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| 00:56.49 | boodu | I try to configure isdn trunk with no success. Call Entrances is ok but call out failed. I don't see any channels ISDN |
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| 00:59.51 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:01.20 | jsidhu | so I've got both of my boxes connected via IAX2 trunks. this works if both sides have static ips since they both register to each other. How can i handle a situation where one side has a dynamic ip? |
| 01:01.50 | chuckf | jsidhu: dyndns is one way |
| 01:03.10 | jsidhu | chuckf: there's no other way besides have the remote side register? since the dynamic side is already registering, cant asterisk use that same trunk? |
| 01:05.40 | *** join/#asterisk blaines (~blaines@75-171-71-26.phnx.qwest.net) |
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| 01:11.15 | jsidhu | .. |
| 01:13.51 | pabelanger-lap | jsidhu: have your dynamic IAX2 IP register with your static IAX2 IP |
| 01:14.42 | *** join/#asterisk coppice (~chatzilla@m121-203-201-64.smartone-vodafone.com) |
| 01:19.34 | jsidhu | pabelanger: yeah i was doing something wrong |
| 01:19.41 | jsidhu | started over and followed http://pbxinaflash.com/forum/showpost.php?p=24595&postcount=3 |
| 01:19.45 | jsidhu | works fo rme |
| 01:27.21 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 01:28.31 | sawgood | How could this be ... I yum removed all the asterisk RPM packages I installed, I rebooted the server, I look and saw the /etc/asterisk folder was empty, but yet I can still launch and run Asterisk? |
| 01:28.53 | *** part/#asterisk |Rain| (rain@ev.il.net) |
| 01:29.21 | chuckf | sounds like a binary didn't get removed |
| 01:30.02 | sawgood | <PROTECTED> |
| 01:30.08 | sawgood | I am looking now to see if there is anything left |
| 01:30.37 | chuckf | 'which asterisk' |
| 01:31.30 | sawgood | 1.6.2.8 |
| 01:31.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:31.47 | sawgood | I found two RPM packages still installed ... I was able to remove one, but not the other |
| 01:31.52 | sawgood | I did a core stop now from the CLI |
| 01:32.04 | sawgood | asterisk16-core-1.6.2.8-1_centos5 |
| 01:32.19 | sawgood | this RPM package will not come off with yum remove asterisk16-core-1.6.2.8-1_centos5 |
| 01:32.41 | sawgood | there is goes ... finally ... gone |
| 01:33.36 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:34.18 | sawgood | even if you do a yum remove of all Asterisk packages, it still leaves the /etc/asterisk directory with several saved copies of your Asterisk .conf files |
| 01:34.25 | sawgood | asterisk.conf.rpmsave |
| 01:35.04 | chuckf | leaving behind user configured (or potentially configured) files is a good option to default to |
| 01:36.06 | sawgood | yeah .. you are right about that |
| 01:36.52 | sawgood | The thing is ... I never installed FreePBX on this box, but yet after an install of the Asterisk RPM files (then removing them to see what is left) ... I see a /var/spool/asterisk/freepbx type file left behind |
| 01:37.08 | sawgood | maybe a long time ago this box had FreePbX on it, and I am just overlooking it |
| 01:40.48 | chuckf | have you tried a 'locate asterisk' search? |
| 01:43.47 | sawgood | I did that ... I think maybe this box had AsteriskNOW 1.5 on it a long time ago |
| 01:44.28 | sawgood | that is the only thing I can think ... I'm not sure ... because I would have simply put CentOS 5.4 on the box and build Asterisk from scratch or RPMs .... and FreePBX would have never been on the box |
| 01:44.49 | sawgood | I am going to take the box from the co-lo; wipe the drive; reinstall CentOS |
| 01:46.44 | sawgood | check this out ... the second I fire up Asterisk (after a fresh RPM build and no config files) ... I am flooded with this message from the CLI |
| 01:46.46 | sawgood | [Jun 22 18:46:11] NOTICE[3013]: chan_sip.c:21625 handle_request_register: Registration from '"user" <sip:user@65.49.22.226>' failed for '71.5.70.79' - No matching peer found |
| 01:46.57 | sawgood | 1000s of the same message scrolling up the screen ... |
| 01:47.35 | sawgood | I do not even have anything set in sip.conf or extensions.conf like that yet (it is a fresh build of the software) |
| 01:48.56 | sawgood | maybe someone is trying to 'hack' the box? |
| 01:49.11 | sawgood | I have no idea who this 71.5.70.79 is |
| 01:49.37 | pabelanger-lap | sawgood: Do you have a firewall enabled? |
| 01:49.58 | pabelanger-lap | sawgood: I would recommend one |
| 01:50.46 | sawgood | I can start iptables I guess ... but I do not know the sytax to stop this for sure (I might be able to figure it out) |
| 01:51.01 | sawgood | its been a while since I wrote an iptables DROP statement |
| 01:52.42 | sawgood | ok got it stopped ... |
| 01:52.50 | sawgood | no more messages scrolling by |
| 01:52.50 | pabelanger-lap | sawgood: Unprotected Asterisk box on a public network is never a good idea |
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| 01:56.20 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 01:57.12 | ManxPower | When using the Zaptel/DAHDI and faxdetect=incoming I have to Answer() then wait. Should I use a Wait(x) or a WaitExten(x)? |
| 01:57.29 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 02:04.32 | pabelanger-lap | ManxPower: Depends, add exten => fax,1,... in your context. You'll have to play with the Wait() value |
| 02:07.18 | ManxPower | So I should use Wait instead of WaitExten? |
| 02:07.51 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
| 02:08.03 | pabelanger-lap | ManxPower: Are you expecting somebody to enter a extension? |
| 02:10.52 | sawgood | <PROTECTED> |
| 02:10.58 | sawgood | in a nutshell what does this mean? |
| 02:11.41 | pabelanger-lap | sawgood: RTP is active |
| 02:12.22 | *** join/#asterisk blaines (~blaines@75-171-71-26.phnx.qwest.net) |
| 02:13.29 | sawgood | I am not able to receive phone calls anymore (my trunk is registered) ... the only change is an upgrade to 1.6.2.8 from 1.6.2.6 (now when I call the DID) nothing populates in the CLI except that one statement |
| 02:13.44 | WIMPy | sawgood: 802.1p for mor information |
| 02:13.47 | sawgood | I can call outbound ... but I cannot receive calls |
| 02:13.49 | WIMPy | +e |
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| 02:27.19 | sawgood | So, I see on the Digium site the 'instructions' for installing Asterisk via RPM (very easy clean nice process) .. and now I know how to 'remove' Asterisk using the yum remove commands (so I can go from having Asterisk 1.6.2.8 to nothing and or from nothing to 1.6.2.8) in a matter of a few minutes |
