguin | "Routes" are network, not asterisk. The only thing you'll do with calls is direct them via peer definition in dial plan, and network layer controls the routing of said calls to said peers. |
| 02:02.28 | citywok | yea, if it were me i'd router all traffic to that subnet through location B |
| 02:02.33 | citywok | s/router/route/ |
| 02:02.36 | p3nguin | If you want calls from A to to to B via C, make A dial C and not B. |
| 02:02.54 | p3nguin | s/to to/to go/ |
| 02:03.39 | p3nguin | Then calls reaching C from A will analyze and either dial a phone locally or dial something on B accordingly. |
| 02:05.00 | woleium | that makes sense. I think layer 3 routing is the way to go though. |
| 02:06.08 | woleium | Is there a recommended method for managing floating accounts? |
| 02:06.36 | woleium | e.g. Bob in sales who rims between all three sites? |
| 02:06.47 | woleium | s/rims/roams :-$ |
| 02:07.41 | woleium | at present they just register with their home server, which isn't really a problem, but doesn't strike me as the most efficient method. |
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| 02:13.33 | citywok | woleium: whatever works for you :) |
| 02:15.58 | woleium | Im wondering if there is any point in thinking about developing a system to handle that situation, or if it makes more sense to just aim to centralise and only have one central (virtualised HA) server |
| 02:27.08 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 02:28.50 | dijib | anybody heard of thirdlane ? |
| 02:29.15 | dijib | thoughts. second thoughts. |
| 02:29.40 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 02:29.58 | dijib | are you going to tell me im in #asterisk and not #asterisk-prettyboy-gui |
| 02:30.00 | dijib | ? |
| 02:33.30 | puzzled | dijib: only know they have been around for a long time |
| 02:36.59 | p3nguin | woleium: You could put all peers in sip realtime and distribute the db across all the machines. |
| 02:37.50 | p3nguin | woleium: Then, perhaps, you can use something like dundi to determine where a device is registered and send calls to the appropriate box. |
| 02:59.52 | dijib | is anybody a webmin shorewall configuration god that looks at that console and has a working firewall |
| 03:03.24 | dijib | someone shoulw just tell me to join #shorewall already |
| 03:03.26 | p3nguin | points and laughs at the thought of webmin. |
| 03:04.04 | dijib | and still no use for a sangoma |
| 03:04.13 | dijib | i like webminui |
| 03:04.34 | dijib | also i cant get it working by editing the conf's |
| 03:04.50 | dijib | so to toy webmin |
| 03:05.09 | dijib | i like webmin. |
| 03:05.17 | *** join/#asterisk jetlag (jetlag@pool-71-168-254-144.cmdnnj.east.verizon.net) |
| 03:05.26 | dijib | im sorry to say, its convient and shiney |
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| 03:10.53 | dijib | p3nguin: why are you being the nelson of the chan? |
| 03:11.19 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 03:15.39 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 03:20.37 | p3nguin | Stop screwing with shorewall and learn how to use iptables. |
| 03:20.50 | p3nguin | You'll have much better results with your firewall when you know what it is doing. |
| 03:21.30 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.166.29.218) |
| 03:23.23 | dijib | what am i going to use for bit shapping? |
| 03:25.19 | dijib | #shorewall is dead anyways |
| 03:25.23 | p3nguin | The same thing everyone else uses, of course. |
| 03:25.42 | dijib | yeah and most likely something you have said specifically to me prior aswell |
| 03:26.11 | woleium | +1 for direct iptables |
| 03:26.48 | p3nguin | For traffic shaping, see here: http://lartc.org/howto/ |
| 03:26.53 | woleium | although pf on bad is good too |
| 03:27.01 | woleium | s/bad/bsd |
| 03:27.09 | p3nguin | Oh, so close! |
| 03:27.23 | p3nguin | s/Oh,/You were/ |
| 03:27.38 | dijib | how do i get htb and tc installed on cent 6.2? |
| 03:28.00 | woleium | blames autocorrect |
| 03:28.29 | woleium | yum install htb tc |
| 03:28.31 | woleium | ? |
| 03:28.37 | dijib | i want to run asterisk on a router/dsl gateway |
| 03:28.44 | dijib | yum missing those |
| 03:29.56 | p3nguin | tc is part of iproute. yum -y install iproute |
| 03:30.40 | dijib | oh yoru right tc is already on the box |
| 03:30.42 | dijib | htb |
| 03:30.46 | woleium | indeed, although it's iptables-htb iirc |
| 03:30.48 | dijib | is not |
| 03:31.06 | dijib | not that |
| 03:31.27 | woleium | it is on rpm.pbone.net |
| 03:32.24 | p3nguin | I'm pretty sure htb is something in your kernel and not a piece of software you add on. |
| 03:33.49 | dym | jesus |
| 03:33.59 | dym | p3nguin: its not like you wouldnt know what shorewall is doing |
| 03:34.05 | dym | since its not doing anything, basically |
| 03:34.18 | dym | it just parses its config and translates into iptables rules. |
| 03:34.38 | dym | dijib: i offered you help earlier, but you seemed unable to perform simple tasks. I suggest hiring a sysadmin |
| 03:36.00 | woleium | or install pfSense |
| 03:36.08 | woleium | and click away :-) |
| 03:36.17 | dym | bit overdosed |
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| 03:42.51 | dijib | hiring a sysadmin |
| 03:42.53 | dijib | lol |
| 03:43.04 | dijib | dude ive got a shoebox server |
| 03:43.48 | dijib | i can follow simple tasks but its not like its a brake fix its a new implimentation and out of the box should just be a drop in with config and go. |
| 03:46.34 | LiuYan | If use AMI to detect incoming call, does it must poll events to achieve this purpose? |
| 03:49.15 | dym | dijib: errr - nothing is "just an implementation", "out of the box" and "just be a drop in with config and go". |
| 03:49.27 | dym | I guess you misunderstood running a server. |
| 03:51.12 | dym | Seeing as you are irc'ing with a root account I can understand how you think it would though. |
| 03:51.22 | dijib | lol #2 thanks |
| 03:51.27 | dijib | possibly 3 |
| 03:51.35 | dijib | i do everything as root why is that a problem? |
| 03:52.06 | dym | and you run an irssi screen session as root |
| 03:52.29 | dym | well, maybe it poses potential security risks to run services with priviledged rights? |
| 03:52.41 | dijib | ok well i have maid some progress |
| 03:52.57 | dijib | im not able to enable iptables and shorewall and have my connection remain |
| 03:53.23 | dijib | but my 80-89 & changed webmin isnt working |
| 03:54.01 | dym | you are not able to have the connection remaining, but your 80-89 and webmin arent working. makes perfect sense... |
| 03:54.11 | dym | "this is not working, but this isnt either." |
| 03:54.14 | dym | ? |
| 03:54.33 | dijib | sorry ssh is working 80-90 not |
| 03:54.50 | dijib | firewall rules match the ssh beyond having 80:89 |
| 03:54.51 | dym | then apply what you did to make it work to that what isnt working. |
| 03:55.31 | dijib | should that be 80-89 80:89 or 80,81,82,83? |
| 03:55.40 | dym | shorewall would error if it was wrong |
| 03:55.48 | dijib | it doesnt |
| 03:55.56 | dym | then it obviously isnt |
| 03:55.58 | dijib | or not in the check does it |
| 03:56.06 | dijib | doesnt |
| 03:56.19 | dijib | uhm log level 3 gave me an error also |
| 03:56.48 | dijib | LOG_VERBOSITY 3 gave error 2 was defaut |
| 03:57.05 | dym | dijib: shorewall.net |
| 03:57.29 | dym | http://www.shorewall.net/two-interface.htm |
| 03:59.00 | dijib | http://www.shorewall.net/MultiISP.html |
| 03:59.17 | dym | What would I want with that? |
| 03:59.27 | dym | I posted the links for your to read and understand |
| 04:24.52 | ChannelZ | jesus Youtube has been sucking it lately |
| 04:32.57 | dijib | updates? |
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| 04:35.40 | ChannelZ | Slow, doesn't respond for 10 seconds, causes buffering.. |
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| 05:50.39 | mpoole | hey, any ideas why an IAX call (using dundi) would get rejected when I'm using dundi:${SECRET} and dbsecret=dundi/secret on in iax.conf ? |
| 05:50.45 | mpoole | I'm a bit stuck :o |
| 05:51.10 | mpoole | everything seems to suggest this should work |
| 05:51.58 | mpoole | remote box returns: CAUSE : No authority found |
| 05:52.14 | mpoole | which would suggest an authentication issue |
| 05:52.48 | mpoole | "dundi:${SECRET}" is in dundi.conf on the source box |
| 05:54.59 | mpoole | I'm connecting asterisk 1.4 to 1.8 if that could make a difference |
| 05:57.29 | mpoole | I might just try setting a static username= and secret= or switch to SIP and have done with it |
| 05:59.55 | jkroon | i |
| 06:00.05 | jkroon | i've done DUNDI => IAX before. |
| 06:00.14 | jkroon | used static usernames though |
| 06:07.18 | mpoole | Its not the username that is dynamic |
| 06:07.24 | mpoole | that is always dundi |
| 06:07.29 | mpoole | it's the password |
| 06:10.36 | jkroon | ok... |
| 06:10.50 | jkroon | no authority found implies IAX/2 can't find the username IE in the call setup request. |
| 06:11.27 | jkroon | you can run iax/2 in debug mode on the receiving side to see exactly why the call is being rejected (it's pretty thorough) |
| 06:24.18 | kaldemar | mpoole: do you have [dundi] in iax.conf? |
| 06:24.25 | mpoole | yea |
| 06:24.38 | mpoole | [dundi] |
| 06:24.39 | mpoole | type=user |
| 06:24.39 | mpoole | dbsecret=dundi/secret |
| 06:24.39 | mpoole | context=dundi-priv-local |
| 06:25.21 | mpoole | in the dialing PBX |
| 06:25.23 | mpoole | priv => dundi-priv-canonical,1,IAX2,dundi:${SECRET}@sanr-pbx.xxx.com/${NUMBER},nounsolicited,nocomunsolicit,nopartial |
| 06:26.04 | mpoole | sanr-pbx is the dialing and shef-pbx is the receiver |
| 06:27.40 | kaldemar | you should have the mapping on the receiving side. |
| 06:28.37 | mpoole | oh |
| 06:29.00 | mpoole | so so the receiving side (shef-pbx) needs to have the sanr-pbx mapping ? |
| 06:29.09 | mpoole | rather than the other way round |
| 06:32.51 | kaldemar | the mapping that you call with the switch or lookup from your dialplan. and the peer definition on the originating side must list it. |
| 06:34.00 | mpoole | all the docs say it should be the local address there |
| 06:40.32 | kaldemar | it's been a while since i've used dundi, but iirc it goes like this: when you do a lookup, all peers in dundi.conf are checked for the given mapping as an include parameter. those who have the mapping, are sent a query. when a dundi query is received, a mapping defines how it is handled. it has the name of the mapping that should match what the incoming query has, and the local context that is used to check if the receiver has a matching ... |
| 06:40.38 | kaldemar | ... extension. |
| 06:42.00 | kaldemar | so you have to have the switch statement or lookup, peer definitions, mappings and IAX settings correct for it to work. maybe you should pastebin your relevant dundi.conf, iax.conf and extensions.conf. mask any secrets that you might have. |
