| 00:05.35 | *** part/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 00:08.40 | *** join/#asterisk Netgeeks (~chris@173.11.68.156) |
| 00:23.44 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 00:26.30 | gusto | i have no peers |
| 00:34.04 | *** join/#asterisk gusto (~gusto@nrbg-4dbe37e1.pool.mediaWays.net) |
| 00:34.18 | gusto | uri="sip:sip:HSS@ims.telefonica.de" |
| 00:34.35 | gusto | i do not like the "sip:sip:..." in the log |
| 00:35.33 | gusto | because the other party says ,domain="sip:HSS@ims.telefonica.de |
| 00:36.42 | p3nguin | Why are you so worried about the domain? Don't you know your credentials for their system? |
| 00:37.55 | gusto | no |
| 00:38.04 | gusto | the other side has sip: only once |
| 00:38.22 | gusto | but in my register requests there are sip:sip: twice |
| 00:38.27 | gusto | maybe that is the problem |
| 00:40.09 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:40.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:44.14 | gusto | Reliably Transmitting (no NAT) to 62.52.148.39:5060: <- why is there a : at the end? |
| 00:44.40 | gusto | something does not fit |
| 00:45.05 | gusto | i am sending the right requests and i am getting also answers from the voip |
| 00:45.20 | gusto | so there seems to be no problem, but it does not authenticate |
| 00:45.37 | gusto | it does not say that it is wrong, it just gets a timeout on the other side |
| 00:45.53 | gusto | while i am sending requests all the time |
| 00:46.02 | gusto | so there must be something that does not fit |
| 00:46.55 | gusto | how do i change this realm? |
| 00:47.17 | gusto | ah, i found that |
| 00:49.59 | gusto | somehow it has no effect changing that |
| 00:51.31 | gusto | where can i adjust the realm name of the other party? can i? |
| 00:53.43 | gusto | is there a documentation for asterisk, or is it possible only to google at it? |
| 00:54.59 | *** join/#asterisk BuenGenio (~Gene@113.28.26.162) |
| 00:55.04 | *** join/#asterisk FainaUkraina (~Gene@113.28.26.162) |
| 00:57.29 | *** join/#asterisk doneir (~var@60-242-117-97.static.tpgi.com.au) |
| 00:59.12 | doneir | hey all, I'm trying to find out a way (in dialplan) to check to see whether a specific channel is currently in use (PRI 30 channels available) - this is for additional code handling in the dialplan later. Unfortunately it seems the current functions (DEVICE_STATUS, ChanIsAvail) focus on trunks/groups (e.g is there a chan availble on dahdi g1), not to see if a channel that was created is currently still in use |
| 00:59.41 | doneir | the reason i need this check is due to a bug in Read() where if a user hangs up during Read() the hangup extension is not executed |
| 01:01.14 | doneir | otherwise i could handle this manually in a create and tear-down process |
| 01:01.31 | WIMPy | And how are you checking for the dialpan for another call? |
| 01:01.37 | WIMPy | Why don;t you listen on AMI? |
| 01:01.59 | doneir | is AMI accessible through AEL? |
| 01:02.20 | WIMPy | No, It's a socket interface. |
| 01:02.39 | doneir | I've done a lot of development to keep the maintanence completely in AEL |
| 01:10.23 | gusto | how do i change this digest authorization preferences? |
| 01:17.27 | ChannelZ | change it how? |
| 01:20.04 | gusto | not to have sip:sip there |
| 01:20.10 | gusto | and only sip! |
| 01:20.15 | gusto | like on the other side |
| 01:21.44 | gusto | what is the difference between domain and uri? |
| 01:31.46 | p3nguin | domain is something like domain.com. URI is the entire resource identifier, such as gusto@domain.com. |
| 01:32.13 | *** join/#asterisk wonderworld (~ww@dsdf-4db54587.pool.mediaWays.net) |
| 01:32.48 | gusto | Authorization: Digest username="4991131042466", realm="ims.telefonica.de", algorithm=MD5, uri="sip:sip:HSS@ims.telefonica.de" |
| 01:33.22 | gusto | WWW-Authenticate: Digest realm="ims.telefonica.de",domain="sip:HSS@ims.telefonica.de |
| 01:33.29 | gusto | SIP/2.0 401 Unauthorized |
| 01:34.25 | gusto | [Mar 5 01:34:10] NOTICE[1681]: chan_sip.c:20027 handle_response_register: Failed to authenticate on REGISTER to '4991131042466@sipproxy.endesha.be-converged.com' (Tries 3) |
| 01:34.29 | p3nguin | You're still sending the wrong information. |
| 01:34.36 | gusto | yes |
| 01:34.39 | gusto | how do i change it? |
| 01:34.44 | ChannelZ | There is no nonce in the authenticate reply, unless you erased it |
| 01:34.55 | gusto | i did not send it |
| 01:34.56 | gusto | w8 |
| 01:35.02 | gusto | Authorization: Digest username="4991131042466", realm="ims.telefonica.de", algorithm=MD5, uri="sip:sip:HSS@ims.telefonica.de", nonce="8a302fd399574273581a2caac4038c21", response="d950dc68bdc698b08866ed8f611c1211", qop=auth, cnonce="7c5af395", nc=00000002 |
| 01:35.16 | gusto | WWW-Authenticate: Digest realm="ims.telefonica.de",domain="sip:HSS@ims.telefonica.de",nonce="f253f915a8a88745ed6c64b613b88dfd",stale=false,qop="auth",algorithm=MD5 |
| 01:35.22 | gusto | SIP/2.0 401 Unauthorized |
| 01:35.23 | gusto | Via: SIP/2.0/UDP 77.190.55.225:5060;branch=z9hG4bK4aaabc7e |
| 01:35.25 | *** join/#asterisk johnwigley (bc4027e7@gateway/web/freenode/ip.188.64.39.231) |
| 01:36.00 | gusto | nonce should match? |
| 01:36.46 | gusto | it does not match here |
| 01:36.50 | gusto | i just saw |
| 01:36.57 | ChannelZ | I don't know where the double sip is coming from. What does your register line and the peer entry look like (pastebin.com and you can XXX out your secret= there) |
| 01:37.23 | gusto | w8 |
| 01:37.31 | gusto | that SIP SIP was not my idea |
| 01:37.32 | johnwigley | Hi, I'd appreciate any help with a question. I've only just started using Asterisk and haven't really got a clue a how to execute specific dialplan code when a channel is connected. What I'm trying to do is to call a particular channel and when they answer then bridge the answering channel with another existing channel. |
| 01:37.40 | gusto | and that does not come from SIP.CONF !!! |
| 01:38.53 | Kobaz | anyone ever set up polycom ldap corporate directory |
| 01:43.11 | johnwigley | Do the options M (execute Macro) & G (execute Gosub) for the Dial cmd fire after the called channel has answered, and would therefore allow me to run a Bridge or ChannelRedirect to join the called channel with another existing channel? |
| 01:44.47 | *** join/#asterisk RickCogley (~anonymous@EM1-112-221-65.pool.e-mobile.ne.jp) |
| 01:46.00 | gusto | why is there realm="ims.telefonica.de", algorithm=MD5, uri="sip:sip:HSS@ims.telefonica.de" ? |
| 01:46.21 | gusto | i did not put it inside the sip.conf ... so it must be somehow from the other party |
| 01:46.39 | gusto | and i seem not to have the tools to overwrite that |
| 01:51.15 | ChannelZ | The realm seen in the WWW-Authenticate header comes from the remote end |
| 01:52.43 | ChannelZ | It seems like they don't like the username and/or password you're trying to auth with. |
| 01:53.18 | gusto | username does not work, that one is rejected by 403 |
| 01:53.21 | gusto | forbidden |
| 01:53.36 | gusto | so i am authorising with my telephone number |
| 01:53.45 | gusto | that goes till the auth sent |
| 01:54.28 | ChannelZ | Well you're apparently not saying what they want to hear, which I can only guess at since I don't know what they require. |
| 01:56.48 | gusto | so ... |
| 01:56.54 | gusto | what do they want to hear? |
| 01:57.01 | ChannelZ | ??? Ask tech support. |
| 01:57.06 | gusto | no |
| 01:57.09 | gusto | they have no idea |
| 01:57.23 | gusto | i can put in everything, but i have to know where |
| 01:57.30 | ChannelZ | Surely they have some information/instructions on configuring Asterisk, or otherwise have authentication information if you were configuring a SIP phone directly or something |
| 01:58.18 | gusto | http://www.avego.de/faq/index.php/faq/39-fragen-zu-den-dsl-komplettpaketen/142-wie-richte-ich-meinen-telefonanschluss-bei-den-komplettpaketen-ein |
| 01:58.21 | gusto | that is all |
| 01:58.30 | ChannelZ | We're just guessing here, we haven't seen your sip.conf containing your register line and the SIP peer you defined, but even so we don't know what specific requirements they might have. |
| 01:59.04 | gusto | register => fromuser@fromdomain:secret@host |
| 01:59.43 | gusto | register => udp://4991131042466@endesha.be-converged.com@sipproxy.endesha.be-converged.com |