| 02:27.27 | sawgood | What I would like to know is if .... |
| 02:27.53 | sawgood | is there a set of instructions for which yum install RPM packages are required if you want to go from nothing to 1.4.x |
| 02:28.06 | sawgood | I think my box has stopped working with 1.6.2.8 ... |
| 02:28.19 | sawgood | It was working with 1.6.2.6 and earlier ... |
| 02:28.59 | sawgood | the yum RPM packages automatically install the latest 1.6.2.x release ... and I want to use 1.4.x (without having to build it from src) |
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| 02:31.36 | boodu | is it possible to send a number to chan isdn from route number ? |
| 02:35.04 | WIMPy | boodu: Can you rephrase that? |
| 02:35.32 | boodu | i can try :D |
| 02:37.06 | boodu | I want a user compose a number as 0751044 and then the isdn call directly 751044. Actually user must dial 0 and after compose the real phone number |
| 02:37.48 | boodu | It's more understandable ? |
| 02:38.18 | WIMPy | Sounds like standard dilplan thing and nothing to do with any channel specific stuff. |
| 02:40.36 | boodu | ok, I must modify dialplan and don't use route with trunk for that |
| 02:43.25 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 02:44.54 | sawgood | wow! I confirmed it ... I took Asterisk 1.6.2.8 off via the RPM remove process ... and I put 1.4.x on via the RPM process, and my box receives phone calls now ... |
| 02:45.08 | sawgood | something 'broke' between 1.6.2.6 and 1.6.2.8 |
| 02:45.10 | sawgood | amazing |
| 02:45.37 | sawgood | so, lesson here is ... if your box works, don't just run yum update because you want the latest release |
| 02:46.04 | WIMPy | What didn't work with 1.6.2.8? |
| 02:46.55 | sawgood | Well, when I try to call the IP PBX with 1.6.2.8 installed, from the CLI ... I would only get the message about CoS 5 (then the line would go dead) |
| 02:47.07 | sawgood | I got nothing else on the CLI (I could not receive a phone call) |
| 02:47.18 | sawgood | When I had 1.6.2.6, calls flowed in just fine |
| 02:47.42 | WIMPy | Ok, so nothing to worry about :-) |
| 02:47.55 | sawgood | So, I guess now ... I will need to get the SRC tarball for 1.6.2.6 and build that on the box |
| 02:48.05 | sawgood | then, if it works ... do not update it |
| 02:48.06 | WIMPy | just upgraded from 1.6.2.0something to 1.6.2.9, but SIP works. |
| 02:48.30 | sawgood | 1.6.2.9 is not in the RPM yet, right? |
| 02:48.40 | WIMPy | nfi |
| 02:49.20 | sawgood | Actually, I'm glad this happened because with my excellent note taking ability ... I can overcome problems like this in the days ahead .. |
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| 03:06.29 | Shaaan | anyone here do predictive dialing?? |
| 03:08.11 | [TK]D-Fender | SIP always worked |
| 03:10.47 | WIMPy | Ugh. Mu CLI just turned black on black. |
| 03:14.20 | sawgood | [TK]D-Fender: I read your archived 'frog' joke while researching Asterisk stuff on the net |
| 03:14.23 | sawgood | funny joke! |
| 03:14.33 | [TK]D-Fender | sawgood: Which? |
| 03:14.46 | sawgood | frog and the princess and the girlfriend |
| 03:15.01 | [TK]D-Fender | sawgood: Link it... I've clearly fogotten |
| 03:15.34 | sawgood | no bookmark here ... it was during late night research on MWI lights and PFK buttons ... |
| 03:15.52 | sawgood | It was in a forum and/or chat room archive format though (on some web page) |
| 03:17.42 | sawgood | Do you think Digium would be interested in knowing (which I have re-confirmed three times) ... my Asterisk box was working just fine under 1.6.2.6, but with a yum update to 1.6.2.8 (I was no longer able to receive an incoming SIP trunk call) ... |
| 03:17.53 | sawgood | Do you think they would like to know about this in some form? |
| 03:19.09 | sawgood | I have rolled back, and I am receiving calls just fine (I re applied 1.6.2.8 and could not receive a call) ... then back and it work ... etc .. etc. |
| 03:21.03 | *** join/#asterisk draiven (~draiven@cable201-232-155-11.epm.net.co) |
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| 03:21.25 | [TK]D-Fender | sawgood: You have absolutely nothing to show us and the weakest description this side of creation. |
| 03:21.57 | [TK]D-Fender | sawgood: Basically... you did something wrong or the packager did. because the release versions would have caused a panic otherwise |
| 03:36.11 | draiven | hi, I am new user of asterisk, I install on ubuntu the asterisk 1.6.2.9 and asterisk-addons 1.6.1.4 from http://downloads.digium.com repositories and I has the error; "ERROR: __ASTERISK_SBIN_DIR__/asterisk not found" when I run service asterisk start |
| 03:36.37 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:38.14 | draiven | somebody can help me?, and sorry for my english, i speak spanish |
| 03:43.10 | draiven | some body here? |
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| 03:45.02 | WIMPy | Probably noone hwo built those packages. |
| 03:48.33 | draiven | i unpack the files in /usr/src/ then i run sudo configure, sudo make, sudo make install, and sudo make samples |
| 03:49.42 | WIMPy | aye |
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| 03:51.21 | draiven | also, I install before the zaptel and the libpri |
| 03:51.43 | [TK]D-Fender | Asterisk 1.6 does not WORK with Zaptel |
| 03:52.02 | WIMPy | zaptel has been replaced by dahdi and won't work with 1.6. |
| 03:52.26 | draiven | mmm |
| 03:52.28 | draiven | ok |
| 03:52.45 | WIMPy | But nfi on that 'service' thing. |
| 03:53.01 | [TK]D-Fender | And you need "make config" for init scripts |
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| 03:55.14 | boodu | ciao |
| 03:55.42 | Kyosh | whats the best way to audit an asterisk box to find any problems to ensure no hackers (or at least minimize potential threats)? |
| 03:56.52 | draiven | I copy de init scripts from contrib/init.d/rc.debian.asterisk to /etc/init.d/asterisk |
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| 04:09.41 | Kyosh | is there any docs on working with T38 and asterisk (specifically v1.4)? |
| 04:09.53 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 04:10.22 | [TK]D-Fender | Kyosh: To do what? |
| 04:10.47 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 04:10.54 | coppice | guide space shuttles home. of course |