| 06:44.32 | mpoole | OK 2mins I'm probably just missing something stupid |
| 06:47.27 | *** join/#asterisk itsurkg (~itsurkg@202.52.236.10) |
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| 06:52.45 | mpoole | fuck |
| 06:52.47 | mpoole | I worked it out |
| 06:52.55 | mpoole | I removed the # from an include by mistake |
| 06:53.27 | mpoole | that sucked :) |
| 06:53.31 | mpoole | kaldemar: thanks |
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| 07:05.39 | Giant_enemy_crab | hello. How do i fix this dialplan: http://paste.org.ru/?t5a399 ? i need music playing after bg message |
| 07:08.48 | kaldemar | beaver_rrr: WaitMusicOnHold is deprecated, you should use MusicOnHold instead. why would you want to play music for 60 seconds and then start dialing? or is it something else you want? |
| 07:09.25 | beaver_rrr | no, i want to play music while dialing |
| 07:11.53 | kaldemar | see option m() in "core show application Dial" |
| 07:12.59 | beaver_rrr | thanks. |
| 07:13.51 | kaldemar | however, if you're dialing multiple peers like that, a queue might serve your purpose better. |
| 07:14.10 | kaldemar | http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ACD.html |
| 07:15.15 | beaver_rrr | thanks again, gonna go read |
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| 07:19.31 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:19.33 | schmidts | good morning |
| 07:20.42 | itsurkg | Hello. How do I automatically play a sound and hangup for outgoing call when all fxo port are busy. I have asterisk gui installed. |
| 07:23.01 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:23.47 | schmidts | itsurkg you should use GROUP and GROUPCOUNT so see how many calls are active and when you hit the limit just playback the file |
| 07:25.26 | ChannelZ | Or you can just look if the Dial returned CHANUNAVAIL or whatever it is |
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| 07:28.44 | itsurkg | schmidts, thanks. Should i have to edit extension.conf file for that? |
| 07:30.48 | schmidts | ChannelZ will it return CHANUNAVAIL only if all ports are busy? |
| 07:30.55 | schmidts | itsurkg yes sure ;) |
| 07:31.41 | ChannelZ | Well it will return that for any number of reasons a channel was unavailable :) |
| 07:32.23 | ChannelZ | The purpose being to tell the person "hey, your call is going nowhere" seems applicable whether it's because all the channels are full, or down... |
| 07:36.15 | itsurkg | Well I am a newbie here and since i haven't much idea, can I have some reference or any link to look at. Thanks |
| 07:36.53 | ChannelZ | Well in the case of Asteris GUI I have no idea what you tell you |
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| 07:41.16 | itsurkg | As you mentioned about GROUP, GROUPCOUNT, CHANUNAVAIL is there any sample to look so I can understand |
| 07:41.59 | kaldemar | application ChanIsAvail can be used to check availability before trying to dial. |
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| 08:24.24 | *** join/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 08:24.40 | AliRezaTaleghani | Hi all |
| 08:24.45 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:24.58 | AliRezaTaleghani | can someone help for for this errors, in asteriskSCF installation: |
| 08:25.01 | AliRezaTaleghani | http://paste.org/46762 |
| 08:25.40 | *** join/#asterisk polartux (~polartux@dy869-1-82-228-25-144.fbx.proxad.net) |
| 08:25.45 | polartux | hi all |
| 08:25.55 | AliRezaTaleghani | when i what to run './cmake/init-cmake.sh' in phase #4 https://wiki.asterisk.org/wiki/display/TOP/Building+on+Linux |
| 08:26.24 | AliRezaTaleghani | under the above link! i will face with problem's which shown here: http://paste.org/46762 |
| 08:29.51 | polartux | someone have troubles with queue and moh inheritance ? |
| 08:32.03 | Yourname` | Hi! Anyone know why eyeBeam pops up for a split second when it receives a call? It happens only a few times.. but too annoying for the users. Queue timeout is set to 15 seconds. |
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| 08:35.55 | kaldemar | Yourname`: asterisk has nothing to do with eyebeam behavior on the desktop. |
| 08:37.46 | Yourname` | kaldemar: I thought I'd make it simple hence the eyebeam mention. But well, Polycom IP 330 is also ringing for a short second. |
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| 08:43.46 | kaldemar | Yourname`: by "it", did you not mean eyebeam but a queue in your first sentence? |
| 08:44.11 | vlt | kaldemar: Hello. Yesterday you said that only one codec per *SIP* session is possible. What about IAX? |
| 08:45.13 | kaldemar | vlt: same limitations apply. |
| 08:45.22 | vlt | kaldemar: Ok, thanks. |
| 08:47.00 | Yourname` | kaldemar: I meant eyebeam (any phone basically) |
| 08:50.01 | kaldemar | Yourname`: explain the whole scenario. |
| 08:51.31 | *** join/#asterisk timahvo1 (~rogue@197.176.105.71) |
| 08:51.36 | Yourname` | kaldemar: A call comes in to the queue. Few agents on eyeBeam and few on Polycoms. Queue timeout is set to 15 seconds, and retry is 2 seconds. On the desktop, the popup that comes up on the corner right side of the screen comes up for a split second and before the agent has a chance to hit "Answer", it's gone. Same with polycom.. before the agent can "life" the receiver, the call is gone. |
| 08:51.58 | Yourname` | kaldemar: This happens only to a few calls. When you call agent to agent, it's fine. Only queue calls causes this issue. |
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| 09:00.46 | kaldemar | Yourname`: what's your queue strategy? ringall? |
| 09:00.54 | Yourname` | kaldemar: rrmemory |
| 09:01.24 | kaldemar | have you caught a CLI output of this? |
| 09:02.13 | Yourname` | kaldemar: Not yet.. figured I'd see if anyone knows whats up |
| 09:08.05 | *** join/#asterisk jkroon (~jkroon@dsl-244-23-143.telkomadsl.co.za) |
| 09:09.10 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 09:10.12 | r3za | hi , i install asteriskNOW freshly , but i cant make user in it ... can anyone help me , i add a SIP Trunk with this config : http://i.imgur.com/aQ2Fc.png but when i want to connect with x-lite it said : 403 forbidden , bad auth |
| 09:10.21 | r3za | is there anyone can help me ? |
| 09:13.50 | kaldemar | r3za: ask in #freepbx |
| 09:14.40 | KNERD | r3za: you don't seup a "user" |
| 09:14.59 | KNERD | you setup an extension |
| 09:15.56 | r3za | kaldemar: hey , actually i config sip-customize.conf with this setting : type=friend secret=1234 username=1004 and host=dynamic but i still got that error |
| 09:16.30 | r3za | KNERD: how can i setup a user , i config sip-customize.conf |
| 09:16.59 | KNERD | I just said you dont setup a user???? |
| 09:17.24 | r3za | KNERD: yup , u said u dont setup a user |
| 09:17.46 | KNERD | and again...to to E X T E N S I O N S |
| 09:18.16 | r3za | KNERD: how can i setup a user , can u help me out ? |
| 09:18.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:18.45 | kaldemar | r3za: continue this in #freepbx |
| 09:18.46 | KNERD | wow..i just told you, but still dont listen |
| 09:18.47 | KNERD | goodbye |
| 09:18.52 | *** part/#asterisk KNERD (~KNERD@99.46.50.247) |
| 09:21.41 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 09:27.45 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 09:36.24 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 09:41.19 | Yourname` | LOL KNerd |
| 09:41.20 | Yourname` | hahaha |
| 09:44.00 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:50.07 | *** join/#asterisk slidesinger (~slidesing@c-174-57-5-70.hsd1.nj.comcast.net) |
| 10:10.45 | *** join/#asterisk paolosupino (~paolosupi@net-2-38-113-188.cust.dsl.vodafone.it) |
| 10:12.43 | paolosupino | Hi, I need some help resolving IAX incoming DID. |
| 10:14.03 | *** join/#asterisk Tim_Toady (~fuzzy@178.128.154.87.dsl.dyn.forthnet.gr) |
| 10:15.15 | paolosupino | I've setup iax.conf for didww.com (as described on their website), mapped the their DID to my PBX and setup the right context in extensions.conf, when I try to dial the DID I can on my PBX console that the call is coming in (but only when I have max set debug on) but nothing happens. |
| 10:16.11 | jacc0 | could you pastebin the cli output? |
| 10:16.34 | *** part/#asterisk r3za (~R3za@89.165.117.254) |
| 10:16.56 | paolosupino | jacco0: do you mean the debug information on the console? |
| 10:17.20 | jacc0 | yes please |
| 10:18.09 | *** join/#asterisk kasanop (~kasanop@109-73-37-220.in-addr.mastertelecom.ru) |
| 10:19.38 | paolosupino | did… labeled IAX debug. |
| 10:20.08 | paolosupino | the numbers and username were changed for privacy reasons :-) |
| 10:20.23 | kaldemar | paolosupino: a link would be appreciated. |
| 10:20.39 | paolosupino | http://pastebin.com/Pcj7KkXZ |
| 10:20.41 | paolosupino | sorry… |
| 10:21.03 | paolosupino | (not used to using pastebin and thus don't know eticket… |
| 10:21.22 | kaldemar | also a debug of a whole call with verbose output. "core set verbose 10" |
| 10:22.10 | kaldemar | don't worry. it's just that there are dozens of pastebin sites so it's easier to use a link. |
| 10:26.02 | paolosupino | that paste was at verbosity 20… (I did "core set verbose 10" and it said it's taking verbosity down from 20 to 10). |
| 10:26.28 | paolosupino | will try again and pate bin again… |
| 10:26.41 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:27.10 | kaldemar | then add the rest of the output. that was just a single IAX message. actually you can disable iax debug. |
| 10:32.47 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 10:32.51 | qakhan | hi all |
| 10:33.09 | jacc0 | hi |
| 10:33.09 | qakhan | can someone guide me about phpagi |
| 10:33.10 | paolosupino | http://pastebin.com/ZMFVsswj this is a complete IAX session (from the start of the call to the moment I hit cancel on my skypeout). |
| 10:33.23 | jacc0 | sure |
| 10:34.16 | jacc0 | rename your .php to .agi |
| 10:34.21 | jacc0 | and put this on top: |
| 10:34.23 | jacc0 | #!/usr/bin/php |
| 10:34.23 | jacc0 | <?php |
| 10:34.40 | jacc0 | or the right path wherever your php binary is |
| 10:34.50 | wdoekes2 | #!/usr/bin/env php |
| 10:35.13 | jacc0 | and then you can just call the .agi from your dialplan |
| 10:35.13 | kaldemar | paolosupino: the verbose output is the point of interest. disable iax debug. do you not see anything in CLI then? |
| 10:35.42 | jacc0 | AGI(/etc/asterisk/scripts/myTestAgi.agi) |
| 10:36.01 | paolosupino | kaldemar: correct. If I disable IAX debug I don't get anything on the scree. |
| 10:36.06 | paolosupino | screen. |
| 10:36.29 | qakhan | jacc0 can you recommend me a book |
| 10:37.55 | paolosupino | kaldemar: I tried dialing without IAX debug and I got a busy signal on my skypeout client |