| 02:01.36 | ChannelZ | try just register => 4991131042466:SECRET@sipproxy.endesha.be-converged.com where SECRET is your password |
| 02:02.08 | gusto | forbidden |
| 02:02.32 | gusto | it works only with user@domain:secret@host |
| 02:02.52 | gusto | and comes that far |
| 02:03.10 | gusto | so it does not matter if secret is there or not |
| 02:03.56 | ChannelZ | I don't speak german or whatever this page is in. is 4991131042466 your auth username from them? This page seems to suggest you'd have a different username |
| 02:04.08 | gusto | no |
| 02:04.23 | ChannelZ | Either you're wrong or this page is lying then. |
| 02:04.35 | WIMPy | Try user@registrar:passwd:user@proxy/user |
| 02:04.41 | gusto | there is also a checkbox what says that you should authenticate with tel.nr. |
| 02:05.44 | gusto | as soon as i take user he says - forbidden - the same he says when you put in some random value |
| 02:06.21 | ChannelZ | well it's not going to tell you you misspelled your username or your password is "close, but not close enough" |
| 02:06.36 | gusto | so the telephone number is right+registar i have+passwd does not matter+user i ve tried+proxy is there+user i have tied out too |
| 02:06.42 | gusto | tried |
| 02:06.43 | ChannelZ | my guess is it's username@endesha.be-converged.com:SECRET:4991131042466@sipproxy.endesha.be-converged.com and that your 'username' is NOT your phone number. |
| 02:07.03 | gusto | well, we can try again |
| 02:07.21 | WIMPy | It says the number is the name. |
| 02:07.48 | WIMPy | So in what I wrote above user=phone#. |
| 02:07.49 | ChannelZ | so what is the "lhre-Kennung@...." in the screenshot? |
| 02:07.54 | ChannelZ | next to "username"? |
| 02:08.43 | gusto | [Mar 5 02:08:35] WARNING[1681]: chan_sip.c:20032 handle_response_register: Forbidden - wrong password on authentication for REGISTER for 'DcZETjN1TbyV3aJ5OgHh' to 'sipproxy.endesha.be-converged.com' |
| 02:09.08 | gusto | i had that setup in there, so i only had to uncomment it |
| 02:09.13 | gusto | that's why it came so fast |
| 02:09.15 | WIMPy | It also has user number for registration checked. |
| 02:09.37 | WIMPy | I don't think the username there makes much sense. |
| 02:09.40 | ChannelZ | OK.. as I said I can't read german |
| 02:09.46 | WIMPy | That would make three domains. |
| 02:10.06 | ChannelZ | WIMPy will be more help to you than I can. I'm not understanding Google's bad translation on the rest of the page, like "As a registrar you always wear endesha.be converged.com-as-proxy always sipproxy.endesha.be converged.com one." |
| 02:10.15 | ChannelZ | (the second sentence below the screenshot) |
| 02:10.21 | gusto | do not care about the provider |
| 02:10.29 | gusto | they are idiots, they do not know how it works |
| 02:10.30 | ChannelZ | sounds like you don't want to be registering to sipproxy.blahblah |
| 02:10.45 | ChannelZ | (and it sounds like if they are idiots you should find another provider, but anyway...) |
| 02:10.52 | gusto | we have to figure it out with the server anyway |
| 02:11.14 | gusto | all ISP's have a diagnosis |
| 02:11.25 | ChannelZ | From what I can tell you should be registering to endesha.be-converged.com and not sipproxy.endesha.be-converged.com |
| 02:11.32 | gusto | they just say .. .well it works on fritzbox ... so it works ... |
| 02:11.37 | gusto | and that would be all |
| 02:11.59 | WIMPy | Have you tried to register as I wrote above? |
| 02:12.00 | gusto | endesha.be-converged.com does not resolve |
| 02:12.11 | WIMPy | That's ok. |
| 02:12.35 | ChannelZ | HMm. Well good luck with this busted-ass thing, I'm going to go watch some zombies. |
| 02:16.01 | *** join/#asterisk RickCogley (~anonymous@EM1-112-221-65.pool.e-mobile.ne.jp) |
| 02:17.35 | gusto | [Mar 5 02:17:17] NOTICE[1681]: chan_sip.c:20027 handle_response_register: Failed to authenticate on REGISTER to '4991131042466@sipproxy.endesha.be-converged.com' (Tries 3) |
| 02:17.56 | gusto | with the telephone number it comes quite far |
| 02:18.15 | gusto | and capitulates on MD5 auth |
| 02:18.41 | WIMPy | So dou you have an additional userNAME? |
| 02:19.08 | *** join/#asterisk koffel (~whome@173-167-212-105-ip-static.hfc.comcastbusiness.net) |
| 02:19.31 | koffel | whatz a good voip provider ? |
| 02:19.37 | gusto | well there is a user name but i can not log in with it, so i am either not using it or using it after/ |
| 02:19.52 | gusto | however, that name does not make any difference |
| 02:20.14 | gusto | that telephone number is the best |
| 02:20.53 | WIMPy | Maybe you need the name for the registrar and the number for the proxy or vice versa? |
| 02:21.35 | WIMPy | I have no idea what would happen if you put a username in to a FB and tick the use number box. |
| 02:22.03 | gusto | well |
| 02:22.17 | gusto | what is the proxy and what is the registar? |
| 02:22.22 | gusto | i do not see any proxy here |
| 02:23.08 | WIMPy | sipproxy... |
| 02:23.08 | gusto | that sipproxy resolves to voip servers under the domain endesha.be-converged.com |
| 02:23.18 | gusto | so ... where is the sipproxy of it? |
| 02:23.45 | WIMPy | "Where"? |
| 02:23.56 | gusto | _sip._udp.sipproxy.endesha.be-converged.com. 6056 IN SRV 1 1 5060 pcscf-brln-de01.endesha.be-converged.com. |
| 02:24.07 | gusto | that is SIP |
| 02:24.09 | WIMPy | Back to step one: Have you tried to register the way I wrote in my first line? |
| 02:24.28 | WIMPy | Why do you care what it resolves to? |
| 02:24.33 | gusto | you have seen the result |
| 02:25.19 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 02:25.32 | WIMPy | Ok, so then I'd try to replace either occurance of the number byt the name. |
| 02:27.58 | p3nguin | He's too hung up on the SRV records to care about actually solving the problem. |
| 02:30.00 | gusto | and what is the problem? |
| 02:34.03 | gusto | Authorization: Digest username="DcZETjN1TbyV3aJ5OgHh", realm="ims.telefonica.de", algorithm=MD5, uri="sip:sip:HSS@ims.telefonica.de", nonce="6b19229367b17592284ef7e1169d2147", response="63043b66b43592423ddcbffb24bcd027", qop=auth, cnonce="6649c2df", nc=00000001 |
| 02:34.28 | gusto | so putting username in there also did not change anything |
| 02:34.43 | gusto | so username as authuser |
| 02:43.29 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 02:43.45 | Yourname` | Would anyone happen to know what could be the reason there's no audio anywhere after reboot of server? |
| 02:44.00 | Yourname` | SIP to SIP extension calls, and no one can hear each other. |
| 02:44.05 | Yourname` | DAHDI calls, same thing. |
| 02:44.12 | Yourname` | All that was done is the server got rebooted. |
| 03:00.36 | *** join/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:08.33 | *** join/#asterisk Scar-G (~enforcer@182.178.23.129) |
| 03:12.28 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:34.06 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 03:39.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 04:15.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 04:30.23 | ChannelZ | Yourname`: firewall that was in one state and got put into another when rebooted? |
| 04:32.20 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
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| 05:29.33 | *** join/#asterisk tyman_ (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 05:53.03 | *** join/#asterisk tehrabbitt_ (47ac59bf@gateway/web/freenode/ip.71.172.89.191) |
| 05:53.14 | tehrabbitt_ | hey, is there anyone here that would be able to send me a test fax? |
| 06:00.06 | [TK]D-Fender | tehrabbitt_: http://www.google.ca/#hl=en&gs_nf=1&cp=8&gs_id=t&xhr=t&q=faxback+service&pf=p&output=search&sclient=psy-ab&pbx=1&oq=faxback+&aq=0&aqi=g1g-s1g2&aql=&gs_sm=&gs_upl=&gs_l=&bav=on.2,or.r_gc.r_pw.r_cp.r_qf.,cf.osb&fp=5a1555c5fd3b3cce&biw=1920&bih=1080 |
| 06:00.09 | [TK]D-Fender | theFIRST LINK |
| 06:00.22 | [TK]D-Fender | tehrabbitt_: FIRST LINK |
| 06:01.05 | tehrabbitt_ | [TK]D-Fender: yeah, I used up my 2 "free tests" earlier today realizing I did not have the license installed from digium, and now that it's installed, i have no way to test lol |
| 06:01.32 | [TK]D-Fender | tehrabbitt_: Find desk. Apply head repeatedly |
| 06:01.55 | tehrabbitt_ | [TK]D-Fender: pretty much |