| 04:11.27 | Kyosh | well i was wondering, i have asterisk v1.4 running. right now i dont know how to get T38 faxing to work. my provider supports T38 fax and my ATA's support T38 fax, but how do i get my asterisk box to support it as well? |
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| 04:12.53 | *** join/#asterisk _pepo_ (~kvirc@190.12.5.106) |
| 04:13.12 | _pepo_ | hi friends |
| 04:13.50 | Kyosh | hi pepo. all my friends have been giving me money tonight. want to be a friend? :) |
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| 04:15.28 | _pepo_ | I pass, thanks |
| 04:15.30 | _pepo_ | :D |
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| 04:15.48 | Kyosh | some friend |
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| 07:14.24 | derbaron | hello @ all |
| 07:15.18 | derbaron | i have a question regarding analog line in the usa .... connecting to asterisk 1.6 |
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| 07:41.19 | stix | Hi guys. Is it impossible to manipulate with the EXTEN variable? |
| 07:41.38 | stix | Trying something like this: Set(EXTEN=11110${EXTEN}) |
| 07:41.45 | stix | but it wont change the variable |
| 07:43.09 | ChannelZ | You can re-write it but that will suddenly change the extension you would be executing if memory serves so your dialplan will probably misbehave |
| 07:43.18 | ChannelZ | What exactly are you trying to accomplish? There is probably a better way |
| 07:44.13 | stix | hmm okay |
| 07:45.47 | stix | well the thing is, that the system I am trying to configure is using freepbx, so I can't just change what I want in the conf's. I have this "outbound route"/exten => _1111. and I want to add a 0 to all that is dialed here. |
| 07:46.02 | stix | Eg exten => _1111.,n,Macro(dialout-trunk,5,${EXTEN:4},,) Becomes: exten => _1111.,n,Macro(dialout-trunk,5,0${EXTEN:4},,) |
| 07:46.46 | kaldemar | stix: you just did what you want |
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| 07:47.32 | stix | yes but I cannot change that line in the conf, freepbx will overwrite it. I can however add my custom code above, and there I wanted to rewrite the EXTEN variable |
| 07:47.42 | stix | guess that can't be done |
| 07:48.43 | kaldemar | ask in #freepbx how that is done. |
| 07:48.54 | stix | yes I better |
| 07:50.31 | kaldemar | btw, the EXTEN var cannot be set in the dialplan. only read. |
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| 07:55.24 | stix | okay thanks |
| 07:59.04 | ChannelZ | Ahhh I'm mis-remembering -- the case I'm thinking of was someone who was using exten qualifiers (not sure what they are called - like 555/111) and he was playing some games with CallerID causing the dialplan to fail |
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| 09:04.20 | ndemir | here is a interesting question: can asterisk be used for voip infrastructure that will serve to 500.000 users? |
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| 09:05.27 | ndemir | and how many servers should i use? |
| 09:06.19 | mort_gib | I have an B410P card, but I seem to only get 3 calls out of 3 ISDN lines??? |
| 09:10.21 | *** join/#asterisk Benwa (~Benwa@17.124-200-80.adsl-dyn.isp.belgacom.be) |
| 09:12.53 | tzafrir | mort_gib, what would you expect to get? 6 calls? |
| 09:13.02 | tzafrir | On which ports do you get those calls? |
| 09:13.10 | mort_gib | for 3 ISDN2 circuits.. Yes |
| 09:14.34 | mort_gib | Looked like port 1-1 |
| 09:14.34 | mort_gib | for the first circuit, not in clients office now, just strange |
| 09:15.34 | mort_gib | tzafrir or, what does channel => 1-2 mean in chan_dahdi.conf if not channel one and two |
| 09:16.02 | tzafrir | yes, 1 to 2 |
| 09:16.06 | ndemir | no answer to me? |
| 09:16.24 | mort_gib | But then I should be able to use channel one and two |
| 09:16.33 | tzafrir | ndemir, "many" |
| 09:16.34 | mort_gib | ndemir: What was your question?? |
| 09:17.02 | ndemir | tzafrir , mort_gib : here is a interesting question: can asterisk be used for voip infrastructure that will serve to 500.000 users? |
| 09:17.12 | tzafrir | For starters, how many concurrent calls do you expect? How many concurrent registrations? |
| 09:17.13 | ndemir | and how many servers should i use? |
| 09:17.25 | mort_gib | ndemir yes, with careful planning |
| 09:17.38 | mort_gib | ndemir and enough to carry the load |
| 09:17.54 | ndemir | mort_gib what are the important points? |
| 09:18.09 | tzafrir | ndemir, that's the wrong question to ask. Start by understanding the exact details |
| 09:18.34 | mort_gib | how many concurrent calls, what bandwidth you have, what codecs etc etc |
| 09:19.33 | tzafrir | "I have to move 500,000 KG from here to there. How many trunks do I need?" |
| 09:19.37 | ndemir | that is the question i ask. suppose 300.00 concurrent calls i have. so what codec should i use for quality and what must be bandwidth? |
| 09:20.40 | ndemir | and i think i need load balancing, i should use may be 10 servers? |
| 09:20.54 | mort_gib | ndemir interesting project, you do realize that you will need funding for proof of concept and proper testing, even before you start answering those questions |
| 09:21.22 | mort_gib | Get some really seasoned Asterisk gurus onboard too! |
| 09:21.27 | ndemir | yes, i know mort_gib. |
| 09:21.42 | mort_gib | :-) Sorry! |
| 09:23.12 | ndemir | mort_gib , why sorry? :) |
| 09:23.16 | mort_gib | ndemir You will certainly need openSER and a clever infrastructure |
| 09:23.46 | mort_gib | Because it is self evident that you will need people who know what they are doing when you do a 500KK user setup |
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| 09:24.53 | mort_gib | I'm looking into doing a 155 office setup now |
| 09:25.06 | mort_gib | With between 15 and 25 users in each office |
| 09:25.59 | ndemir | mort_gib , can you tell what are you doing for infrastructure? basically? |
| 09:26.07 | mort_gib | In a setup like that it's not only the amount of users, but also how many changes you get to the setup and how you handle them |
| 09:26.17 | mort_gib | Well, early stages |
| 09:26.40 | mort_gib | I will be using an external provider for the numbers |
| 09:26.53 | mort_gib | DDI's that is |
| 09:27.00 | ndemir | dou you need balancing for your project? |
| 09:27.27 | mort_gib | I'm playing with Asterisk running in a cloud, but waiting for the servers to arrive |
| 09:27.50 | mort_gib | That could is for other projects too, I know there are issues getting Asterisk to run in a cloud |