| 10:38.12 | jacc0 | ~thebook |
| 10:38.13 | 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 |
| 10:38.16 | paolosupino | kaldemar: I turned on IAX debug again and the busy signal disappeared.. |
| 10:38.20 | jacc0 | lol |
| 10:39.18 | paolosupino | kaldemar: I'm using asterisk 1.8.8.0 installed from asterisk debian repo. |
| 10:39.23 | qakhan | jacc0 what happened? |
| 10:40.04 | jacc0 | you asked me to recommend a book |
| 10:40.04 | kaldemar | paolosupino: what do your iax configs look like? |
| 10:40.07 | jacc0 | ~thebook |
| 10:40.08 | 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 |
| 10:41.03 | qakhan | infobot i read this but there is not much detail |
| 10:41.03 | infobot | You read this but there is not much detail? |
| 10:41.15 | qakhan | yes |
| 10:41.23 | qakhan | phpagi |
| 10:41.34 | qakhan | or i may not understand it |
| 10:42.08 | *** join/#asterisk ks3 (ks3@2600:3c02::f03c:91ff:fedf:e474) |
| 10:42.36 | jacc0 | talking to a bot? |
| 10:43.33 | jacc0 | :P |
| 10:44.16 | kaldemar | infobot never stops being helpful. |
| 10:44.22 | jacc0 | :) |
| 10:48.54 | paolosupino | kaldemar: here's the IAX configuration.. http://pastebin.com/H4jdwBuN |
| 10:49.59 | paolosupino | kaldemar: I removed the g729 from the didww.conf, but it didn't resolve the problem. |
| 10:51.42 | jacc0 | @paolosupino: it might not be so (security) wise to post all ip addresses of your clients online |
| 10:52.35 | jacc0 | there are also some secrets in there |
| 10:52.46 | paolosupino | jacc0: I changed most of the IP addresses but I missed a few places… next time I will sanitize is better. |
| 10:53.18 | paolosupino | jacc0: I also changed extension numbers, usernames and passwords :-) |
| 10:53.54 | paolosupino | jacc0: just missed a few places of 1 client in one of the pastes. |
| 10:54.29 | paolosupino | Jacc0: thanx for the heads up, thug :-) |
| 10:54.40 | paolosupino | Jacc0: though… |
| 10:56.32 | kaldemar | there is no such option in iax.conf as "insecure". |
| 10:59.10 | kaldemar | also, don't use [<ip>] as peers, you should have a username between []. |
| 11:00.22 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 11:00.31 | paolosupino | Kaldemar: I tried to change that, but it didn't work. so I ended up copying what the www.didww.com gives on their support website (and still it doesn't work). |
| 11:01.59 | jacc0 | i see: USERNAME : myuser |
| 11:02.00 | jacc0 | in you log |
| 11:02.12 | jacc0 | but not in your iax.conf |
| 11:02.38 | jacc0 | sorry, i do |
| 11:02.41 | jacc0 | :P |
| 11:02.57 | kaldemar | paolosupino: if that's what www.didww.com gives as samples, i wouldn't trust them with configurations. |
| 11:03.19 | jacc0 | sorry, no I don't |
| 11:04.20 | paolosupino | kaldemar: I understand what you mean, but since my configuration wasn't working I decided to put what they give. |
| 11:04.54 | paolosupino | jacc0: I tried adding username/secret to the IAX configuration, but it didn't change… |
| 11:04.56 | kaldemar | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html |
| 11:05.43 | paolosupino | kaldemar: that is the example I used to connect to the other PBX :-) |
| 11:06.01 | paolosupino | and calling between the 2 PBXes is working wonderfully (thank you ;-) ) |
| 11:06.12 | kaldemar | paolosupino: change [46.19.209.76] to [myuser] and type=peer instead of type=friend. |
| 11:06.46 | paolosupino | kaldemar: do you want me to do that to all the IP addresses, or should I remove all other IP addresses? |
| 11:07.57 | kaldemar | paolosupino: do the calls come from multiple addresses or just one? |
| 11:08.29 | kaldemar | paolosupino: what version of asterisk are you using? |
| 11:09.25 | paolosupino | kaldemar: core show version |
| 11:09.26 | paolosupino | Asterisk 1.8.8.0~rc4-1digium0+1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-11-23 17:02:24 UTC |
| 11:13.43 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:15.25 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 11:18.57 | paolosupino | kaldemar: can it be that some feature/module is missing from asterisk's configuration? |
| 11:20.15 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:21.00 | paolosupino | kaldemar: my new didww.conf: http://pastebin.com/QVBNbZ5q |
| 11:21.13 | paolosupino | kaldemar: didn't change anything :-( |
| 11:24.12 | paolosupino | kaldemar: I noticed that username is redundant so I removed it, still doesn't work… |
| 11:28.15 | jacc0 | @paolosupino: maybe they require you to register? |
| 11:28.24 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:28.33 | paolosupino | jacc0: what do you mean? |
| 11:31.39 | *** join/#asterisk iulhk (~iulhk@116.71.188.148) |
| 11:35.20 | paolosupino | I'm starting to think that maybe there's a problem with Skypeout service… I'm getting inconsistent responses even though I didn't change the configuration the PBX side. |
| 11:37.56 | iulhk | hello all , using asterisk-10.2.0, sip to sip video call channel is established, i want to broadcast and want to watch this live call at some media player ? |
| 11:39.09 | *** join/#asterisk cstachris (~chrismylo@114.73.114.84) |
| 11:39.50 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 11:41.45 | *** join/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 11:42.23 | ziz212 | hi there, if i create custom destination, which file it get updates? Is it extensins.conf or any other ....I have searched this in the local machine and internet, but couldnt find any answer for that. |
| 11:43.47 | paolosupino | asked friends to use different services… all had the same problem as I did… This rules out SkypeOut problems… |
| 11:44.52 | *** join/#asterisk Atlan (~mm@h-213.61.221.58.host.de.colt.net) |
| 11:45.31 | *** join/#asterisk dunlingli (~dunlingli@mail.unleashbts.com) |
| 11:46.48 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 11:48.50 | kaldemar | ziz212: #freepbx |
| 11:49.35 | ziz212 | where is it the file ? |
| 11:50.07 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 11:50.29 | kaldemar | ziz212: you're on the wrong channel, ask in #freepbx. |
| 11:50.58 | ziz212 | ok thanks |
| 11:51.13 | paolosupino | kaldemar: something is very wrong with my PBX… now I don't have sip command in CLI… |
| 11:51.14 | *** part/#asterisk ziz212 (~chatzilla@203.115.2.22) |
| 12:01.24 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 12:02.32 | paolosupino | what does this message mean: [Mar 14 12:01:39] ERROR[18961]: chan_iax2.c:4911 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 46.19.210.10 in the calltokenoptional list or setting user didww requirecalltoken=no |
| 12:02.34 | paolosupino | ??? |
| 12:05.39 | kaldemar | it means that you require calltokens (a security feature introduced in the IAX2 protocol a few years ago) but the other end is probably running an old version that does not support tokens. |
| 12:09.02 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:12.05 | *** join/#asterisk ke-esc (~ke-esc@155.229.209.170) |
| 12:15.03 | ke-esc | Morning all- anyone know if its possible to support server side DND using the button on the Polycom phone? With SIP debugging on, I see that the phone sends a subscribe message to * with a SetDoNotDisturb XML tag in the content.. I'm guessing I'd have to use a custom device state but not sure how |
| 12:18.01 | [TK]D-Fender | ke-esc, this is not a "standard" and * does not support it |
| 12:19.51 | ke-esc | [TK]D-Fender, ah boo.. is it perhaps possible to create a script that will get called to process subscribe messages? |
| 12:21.42 | iulhk | any posibilies regarding , using asterisk-10.2.0, sip to sip video call channel is established, i want to broadcast and want to watch this live call at some media player ? |
| 12:26.44 | [TK]D-Fender | k3-no |
| 12:26.49 | [TK]D-Fender | ke-esc, no |
| 12:29.23 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 12:29.24 | *** join/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 12:32.24 | leifmadsen | iulhk: that feature is not yet available, but has been filed and has some code (although probably old now). It's called chan_rtmp |
| 12:32.27 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 12:32.49 | leifmadsen | iulhk: https://issues.asterisk.org/jira/browse/ASTERISK-14451 |
| 12:36.09 | [TK]D-Fender | leifmadsen, Does Chanspy also hand over both kinds of media? |
| 12:36.57 | [TK]D-Fender | leifmadsen, And how does that new channel type hook into the SIP channel for the implementation iulhk was looking to do? |
| 12:37.45 | dunlingli | I start to learn the functionality of Asterisk SNMP. Based on the book: Asterisk the definitive guide, I am able to use OpenNMS to see the graphs of Asterisk active channels, Asterisk SIP channels, In/Out traffic utilization, etc. Now, I connect iPhones using Bria soft phone app to an Asterisk server. Can Asterisk SNMP show the traffic on iPhones if iPhones are not SNMP enabled? If so, how can I set it up? |
| 12:38.23 | leifmadsen | dunlingli: well OpenNMS monitors the channels on Asterisk, so it doesnt' care what the device is, just the channel type |
| 12:38.36 | leifmadsen | if the call is being handled by asterisk, then opennms can see it |
| 12:38.42 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:39.04 | leifmadsen | it isn't monitoring the phone itself, just the information reported by asterisk |
| 12:39.14 | leifmadsen | it == snmp monitoring service |
| 12:39.19 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:39.58 | *** join/#asterisk Li7h (~dps@71-81-21-14.static.gwnt.ga.charter.com) |
| 12:40.37 | paolosupino | kaldemar: DIDWW support page has the following: Under 'iax_general_custom.conf.' add following lines: |
| 12:40.37 | paolosupino | requirecalltoken=no |
| 12:40.39 | paolosupino | calltokenoptional = 46.19.209.0/255.255.255.0 |
| 12:40.39 | paolosupino | <PROTECTED> |
| 12:40.44 | *** join/#asterisk gonewage (~gonewage@72.2.130.205) |
| 12:41.10 | paolosupino | can you explain what it means and where I should put requirecalltoken=no and calltokenoptional = 46.19.209.0/255.255.255.0 ? |
| 12:41.22 | leifmadsen | paolosupino: iax.conf |
| 12:41.34 | paolosupino | in the general section? |
| 12:42.27 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-ksnmlacaoakpidhn) |
| 12:42.40 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:42.43 | leifmadsen | paolosupino: check the iax.conf.sample for examples of where it can go |
| 12:43.24 | paolosupino | leifmadsen: thanx. I forgot about the sample files :-/. going to look. |
| 12:43.25 | kaldemar | paolosupino: requirecalltoken=no is put under peer definitions. calltokenoptional is a global option that goes under [general]. |
| 12:45.02 | kaldemar | http://downloads.asterisk.org/pub/security/AST-2009-006.html |
| 12:45.05 | *** part/#asterisk Bullmoose (~Bullmoose@65-129-4-69.bois.qwest.net) |