| 06:03.06 | tehrabbitt_ | I was being stupid this morning being all "why is it not accepting the fax / saying that the module cannot be found" then I remembered, I upgraded from 1.6 to 10.0 a while back, so I figured, "meh i'll just upgrade to 10.1 and install fax module again" which I did, and if I dial the #, I get lovely fax tones, but idk how well it will receive |
| 06:05.40 | tehrabbitt_ | [TK]D-Fender: do you know if it's possible to save the incoming faxes as PNG or JPG rather than TIFF? |
| 06:06.05 | [TK]D-Fender | Convert after the fact |
| 06:06.12 | [TK]D-Fender | the app does what it does. |
| 06:07.03 | tehrabbitt_ | ah okay |
| 06:11.53 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xhizfwhvjjzqvgsu) |
| 06:19.52 | Yourname` | ChannelZ: That helped, thanks!! |
| 06:22.27 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:48.23 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
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| 06:55.18 | ChannelZ | Yay! |
| 07:00.07 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:00.25 | Henchman21 | i somehow broke asterisk |
| 07:00.36 | Henchman21 | installed the dahdi stuff |
| 07:00.47 | Henchman21 | now nothing works |
| 07:03.55 | ChannelZ | Cool. |
| 07:04.16 | ChannelZ | Define "nothing works" |
| 07:05.53 | Henchman21 | calls work sounds or any feedback dont |
| 07:07.24 | Henchman21 | its cool though i /etc/init.d/dahdi stop (removed all the dahdi modules ) |
| 07:07.28 | Henchman21 | and its working again |
| 07:07.33 | ChannelZ | on what kind of channels? You didn't have DAHDI at all before |
| 07:07.44 | Henchman21 | nah |
| 07:07.53 | Henchman21 | im trying to install ztdummy |
| 07:08.03 | Henchman21 | but its like gone now or something |
| 07:08.15 | ChannelZ | basically yes |
| 07:08.25 | Henchman21 | things have changed it seems |
| 07:08.43 | ChannelZ | it's dahdi_dummy and it's somewhat internalized |
| 07:14.44 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 07:30.35 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:31.48 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 07:33.09 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:33.11 | schmidts | good morning |
| 07:34.38 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 07:35.19 | ChannelZ | whips off his pants |
| 07:38.33 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 07:41.47 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 07:48.17 | wdoekes2 | good morning |
| 07:51.08 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-142-199.w90-58.abo.wanadoo.fr) |
| 07:53.38 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:53.46 | v0lZy | hi |
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| 08:06.08 | atan | Inside the dialplan how do you access the SIP device ID? Like say you register the phone using 1234. I want to get that 1234 inside my dialplan. I did look here http://www.voip-info.org/wiki/view/Asterisk+variables but wasn't able to locate it. |
| 08:06.29 | atan | If it's of any help it seems the 1234 I am after is stored within square brackets in sip.conf. |
| 08:09.41 | LiuYan | atan: maybe you want to see 'core show function CALLERID' |
| 08:09.48 | jacc0 | ${CALLERID(number)} |
| 08:09.55 | *** join/#asterisk freeman_u (~freeman@gw.quart-soft.net) |
| 08:09.57 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:10.05 | jacc0 | good morning all :) |
| 08:11.00 | atan | Hmm, well see more than one device is sending the same caller ID num. I think I will end up with ${CHANNEL:0:9} and looking for like SIP/1234- to match |
| 08:11.14 | atan | I just figured there must be some other shorthand way of doing it lol :-) |
| 08:11.18 | kaldemar | atan: use the CHANNEL function. |
| 08:11.26 | atan | Thanks for the idea there though jacc0 :-) |
| 08:11.46 | kaldemar | CHANNEL(peername) is what you want. |
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| 08:12.33 | atan | Oh snap! ^5 kaldemar rock on :-) works like a charm |
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| 08:25.24 | atan | Well color me confused. I played around with MixMonitor() in the outbound section of my dialplan and now my outbound calls do not connect the audio. I suppose I skipped right over some important part somewhere then? :-) |
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| 08:25.46 | atan | I did restore my extensions.conf back to exactly how it was before mucking around with it and the audio still does not exist. Meh. |
| 08:39.10 | *** part/#asterisk cstachris (~chrismylo@42.241.120.109) |
| 08:40.16 | atan | Interesting. Issue seems resolved when I pipe calls over another provider... perhaps something's up @ flowroute right now. Meh. |
| 08:41.08 | kaldemar | you'll see what's going on by enabling sip debug for a call. |
| 08:41.51 | ChannelZ | NAT issues? |
| 08:42.43 | atan | Oh probably :P hah. Asterisk box is sitting public, so are the other servers I'm connecting to so I assumed not =\ have always worked fine |
| 08:42.53 | atan | No big deal though I can use the other provider :-) |
| 08:43.08 | atan | Now just to figure out how to get these MixMonitor files into something smaller :D |
| 08:44.10 | ChannelZ | Hammersz |
| 08:44.14 | atan | Right now it's looking like almost 1MB/min which could work if it had to I don't need top quality audio surely there's a way to adjust :D |
| 08:44.57 | ChannelZ | doesn't it work by file extension? |
| 08:45.22 | atan | Oh perhaps. Let me rename it to .gsm and see what happens. |
| 08:45.26 | ChannelZ | IE you can give Foo.gsm to the filename |
| 08:46.43 | atan | Is there a diff between GSM & gsm? |
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| 08:47.29 | ChannelZ | well GSM is what it's called but gsm is the extension.... ? |
| 08:49.40 | atan | When I use uppercase it doesn't even save the file. Meh. Now, VLC doesn't seem to want to play the .gsm file do you happen to have any tips on how one could listen to the recording? |
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| 08:49.55 | ChannelZ | like I said.. the extension is .gsm |
| 08:49.57 | ChannelZ | not .GSM |
| 08:50.31 | ChannelZ | What platform are you trying to listen on? |
| 08:50.35 | atan | Windows |
| 08:51.30 | ChannelZ | you might use .WAV then which is like gsm in a WAV wrapper that I'm pretty sure Windows Media Player can play |
| 08:53.31 | atan | Hmm. How did I skip over that. I must have put .wav which made quite a large file. However, .WAV like you suggest does leave me with a reasonable sized file. |
| 08:53.32 | atan | Thanks! |
| 08:54.01 | ChannelZ | yup |
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| 09:00.36 | iulhk | using asterisk 10.1.2 , machine specifications are (model name Intel(R) Xeon(R) CPU E31270 @ 3.40GHz), (cache size : 8192 KB), (MemTotal:4138904 kB), making sip to sip video call by using (gsm and h263 codecs), there is no any transcoding, just need to know any formula, or need to know that how many simultaneous calls this server can handle ? |
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| 09:02.50 | kaldemar | iulhk: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
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| 11:58.37 | jacc0 | I'm trying to make sound files for an IVR; is there a window tool that can make the right format? |
| 11:58.44 | jacc0 | *Windows |
| 11:59.16 | WIMPy | audacity |
| 11:59.36 | jacc0 | okay, then I have the right tool |
| 11:59.56 | jacc0 | does asterisk 10 support more wav formats? |
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| 12:06.49 | WIMPy | Probably. But I haven't looked in to that. |
| 12:07.03 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats |
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| 12:08.58 | jacc0 | " file format for all sampling rates" |
| 12:09.06 | jacc0 | that sounds good |
| 12:09.29 | kaldemar | guess wav is still only 8000 Hz, 16-bit, and mono. |
| 12:09.49 | jacc0 | O |
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| 12:11.17 | jacc0 | I want to allaw users to upload there .wav files; I guess I will have to convert them |
| 12:11.46 | jacc0 | because almost no windows program can save .wav in 8000Hz |
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| 12:51.24 | dxd828 | hey guys, little problem. My asterisk server was hacked, or an extension was not sure yet. But I keep seeing this in the logs "Received incoming SIP connection from unknown peer" then a call is being made out. How is this possible when i have unregistered calls disabled. Only one extention was making out bounds calls 100 which has a massive password on it |