| 09:27.51 | tzafrir | (openser has forked into kamailio and opensips) |
| 09:27.59 | mort_gib | tzafrir I know |
| 09:28.14 | tzafrir | ndemir might not know |
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| 09:28.31 | mort_gib | Well thanks for correcting me :-) |
| 09:28.38 | tzafrir | <mort_gib> I'm playing with Asterisk running in a cloud, but waiting for the servers to arrive |
| 09:28.52 | tzafrir | Sounds like a definition of "asterisk in the could" :-) |
| 09:29.05 | mort_gib | tzafrir Yes heh |
| 09:29.25 | mort_gib | Well I ordered 5 T310's from Dell last week |
| 09:30.17 | mort_gib | interesting to see how it works, and if it scales nicer than just buying a monster server |
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| 10:15.45 | Godfather_ | how can i see if my config in cdr_mysql.conf if connected to my asterisk? |
| 10:15.56 | Godfather_ | to see the "status" |
| 10:16.02 | Godfather_ | (1.6.0) |
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| 10:18.00 | Godfather_ | No such command 'cdr mysql status' (type 'help cdr mysql status' for other possible commands) |
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| 10:20.46 | kaldemar | Godfather_: does "module show like cdr" list cdr_mysql.so? |
| 10:21.39 | Godfather_ | kaldemar, http://pastebin.com/2JhJKCPc |
| 10:22.12 | kaldemar | cdr_addon_mysql.so seems to be the correct module name... |
| 10:22.50 | Godfather_ | kaldemar, i installed it from addons yes... |
| 10:23.11 | kaldemar | unload the module and load it again to see if you get a warning of some sort. |
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| 10:26.52 | Godfather_ | kaldemar, was a mistake on cdr_mysql.conf |
| 10:27.02 | Godfather_ | i didnt uncomment [general] :| |
| 10:27.08 | Godfather_ | Not currently connected to a MySQL server. |
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| 11:07.45 | Godfather_ | kaldemar, http://pastebin.com/R5B0GS19 |
| 11:07.46 | Godfather_ | any ideas? |
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| 11:53.32 | Martinblr | Does anybody knows BRI tone settings for greece? |
| 11:56.00 | kaldemar | Martinblr: indications.conf has some |
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| 12:08.24 | Martinblr | kaldemar: is there anything like power db for busy tone? |
| 12:13.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:13.27 | kaldemar | Martinblr: what do you mean, exactly? |
| 12:14.16 | Martinblr | something like power db settings |
| 12:14.43 | kaldemar | and by power db you mean? |
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| 12:17.57 | Martinblr | I suspect it could be the power settings for the line |
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| 12:56.41 | ndemir | which codec should be used for ARM926 processor? |
| 13:00.11 | kaldemar | ndemir: one that does not require the processor to transcode. |
| 13:00.25 | pabelanger | Agree |
| 13:00.38 | ndemir | how is it transcoded? |
| 13:01.03 | kaldemar | if two legs of a call don't use the same codec. |
| 13:02.11 | ndemir | if two peers of calls use the same codec, the processor is not used. Is that true? |
| 13:02.13 | pabelanger | ndemir: Ideally RTP should not be bridge via Asterisk. |
| 13:02.41 | pabelanger | ndemir: correct, no transcoding is required |
| 13:02.52 | kaldemar | ndemir: not true as you put it, but the processor is not used for transcoding. |
| 13:03.11 | kaldemar | ndemir: which quite obviously was what you meant. :) |
| 13:11.52 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:15.15 | [TK]D-Fender | RTP forwarding is a very small load. Transcoding is not |
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| 13:29.00 | radic | http://bravo.hopto.org/extensions.conf |
| 13:29.31 | radic | if I cakk 0900XXXXXX from context 373 It sohld be end in the context filter |
| 13:30.12 | radic | why asterisk let do the call? |
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| 13:32.05 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 13:32.07 | wcselby | o/ |
| 13:32.16 | kaldemar | radic: exten => _0ZXXX.,1,Dial(SIP/${EXTEN}@${OUTA}${CONTEXT}) is a better match than the include. |
| 13:32.47 | pabelanger | radic: *CLI> dialplan show 00900XXXXXX@373 |
| 13:32.51 | kaldemar | radic: inside contexts, extensions are matched first, then includes in the order they appear in the dialplan. |
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| 13:33.43 | radic | kaldemar: Is there a way that the include is processed befor the last extension? |
| 13:34.01 | kaldemar | radic: btw, is this: "exten => _090O.,1,GoTo(filter,1)" 090O and not 0900 on purpose? |
| 13:34.11 | kaldemar | radic: no. |
| 13:34.40 | kaldemar | radic: you need to put it in a separate context and include it before "filter". |
| 13:36.54 | pabelanger | radic: A debug log of your call will be help full too |
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| 13:37.04 | pabelanger | ~collectdebug |
| 13:37.14 | infobot | it has been said that collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 13:45.04 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:50.44 | radic | I have to list all nubers that shouldn't dialed in all 5 contexts? |
| 13:51.59 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 13:53.14 | E-bola | Have anybody ever heard/seen anybody use asterisk as a PA/Announcer type system. I have a new client who has an intercom system already, which allows the manager to "speak" to all basestations at the same time, so they can hear him in every room |
| 13:53.38 | E-bola | I've setup asterisk and snom's intercom in combination which lets you do point A->B intercom |
| 13:53.46 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 13:53.46 | E-bola | but they need point A->many |
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| 13:56.41 | pabelanger | E-bola: app_page? |
| 13:59.16 | *** join/#asterisk ndemir (~ndemir@94.121.167.175) |
| 13:59.42 | [TK]D-Fender | radic: No, it looks in the DIRECT context first, then through the INCLUDE'sin the order you included them |
| 13:59.49 | [TK]D-Fender | radic: Get your order right |
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| 14:00.13 | [TK]D-Fender | pabelanger: Indeed |
| 14:00.22 | [TK]D-Fender | pabelanger: They should name it something more obvious. |
| 14:01.27 | ndemir | how to enable SIP tcp transport? |
| 14:02.10 | kaldemar | ndemir: you'll find the parameters, among others, in the sample sip.conf |