| 12:46.16 | iulhk | another query, by using chanspy, spy can hear audio calls, so do we have any functionality that chanspy can just watch video calls too ? |
| 12:47.07 | paolosupino | kaldemar: Thank you. going to read… |
| 12:47.09 | leifmadsen | iulhk: chanspy does not support video afaik |
| 12:47.30 | leifmadsen | iulhk: either point-to-point, or via ConfBridge() is all |
| 12:50.36 | iulhk | <@leifmadsen>: so by using ConfBridge, if 2 peer 100 & 200 are in confbridge then 3rd peer 300 can join in a silence mode and able to watch both video and audio ? |
| 12:50.52 | leifmadsen | iulhk: perhaps, yes |
| 12:51.44 | iulhk | thanx, let me test |
| 12:53.50 | bulkorok | hi... I want to check if a fax was received succesfully with faxopt(status). if it's ok it has "success" but what are the other options?! is there any documentation?! |
| 12:54.09 | *** join/#asterisk andresmujica (~andres@ubuntu/member/andresmujica) |
| 13:00.30 | *** join/#asterisk aberrios_ (5c1b1c5d@gateway/web/freenode/ip.92.27.28.93) |
| 13:01.17 | aberrios_ | I've got a RSeries 850 nearly setup. But there seems to be no heartbeat coming for either node to the R850 device.. in otherwords the LEDs for the primary and secondary are red on the device |
| 13:01.19 | aberrios_ | any ideas |
| 13:01.24 | leifmadsen | bulkorok: there is always the code :) |
| 13:02.09 | bulkorok | leifmadsen: I tried to read... but I'm not the good programmer.. |
| 13:02.26 | *** join/#asterisk timahvo1 (~rogue@41.90.151.139) |
| 13:02.35 | bulkorok | I only find the functions that read or sets the faxopt(x) values... |
| 13:02.40 | aberrios_ | it may be linked to when I disconnect the primary node the secondar attempts to come up but never goes Primary or mounts the drbd device. The primary node goes secondary in drbd, when i reconnect the primary it goes to primary again and mounts/starts * fine. |
| 13:02.47 | leifmadsen | just find the section where faxopt is set along with the one return value you do know by greping, and in the same area see if there are other return values |
| 13:02.55 | *** part/#asterisk shihan (shihan@pjr.cc) |
| 13:05.41 | dunlingli | leifmadsen: Thanks. Since the traffic between phones are mainly RTP traffic, can OpenNMS see the traffic through Asterisk RTP channels? |
| 13:05.56 | leifmadsen | dunlingli: define, "see" |
| 13:06.25 | leifmadsen | dunlingli: it's not going to allow you to look at the audio, or record it, via SNMP |
| 13:06.37 | leifmadsen | it might show you the amount of traffic passing through Asterisk in terms of bitrate... |
| 13:06.52 | leifmadsen | but that would be all, if it handles that at all |
| 13:08.10 | dunlingli | lefmadsen: That will be enough information for me. How can I set it up? |
| 13:08.38 | leifmadsen | dunlingli: you follow the instructions in Asterisk: The Definitive Guide |
| 13:08.42 | paolosupino | With my last configuration I'm now getting a busy signal … (I was never this happy to hear a busy signal ;-) ) and I'm trying to understand who is the cause of the busy signal: Asterisk or the ITSP. Is there a way I can find out? |
| 13:09.02 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 13:10.32 | jacc0 | you could use tshark |
| 13:10.40 | jacc0 | tshark -R sip |
| 13:11.20 | kaldemar | leifmadsen: now that you're here, is the IAX2 section intentionally that brief or am i missing something: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291291 |
| 13:11.45 | leifmadsen | it is intentional |
| 13:11.45 | dunlingli | lefmadsen: Yes. I fellow it all the way through. However, after Astersik established the call, traffic between phones will not pass through Asterisk server, so the traffic will not shown on in/out traffic utilization |
| 13:12.18 | leifmadsen | dunlingli: then you have to make it pass through Asterisk -- you're monitoring Asterisk, and if Asterisk can't see the data, then Asterisk can't report the data. |
| 13:13.06 | leifmadsen | kaldemar: none of the authors use IAX2 very much, the configuration is similar to SIP, and personally, I don't like the IAX2 implementation in Asterisk, so I never use it. |
| 13:17.05 | ke-esc | Is there an IAX implementation aside from asterisks? |
| 13:17.25 | dunlingli | leifmadsen: By default SIP protocol is that voice/video traffic does not go through the server after peer-to-peer communication is established. How to configure asterisk and make the traffic go through Asterisk? |
| 13:17.48 | leifmadsen | dunlingli: directmedia=yes |
| 13:17.55 | leifmadsen | s/yes/no/ |
| 13:17.58 | kaldemar | leifmadsen: ok. the reason i was asking is because i ran into sip.conf options in a sample IAX config given by a provider (again) and was trying to hand out an example on how to configure IAX. but the 2nd edition is still online. |
| 13:18.09 | kaldemar | ke-esc: libiax |
| 13:18.19 | leifmadsen | kaldemar: wherever you're looking is wrong then |
| 13:18.25 | leifmadsen | kaldemar: third edition is posted on asteriskdocs.org |
| 13:18.35 | dunlingli | leifmadsen: thanks. I will try it |
| 13:18.44 | leifmadsen | kaldemar: in fact you linked to the 3rd edition |
| 13:18.53 | ke-esc | kaldemar, ahh.. i always just assumed "Inter-Asterisk eXchange" was specific to asterisk |
| 13:19.47 | leifmadsen | ke-esc: IAX is also an RFC with the IETF |
| 13:19.49 | kaldemar | leifmadsen: yes, http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html is still functional though. |
| 13:20.01 | leifmadsen | kaldemar: functional, yes.... |
| 13:20.08 | kaldemar | there's an OpenIAX also: http://www.openiax.org/ |
| 13:21.12 | kaldemar | heh, but no releases since 2006. |
| 13:21.45 | paolosupino | kaldemar: the sip options weren't given by a provider… They were wrongly put there by the copier (that would be me) in a moment of distraction… |
| 13:25.12 | kaldemar | paolosupino: gotcha. but that has been seen before |
| 13:25.49 | paolosupino | Kaldemar: so even here I didn't invent anything new??? Sigh… ;-) |
| 13:27.45 | kaldemar | paolosupino: you can withdraw the patent application. :) |
| 13:28.59 | paolosupino | Kaldemar: I don't believe in patents :-) |
| 13:33.17 | *** join/#asterisk blitzrage (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:33.17 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:36.31 | Tamo | I |
| 13:39.01 | *** join/#asterisk TimeRider (~steve@92.41.233.134.threembb.co.uk) |
| 13:39.39 | jacc0 | you |
| 13:47.42 | *** join/#asterisk awclin (~alinford@host86-170-114-41.range86-170.btcentralplus.com) |
| 13:47.56 | *** part/#asterisk awclin (~alinford@host86-170-114-41.range86-170.btcentralplus.com) |
| 13:48.37 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:50.16 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 13:53.02 | *** join/#asterisk dxd828 (~dxd828@dxdservers-3-pt.tunnel.tserv10.par1.ipv6.he.net) |
| 13:53.12 | dxd828 | Anyone got BLF working with cisco phones? |
| 13:59.06 | *** part/#asterisk AliRezaTaleghani (~taleghani@unaffiliated/AliRezaTaleghani) |
| 14:00.21 | Dovid | hi all |
| 14:02.28 | paolosupino | anyone has any more thoughts on resolving me peering issue with DIDWW? |
| 14:04.06 | *** join/#asterisk fred9999 (~fred@59.pool85-59-166.dynamic.orange.es) |
| 14:04.46 | fred9999 | Hi I a question concerning conference across multiple boxes |
| 14:05.52 | fred9999 | Is that the correct channel |
| 14:05.54 | fred9999 | ? |
| 14:06.04 | pabelanger | yes |
| 14:06.22 | pabelanger | however, you can't have 1 conference across 2 servers |
| 14:08.17 | fred9999 | here is the situation |
| 14:09.29 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:10.11 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:10.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:10.13 | *** join/#asterisk ulogic (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 14:13.24 | fred9999 | I have 2 asterisks. operators are connected each to one asterisk and work for a call-center. an outgoing call is made from the first asterisk. Operator1 is busy, and operator2 sitting on the 2nd asterisk (same LAN no NAT) is free. When I say sitting, he has been called by the * and waiting for an incoming call to be bridged. The question is how can operator2 can pickup the call on the first |
| 14:13.24 | fred9999 | * ? |
| 14:16.31 | *** join/#asterisk grEvenX (~even@84.49.69.109) |
| 14:16.58 | fred9999 | The ideas I have are the following: |
| 14:16.59 | fred9999 | transfer the call of the 2nd operator to an extension of the 1st that would redirect to a conference room on *1 where |
| 14:16.59 | fred9999 | the incoming call would go as well. |
| 14:17.28 | *** join/#asterisk krotos (~d3v1l@host249-34-dynamic.2-87-r.retail.telecomitalia.it) |
| 14:17.32 | krotos | hi all ;) |
| 14:18.33 | fred9999 | using the meetme |
| 14:18.34 | fred9999 | or use chan_spy from the second * to "monitor" the channel on the first *. |
| 14:18.34 | fred9999 | I thought I would have to define a peer sip trunk between the 2 asterisk |
| 14:19.01 | *** join/#asterisk serafie1 (~erin@75.76.38.159) |
| 14:19.47 | fred9999 | something similar to a supervisor sitting on one * and trying to check communication |
| 14:19.47 | fred9999 | <PROTECTED> |
| 14:21.06 | fred9999 | <PROTECTED> |
| 14:23.56 | *** join/#asterisk ulogic_ (421fc7ab@gateway/web/freenode/ip.66.31.199.171) |
| 14:23.59 | krotos | I'm having some trouble with Transfer application, someone can help me? I'm try to set up on my home pbx a call transfer to my phone if no answer. i'm on asterisk 1.8 and the interconnetcion with my providere is in SIP trunk |
| 14:26.02 | [TK]D-Fender | krotos, show us what you're attempting |
| 14:26.04 | [TK]D-Fender | ~pb |
| 14:26.04 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:26.05 | [TK]D-Fender | ^^^ |
| 14:26.48 | krotos | [TK]D-Fender: ok, let me paste dialplan on pb |
| 14:26.50 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
| 14:27.54 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 14:29.46 | krotos | [TK]D-Fender: here is the dialplan that i'm testing..http://pastebin.com/bf0nbqYw |
| 14:36.20 | krotos | i'm thinking to make someting like Dial(SIP/prov-1/393406693887), but i see the Transfer application an i've thinked that shoud work.. |
| 14:36.34 | [TK]D-Fender | krotos, Show the call |
| 14:42.56 | krotos | [TK]D-Fender: here is : http://pastebin.com/i9msaxFm |
| 14:43.40 | krotos | i have access also on prov-1 asterisk box, and show me after the transfer application |
| 14:44.14 | krotos | http://pastebin.com/kSdipbwm |
| 14:44.28 | [TK]D-Fender | krotos, you should have SIP DEBUG enabled for that entire call.. you aren't loking at what's really going on there... |
| 14:45.56 | [TK]D-Fender | krotos, so prov-1 is another * box you control? go verify that the peer that is calling you even has access to what you told it to transfer to. I'm betting the contexts disallow it |