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| 12:52.56 | jondot | hi guys. anyone know if a TTS engine can also output when it starts to say each word? |
| 12:55.04 | kaldemar | dxd828: what do you mean by "unregistered calls disabled"? |
| 12:55.44 | dxd828 | kaldemar, anonymous inbound sip calls |
| 12:59.08 | kaldemar | dxd828: what did you do to disable them? |
| 13:00.31 | dxd828 | Im using the freepbx front end which has an option |
| 13:01.12 | kaldemar | you better continue in #freepbx then. |
| 13:02.11 | dxd828 | kaldemar, im also in there but no one is about :/ |
| 13:05.59 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:07.21 | *** join/#asterisk dom| (~domi@mail.tas.de) |
| 13:09.20 | dom| | Asterisk 1.8 ignores the values of callgroup and pickupgroup in the sip.conf. is there any other way to limit the users that can pickup a call? |
| 13:12.31 | dom| | one way is to use several subscibe-contexts, but then our telephone exchange is not able to see weather an extension is free or not |
| 13:13.25 | dom| | and with different subscribe-context, its not possible to let someone pickup calls from two or more groups |
| 13:18.26 | kaldemar | dom|: how did you verify that asterisk 1.8 would ignore your settings in sip.conf? |
| 13:19.17 | [TK]D-Fender | "one way is to use several subscibe-contexts, but then our telephone exchange is not able to see weather an extension is free or not" <- what's what INCLUDE's are for |
| 13:24.07 | dom| | kaldemar, i have three extensions in my testing environment. two of them with the same call- and pickupgroup, one with others. if one of the first two extensions calls the other, the thirt can pickup the call |
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| 13:25.27 | dom| | [TK]D-Fender, mhh this should work, right.... |
| 13:25.35 | kaldemar | i'd be interested to see that in action. as in "sip show peer <peer>" for all three and a picked up call. |
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| 13:34.28 | dom| | kaldemar, may i send you the links private? |
| 13:34.40 | kaldemar | dom|: if you insist |
| 13:36.16 | p3nguin | OOOoooooooooohhhh, secrets. |
| 13:38.07 | kaldemar | dom|: and you picked up by dialing *8 or the exten that is defined in features.conf as pickupexten? |
| 13:38.38 | dom| | i picked up with the flashing extension-button on a snom phone |
| 13:39.33 | [TK]D-Fender | dom|, that isn't a "pickup" |
| 13:39.43 | [TK]D-Fender | dom|, that is a raw dial. |
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| 13:44.51 | dom| | [Mar 5 14:42:23] NOTICE[1696]: app_directed_pickup.c:186 pickup_by_exten: SIP/4921-000003af pickup by SIP/317-000003b0 |
| 13:44.51 | dom| | [Mar 5 14:42:23] -- SIP/317-000003b0 answered SIP/4901-000003ae |
| 13:44.58 | dom| | [TK]D-Fender, looks like a pickup |
| 13:45.24 | dom| | the invite is with the calling extension in the to-field, not the extension the button is monitoring |
| 13:45.28 | [TK]D-Fender | dom|, Show us all the calls & config |
| 13:46.08 | dom| | sip.conf -> http://pastebin.com/kMPY7iEs |
| 13:46.09 | dom| | sip show peer 317 -> http://pastebin.com/afvfLMsV |
| 13:46.09 | dom| | sip show peer 4901 -> http://pastebin.com/cD3AZq21 |
| 13:46.09 | dom| | sip show peer 4921 -> http://pastebin.com/zJ579ZvX |
| 13:46.09 | dom| | call from 4901 to 4921, picked up by 317 -> http://pastebin.com/xhQx36Q4 |
| 13:46.50 | dom| | sip trace, taken on 317 -> http://pastebin.com/CgFhBELC |
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| 13:47.43 | dom| | if call- and pickupgroup would work, 317 would never get the pickup-info |
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| 13:53.51 | kaldemar | dom|: you should be using the features-provided pickup method for the callgroup and pickupgroup to take effect. |
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| 13:55.18 | kaldemar | dom|: instead of dialing 490, make the button dial *8. |
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| 13:56.14 | dom| | indeed... if i dial *8 the settings take effect... in previous versions (bristuffed) it worked with the extension... |
| 13:56.24 | kaldemar | dom|: app Pickup does not care about callgroup or pickupgroup settings in sip.conf. |
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| 13:58.53 | dom| | mhh ok ... if i set the key to <exten>|*8 then the phone dials the (non existing) extension *8<exten> |
| 13:59.17 | kaldemar | the app from bristuff works different from the native one. |
| 13:59.36 | leifmadsen | dom|: alternatively change features.conf to change the pickup extension to match what you're dialing from the phone |
| 13:59.55 | leifmadsen | results are the same |
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| 14:20.49 | p3nguin | In a case where a call is in a BackGround() or a WaitExten(), and there is an include for another context containing the extension which the caller is going to enter, is there a way to manipulate the ${EXTEN}? I can't see any way to do it directly; I can only think of adding the extensions to the current context and altering it there before sending it off to another context. |
| 14:21.46 | WIMPy | [l|e]switch => loopback/... |
| 14:22.32 | p3nguin | I don't understand. |
| 14:23.31 | WIMPy | With a loopback switch you can try modified versions of ${EXTEN} on the fly. |
| 14:25.00 | p3nguin | I'll see if I can figure it out. I have never used anything similar to your example above. |
| 14:28.31 | p3nguin | It's not exactly easy to locate information about "switch" online. |
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| 14:35.58 | WIMPy | No, but if you want to "match-as-you-go" it is absolutely essential. |
| 14:42.12 | p3nguin | It's not in Asterisk-Admin-Guide-1.8, either. |
| 14:45.02 | p3nguin | Wow, WTF? The people at 8887878722 need a new Asterisk admin. |
| 14:45.20 | p3nguin | Waiting in queue, there's a periodic announcement every 1.0 second. |
| 14:45.26 | leifmadsen | nice |
| 14:45.43 | p3nguin | .9 seconds of default moh in between. |
| 14:47.46 | p3nguin | It keeps switching back and forth between Allison saying the call is Nth in line, and their in-house prompt person saying how much she appreciates your patience. |
| 14:48.23 | leifmadsen | p3nguin: they REALLY appreciate it and need you to know just how much |
| 14:48.51 | p3nguin | I'd shoot myself if I had to listen to this more than once. |
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| 14:54.23 | phr3ak | hi, how could i dial an external sip uri from asterisk cli? |
| 14:55.26 | phr3ak | ok, i found originate |
| 14:55.31 | p3nguin | channel originate Local/your-extension@your-phone-context application Dial SIP/other-person@hisdomain.com |
| 14:55.37 | phr3ak | thank you |
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| 14:56.31 | p3nguin | or, adversely, channel originate SIP/other-person@hisdomain.com extension your-extension@your-phone-context |
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| 15:00.43 | p3nguin | And there's only a single reference to switch statements on http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html, which isn't really helpful to me. |
| 15:01.22 | Kobaz | is there anything special i need to do, to get polycom phones provisioning over https |
| 15:01.36 | Kobaz | the phone connects to the server but doesn't access any files |
| 15:01.44 | Kobaz | i think it's being put off by the self-signed cert |
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| 15:13.38 | kaldemar | p3nguin: did you look at the sample extensions.conf? |
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| 15:15.14 | DelphiWorld | hi ;) |
| 15:15.19 | DelphiWorld | anyone from Oman or UAE? |
| 15:18.06 | p3nguin | I just checked it, and I do see some examples of lswitch and eswitch! I was looking everywhere that I figured might have it and never gave extensions.conf a thought. |
| 15:20.25 | p3nguin | Is the lswitch statement using Loopback/ going to operate just like an include? |
| 15:21.06 | WIMPy | It can do what a include can do and more. |
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| 15:21.25 | WIMPy | You can use patterns and you can modify the extension during lookup. |