| 14:02.14 | pabelanger | ndemir: transport=tcp,udp |
| 14:13.31 | ndemir | pabelengar thanx |
| 14:14.44 | ndemir | pabelanger: will it open tcp port 5060? |
| 14:15.51 | puzzled | hi |
| 14:16.09 | pabelanger | ndemir: by default yes. You can change it using binaddr= |
| 14:16.33 | pabelanger | s/binaddr/bindaddr |
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| 14:17.20 | ndemir | pabelanger: ok i have to add tcpenable=yes |
| 14:18.01 | pabelanger | ndemir: sip.conf.samples has all the information |
| 14:23.04 | E-bola | pabelanger: sounds like the right thing |
| 14:23.13 | E-bola | although i get a WARNING[29159]: app_meetme.c:774 build_conf: Unable to open pseudo device |
| 14:23.34 | pabelanger | E-bola: DAHDI installeD? |
| 14:23.58 | E-bola | not intentionally atleast. We've never used conferences before, but it looks like page needs that? |
| 14:25.59 | pabelanger | E-bola: not sure about app_page.c but your warning is from app_meetme.c. Install DAHDI, recompile / install Asterisk |
| 14:27.32 | WIMPy | Or use ConfBridge instead. |
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| 14:34.04 | [TK]D-Fender | app_pagUses meetme. |
| 14:34.55 | *** part/#asterisk Mhaddog_NetB (~Mhaddog@173-146-109-198.pools.spcsdns.net) |
| 14:40.29 | pabelanger | [TK]D-Fender: learn something new everyday |
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| 14:41.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 14:46.20 | Godfather_ | hi |
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| 14:57.13 | houms | can anyone help me with openfire asterisk-im? everything works great except for incoming call notifications in the spark client? |
| 14:57.26 | Godfather_ | Im trying to connect cdr_mysql to asterisk, but i can't. i'm getting "failed to connect to database" |
| 14:57.33 | Godfather_ | here is some output... http://pastebin.com/6aLaYQEX |
| 14:58.09 | Naikrovek | houms: i have that working. you need to associate an asterisk extension with the spark username |
| 14:58.17 | Naikrovek | i need to update those, actually |
| 14:58.21 | houms | what do you mean? |
| 14:58.26 | Naikrovek | hang on |
| 14:58.30 | houms | currently I have the phone mappings done |
| 14:58.34 | leifmadsen | Godfather_: check the logs on the mysql side of things to see what part of the authentication is failing. From the Asterisk side you're not going to get much information. |
| 14:58.35 | [TK]D-Fender | Godfather_: sock=/tmp/mysql.sock <------ is it actually there? |
| 14:59.11 | houms | and logging in to spark client works fine, when on call the status works properly, but no pop-ups show when incoming calls are received |
| 14:59.40 | Godfather_ | [TK]D-Fender, vitto:/etc/asterisk# updatedb vitto:/etc/asterisk# locate mysql.sock |
| 14:59.43 | Godfather_ | i get no output |
| 15:00.14 | houms | Naikrovek how can i troubleshoot this? Or is there another configuration piece i am missing? |
| 15:00.28 | Naikrovek | houms: i'm looking at mine now |
| 15:00.45 | houms | thanks Naikrovek, your help is appreciated |
| 15:01.07 | Godfather_ | leifmadsen, no output, http://pastebin.com/BwgLx3Ki |
| 15:01.41 | radic | how can I get the number auf a incoming call? |
| 15:02.07 | Godfather_ | [TK]D-Fender, i see in my.conf .. |
| 15:02.11 | Godfather_ | [client] |
| 15:02.11 | Godfather_ | port = 3306 |
| 15:02.11 | Godfather_ | socket = /var/run/mysqld/mysqld.sock |
| 15:02.36 | Godfather_ | should i remplace sock to /var/run/mysqld/mysqld.sock ? |
| 15:03.49 | Godfather_ | [TK]D-Fender, http://pastebin.com/Gq3XNNiv |
| 15:04.49 | Corydon76-dig | leifmadsen: aren't you on vacation? |
| 15:05.02 | Naikrovek | since when did vacation prevent people from working |
| 15:05.02 | *** join/#asterisk s0lid (~s0lid@121.96.69.36) |
| 15:05.21 | Naikrovek | houms: i don't know why yours isn't working. after looking at mine i'm not sure why it *is* working... |
| 15:05.27 | leifmadsen | Corydon76-dig: not really today... I'm supposed to be, but it's going to thunderstorm this afternoon and it rained yesterday, so we came back |
| 15:05.37 | Godfather_ | [TK]D-Fender, you were right, now is loading, Loaded cdr_addon_mysql.so |
| 15:06.01 | Godfather_ | Connected to asteriskcdr@localhost, port 3306 using table cdr for 49 seconds. |
| 15:06.01 | Godfather_ | :D |
| 15:06.52 | houms | lol, I just setup the asterisk-im plugin and added the server. and added an entry to manager.conf. do you mind sharing what user the openfire server is using to talk to asterisk? are you using admin? |
| 15:06.53 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 15:07.06 | wcselby | funny how putting thigns like the right path in your config files seems to make things work |
| 15:07.10 | houms | Naikrovek are you using the admin user? |
| 15:07.58 | wcselby | ugh, i have a client that thinks the answer to any phone system issue is to reboot the phone server. which wouldn't be so bad, if I didn't get emails notifying me of the fact, every time. |
| 15:08.07 | [TK]D-Fender | Godfather_: Sometimes you just have to read the big print with the giant neon sign and flashing arrow on it... |
| 15:08.28 | Godfather_ | lol |
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| 15:52.15 | wcselby | miracle goal in penalty time and USA advances |
| 15:52.29 | wcselby | well, not a miracle, but a damn fine goal |
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| 15:58.20 | ndemir | is there a way to calculate maximum number of concurrent call for a asterisk server? |
| 15:58.48 | Chainsaw | ndemir: It is highly dependent upon whether any transcoding is required. |
| 15:59.08 | [TK]D-Fender | ndemir: Yes. Reach it. |
| 16:00.20 | ndemir | Chainsaw: suppose no transcoding is needed. |
| 16:00.32 | Chainsaw | ndemir: You've actively prevented it? |
| 16:00.42 | ndemir | Chainsaw: yes. |
| 16:01.01 | fenrus | there's benchmarking software that you can use :) |
| 16:01.27 | ndemir | fenrus: which benchmark is suitables? |
| 16:02.41 | xheliox | USA! USA! USA! :) |
| 16:02.47 | xheliox | That was awesome. |
| 16:04.35 | fenrus | ndemir, sipp |
| 16:04.50 | fenrus | sipp.sf.net |
| 16:05.26 | ndemir | fenrus: thanks |
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| 16:07.27 | ndemir | I found this: The figure of 200 concurrent channels is based on a dual Xeon 2.8 Ghz system with 1 GB of RAM performing no echo cancellation and no codec transcoding. |