| 14:46.32 | [TK]D-Fender | krotos, No such extension/context irpinia-wholesale@whs-out-pri while calling Local channel <--- not looking like the request is getting formed correctly |
| 14:46.41 | krotos | yes, is another * box that i control |
| 14:46.48 | [TK]D-Fender | krotos, I'm suspecting you'll simply have to DIAL() it back instead of Transfer() |
| 14:49.00 | krotos | [TK]D-Fender: yes, if i Dial() back it work, but on the cell phone i can't see the icon that inform me that the call was transfered ( such as a opreator do) |
| 14:59.05 | [TK]D-Fender | krotos, I'd check the remote end of this |
| 14:59.43 | [TK]D-Fender | krotos, No such extension/context irpinia-wholesale@whs-out-pri while calling Local channel <- doesn't look like the REFER it should send back is formed right |
| 15:03.12 | krotos | right, there are no extension for called "irpinia-wholesale", but i expect that the transfered call come on "prov-1" box with "393406693887" , not with "irpinia-wholesale" |
| 15:03.19 | krotos | or i'm wrong? |
| 15:04.15 | p3nguin | Watching the console with the correct debug enabled would show you that. |
| 15:05.28 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:15.14 | jaytee | anyone know how to force CentOS to install with a standard kernel instead of PAE? I changed out memory to drop the RAM to 2GB but a reinstall still installs the PAE kernel. |
| 15:16.08 | coppice | what's wrong with a PAE kernel? |
| 15:17.09 | jaytee | aha! nevermind |
| 15:17.22 | p3nguin | Why bother? |
| 15:19.32 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 15:19.36 | *** join/#asterisk l0st-soul (~nqezhw@nat.msfsupply.be) |
| 15:20.27 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 15:22.37 | l0st-soul | hello |
| 15:24.04 | FlashDeluxe | hi @all! i am using asterisk 1.6.2 with chan_capi and i got a problem: if i restart my server the chan_capi module isn`t loaded, i get an error "CAPI not installed, chan_capi disabled!" BUT if i log in to cli and do a module reload chan_capi.so everything works fine... i wrote a start script with "asterisk -rx "module reload chan_capi.so"" but this doesn`t work either. it just works if i connect manually to the cli and do a reload, |
| 15:24.05 | FlashDeluxe | <PROTECTED> |
| 15:24.54 | l0st-soul | i heard Skype4Asterisk is not going to be supported/working anymore |
| 15:24.57 | [TK]D-Fender | FlashDeluxe, I'm suspecting there is a load-order conflict with your modules... |
| 15:25.06 | l0st-soul | is there an alternative? I only need the ability to receive calls from skype users |
| 15:25.08 | [TK]D-Fender | l0st-soul, that has long been dead. |
| 15:25.18 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 15:25.35 | [TK]D-Fender | l0st-soul, And I'm seeing every "alternative" dying as they go. Skype is closed and they are liking keeping it that way |
| 15:26.13 | l0st-soul | [TK]D-Fender: i'm still using it on a 1.4, and it works .. but i need to create another server now so i was wondering |
| 15:26.46 | l0st-soul | and i'm not interrested in placing calls TO skype, just answer calls from skype |
| 15:27.57 | p3nguin | You can use their SIP gateway product. |
| 15:28.14 | puzzled | l0st-soul: afaik since the Digium skype module was EOL'd only FreeSWITCH can do that |
| 15:28.19 | *** join/#asterisk oneadvent (~oneadvent@ip68-105-63-31.pn.at.cox.net) |
| 15:28.38 | p3nguin | Or you can use Skype's SIP gateway product. |
| 15:29.20 | FlashDeluxe | [TK]D-Fender: here is my modules.conf: http://paste.debian.net/159718/ there are not many modules loaded.. :S |
| 15:29.53 | l0st-soul | p3nguin: skype connect uh .. indeed, it seems to be free to receive calls |
| 15:30.02 | l0st-soul | i'll have a go with that and we'll see.. thanks |
| 15:30.14 | p3nguin | Report back. |
| 15:30.29 | l0st-soul | ah its still $6.95/month/canal |
| 15:31.08 | p3nguin | How bad do you want to use skype with asterisk? |
| 15:31.44 | p3nguin | I don't want to use it enough, I know that. |
| 15:33.10 | l0st-soul | well at that price i'm not sure i want it neither :) |
| 15:35.36 | *** join/#asterisk GameGamer43 (u5533@gateway/web/irccloud.com/x-zurnagwwrjtmjwfc) |
| 15:35.39 | l0st-soul | damn microsoft |
| 15:35.53 | leifmadsen | it had nothing to do with microsoft |
| 15:36.07 | leifmadsen | (discontinuation of chan_skype anyways) |
| 15:36.56 | coppice | microsoft will make skype wither and die just by being themselves. |
| 15:37.57 | [TK]D-Fender | FlashDeluxe, load => chan_capi.so <- shouldn't have to , and what is chan_capi.so=yes ? |
| 15:39.27 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 15:40.56 | FlashDeluxe | [TK]D-Fender: i don`t know exactly :( the documentation of chan_capi says that this is both needed to load chan_capi |
| 15:48.34 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 15:50.20 | bowzak | how can i increase the delay to wait for the SIP response ? i guess I don't always get the reply soon enough, then it sends like 6 retransmits and the provider blacklists the ip for 20 secs |
| 15:58.08 | paolosupino | moving to SIP to all my trouble (damn it)… |
| 16:13.46 | *** join/#asterisk paolosupino (~paolo-sup@109.112.159.184) |
| 16:13.50 | paolosupino | hi |
| 16:14.21 | paolosupino | what do I need in order for asterisk to work if both asterisk server and SIP clients are behind (different) NATs? |
| 16:14.38 | Qwell | ~nat |
| 16:14.38 | infobot | nat is probably Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 16:15.57 | paolosupino | Qwell: thanx, but externip is different for different clients…. |
| 16:16.03 | paolosupino | how do I overcome that? |
| 16:16.15 | Qwell | That's why clients register.. |
| 16:17.50 | p3nguin | externaddr (or externip in older versions) is YOUR external address. It has nothing to do with your phones in remote networks. |
| 16:29.14 | paolosupino | p3nguin: what do you mean? |
| 16:29.32 | p3nguin | I meant what I said. |
| 16:29.52 | paolosupino | I apologize for being a pain in the ass for in this, but I'm having a hard time picturing the scenario in my head… |
| 16:31.59 | kaldemar | ~sipnat |
| 16:32.00 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 16:32.39 | autofsckk | paolosupino: try to put nat=yes in the sip.conf at the "user" context |
| 16:33.12 | paolosupino | autofsckk: I did put it in the user context and not the general context. |
| 16:33.30 | autofsckk | i dont use maybe the appropiate term, but where you create your user-extension put there nat=yes and put that too in the general tab, maybe that can help you |
| 16:33.49 | paolosupino | kaldemar: thanx. I will go educate myself… |
| 16:35.24 | autofsckk | the general context specify the conditions of your asterisk box, and if its behind nat you should tell that in general, and in the specific context of different user context you should put that too |
| 16:41.33 | *** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com) |
| 16:43.44 | p3nguin | paolosupino: To configure your asterisk to work with NAT, you will put nat=yes in the general section, you will configure localnet according to the local network's address range, and you will configure either externaddr or externhost, depending on if you have a static public IP address or a dynamic one using dynamic DNS. |
| 16:45.19 | p3nguin | Doing these things will allow your asterisk to work behind a NAT with phones also behind a NAT. |
| 16:46.18 | p3nguin | It would also be a reasonable idea to explicitly define directmedia=no |
| 16:51.00 | *** join/#asterisk dijib (~root@bas10-kitchener06-1176001626.dsl.bell.ca) |
| 16:53.39 | paolosupino | p3nguin: and for phones that aren't behind NAT? |
| 16:54.39 | paolosupino | p3nguin: I set directmedia=no. |
| 16:54.43 | p3nguin | Don't worry about them. You only have to make special configuration for the system to support phones which ARE behind NAT. Phones which are not behind NAT will already work normally and will continue to work normally. |
| 16:57.28 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:04.46 | *** join/#asterisk alexoc (~alexoc@187.106.48.119) |
| 17:07.28 | autofsckk | p3nguin: the directmedia=no what does it do? |
| 17:08.12 | p3nguin | It prevents media from going direct between two devices, keeping asterisk in the media stream. |
| 17:08.27 | p3nguin | If your devices are behind NAT, this is almost already a must. |
| 17:08.37 | p3nguin | s/already/always/ |
| 17:08.57 | *** join/#asterisk paolosupino (~paolo-sup@109.114.97.86) |
| 17:12.07 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 17:13.30 | autofsckk | oh i see, thanks |
| 17:16.15 | p3nguin | The directmedia setting replaced canreinvite. |
| 17:19.09 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 17:30.26 | rrittgarn | anyone know if there is a difference as far as context goes for a transfer and a blind transfer? |
| 17:30.59 | rrittgarn | I'm having an issue where if I blind transfer someone to a parking extension, I can't retrieve the call. If i do a supervised transfer, Everything works as expected |
| 17:31.52 | [TK]D-Fender | rrittgarn, Show us |
| 17:31.55 | [TK]D-Fender | ~pb |
| 17:31.55 | 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. |
| 17:31.57 | [TK]D-Fender | ^^ |
| 17:34.45 | p3nguin | Use attended transfers for parking so you know what parking space the call has been put in. |
| 17:35.25 | rrittgarn | fender: http://pastebin.com/QJH9nzZP (regular transfer) Blind inc |
| 17:35.59 | rrittgarn | p3nguin, I do, and it works. We have a client who wants just one button press though to park a call. They have polycom 650's with sidecars, so they can see the parks as they come in |
| 17:37.07 | rrittgarn | http://pastebin.com/mMdf8mTL - Blind xfer |
| 17:39.58 | p3nguin | What is your parking lot extension? |
| 17:40.13 | rrittgarn | 70 |
| 17:40.26 | *** join/#asterisk nick1234567 (~nick12345@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 17:41.04 | p3nguin | And what is your parking space range? |
| 17:41.09 | rrittgarn | 71-74 |
| 17:41.38 | p3nguin | If you blind transfer a call to 70, does it not park the call? |
| 17:41.48 | rrittgarn | it does, but i can't retrieve it |
| 17:41.58 | rrittgarn | its like its going to a different context / parkinglot |
| 17:42.09 | p3nguin | It should announce the space to the other side of the call when you do that, but I would have thought you could still pick it up. |
| 17:42.40 | rrittgarn | I have it set to announce only to callee i believe, as on my test phones I'm not hearing it unless i do a supervised transfer |
| 17:43.21 | p3nguin | But when you blind transfer to parking, it usually announces to the OTHER side of the call, since it has been transferred improperly. |
| 17:43.38 | rrittgarn | it doesnt though |
| 17:44.24 | p3nguin | After you blind transfer to parking, have you checked to see where the call is with the asterisk CLI? |