| 15:22.13 | p3nguin | Since I can't find a full explanation of switch statements, it's kind of hard for me to guess how it will behave. |
| 15:23.27 | WIMPy | switch => Loopback/extension@context/pattern |
| 15:23.37 | WIMPy | pattern is just like for extensions |
| 15:23.51 | WIMPy | and extension can be anything including ${EXTEN} |
| 15:24.54 | WIMPy | Like e.g. Loopback/${EXTEN:1}@context/_1! to cut of a leading 1. |
| 15:25.04 | p3nguin | So I could use 'lswitch => Loopback/9${EXTEN}@internal/_NXX' to prepend a 9 on the front of dialed number 762? 762 would normally match the include for internal, otherwise. |
| 15:25.32 | WIMPy | yes |
| 15:25.44 | WIMPy | And switches have a higher priority than includes. |
| 15:26.03 | p3nguin | I don't have to remove the include, then, I suppose. |
| 15:26.51 | bbourdage | I am trying to hire a full time developer for some Asterisk apps, and also a full time asterisk support person. I have put an add in the asterisk Forum, but that does not seem to get a lot of activity. Does anyone have any reccomendations for locations to advertise ?. Monster, career builder, typically have not worked for me either. I currently also have adds in craigs list(Which usuallly works the best) I am also loo |
| 15:27.35 | WIMPy | bbourdage: The asterisk-biz mailing list is said to be the right place. |
| 15:27.44 | p3nguin | Does your dev person and support person need to be one, or are you looking for two people? |
| 15:28.08 | _Corey_ | bbourdage: You might try the dCAP group on LinkedIn |
| 15:29.25 | dom| | kaldemar, after some tests... pickup with *8 works as espected (with limitations from call- and pickupgroup. to do this i have to set "<exten>|*8" for the extension key. fine. but if the user will set only <exten> for the key, he would be able to pickup any call again. |
| 15:30.02 | bbourdage | Thank you. |
| 15:30.04 | dom| | the only way i found to disable this, is to use different subscribecontexts or to disable notifycid |
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| 15:31.38 | dom| | with the first way the blf will not work context-overlapping and with the second way no user will see who is calling the monitored extension |
| 15:35.47 | kaldemar | dom|: you can remove the extension that calls the Pickup app from your dialplan. |
| 15:38.49 | Sparky-UK | Can anyone tell me if the agents in a queue have to be extensions on the same system, or can they be externally located endpoints? |
| 15:39.48 | kaldemar | dom|: or do a check for the callgroup and pickupgroup in your dialplan maybe. ${SIPPEER(${CHANNEL(peername)},pickupgroup)} would for example get you the pickupgroup of the calling peer. |
| 15:39.54 | leifmadsen | Sparky-UK: if it's SIP, it's just SIP |
| 15:40.07 | leifmadsen | Sparky-UK: so yes, they can be located anywhere |
| 15:40.39 | p3nguin | If you're using agents, extensions are not evaluated, are they? |
| 15:40.40 | kaldemar | dom|: anyway, for anyone to be able to help you precisely, you'd have to show how you have implemented pickups in your dialplan. |
| 15:40.45 | p3nguin | chan_agent |
| 15:40.46 | [TK]D-Fender | Sparky-UK, You can add whatever device youw ant as a memeber |
| 15:41.18 | Sparky-UK | Right, I am getting a bit fustrated with FreeSwitch's FIFO and CallCenter Implementation for Multi-tennant and I already have a version in Asterisk that works, so I think I will just use that thanks |
| 15:41.27 | dom| | kaldemar, the problem is, that pressing the extension-key (set to "<ext>") will not start any dialplan app, it's some magic |
| 15:41.56 | WIMPy | black magic? |
| 15:42.03 | dom| | kaldemar, so there is nothing related to pickups in my dialplan |
| 15:42.43 | Sparky-UK | [TK]D-Fender, leifmadsen, thanks |
| 15:43.13 | dom| | kaldemar, as you can see in http://pastebin.com/xhQx36Q4 the pickup is executed without any context/extension start |
| 15:44.11 | dom| | the asterisk get an invite with a reference to the dialog-info and will do the pickup, without taking care of call-/pickupgroups :/ |
| 15:44.37 | kaldemar | dom|: ok. it's a notify with pickup information and a dial directly to the peer. |
| 15:46.11 | akrohn | easy question. in my attendant menu, i have 'press 5 for billing'. I also have extensions that start with 5. can i make the input not go to 5 immediately? |
| 15:46.51 | p3nguin | If you have other extensions starting with 5, it should never go to 5 immediately. |
| 15:47.07 | _Corey_ | akrohn: Not if you want to have the 5+ extensions dialable.. Asterisk won't read minds ;) |
| 15:47.27 | p3nguin | If you have 5 and 500 in the context, and the caller presses 5 and waits 1 second before pressing 00, it should still end up at 500. |
| 15:47.38 | p3nguin | At least this is how it works for me! |
| 15:47.45 | akrohn | excellent. thanks p3nguin and _Corey_ |
| 15:47.48 | p3nguin | I used to want to make it dial 5 instantly. |
| 15:47.58 | p3nguin | because it would always wait for more digits. |
| 15:48.24 | _Corey_ | akrohn: I may have misread your question actually... p3nguin explained it more thoroughly |
| 15:49.22 | p3nguin | For me, the overlap always caused delay. |
| 15:51.00 | dom| | kaldemar, do you see any way to disable this without disabling the dialog-info? |
| 15:53.15 | NotHere | when I monitor a call (*1 during the call), the produced wav is really weird. First it plays what I say only and then it plays what the other person says only. That means a 15 sec conversation results in a 30sec wav non-duplex |
| 15:53.41 | p3nguin | Do you have a lot of load on your system? |
| 15:53.47 | NotHere | no load at all |
| 15:54.10 | NotHere | does the wav need some sort of postprocessing? |
| 15:54.53 | p3nguin | If you are using Monitor(), there's an option to mix when the recording stops. |
| 15:55.03 | p3nguin | If you are using MixMonitor(), it is dones on-the-fly. |
| 15:55.11 | p3nguin | s/dones/done/ |
| 15:55.25 | WIMPy | is wondering if the priorty lable not being found is a result of my tidying up my dialplan or if there is a regression on the switch statement. |
| 15:55.27 | NotHere | i'm using whatever FreePBX uses... |
| 15:55.33 | p3nguin | ~freepbx |
| 15:55.34 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:56.07 | NotHere | so this is probably a FreePBX bug... |
| 15:56.12 | NotHere | thanks |
| 15:56.14 | p3nguin | I doubt it. |
| 15:56.41 | p3nguin | The monitor/recording apps are part of Asterisk. |
| 15:57.13 | p3nguin | But I don't have the foggiest idea how FreePBX has set up anything for use. |
| 15:57.18 | NotHere | but you are saying that what I see is the expected behaviour of Monitor(), do you? |
| 15:57.38 | NotHere | and MixMonitor() should be used instead... |
| 15:57.42 | kaldemar | producing configurations that cause audio to not be mixed is a freepbx issue entirely. |
| 15:57.48 | p3nguin | If you are using the option to mix, wait a few seconds after the call ends so it can mix, then listen to the recording. |
| 15:58.23 | p3nguin | Or use MixMonitor instead of Monitor. |
| 15:58.30 | *** join/#asterisk wonderworld (~ww@dsdf-4db54587.pool.mediaWays.net) |
| 15:59.14 | NotHere | Monitor() just does the mixing after the call? |
| 15:59.19 | kaldemar | dom|: i don't know, sorry. probably not if the phone is set to work in one way. you could probably work your way around it all by implementing the pickups yourself. |
| 15:59.22 | phr3ak | do i need configure all domains one by one (like ekiga.net) if i want access other domains from sip clients? |
| 16:00.01 | p3nguin | If you are using Monitor, you have to use the correct option to mix or you end up with two recordings (one for in and one for out). |
| 16:00.13 | kaldemar | phr3ak: if digits is all the phones can dial, then yes. |
| 16:00.31 | kaldemar | phr3ak: or clients as you said, not phones. |
| 16:00.33 | p3nguin | If you are using the mixing option, the two channels will be mixed after the call has ended. |
| 16:01.22 | jzaw | <PROTECTED> |
| 16:01.22 | jzaw | <PROTECTED> |
| 16:01.34 | jzaw | gives mono |
| 16:01.53 | p3nguin | We still don't know if he's using Monitor or MixMonitor... because he's using FreePBX. |
| 16:02.01 | jzaw | fair enough |
| 16:02.45 | jzaw | i think with monitor i had to use sox to mix the left and right recordings |