| 16:07.38 | jcims | quick question. should the reply to a SIP REGISTER command go to the source port that the SIP REGISTER command was sent from, or to port 5060 on the phone that originated the request. |
| 16:08.11 | troffasky | I remember seeing something about source port brokenness on cisco handsets |
| 16:08.18 | troffasky | but I can't find it now |
| 16:08.29 | jcims | ok...i'll look around. |
| 16:08.31 | jcims | thanks |
| 16:08.37 | troffasky | I've been searching for it since you asked in #cisco |
| 16:09.01 | troffasky | I read it when I was setting up a 7940G to work wit asterisk |
| 16:09.18 | *** join/#asterisk Bladerunner05 (~Bladerunn@81-174-56-54.static.ngi.it) |
| 16:09.51 | jcims | lol, sorry about that. :) |
| 16:10.56 | Bladerunner05 | hello all, with tdm410p and dahdi (at the latest version) if I do dahdi_scan I see the card and my 4 fxo port. but on cli if I do dahdi show channels I don't see any channel... |
| 16:14.13 | russellb | have you configured it in both dahdi and asterisk? |
| 16:14.22 | russellb | <PROTECTED> |
| 16:14.26 | [TK]D-Fender | clearly not |
| 16:14.28 | russellb | if not, do that. :-) |
| 16:14.58 | russellb | and if you have further trouble getting the config right, http://www.digium.com/en/supportcenter/ |
| 16:15.39 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.219.69.dsl.dyn.forthnet.gr) |
| 16:16.11 | Bladerunner05 | russelb: /etc/dhadi/system.conf yes, but leave untouched chan_dahdi.. |
| 16:16.25 | [TK]D-Fender | Bladerunner05: that is your job. Go configure your channels |
| 16:18.29 | russellb | that is your mission, should you choose to accept it. |
| 16:18.30 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:18.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:18.38 | russellb | should you not, no calls 4 u! |
| 16:18.47 | carrar | I thought the job of blade runner is to track do wn and terminate replicants? |
| 16:18.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:19.24 | p3nguin | Are you asking him if that's what you thought? |
| 16:21.34 | tzafrir | Bladerunner05, also: what's the output of lsdahdi ? |
| 16:24.48 | *** join/#asterisk imox1234 (~imox1234@p4FC5C519.dip0.t-ipconnect.de) |
| 16:25.56 | *** join/#asterisk `Sauron (sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 16:26.35 | *** join/#asterisk imox1234 (~imox1234@p4FC5C519.dip0.t-ipconnect.de) |
| 16:27.18 | jcims | troffasky: gah! it was the voice_control_port setting buried in the sipdefault.cnf i was using. it was set to 5061, which for some reason caused the phone to send queries from any port rather than 5061 |
| 16:27.26 | *** join/#asterisk imox1234 (~imox1234@p4FC5C519.dip0.t-ipconnect.de) |
| 16:27.28 | jcims | changed it to 5060, reboot, phone works |
| 16:27.31 | troffasky | ah ok |
| 16:27.37 | jcims | thanks for the help |
| 16:27.43 | troffasky | there's a voip-info.org page about the 79xx series |
| 16:28.19 | troffasky | listing all the interesting bugs in each release |
| 16:29.04 | jcims | i'll check it out. i'm on a crusty release, but no smartnet, so i'm just making due. :) |
| 16:29.24 | jcims | thanks again... |
| 16:29.56 | *** join/#asterisk eliel (~eliels@host244.200-43-6.telecom.net.ar) |
| 16:33.14 | *** join/#asterisk jasonwert (~jasonwert@99-27-170-70.lightspeed.cicril.sbcglobal.net) |
| 16:33.20 | *** join/#asterisk neurosys (~neurosys@69.199.204.14) |
| 16:35.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
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| 16:39.20 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 16:43.04 | *** join/#asterisk sputnick (~sputnick@unaffiliated/sputnick) |
| 16:43.08 | sputnick | hi there |
| 16:44.12 | sputnick | is there an implementation or a feature to automaticly and dynamicly change extensions for the lowest price route ? |
| 16:44.41 | Qwell | sputnick: sure. it's called dialplan. |
| 16:44.50 | sputnick | or I need to be aware of prices from SIP providers and maintain a MySQL DB ? |
| 16:45.20 | [TK]D-Fender | sputnick: There is no AOC for SIP in * |
| 16:45.52 | [TK]D-Fender | sputnick: and * isn't psychic. Indeed the diallpan does exactly what you tell it to. If you have some table you can access for your vendors, then by all means get coding |
| 16:46.14 | [TK]D-Fender | sputnick: What you use as a pricing reference is up to you. No-one else said you had to use SQL for this |
| 16:47.33 | sputnick | [TK]D-Fender: by AOC you mean http://en.wikipedia.org/wiki/Autonomic_Computing ? |
| 16:48.07 | [TK]D-Fender | sputnick: No. |
| 16:48.12 | sputnick | sorry |
| 16:48.14 | [TK]D-Fender | sputnick: Advice Of Charge |
| 16:48.26 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:48.55 | [TK]D-Fender | sputnick: there is standard way to know which resource will be cheapest. You'll have to mash it up yourself |
| 16:49.09 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 16:52.02 | sputnick | [TK]D-Fender: by "standard way", what did you thought ? |
| 16:52.11 | sputnick | web scraping ? |
| 16:53.01 | [TK]D-Fender | sputnick: Every vendor lists their prices however they feel like. Some may be a parseable web page. Other more direct CSV. Some direct DB driven. Who knows. the thing is EVERY one will be different |
| 16:53.08 | [TK]D-Fender | sputnick: Good luck keeping up with them all. |
| 16:53.16 | sputnick | :) |
| 16:53.20 | leifmadsen | I read "keeping" as "sleeping" |
| 16:54.09 | sputnick | thanks for the explanation [TK]D-Fender |
| 16:56.13 | sputnick | I think to deal only with a small number of providers to start. If that's starts with 33686xxxxxx -> provider foo ; If that's starts with 33656xxxxxx -> provider bar |
| 16:56.47 | [TK]D-Fender | leifmadsen: Slut-tastic :-) |
| 16:56.59 | *** join/#asterisk ndemir (~ndemir@94.121.167.175) |
| 16:57.19 | [TK]D-Fender | leifmadsen: I know... I'm just whore-able :p |
| 16:57.26 | [TK]D-Fender | turns his puns to "11" |
| 16:57.27 | leifmadsen | ba-doom-chik |
| 16:57.36 | ndemir | another question: when should i use isdn pri card? |
| 16:57.48 | Qwell | ndemir: because you have an ISDN PRI |
| 16:57.56 | [TK]D-Fender | ndemir: When you want to use an ISDN PRI with Asterisk |
| 16:57.58 | Qwell | err, when |
| 16:58.02 | leifmadsen | Qwell: I was going to say something like that :) |
| 16:58.32 | [TK]D-Fender | Doctor, Doctor! It hurts when I raise me arrrr..... awww FUKKIT |
| 16:58.35 | [TK]D-Fender | gives up |
| 17:00.49 | ndemir | <PROTECTED> |
| 17:01.57 | [TK]D-Fender | ndemir: That's the idea. |