| 17:44.46 | rrittgarn | core show channels right? |
| 17:44.53 | p3nguin | parkedcalls show |
| 17:44.58 | rrittgarn | ah sec |
| 17:45.03 | *** join/#asterisk wonderworld (~ww@dsdf-4db557b8.pool.mediaWays.net) |
| 17:45.49 | nick1234567 | Hi, I'm new here... Can we ask a question whenever or is there a line? |
| 17:46.27 | p3nguin | ~ask |
| 17:46.27 | 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. |
| 17:46.44 | rrittgarn | penguin: http://pastebin.com/hrBRAkgT |
| 17:46.53 | rrittgarn | so its going to a different context/parkinglot |
| 17:47.05 | rrittgarn | or wait |
| 17:47.22 | paolosupino | p3nguin: anything about nat? |
| 17:47.31 | p3nguin | paolosupino: Sure. |
| 17:48.06 | rrittgarn | lot default shouldn't be where it ended up right? Do i need to have different parking extensions for each context? Default parking lot is 70 as well... |
| 17:48.10 | paolosupino | S**** I mist it… |
| 17:49.19 | p3nguin | rrittgarn: It is parked in default 71. If you didn't want it in default, I guess you'll have to reconfigure the parking settings. |
| 17:49.33 | p3nguin | paolosupino: I told you everything you needed to know. |
| 17:49.44 | rrittgarn | odd that it would work properly when supervised |
| 17:49.56 | p3nguin | (1153.39) <paolosupino> p3nguin: and for phones that aren't behind NAT? |
| 17:49.59 | p3nguin | (1154.43) <p3nguin> Don't worry about them. You only have to make special configuration for the system to support phones which ARE behind NAT. Phones which are not behind NAT will already work normally and will continue to work normally. |
| 17:50.39 | p3nguin | rrittgarn: I didn't see the parking settings, so I don't know if it's odd or not. |
| 17:50.53 | paolosupino | p3nguin: but the PBX is also behind NAT. |
| 17:51.04 | p3nguin | paolosupino: Great. Go configure it as I've told you. |
| 17:53.38 | rrittgarn | p3nguin: http://pastebin.com/CirBJTUc Features.conf |
| 17:53.43 | nick1234567 | I'm running 1.8 on CentOS 6. I have "ringinuse = no" set in queues.conf, but agents who are on the phone still hear a call waiting beep. Is there another option that needs to be set? My queue has defined SIP members. |
| 17:54.36 | Qwell | nick1234567: I believe you need busy limits set in sip.conf |
| 17:56.26 | nick1234567 | like "busylevel =1" |
| 17:57.20 | nick1234567 | there's also "call-limit" parameter... what is the difference? |
| 17:57.22 | Qwell | nick1234567: it can be higher than 1 - it just needs to be set |
| 17:57.43 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 17:57.58 | nick1234567 | but if an agent can only handle one call at a time it should be 1, right? |
| 17:58.08 | Qwell | sure |
| 18:00.02 | mpoole | hey guys, I was wondering if anyone could help me with a strange dundi/iax problem between an asterisk 1.4 and 1.8 box: http://pastebin.ca/2128150 |
| 18:00.47 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
| 18:01.01 | mpoole | it looks like the recieving server sends an ACK |
| 18:01.10 | mpoole | but the sending server sees a REJECT |
| 18:01.25 | mpoole | I'm wondering if this is an imcompatibility between 1.4 and 1.8 |
| 18:01.31 | mpoole | incompatibility* |
| 18:02.05 | p3nguin | rrittgarn: You've got all sorts of overlapping going on. |
| 18:02.45 | nick1234567 | Thanks Qwell |
| 18:02.56 | p3nguin | nick1234567: If you set the call-limit to 1, that also prevents the user from putting one call on hold and making another call. |
| 18:03.13 | rrittgarn | Yeah, i figured different contexts, so it wouldn't matter. |
| 18:03.51 | p3nguin | It matters. Parking does not obey contexts. |
| 18:04.02 | rrittgarn | isn't that just splendid |
| 18:04.39 | p3nguin | Every parking lot should have its own parking spaces. |
| 18:05.03 | rrittgarn | does each lot need to be unique as well? |
| 18:05.34 | navaismo | mpoole, do you check the requirecalltoken in 1.8? |
| 18:05.38 | nick1234567 | ah, so "busylevel" is how many until the user is considered busy for Q purposes and call-limit is max concurrent calls |
| 18:05.44 | p3nguin | Well... since you can't have two lots with the same name... of course it needs to be unique. |
| 18:07.03 | mpoole | navaismo: no I didn't |
| 18:07.39 | p3nguin | Oh, but you can also set parking context per lot, and then only include THAT parking context within your phones' dialable context. Did you try that? |
| 18:08.02 | mpoole | navaismo: thanks mate it works now |
| 18:08.12 | mpoole | you just saved me so much time |
| 18:08.17 | p3nguin | I have a single parking lot. In my phones' context, I include => parkedcalls. Now the phones have access to the lot with the parking context of parkedcalls. |
| 18:08.19 | navaismo | np |
| 18:12.45 | rrittgarn | p3nguin, I have it configured that way, unique names on each context/parking lot. and am able to park within each lot |
| 18:12.56 | rrittgarn | the problem only happens when i blind transfer |
| 18:13.06 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 18:13.33 | p3nguin | But did you include the park contexts correctly? Since you failed to provide all of the relevant configuration up-front as requested, I don't know what you've configured and how you've configured it. |
| 18:15.14 | rrittgarn | my apologies, yes, I do not have the default parkedcalls context included, but i do have the specific parking lot included |
| 18:15.54 | p3nguin | Does each group of phones have their own context? And each phone group's context includes only its own park context? |
| 18:16.33 | rrittgarn | correct |
| 18:17.30 | p3nguin | If it works 100% correctly when doing an attended transfer, stop doing unattended transfers. |
| 18:17.48 | rrittgarn | haha I really wish it was that easy |
| 18:18.37 | *** join/#asterisk woleium (~woleium@208.53.145.169) |
| 18:19.23 | paolosupino | p3nguin: can you help me resolve a IAX turnk problem? |
| 18:21.06 | p3nguin | I don't know if I can or can't. Did you describe the problem and paste relevant configuration files and debug information? |
| 18:21.26 | *** join/#asterisk libryder (~libryder@libryder.com) |
| 18:21.40 | libryder | you guys know of any open api service that would allow me to get a list of incoming local numbers for a given npa-nxx? |
| 18:21.42 | paolosupino | everyone: I have a problem with IAX channel set in trunk: Asterisk processes requests minutes after a call was attempted… |
| 18:23.47 | libryder | something similar to this: http://www.localcallingguide.com/xmlquery.php |
| 18:24.01 | autofsckk | im having problems with a pap that gets disconnected, i turn on the pap, it connects right, and after a few minutes it get disconnected and shows unreachable at cli, how can that be? or how can i debug that? |
| 18:26.28 | navaismo | autofsckk, The pap its outside the network? The pap has enabled the option nat keep alive? |
| 18:26.31 | p3nguin | libryder: Are you looking for numbers that may have called you, or are you looking for DIDs? |
| 18:27.01 | libryder | when someone signs up to our service we have a field for choosing a phone number local to them "Enter your phone number" which hits this: http://www.localcallingguide.com/xmlquery.php to get a list of all NPA-NXX's local to them then searches our own inventory for a match |
| 18:27.09 | libryder | the problem with that service is that it's so sloooow |
| 18:27.36 | p3nguin | autofsckk: It could be a NAT that is slamming shut. Enable qualify times in asterisk for that peer and/or enable NAT keepalives in the ATA. |
| 18:28.01 | p3nguin | Do not set NAT traversal to yes in the ATA -- let asterisk handle NAT for the device. |
| 18:28.11 | p3nguin | And don't forward ports to the ATA. |
| 18:29.17 | p3nguin | libryder: So you're looking for available DIDs. |
| 18:29.46 | autofsckk | i have qualify=3000 doint the nat keepalives in ata, and ports arent forwarded |
| 18:30.01 | libryder | p3nguin: yep |
| 18:30.09 | libryder | well just the npa-nxx |
| 18:30.29 | p3nguin | What's the point of knowing half of a phone number? |
| 18:31.14 | libryder | i just explained it :) to be able to search our own inventory for a number local to a number entered |
| 18:32.17 | libryder | check out that link.. their api only needs npa/nxx |
| 18:33.26 | rrittgarn | p3nguin: been doing some testing of my own, even if i set up an extension to call Park() and it functions properly when not blind xfer, it still goes to default when blind... so I'm out of ideas |
| 18:33.38 | autofsckk | p3nguin: i did the things you told me but after 3 minutes it just got disconnected again |
| 18:33.40 | rrittgarn | i may have to explain to my boss thats how it has to be done |
| 18:34.28 | p3nguin | Tell him you're not supposed to blind transfer to park. It doesn't work if you do. |
| 18:34.30 | autofsckk | p3nguin: i couldnt find the NAT keepalives in the ATA |
| 18:34.52 | rrittgarn | specific site we're working with is a car dealership... so they are used to just "put it on 1" mentailty |
| 18:34.58 | rrittgarn | they don't want to sit and listen to park numbers |
| 18:35.53 | p3nguin | Polycom phone should be able to have a park button configured which parks correctly, but if you don't know the space number, I don't know how you can pick up the call. |
| 18:36.16 | rrittgarn | we have the sidecars with BLFS on the parking lots |
| 18:37.06 | rrittgarn | currently using the nifty little "automata" function, press park (parking extension), then watch it light up n the phone and then page |
| 18:42.26 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 18:42.51 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 18:42.57 | p3nguin | I just looked in my user training materials, and it says: To use Call Parking, with your call on the line, press the Transfer soft-key, and then press the Park line key. To retrieve the call, press the Park line key. |
| 18:44.13 | p3nguin | I know that two of the line keys on the phone say Park1 and Park2. |
| 18:44.47 | autofsckk | navaismo: that nat keep alive i couldnt find, i find the option to put 90 in te NAT Keep Alive Intvl space, will that help? or where can i put the keep alive option? |
| 18:44.55 | p3nguin | I unfortunately do not know what those keys are configured to do. |
| 18:47.10 | autofsckk | well its now 5 minutes and it is still connected |
| 18:47.26 | autofsckk | well not anymore :( |
| 18:49.40 | navaismo | is a pap2t or only pap? |
| 18:50.48 | autofsckk | pap2 3.1.23(LS) |
| 18:50.56 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 18:51.26 | autofsckk | navaismo: i found the nat keep alive, the thing i cant find is the nat traversal thing |
| 18:51.39 | p3nguin | It should be near it. |
| 18:52.32 | navaismo | enabled the nat keep alive and nat mapping |
| 18:53.02 | *** join/#asterisk philippel_mac (~p_lindhei@c-24-19-228-236.hsd1.wa.comcast.net) |