| 16:02.45 | *** join/#asterisk gain_ (~gain@mail.ufficyo.com) |
| 16:03.05 | p3nguin | It should do that automatically when you use the mixing option. |
| 16:03.54 | phr3ak | kaldemar: so i cannot call my @ekiga.net account if i registered in asterisk |
| 16:04.27 | p3nguin | Or set MONITOR_EXEC to something other than sox, and it'll use that instead of sox. |
| 16:06.35 | kaldemar | phr3ak: sure you can. |
| 16:06.48 | kaldemar | phr3ak: what exactly do you want to do? |
| 16:08.58 | leifmadsen | so.... what is.... you.... DO.... here? |
| 16:11.11 | kaldemar | leifmadsen: did your load average skyrocket momentarily? |
| 16:11.27 | leifmadsen | kaldemar: eh? |
| 16:11.28 | phr3ak | i have a client1 registered in ekiga.net. i can call it from client2 if i'm not registered in my asterisk. because asterisk think this is a local user |
| 16:13.25 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-375536.home.otenet.gr) |
| 16:13.52 | kaldemar | phr3ak: register asterisk in ekiga.net, handle incoming calls from ekiga.net and make an extension that allows you do dial ekiga.net clients. |
| 16:14.25 | kaldemar | leifmadsen: referred to the... dotted... writing... |
| 16:15.16 | jaytee | leif, that sounded like John McGinley in the movie Office Space |
| 16:15.31 | Henchman21 | wtf!!! this stupid spa3000 |
| 16:15.41 | leifmadsen | jaytee: ding ding ding, you win! |
| 16:16.01 | phr3ak | and what can i do if i'd like to call other user in other domain? |
| 16:16.58 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 16:17.32 | kaldemar | phr3ak: make an extension that allows you to. |
| 16:18.22 | *** join/#asterisk Rewris (~Rewris@201008081252.user.veloxzone.com.br) |
| 16:19.28 | phr3ak | so i cannot call arbitary peer without asterisk reconfiguration? |
| 16:19.56 | p3nguin | If you have an extension that provides you that capability, you can do it. |
| 16:21.18 | kaldemar | phr3ak: back to square one. "< kaldemar> phr3ak: if digits is all the phones can dial, then yes." |
| 16:22.12 | kaldemar | that of course means calling through asterisk only. |
| 16:24.17 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 16:25.09 | akrohn | http://arstechnica.com/business/news/2012/03/the-pwn-plug-is-a-little-white-box-that-can-hack-your-network.ars |
| 16:25.16 | akrohn | sorry wrong window! |
| 16:25.39 | akrohn | pretty neat tool though =P |
| 16:45.39 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 16:46.36 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
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| 16:47.44 | DelphiWorld | hey leifmadsen |
| 16:47.53 | leifmadsen | hey DelphiWorld |
| 16:49.33 | *** join/#asterisk flan (~flan@23.17.142.228) |
| 16:53.19 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v021-062.mobile.uci.edu) |
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| 17:01.44 | *** join/#asterisk Rewris (~Rewris@201008081252.user.veloxzone.com.br) |
| 17:03.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:03.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:05.50 | phr3ak | why doesn't created this extension if i write this in sip.conf |
| 17:05.51 | phr3ak | regexten=1001 |
| 17:05.51 | phr3ak | context=default |
| 17:06.28 | p3nguin | In an SPA-942, if I press Submit All Changes while there is a call active, will the call be affected? |
| 17:06.49 | p3nguin | phr3ak: dialplan show 1001@default |
| 17:07.42 | p3nguin | A terrible choice of context name, by the way. |
| 17:07.56 | Dovid | hello all |
| 17:08.46 | phr3ak | There is no existence of 1001@default extension |
| 17:08.57 | p3nguin | Did the device register? |
| 17:09.07 | phr3ak | yes |
| 17:09.34 | p3nguin | Did you remember to save and run "sip reload" before it registered? |
| 17:11.20 | phr3ak | ofcourse. i see the peer at sip show peers |
| 17:11.28 | kaldemar | phr3ak: where did you write it? do you have a regcontext configured? |
| 17:11.55 | phr3ak | regcontext not counfigured |
| 17:12.14 | [TK]D-Fender | Maybe you should change that |
| 17:15.42 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 17:18.40 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 17:21.22 | *** join/#asterisk caplink (~w0rm@adsl-65-3-67-6.mia.bellsouth.net) |
| 17:22.58 | Henchman21 | i have this spa3000, getting it to dialout the house line has been a headache and i've noticed if i attempt an outgoing call and it fails i'll call the line with my cellphone and it picks up dials the number it failed at me, and procedes to handoff the call to my lil asterisk IVR... weird i know wtf is wrong |
| 17:23.36 | *** join/#asterisk albertoandrade (~albertoan@187.112.133.55) |
| 17:26.11 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
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| 17:37.21 | phr3ak | kaldemar: it's helped but now i got SIP/2.0 603 Declined |
| 17:38.47 | [TK]D-Fender | phr3ak, because * didn't answer it |
| 17:39.01 | phr3ak | i'm calling myself |
| 17:39.13 | [TK]D-Fender | Yes, and you aren't looking at what your dialplan is doing |
| 17:53.33 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 17:55.55 | *** join/#asterisk jackson5 (~John@adsl-75-24-203-14.dsl.scrm01.sbcglobal.net) |
| 17:56.07 | jackson5 | hello everyone |
| 17:56.35 | jackson5 | i have a few questions about a simple dialplan i'm developing |
| 17:56.45 | jackson5 | does anyone have a few minutes to help me out? |
| 17:57.12 | [TK]D-Fender | Ask your question and see who responds |
| 17:57.12 | jackson5 | it would be greatly appreciated :) |
| 17:57.18 | jackson5 | ^ ok |
| 17:58.06 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:58.32 | jackson5 | What I'm trying to do is make an outgoing call, when the call is answered play an audio file. Also I would like to update the dial status to answered, hangup, no answer, voicemail |
| 17:59.07 | [TK]D-Fender | jacyou don't update DIALSATATUS. it's read-only |
| 17:59.16 | [TK]D-Fender | jackson5, you don't update DIALSATATUS. it's read-only |
| 17:59.41 | [TK]D-Fender | jackson5, Next please clarify how you are "dialing out" what is the point of origin for this call? |
| 18:00.06 | jackson5 | through a call file |
| 18:00.18 | jackson5 | i mean, update the status in my mysql database |
| 18:01.05 | jackson5 | i'm using AGI(*.php) to update it to mysql |
| 18:01.14 | [TK]D-Fender | What you put into yor own DB is of course up to you |
| 18:01.52 | jackson5 | ok |
| 18:02.41 | jackson5 | let me show you my current dialplan |
| 18:03.05 | jackson5 | this is my call file |
| 18:03.07 | jackson5 | Channel: SIP/.../... |
| 18:03.07 | jackson5 | MaxRetries: 0 |
| 18:03.07 | jackson5 | WaitTime: 20 |
| 18:03.07 | jackson5 | Context: myContext |
| 18:03.07 | jackson5 | Extension: s |
| 18:03.08 | [TK]D-Fender | ~pb |
| 18:03.08 | 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. |
| 18:03.09 | [TK]D-Fender | ^^^^^^^^^ |
| 18:03.16 | [TK]D-Fender | PASTEBIN. Do not flood in here |
| 18:03.21 | p3nguin | Must be someone's first day on IRC. |
| 18:03.22 | jackson5 | sorry |
| 18:04.28 | jackson5 | http://pastecode.com/ty |
| 18:08.34 | jackson5 | sometimes, the dialplan will play audio to the callee's voicemail |
| 18:08.45 | [TK]D-Fender | Absolutely. |
| 18:09.41 | jackson5 | is there anyway to stop this from happening? |
| 18:10.07 | [TK]D-Fender | jackson5, "core show application dial", "core show application amd" |
| 18:10.10 | p3nguin | I have a design question. In a situation where the receptionist has two line keys (one is used for the DID, one is used for internal), would you create two voice mail boxes (one for the DID and one for herself) or send all calls destined for her phone into her own mail box? |
| 18:11.16 | jackson5 | standby [tk]d-fender |
| 18:11.35 | _Corey_ | p3nguin: I usually find it easier for the user to manage one mailbox... |
| 18:11.39 | [TK]D-Fender | jackson5, that wasn't a request for you to show me those instructions. It was for you to read them |
| 18:12.11 | *** part/#asterisk NotHere (~steve@geozix.static.otenet.gr) |
| 18:12.51 | jackson5 | ah, i understand |
| 18:13.26 | p3nguin | That's how I set it up, but then I was second guessing that, thinking maybe calls from outside needed to be in one box and calls from the inside needed to be in her personal box. |
| 18:13.54 | _Corey_ | p3nguin: No rule saying you have to play the same greeting for all scenarios |