| 17:02.56 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 17:03.04 | ndemir | for example in this picture (http://www.voip-info.org/img/wiki_up//asterisk_1.png), asterisk servers connect each other via ISDN. Why do i need this instead of connecting via IP address? |
| 17:05.09 | [TK]D-Fender | ndemir: Depends where A & B *are* |
| 17:05.23 | [TK]D-Fender | ndemir: SIP over the internet = massively unreliable |
| 17:06.04 | [TK]D-Fender | ndemir: locally I might be inclined to use SIP on a secondary NIC, or perhaps TDMoE if I found it necessary. Neither requires a special card, just a NIC |
| 17:08.05 | [TK]D-Fender | ndemir: And your picture didn't show a fax at all BTW |
| 17:08.34 | ndemir | [TK]D-Fender: it is not my picture. i just found it. |
| 17:09.36 | [TK]D-Fender | ndemir: Hi, here's an unrelated picture, now what's wrong with ME systeM?!?!?! |
| 17:09.39 | [TK]D-Fender | MY* |
| 17:10.25 | ndemir | [TK]D-Fender: i am trying to understand when to use ISDN. |
| 17:11.02 | [TK]D-Fender | ndemir: When you need to interface with one. |
| 17:11.24 | drmessano | Don't drive your car to the kitchen, the hallway won't like it |
| 17:11.28 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:12.51 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 17:15.48 | ndemir | [TK]D-Fender: now i have an asterisk working with SIP (over IP), it works. On the other hand, i have an ISDN card, i installed it, compiled kernel modules, created conf with genzatelconf. Now what to do? |
| 17:17.07 | [TK]D-Fender | ~nowwhat |
| 17:17.08 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
| 17:20.29 | pabelanger | infobot: Get a pepsi |
| 17:20.30 | infobot | ACTION fetches a pepsi |
| 17:20.51 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:22.43 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 17:25.20 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 17:28.40 | eduzimrs | [TK]D-Fender is prefered to use IAX over internet? |
| 17:30.12 | t_dot_zilla | does asterisk adjust the volume of MOH ? the MOH was too loud, so we lowered the volume in audacity and replaced the MOH in asterisk but it seems to be the same volume |
| 17:30.28 | Qwell | t_dot_zilla: the phones likely do |
| 17:30.46 | Chainsaw | t_dot_zilla: It depends on how you produce the music-on-hold, but yes, the other end may be adjusting the volume as well (automatic gain control). |
| 17:30.51 | pabelanger | eduzimrs: just easier to manage |
| 17:31.01 | Chainsaw | t_dot_zilla: I have a volume adjustment on mine, which is pretty far down. |
| 17:31.20 | t_dot_zilla | we are calling in from TDM and it seems to be too loud |
| 17:31.27 | WIMPy | also finds that he has to save MOH files at extremely low volume in order to get acceptable results. |
| 17:31.29 | Chainsaw | t_dot_zilla: application=/usr/bin/dumbout /etc/asterisk/LINX/TheBlueValley.s3m -m -s 8000 -r 2 -v 0.2 -o - |
| 17:31.59 | Chainsaw | t_dot_zilla: The -v switch on that application controls the volume. It seems to have quieted down the complaints of hold music being "excruciatingly loud". |
| 17:32.40 | Qwell | s3m? |
| 17:32.54 | Chainsaw | Qwell: Yes, it's a sequenced music format. Consider it like MIDI with embedded samples. |
| 17:33.09 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 17:33.20 | t_dot_zilla | Chainsaw: i'm confused by your application= i tried something like /usr/bin/play fpm-calm-river.wav | sox -r 16000 -t wav - -r 8000 -c 1 -t raw - vol 0.10 |
| 17:33.28 | t_dot_zilla | i got errors in asterisk |
| 17:33.28 | Chainsaw | Qwell: -rw-r--r-- 1 tony users 553K 2009-10-23 14:21 /etc/asterisk/LINX/TheBlueValley.s3m |
| 17:33.37 | Chainsaw | Qwell: Not bad for 12+ minutes of music. |
| 17:33.55 | Chainsaw | t_dot_zilla: Do not pipe, use a single application. |
| 17:34.05 | t_dot_zilla | Chainsaw: what is in /usr/bin/dumbout |
| 17:34.26 | Chainsaw | t_dot_zilla: That is an application to play sequenced music files. |
| 17:34.39 | Chainsaw | t_dot_zilla: /usr/bin/dumbout: ELF 64-bit LSB shared object, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.9, stripped |
| 17:35.20 | Chainsaw | t_dot_zilla: If you have an application called DUMB in your repository... I can get you the S3M and you can try that? |
| 17:35.30 | [TK]D-Fender | eduzimrs: IAX2 is preferred when it is needed. When it isn't then SIP is |
| 17:35.53 | Chainsaw | eduzimrs: IAX2 ability to penetrate NAT & firewalls is not to be underestimated. |
| 17:36.58 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 17:37.40 | Micc | I need to find a digium reseller in the Seattle area. |
| 17:37.50 | Micc | We need to get a TE122p today if possible. |
| 17:38.19 | Qwell | Micc: https://www.digium.com/en/ecosystem/resellers/locate.php |
| 17:38.25 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 17:39.22 | [TK]D-Fender | Chainsaw: Was it good for you? |
| 17:39.22 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:39.24 | [TK]D-Fender | lights up |
| 17:40.08 | grinder13 | hello! i 've compiled and installed asterisk from svn. while trying to connect to the asterisk CLI I get the following error: "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)" |
| 17:40.18 | Qwell | grinder13: does it exist? |
| 17:40.32 | Chainsaw | grinder13: What user is the Asterisk daemon running as? What user are you calling asterisk -r as? |
| 17:40.40 | grinder13 | i discovered that the problem is here: "asterisk[1317]: segfault at d5 ip 08066032 sp bf904684 error 4 in asterisk[8048000+1af000]" |
| 17:41.04 | Chainsaw | grinder13: You've broken it already? That's quick. How did you get this one? Installed from a package manager? |
| 17:41.05 | *** join/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk) |
| 17:41.22 | grinder13 | as i said I compiled from svn |
| 17:41.55 | grinder13 | any hints? |
| 17:41.57 | *** part/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk) |
| 17:41.58 | Chainsaw | grinder13: Asterisk is developed by humans, not robots. Try the latest release to see if it a problem in the source or the way you've compiled it? |
| 17:42.09 | Qwell | grinder13: what branch did you install? |
| 17:42.21 | Chainsaw | transfers the call to Qwell |
| 17:42.29 | Qwell | presses DND |
| 17:43.23 | grinder13 | that's what i thought Chainsaw. Qwell, I 've used the SVN trunk code from here: http://svn.asterisk.org/svn/asterisk/trunk/ |
| 17:43.40 | Qwell | don't use trunk |
| 17:43.46 | Qwell | use branches/1.6.2/ |
| 17:44.04 | grinder13 | i need to use svn because of SRTP |