| 18:53.43 | autofsckk | ok |
| 18:53.43 | autofsckk | done |
| 18:58.03 | autofsckk | it isnt working |
| 18:58.12 | autofsckk | [Mar 14 14:49:11] NOTICE[27860]: chan_sip.c:20764 handle_response_peerpoke: Peer '0014bf4efa91-hub' is now Reachable. (125ms / 3000ms) |
| 18:58.15 | autofsckk | [Mar 14 14:57:12] NOTICE[27860]: chan_sip.c:26141 sip_poke_noanswer: Peer '0014bf4efa91-hub' is now UNREACHABLE! Last qualify: 158 |
| 18:59.11 | p3nguin | Are you having network problems? If you let it be for a while, does it ever go reachable again? |
| 18:59.11 | autofsckk | the pap2 its behing pfsense, im not natting, i just opened a hole at the firewall |
| 18:59.22 | *** part/#asterisk mateu (~mateu@missoula.org) |
| 18:59.22 | p3nguin | What is the IP address of the pap2 device? |
| 18:59.42 | autofsckk | the network is working fine, well i can leave it for a while to see if it gets connected again |
| 18:59.55 | autofsckk | internal or external? |
| 19:00.05 | p3nguin | The address of the device |
| 19:00.16 | autofsckk | ok 192.168.10.102 |
| 19:00.17 | p3nguin | Then you are behind NAT. |
| 19:00.28 | autofsckk | yes |
| 19:00.31 | p3nguin | whether you like it or not. |
| 19:00.45 | p3nguin | <autofsckk> the pap2 its behing pfsense, im not natting, i just opened a hole at the firewall <-- wrong |
| 19:01.12 | navaismo | nat=yes in the sip peer definition? |
| 19:01.16 | autofsckk | you told me minutes ago that i didnt have to make natting in the fw |
| 19:01.22 | autofsckk | navaismo: yes |
| 19:01.38 | autofsckk | so i should do nating in the pfsense? |
| 19:01.46 | p3nguin | nat= should not be any any peer definition. It goes in general section only. |
| 19:01.58 | p3nguin | You're already natting. |
| 19:02.08 | p3nguin | Stop trying to confuse yourself. |
| 19:02.13 | navaismo | really? I always set per device not general |
| 19:02.22 | p3nguin | I guess you didn't get the memo. |
| 19:02.33 | navaismo | nope |
| 19:02.37 | navaismo | :p |
| 19:02.43 | p3nguin | What asterisk version are you using? |
| 19:02.57 | navaismo | 1.6 and 1.8 branches |
| 19:03.05 | p3nguin | VERSION |
| 19:03.29 | navaismo | 1.8.7.2 |
| 19:03.44 | navaismo | 1.6.2.11 |
| 19:03.53 | p3nguin | I think that's why you didn't get the memo. I think it was added in 1.8.8.0. |
| 19:04.26 | navaismo | 1.8.9.3 |
| 19:04.52 | p3nguin | It's a security thing. Set nat=yes in the general section if you need to support any phones behind NAT. Do not set it in each peer entry. |
| 19:05.00 | navaismo | so for > 1.8.8.0 |
| 19:05.01 | navaismo | ok |
| 19:05.14 | p3nguin | It's true for all versions, but the memo and checking was added later. |
| 19:05.59 | p3nguin | Let me see if I can find the info for you. |
| 19:06.13 | autofsckk | this fckn telephone was working right yesterday, i sometimes hate this :S |
| 19:08.49 | philippel_mac | Asterisk 10.2.0 seems to not honor ringinuse=yes on a queue, if the phone is inuse regardless of the setting or removing the setting completely it never tries to call the phone. Anyone know if there is something new needed in 10 along with this? |
| 19:09.53 | philippel_mac | I notice the mention of an "ignorebusy" flag in the sample conf file which seems to imply a per agent configuration that can have a similar affect but there is nothing describing how to use it other than some vague references? |
| 19:15.24 | autofsckk | p3nguin: do you use siproxd with pfsense? |
| 19:16.23 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 19:16.33 | p3nguin | http://downloads.asterisk.org/pub/security/AST-2011-013.html |
| 19:16.52 | p3nguin | I can't find the exact memo unless I look in chan_sip I guess. |
| 19:16.54 | p3nguin | autofsckk: No. |
| 19:17.11 | p3nguin | I don't use siproxd, and I don't use pfsense. |
| 19:17.34 | [TK]D-Fender | <p3nguin> nat= should not be any any peer definition. It goes in general section only. <- pardon? |
| 19:17.57 | p3nguin | You must have missed the memo, too. |
| 19:18.03 | navaismo | LOL |
| 19:18.10 | navaismo | let me read that |
| 19:18.13 | [TK]D-Fender | What memo? |
| 19:19.14 | [TK]D-Fender | While I see that post as a potential issue.. it has no resolution, and doesn't preclude the need. |
| 19:19.36 | [TK]D-Fender | That looks tantamount to "life sucks and we don't have a fix" |
| 19:20.31 | navaismo | LMAO |
| 19:20.32 | p3nguin | http://pastebin.com/Abb4X3Ch |
| 19:20.57 | p3nguin | "use the global 'nat' setting and do not set 'nat' per peer/user." |
| 19:21.51 | *** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66) |
| 19:23.36 | navaismo | "The default NAT setting has been changed to what we believe the most commonly used setting for the respective version in Asterisk 1.4.43, 1.6.2.21, and 1.8.7.2. |
| 19:23.38 | navaismo | " |
| 19:23.53 | *** part/#asterisk libryder (~libryder@libryder.com) |
| 19:24.00 | navaismo | so versions > 1.8.7.2 fixed it? |
| 19:24.42 | p3nguin | While there is a difference between "do not" and "not able to," I'm going to recommend the "DO NOT" part rather than the "do it if you want to" part. |
| 19:25.03 | p3nguin | In versions > 1.8.7.2, you get the warning when chan_sip loads. |
| 19:25.17 | p3nguin | I pasted the warning. |
| 19:25.23 | navaismo | In versions > 1.8.7.2, you get the warning when chan_sip loads |
| 19:25.31 | bent_screwdriver | having trouble with vpm echo cancelation module on a digium t1 card. thinking of doing away with it and going to sw echo cancellation. has anyone had good experience with sw echo cancellation? |
| 19:25.41 | navaismo | me stupid mouse |
| 19:26.54 | [TK]D-Fender | p3nguin, It's a warning. It doesn't mean it isn't a NECESSARY evil |
| 19:27.23 | navaismo | argh |
| 19:27.28 | navaismo | stupid me |
| 19:30.22 | p3nguin | I enforce best practice whenever possible. |
| 19:32.40 | [TK]D-Fender | p3nguin, You don't pick a "nat" setting just "because". You pick the one you need for it to work. So it's not a question of "can have a back-draw", it's "won't work without it" |
| 19:33.38 | *** join/#asterisk chasing`Sol (~cS@197.132.91.187) |
| 19:34.25 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:34.33 | navaismo | mom im very confused now |
| 19:35.14 | p3nguin | Of course you don't pick a setting just because. If you need to support any devices behind a NAT, set nat=yes in the general section. If you do not need to support any devices behind NAT, set nat=no in the general section. It is very clear and very easy to follow. |
| 19:35.23 | p3nguin | Don't be confused. Just read what I've said. |
| 19:36.53 | autofsckk | the pap is working now, thanks a lot |
| 19:37.44 | navaismo | what was the problem? |
| 19:40.52 | p3nguin | I'd bet on pfSense. I think I've seen more people with pfSense having problems than people with a Linux-based firewall/router/gateway having problems. |
| 19:41.11 | autofsckk | navaismo: i really dont know, yesterday the pap was working right without the things p3nguin told me to change, i change them but there was a problem with my pfsense firewall i think |
| 19:41.30 | Nugget | some versions of linux have trouble passing traffic through pf if it has scrub on |
| 19:41.42 | Nugget | linux emits packet fragments with the no-fragment bit on |
| 19:42.15 | p3nguin | And scrubbing is default for most deployments. |
| 19:42.37 | p3nguin | By default for most deployments, I mean almost no one leaves it out. |
| 19:43.27 | [TK]D-Fender | p3nguin, nat=yes isn't about devices behind NAT primarily. It is so * knows that ititself is behind NAT and applies localnet settings, and also serves as a default treatment of unknown peers. |
| 19:43.30 | [TK]D-Fender | (as is won't match ones" |
| 19:45.09 | navaismo | ?_? |
| 19:45.37 | autofsckk | i have recently read something about that NAT thing, and it was very similar of that [TK]D-Fender is talking about |
| 19:47.03 | *** join/#asterisk chasing`Sol (~cS@197.132.141.130) |
| 19:48.31 | navaismo | is there some punctual information about it, I alwyas used to set nat=yes for those devices outside my lan |
| 19:48.39 | navaismo | only |
| 19:49.53 | p3nguin | "and also serves as a default treatment of unknown peers." <--- The nat setting in the general section serves as a default treatment of all peers configured in sip.conf, too. If you NEED to override the general setting, you CAN do so in a peer's definition. |
| 19:52.23 | [TK]D-Fender | p3nguin, Yes, but * also needs to know that it itself is behind NAT. It's a trigger as to whether or not * checks the localne's./ |
| 19:57.23 | p3nguin | It is also used when Asterisk it not behind a NAT itself: "The nat= setting is used when Asterisk is on a public IP, communicating with devices hidden behind a NAT device (broadband router). If you have one-way audio problems, you usually have problems with your NAT configuration or your firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP ports for incoming audio in rtp.conf" |
| 19:57.48 | autofsckk | my pap2 is now working with 1 line, if i want to enable the other line, should i change the port? i have to do the nat keep alive and those things you told me about? |
| 19:58.23 | p3nguin | Or in newer sip.conf samples: "If Asterisk is on a public IP, and the phone is inside of a NAT device you will need to configure nat option for those phones. Also, turn on qualify=yes to keep the nat session open" |
| 19:59.00 | p3nguin | The general nat setting isn't only for if Asterisk is behind nat or not. |
| 19:59.57 | p3nguin | autofsckk: If you go to line 2's tab, its sip port should already be set to 5061. In the peer entry that you are going to create in sip.conf, just set port=5061 for it. |
| 20:00.08 | [TK]D-Fender | Correct, but it is needed for that case regardless of the other |
| 20:03.52 | *** join/#asterisk bowzak (~bowzak@95.170.203.162) |
| 20:03.56 | *** part/#asterisk bowzak (~bowzak@95.170.203.162) |
| 20:04.43 | autofsckk | chan_sip.c:28645 reload_config: No valid transports available, falling back to 'udp' i get this error, do you know what it is about? i mean, where do i fix that?} |
| 20:05.46 | p3nguin | You can ignore it, or you can change your register statement to include the transport of udp:// |
| 20:06.02 | p3nguin | I don't think it's an error, as you say, though. |
| 20:06.09 | p3nguin | I think it's a NOTICE or WARNING. |
| 20:09.32 | autofsckk | yes, warning |
| 20:10.55 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:10.55 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.0 (2012/03/05), 1.8.10.0 (2012/03/05), 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 |
| 20:11.15 | autofsckk | i cant connect line2 with asterisk |
| 20:11.21 | p3nguin | Why not? |
| 20:11.26 | autofsckk | is there a way to debug pap? |
| 20:11.42 | p3nguin | Just configure it the same way as the other. |
| 20:11.43 | autofsckk | well i dont know, it doesnt even show in cli the attempt to connect |