| 18:14.14 | p3nguin | I'm skipping the greeting when the call came in from outside. |
| 18:14.16 | jackson5 | thanks [tk]d-fender |
| 18:14.28 | p3nguin | It's really about vm management more than anything else. |
| 18:15.54 | [TK]D-Fender | jackson5, First tip. Don't dial SIP directly <- You have no control over the flow at that point. Go read up on *'s othr channel types. ONE will give you the ability to do what you still need to do and give you more options at the same time... |
| 18:17.11 | jackson5 | <PROTECTED> |
| 18:17.23 | jackson5 | <PROTECTED> |
| 18:18.17 | [TK]D-Fender | jackson5, Keep in mind the pattern you're using... |
| 18:20.35 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 18:21.03 | jackson5 | Ok |
| 18:28.25 | woleium | hola peeps :-) |
| 18:28.57 | woleium | is there any way to disable the park announcement when a caller is blind transferred to park? |
| 18:29.43 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:30.36 | woleium | a normal transfer is fine (it announces to the internal user), but when you blind transfer it announces the slot to the external user (customer) |
| 18:30.43 | p3nguin | Why would anyone want to blind transfer to park? |
| 18:31.03 | woleium | it's quicker |
| 18:31.04 | p3nguin | Of course it does, because you aren't supposed to blind transfer to it. |
| 18:31.32 | woleium | when using FOP you can see what slot the call goes into without having to wait for the announcement |
| 18:31.38 | p3nguin | And if I had employees doing it, I'd yell at them until they stopped doing it. |
| 18:32.08 | woleium | it means that the receptionists can handle more calls. they love it |
| 18:32.45 | p3nguin | It takes approximately 3 seconds to transfer the correct way and hear the slot number. |
| 18:33.03 | p3nguin | The caller can listen to the phone ringing for 3 more seconds without a problem. |
| 18:33.19 | p3nguin | Or learn how to use queues. |
| 18:33.23 | woleium | I understand your point of view, but I have had a request from a customer to do it this way…. is it possible? |
| 18:33.40 | woleium | or is there an alternative? |
| 18:33.49 | p3nguin | The alternative is to train them to do it correctly. |
| 18:33.54 | woleium | hehe |
| 18:33.59 | woleium | indeed |
| 18:34.46 | woleium | I don;t see what's wrong with blind transferring to park if you can use another means (lash operator panel) to obtain the park slot |
| 18:35.01 | _Corey_ | woleium: You could program a Polycom EFK key to do Park w/one button that will properly do the transfer |
| 18:35.39 | woleium | but i do understand why this is not expected (or designed for) betavior |
| 18:37.01 | *** join/#asterisk doolittlework (~doolittle@41-134-22-14.dsl.mweb.co.za) |
| 18:37.47 | woleium | _Corey_: I guess… they do use polycoms. I'll look into it. ty :-) |
| 18:38.22 | doolittlework | can anyone shed some light on the timing issue when one uses mixmonitor instead of monitor application( i am using mixmonitor and the call quality is great, but when i listen to recording the far side sounds terible)?? |
| 18:39.03 | doolittlework | if i use monitor the sound is good on the recordings? |
| 18:39.46 | doolittlework | <PROTECTED> |
| 18:39.57 | _Corey_ | woleium: I believe the one I use does Transfer, 7, 0, 0, (pause 3s), Transfer |
| 18:40.09 | _Corey_ | or something very similar |
| 18:41.43 | [TK]D-Fender | woleium, ParkAndAnnounce |
| 18:45.01 | *** join/#asterisk WildPikachu (~nkukard@about/linux/staff/wildpikachu) |
| 18:45.03 | woleium | [TK]D-Fender: that looks good :-) http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce seems to have a solution |
| 18:45.12 | woleium | ty |
| 18:45.39 | WildPikachu | I think I'm doing something wrong, when I use Dial() and the option T , it seems that the remote side does not get RFC2833 DTMF, Asterisk 1.4.19 |
| 18:46.08 | p3nguin | Do you know what the T does? |
| 18:46.22 | WildPikachu | allows the calling party to transfer the call and intercepts DTMF |
| 18:46.28 | p3nguin | Do you try to have direct media with a re-invite? |
| 18:46.39 | WildPikachu | its NAT :) |
| 18:46.51 | p3nguin | I guess that's probably a "no." |
| 18:47.07 | doolittlework | lol |
| 18:47.30 | doolittlework | sorry but i had to grin |
| 18:48.22 | p3nguin | My first thought was that you had configured either the phone or the peer to use inband DTMF and using the T option would have been keeping asterisk in the media path, potentially causing an unwanted behavior. But that probably isn't the case after all. |
| 18:49.10 | WildPikachu | p3nguin, yep, total no on the reinvite, I have canreinvite=no to keep asterisk in the path for call recording, and because the internal phones don't have a path to the destination IP wise |
| 18:49.42 | p3nguin | You've set all peers involved to use dtmfmode=rfc2833? |
| 18:49.57 | *** join/#asterisk learath (47f6db23@gateway/web/freenode/ip.71.246.219.35) |
| 18:50.20 | doolittlework | what does asterisk use for timing did they do away with dahdi_dummy as timing source in version 1.8? |
| 18:50.43 | doolittlework | if you have no dahdi hardware that is |
| 18:50.52 | WildPikachu | p3nguin, affirmative ... and the thing is I see the RTP packet going out from asterisk to the voip provider, but its not ending up at the destination ... the second I remove T it does |
| 18:51.01 | p3nguin | doolittlework: I still use dahdi for timing. |
| 18:51.08 | WildPikachu | the only difference I see from the UDP packet is the reserved = 1 vs reserved = 0 |
| 18:51.35 | doolittlework | strange i dont see it in cat /proc/dahdi? |
| 18:51.45 | doolittlework | nor in dahdi show channles in cli |
| 18:52.01 | p3nguin | # ls -la /proc/dahdi/ |
| 18:52.01 | p3nguin | total 0 |
| 18:52.19 | p3nguin | Is your dahdi kernel module loaded? |
| 18:52.26 | WildPikachu | p3nguin, the other thing is, my Digium hardware card for BRI works 100% fine .... so I'm thinking either the voip provider is buggy, or its something to do with me keeping asterisk in the media path |
| 18:53.20 | p3nguin | doolittlework: Does "dahdi show channels" show the pseudo channel is present? |
| 18:53.53 | doolittlework | <PROTECTED> |
| 18:53.53 | doolittlework | ? |
| 18:54.24 | doolittlework | does that mean the timing is set p3nguin? |
| 18:54.35 | p3nguin | Does "module show like timing" show that you have res_timing_dahdi.so loaded? |
| 18:55.33 | doolittlework | res_timing_dahdi.so DAHDI Timing Interface 1 |
| 18:55.38 | p3nguin | I would say that your dahdi timing source is working correctly. |
| 18:55.50 | doolittlework | 1 modules loaded |
| 18:55.56 | doolittlework | is that ok |
| 18:55.57 | doolittlework | ? |
| 18:56.14 | p3nguin | It's the only timing source that you have available at this time. |
| 18:56.29 | p3nguin | I actually have dahdi, timerfd, and pthread all loaded and available. |
| 18:57.08 | doolittlework | can this be the cause of my mixmonitor application recording the far side like crap? |
| 18:57.23 | Qwell | WildPikachu: What channel driver are you using for that card? |
| 18:57.59 | p3nguin | doolittlework: I wouldn't necessarily rule it out just yet. Load a different timing module and see how it changes. |
| 18:58.05 | WildPikachu | Qwell, i'll have to check, but its working 100% ... my problem is sip only |
| 18:58.12 | WildPikachu | sip-asterisk-sip |
| 18:58.31 | doolittlework | p3nguin: what would you recommend? |
| 18:59.22 | p3nguin | I think timerfd is pretty popular. |
| 18:59.34 | doolittlework | ta let me try |
| 19:02.38 | WildPikachu | thanks for your time p3nguin , at this point I'm just going to assume the provider is buggy and remove T :) |
| 19:02.51 | doolittlework | p3nguin: how does one activate it i see res_timing_pthread.so is one of my modules? |
| 19:03.07 | p3nguin | That's pretty odd that adding option T would break DTMF. |
| 19:03.39 | p3nguin | If it is available, load it with "module load res_timing_timerfd.so". |
| 19:05.47 | doolittlework | p3nguin: Command 'module load res_timing_pthread.so ' failed. |
| 19:06.19 | p3nguin | Then don't load pthread. |
| 19:09.57 | doolittlework | p3nguin: sorry to bug you but where can i find timerfd? |
| 19:10.20 | Qwell | doolittlework: Have a kernel that supports it. |
| 19:11.26 | p3nguin | If you do not have res_timing_timerfd.so, you did not compile and install it with asterisk. |