| 20:12.58 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:12.58 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.0 (2012/03/05), 1.8.10.0 (2012/03/05), 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 |
| 20:14.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 20:14.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.0 (2012/03/05), 1.8.10.0 (2012/03/05), 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 |
| 20:15.10 | p3nguin | You shouldn't have to enable nat mapping on a device. If you do, it will usually break asterisk's ability to correctly compensate for nat. |
| 20:15.25 | autofsckk | it takes too long to make a call, i dial and then it takes maybe 10 secs to start the actual dialing, could that be fixed? |
| 20:15.39 | p3nguin | That's a result of poor dial plan in the ATA. |
| 20:15.49 | autofsckk | p3nguin: ok i can put it back to not doing nat, but it wasnt even connecting |
| 20:16.16 | p3nguin | Pastebin your config for those two peers and take screenshots of line1 and line2 pages. |
| 20:16.33 | autofsckk | im sure its that bad dial plan, im testing different things |
| 20:16.34 | kaldemar | autofsckk: set transport=udp under [general] to get rid of the warning. |
| 20:17.05 | autofsckk | thanks |
| 20:17.41 | autofsckk | i have to go and get some food, thanks for the help, ill be back in maybe 2 hours :D see you and thanks again |
| 20:17.57 | p3nguin | Ah, I remember having to set transport=udp in my ITSP peer entry. |
| 20:20.42 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-likddhrgqvdosfls) |
| 20:20.42 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 20:23.00 | *** join/#asterisk jrose_atDigium (~jrose_atD@nat/digium/x-xrafeuuraoyabkou) |
| 20:23.39 | kaldemar | in asterisk 10 you get the warning if the option is not found under general. even if e.g. udpbindaddr is used. |
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| 20:25.26 | p3nguin | As a try to recall the configuration, the reason it complained about transport type in my register statement was because of a colon (:) in my password. |
| 20:25.53 | p3nguin | s/a/I/ |
| 20:29.43 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 20:29.43 | [TK]D-Fender | checkout time, later all |
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| 20:35.28 | *** part/#asterisk gonewage (~gonewage@72.2.130.205) |
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| 20:50.09 | *** join/#asterisk diegocn (~diegocn@unaffiliated/diegocn) |
| 20:52.07 | diegocn | Hello ppl. Is possible run multiple instances os asterisk in the same machine? (without virtualization) |
| 20:53.56 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 20:54.43 | *** join/#asterisk nighty^ (~nighty@static-68-179-124-161.ptr.terago.net) |
| 20:56.18 | Gugge | diegocn: yes |
| 20:56.33 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:57.31 | diegocn | Gugge, do you have some link to share about that? I'm looking at google but there are not much stuff about that. |
| 20:57.37 | p3nguin | diegocn: Yes, but you may want to maintain separate asterisk.conf files for each. |
| 20:58.05 | p3nguin | You'll also have to choose a different listen port for each, or bind each to a unique IP address. |
| 20:58.22 | paolosupino | what module enables the s (start) extension in extensions.conf? |
| 20:58.48 | diegocn | p3nguin, that is it... one ip per instance |
| 20:59.08 | p3nguin | paolosupino: nothing |
| 20:59.43 | p3nguin | Modules so not enable extensions. The administrator configures extensions. |
| 20:59.53 | p3nguin | s/so/do/ |
| 21:01.17 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 21:02.04 | sawgood | Hi: kind of seeing something 'different' here ... I have 5 or 6 remote SIP phones which have always worked to an * 1.8.7.x box (today ... for some reason x201 will no longer 'register') ... |
| 21:02.29 | sawgood | Inside of my SIP set debug messages, I see something labelled as, "loop detected" in the SIP message header |
| 21:02.37 | sawgood | I have not seen this type of thing before (any tips)? |
| 21:13.32 | *** join/#asterisk remuda (~Tim@173.243.243.172.rhinocommunications.net) |
| 21:13.46 | ChannelZ | Does it actually say detectED or just detect? |
| 21:17.16 | sawgood | I am trying to register the phone to another system, and I am not able to ... this phone had been working for many months (day in day out). |
| 21:17.34 | sawgood | Maybe a user did something inside the GUI of the phone, so I'll try to factory reset it |
| 21:27.44 | paolosupino | Question: If I put in the dial plan the 's' extension dialing the DID sounds busy, but if I put the DID's number in dial plan the dial plan is execute properly. What am I doing wrong? |
| 21:29.24 | p3nguin | paolosupino: What you are doing wrong is trying to use extension s where it doesn't belong. Go read the book to understand extensions and dial plan. |
| 21:29.27 | p3nguin | ~book |
| 21:29.27 | 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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| 21:37.31 | p3nguin | autofsckk: http://imagebin.org/203512 |
| 21:47.44 | *** part/#asterisk serafie (~erin@nat/digium/x-zxxxxffawgjnprom) |
| 22:05.08 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 22:14.54 | p3nguin | How can I get my perl script being run by AGI to output full verbose? |
| 22:15.31 | KNERD | agi set debug on, or somethign like that |
| 22:16.22 | p3nguin | I did that and it doesn't output the actual perl script being executed. |
| 22:16.35 | p3nguin | It just outputs the AGI execution. |
| 22:18.05 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 22:19.33 | KNERD | then you will have to have it output it to the screen |
| 22:19.41 | navaismo | in phpagi i use verbose in the script any function like that in perl? |
| 22:19.48 | KNERD | by executing it yourself |
| 22:20.27 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 22:25.24 | [TK]D-Fender | p3nguin: You don't get to see the actual code from an AGI run... you can only see what you with STDIN STDOUT, STDERR |
| 22:25.42 | p3nguin | pewp |
| 22:28.41 | KNERD | wel you can tell it to output what it has at that moment |
| 22:28.51 | KNERD | during execution |
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| 22:44.55 | phix | p3nguin: yeah do some logging |
| 22:45.09 | p3nguin | Logging of what sort? |
| 22:46.24 | phix | well I am sure what ever scripting or programming language you are using has a logging API |
| 22:47.28 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 22:47.28 | phix | Logger.getLogger("someNameForALogger").log(Level.WARNING, "example for java and similar syntax for python as well", exceptionToLogStackTrace); |
| 22:47.33 | p3nguin | I don't know perl, so I know nothing about how to log by any perl API. I am not a programmer, but I am trying to hack together some crap in a perl script and it doesn't work, so I need to see what exactly it is doing when it executes if at all possible. |
| 22:47.56 | phix | Why are you using AGI if you cant code? |
| 22:48.05 | p3nguin | I need the output, obviously. |
| 22:49.21 | phix | hmm, what does the AGI do? |
| 22:50.28 | phix | for starters you can see if you perl script has a use warnings; line or -w / -W flag set as a perl option |
| 22:50.51 | phix | that will give you more debuggy output |
| 22:51.20 | phix | whether it gives out relevant or useful information is another thing |
| 22:52.28 | [TK]D-Fender | [18:47]p3nguinI don't know perl, so I know nothing about how to log by any perl API. I am not a programmer, but I am trying to hack together some crap in a perl script and it doesn't work, so I need to see what exactly it is doing when it executes if at all possible. <- you never get to see that in real programming at all as it is, let alone AGI |
| 22:52.44 | Kobaz | haha |
| 22:52.48 | Kobaz | perfect |
| 22:53.17 | [TK]D-Fender | p3nguin: That's why you use the standard output devices to call things like NoOp with messages to indicate your progress, etc. You'll never see regular actual code execute, just your output lines |
| 22:54.46 | phix | [TK]D-Fender: hai! |
| 22:55.02 | [TK]D-Fender | phix: herro |
| 22:55.28 | phix | I got me a snom 320, it has a record button, I push said record button and in my logs it tells me recording isn't enabled, URL me please :) |
| 22:56.22 | phix | I could google it but I am tired and slightly thirsty |
| 22:56.26 | dym | lol |
| 22:56.38 | dym | "Please spoonfeed - too thirsty to search..." |
| 22:56.43 | phix | :P |
| 22:56.45 | dym | i see what u did there |
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| 22:57.56 | phix | Being dehydrated can really send your motivation levels downhill fast |
| 22:58.05 | dym | your point beeing? |
| 22:58.12 | dym | that automagically makes other people more responsive? |
| 22:58.22 | citywok | phix: especially if the dehyrdation is from beer |
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| 23:05.41 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 23:06.39 | [TK]D-Fender | concurs .... beer makes others more responsive... even to questions you never asked... |
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| 23:31.39 | phix | citywok: yeah, I wish it was beer related |
| 23:32.30 | phix | but enough about me, lets asterisks! |
| 23:37.01 | navaismo | wich is the separator for the weekdays in app GOTOIFTIME for no ranges |
| 23:37.32 | navaismo | s/wich/which/ |
| 23:38.35 | navaismo | i use per example mon-fri to use mondat to friday inlcuding the day between |
| 23:38.52 | navaismo | but if i want to use only monday and frida, I use the comma? |
| 23:38.55 | navaismo | the pipe? |
| 23:38.59 | navaismo | ??? |
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| 23:54.36 | phix | navaismo: pipe is old school, but I will look at my config, as I use it |
| 23:56.27 | [TK]D-Fender | navaismo: probably isn't an option for multiple. Betting you have to do them on separate lines |
| 23:56.48 | phix | exten => s,1,GotoIfTime(09:00-16:59,Mon-Fri,*,*?DialMe,1:OutOfBusinessHrsMessage,1) |
| 23:57.11 | phix | hmmmm |
| 23:57.22 | phix | I guess that doesnt help you |
| 23:57.55 | navaismo | yep i do it per day |
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| 23:58.21 | phix | navaismo: perhaps have two lines, if the first one doesn't match, goto the next one |
| 23:58.57 | navaismo | i need the five days, only mon and fri have same settings |
| 23:59.04 | datruth | i should give asterisk a try today see if I cant get it functioning |
| 23:59.18 | phix | exten => s,1,GotoIfTime(09:00-16:59,Mon,*,*?SomeContext,1) |
| 23:59.32 | phix | exten => s,2,GotoIfTime(09:00-16:59,Fri,*,*?SomeOtherContext,1) |
| 23:59.49 | phix | exten => s,3,GotoIfTime(09:00-16:59,Tue-Thur,*,*?SomeOtherOtherContext,1) |