| 19:11.46 | doolittlework | <PROTECTED> |
| 19:12.02 | Qwell | doolittlework: CentOS 5 kernels are ancient. |
| 19:12.06 | doolittlework | p3nguin is it there by make menuselect? |
| 19:12.36 | doolittlework | when i installed asterisk |
| 19:14.07 | doolittlework | cd /etc/asterisk |
| 19:14.14 | doolittlework | lol sorry wrong screen |
| 19:16.38 | *** join/#asterisk wonderworld (~ww@dsdf-4db5cc94.pool.mediaWays.net) |
| 19:17.29 | *** join/#asterisk Bullmoose (~Bullmoose@75-174-77-145.bois.qwest.net) |
| 19:18.02 | anonymouz666 | 2.6.18 is from 2006, I think |
| 19:22.22 | leifmadsen | that's what CentOS 5 uses |
| 19:22.39 | leifmadsen | 2.6.18-274.18.1 is the newest on a CentOS 5.7 system |
| 19:22.50 | Qwell | that's still not going to have timerfd |
| 19:22.55 | leifmadsen | agreed |
| 19:23.08 | leifmadsen | you need 2.6.19 I think or newer |
| 19:23.19 | p3nguin | I have it in 2.6.32. |
| 19:23.24 | Qwell | 27 |
| 19:23.32 | Qwell | wait no |
| 19:23.39 | Qwell | 25 |
| 19:23.48 | leifmadsen | Qwell: oh that late? thought it was earlier than that |
| 19:23.58 | leifmadsen | regardless, nothing beats res_timing_dahdi to stability |
| 19:26.49 | p3nguin | If the other sources are available, should they be unloaded and/or "noload"ed so they don't get used? |
| 19:30.09 | Qwell | no, they have priorities |
| 19:31.51 | *** part/#asterisk WildPikachu (~nkukard@about/linux/staff/wildpikachu) |
| 19:32.48 | p3nguin | Where are those configured? |
| 19:32.54 | Qwell | they aren't |
| 19:32.57 | Qwell | it's hardcoded |
| 19:33.03 | p3nguin | What is the order? |
| 19:34.50 | Qwell | one, then another, then the third |
| 19:34.54 | Qwell | (I have no idea) |
| 19:36.35 | p3nguin | If the hard-coded order is not desired, then unload and noload would be the easiest way, I guess. Altering the source code would be the other way. |
| 19:40.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 19:40.42 | *** topic/#asterisk by mjordan -> 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 |
| 19:41.04 | *** join/#asterisk Li7h (~dps@71-81-21-14.static.gwnt.ga.charter.com) |
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| 20:00.37 | anonymouz666 | 1.8.10.0 is out! |
| 20:00.46 | anonymouz666 | waiting for 1.8.11-rc1 ! |
| 20:02.11 | Qwell | anonymouz666: You're gonna be waiting a while |
| 20:02.30 | p3nguin | I'm stuck on 1.8.8.something for now. When I tried to go up to 1.8.9.1, I had an unacceptable problem with jabber which apparently was not reproducible by the person who tried to reproduce it. I'm going to try 1.8.10.0 today. |
| 20:02.31 | ajg | 1.8 is the LTS right? |
| 20:02.39 | Qwell | ajg: Yes |
| 20:02.44 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 20:03.07 | anonymouz666 | Qwell: any particular reason? |
| 20:03.13 | ajg | as I'm setting up our new system, it's amazing how much functionality was blocked by our old provider |
| 20:03.18 | Qwell | anonymouz666: unless things have changed, it's being skipped |
| 20:10.37 | pabelanger | o.0 |
| 20:14.05 | p3nguin | Ugh, there's no way to redirect a channel in 1.4? |
| 20:14.19 | anonymouz666 | you mean channelredirect? |
| 20:17.36 | p3nguin | I think I have a stuck channel and I was going to redirect it back to the phone that put it on hold, but I guess there is no way in 1.4 CLI. |
| 20:18.13 | [TK]D-Fender | AMI < |
| 20:18.28 | p3nguin | <-- not a programmer |
| 20:19.10 | p3nguin | I just killed the channel. If it wasn't supposed to be hung up, they'll call again. |
| 20:21.28 | p3nguin | In the case where the channel is on hold and there is hold music playing, rtptimeout and rtpholdtimeout will never be enforced, correct? |
| 20:22.00 | Qwell | p3nguin: I think that's incorrect, since it should be incoming packets, not outgoing. |
| 20:22.21 | p3nguin | Okay, then I should have waited 300 seconds to see if it died on its own. |
| 20:22.23 | Qwell | So if Asterisk receives no RTP packets in $rtpholdtimeout seconds, it'll die. |
| 20:22.26 | p3nguin | But I didn't. |
| 20:22.28 | Qwell | I could be wrong though. |
| 20:22.58 | p3nguin | I just don't want to be billed excessively for a dead channel which still has audio on it. |
| 20:23.36 | p3nguin | Next time I will wait 301 seconds before I consider freaking out. |
| 20:24.04 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:26.59 | [TK]D-Fender | p3nguin, You don't need to be a programmer. |
| 20:27.07 | [TK]D-Fender | type a few lines into telnet. The end. |
| 20:27.08 | Nugget | telnet is eeeeeeevil! |
| 20:27.25 | [TK]D-Fender | pushes some more of Nugget |
| 20:27.29 | [TK]D-Fender | 's buttons |
| 20:27.38 | p3nguin | Oh. I was under the impression I had to know some kind of programming language to interact via AMI. |
| 20:28.07 | [TK]D-Fender | p3nguin, well.. not to get flodded or maybe do some more meaningful stuff... sure, but for this need, no |
| 20:28.12 | [TK]D-Fender | no |
| 20:28.58 | p3nguin | Is there a newbies' resource for such administration tasks? |
| 20:32.16 | fprior | hi, I've received a Grandstream GXW4108 (8 fxo ports) gateway; do you used this gw ? what about it ? |
| 20:32.31 | Qwell | what about what about it? |
| 20:34.12 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 20:34.16 | [TK]D-Fender | p3nguin, it's in the book. |
| 20:34.34 | [TK]D-Fender | p3g3 lines to login. then the raw text for the AMI redircet |
| 20:34.37 | p3nguin | Okay. I will check there. |
| 20:35.19 | [TK]D-Fender | p3nguin, think of it as just opening the hood and smacking the engine :) |
| 20:36.00 | p3nguin | That's precisely what I want to do. I don't want to have to know how to connect the diagnostic machine to the car's computer or know how to read trouble codes. |
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| 21:05.51 | dynamite | err woops -> app_meetme.c:1228 build_conf: Unable to open DAHDI pseudo device |
| 21:05.53 | dynamite | any idea on that one? |
| 21:08.38 | Nivex | I see 1.8.10.0 is out. Any ETA on .deb packages in the repo? |
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| 21:21.31 | pabelanger | Nivex: should be able to roll them for tomorrow |
| 21:47.17 | rossand | when I "sip show registry" I see dramatically different refresh values for some of my sip providers. 45 vs. 7185. The one that's set to 45 keeps timing out. Which setting does this correspond to in sip.conf? |
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| 22:17.08 | leifmadsen | mjordan: ping |
| 22:17.35 | mjordan | pong |
| 22:18.11 | leifmadsen | mjordan: I'd like to make a suggestion about Asterisk releases. I think it would be a good policy to adopt the stance that the release should synchronize both source tarball and .rpm / .deb availability |
| 22:18.40 | mjordan | leifmadsen: agreed. We seem to be getting e-mails about that :-) |
| 22:18.57 | mjordan | I think we'll make that a priority for 1.8.11 |
| 22:19.00 | leifmadsen | mjordan: just noticing it a bit lately in here, and think that step should just be added to that wiki page near the bottom :) |
| 22:19.06 | leifmadsen | mjordan: sounds good, thanks! |
| 22:19.08 | mjordan | why, what wiki page? :-D |
| 22:19.16 | leifmadsen | mjordan: the secret one! :) |
| 22:19.37 | mjordan | leifmadsen: I think that will put it over step bajillion |
| 22:19.49 | leifmadsen | mjordan: I think it's closer to 18 or 19 :) |
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| 22:35.30 | Li7h | is there a way to disable the parking slot announcement when parking a call? |
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| 23:11.12 | kfife | Hey guys. I notice that 1.8.8 - 1.8.10 removes the group write permission on /etc/asterisk/. |
| 23:11.30 | kfife | This is unexpected, and different than 1.4 and 1.6, all versions |
| 23:12.30 | kfife | Should I report this to the bug tracker, or is this in fact expected behavior? |
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| 23:28.08 | *** mode/#asterisk [+o file] by ChanServ |
| 23:43.29 | kfife | Hey guys. I notice that 1.8.8 - 1.8.10 removes the group write permission on /etc/asterisk/. |
| 23:43.40 | kfife | This is unexpected, and different than 1.4 and 1.6, all versions |
| 23:43.46 | kfife | Should I report this to the bug tracker, or is this in fact expected behavior? |
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