| 17:44.11 | paulc | while I'm on hold for some more bad customer service from Dell... How can I tell if dahdi_dummy is properly loaded, and where it's getting its timing source from? |
| 17:44.27 | Qwell | paulc: what version of dahdi? |
| 17:44.32 | *** join/#asterisk Draiven (~draiven@cable201-232-155-11.epm.net.co) |
| 17:44.48 | Chainsaw | paulc: The dahdi_dummy kernel module should show up in lsmod. |
| 17:45.19 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:45.30 | t_dot_zilla | in the musiconhold.conf, if you use the application=something, does mode=custom or can it be =files will the application still run? |
| 17:45.37 | Chainsaw | t_dot_zilla: mode=custom |
| 17:46.11 | *** join/#asterisk Jumpie (n3rdz@ip68-98-28-19.ph.ph.cox.net) |
| 17:46.18 | paulc | @Qwell 2.3.0.1+2.30 |
| 17:46.31 | Qwell | then dahdi_dummy does not exist |
| 17:46.38 | Qwell | if dahdi is loaded, it's providing timing |
| 17:47.47 | paulc | lsmod shows dahdi but no dahdi_dummy |
| 17:48.26 | Chainsaw | paulc: Based on the version you have, you shouldn't have dahdi_dummy at all. Listen to Qwell. |
| 17:48.31 | grinder13 | so regarding my segfault error in the syslog, no other tips/solutions apart from recompiling with the latest SVN release? |
| 17:48.48 | Chainsaw | grinder13: You should use the 1.6.2 branch, not trunk. |
| 17:49.03 | Chainsaw | grinder13: Or a release tarball, of course. Those work well. |
| 17:49.16 | *** join/#asterisk sekil (~sekil@78.24.111.218) |
| 17:49.22 | paulc | So just seeing "dahdi" in my lsmod is good enough and I'm all good for timing with no hardware |
| 17:49.36 | Qwell | yes |
| 17:49.42 | Chainsaw | paulc: Yes, the "core timer" is now enabled by default. You'll be fine. |
| 17:50.02 | Chainsaw | paulc: (I used to have to patch it in Gentoo, but it's the default) |
| 17:50.10 | grinder13 | Chainsaw, as I said, I need the SRTP feature for my project. Unfortunatelly, I can't do without it. If it wasn't the SRTP I would install directly from the packages of my Linux distribution |
| 17:50.31 | Chainsaw | grinder13: And you are sure that this feature is not in the 1.6.2 branch? |
| 17:50.44 | grinder13 | yes, it's not there |
| 17:50.58 | Chainsaw | grinder13: That's a shame. You are unlikely to get support on trunk builds. |
| 17:51.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:51.28 | grinder13 | well, will see then |
| 17:51.52 | grinder13 | i 'll try my luck with the latest SVN release |
| 17:52.00 | Chainsaw | grinder13: If you got me a backtrace or something more substantial... |
| 17:52.03 | drmessano | That's all you can hope for, is luck |
| 17:52.38 | Chainsaw | grinder13: Then perhaps we could all give you some educated guesses of what's going wrong. But saying "it exploded" and giving me a report that says "exhibit A exploded"... doesn't leave me with much. |
| 17:53.01 | drmessano | Chainsaw: It exploded and it was bad |
| 17:53.30 | Chainsaw | drmessano: It was bad? This is how it responded to "core restart gracefully" barely 4 versions ago. |
| 17:53.40 | drmessano | lol |
| 17:53.41 | grinder13 | you are right, but how can I get a backtrace? sorry, I am not familiar with these programming stuff (feeling embarassed already). networking guy here... |
| 17:53.47 | *** join/#asterisk Z_God (~julius@wlan224088.mobiel.utwente.nl) |
| 17:53.57 | Chainsaw | grinder13: Okay, make sure you haven't stripped the binaries and run it through gdb. |
| 17:54.05 | t_dot_zilla | can someone tell me what this means: |
| 17:54.08 | t_dot_zilla | [Jun 23 13:53:36] NOTICE[19880]: res_musiconhold.c:602 monmp3thread: Request to schedule in the past?!?! |
| 17:54.10 | t_dot_zilla | [Jun 23 13:53:36] WARNING[19880]: res_musiconhold.c:620 monmp3thread: Unable to send a SIGHUP to MOH process?!!: No such process |
| 17:54.23 | [TK]D-Fender | t_dot_zilla: means your timing is off and you're using MPG123 for MoH |
| 17:54.35 | t_dot_zilla | i'm using /usr/bin/play |
| 17:54.38 | [TK]D-Fender | t_dot_zilla: change to NAtive MoH (mode=files) |
| 17:54.45 | Chainsaw | grinder13: Once it explodes, you'll get a gdb prompt. Type bt full and stick the result on pastebin somewhere. |
| 17:54.53 | [TK]D-Fender | t_dot_zilla: same diff |
| 17:55.11 | t_dot_zilla | <[TK]D-Fender>: i'm trying to use a customer app in MOH to lower the volume |
| 17:55.20 | t_dot_zilla | *custom |
| 17:55.44 | Chainsaw | grinder13: If we're lucky, you have debugging symbols and the output will be useful in determining where things derail. |
| 17:56.06 | [TK]D-Fender | t_dot_zilla: then jsut normalize your files |
| 17:56.15 | Chainsaw | grinder13: (Just "gdb asterisk" should get you going) |
| 17:56.21 | [TK]D-Fender | t_dot_zilla: You're wasting processing load, processes, etc |
| 17:57.21 | grinder13 | thanx Chainsaw, I am reading the debbuging info on voip-info.org. will get back |
| 17:57.29 | *** join/#asterisk southtel_ (~slester@c-24-126-177-12.hsd1.ga.comcast.net) |
| 17:58.18 | *** join/#asterisk Acidshock (~none@cpe-75-84-10-22.socal.res.rr.com) |
| 17:58.19 | Chainsaw | grinder13: Okay. |
| 17:59.20 | southtel_ | Can anyone suggest ways to troubleshoot CallerID problems for an analog system out in a rural area? |
| 18:00.09 | southtel_ | The phone lines out here are not great, but with a regular phone, the CID shows up fine. |
| 18:00.20 | paulc | Chainsaw / @Qwell : Thanks for your help (delayed reply, Dell support were consuming my energy.. aka will to live..) |
| 18:00.27 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
| 18:00.40 | southtel_ | But, on my * box (w/Digium wildcard), I only "see" the CallerID about 50% of the time. |
| 18:00.49 | Acidshock | Can anyone help me out with a dialplan issue I am having? I cant seem to get a sip trunk up and running. Keep getting error cause 3 no route to destitination errors |
| 18:01.06 | Acidshock | lol destination even :P |
| 18:01.19 | [TK]D-Fender | Acidshock: "sip show peer [yourpeerhere]" <- PASTEBIN it |
| 18:01.21 | [TK]D-Fender | ~pb |
| 18:01.21 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:01.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^66 |
| 18:01.49 | grinder13 | hmm, as I am reading stuff in voip-info.org I noticed one thing: The damn fool I forgot to select the DON'T OPTIMIZE compiler flag!!!!! ARGHHHH!!! |
| 18:02.04 | southtel_ | I've tried tweaking the cid_rxgain settings, but they don't _seem_ to make a difference. |
| 18:02.10 | Acidshock | Fender, I am using IP Authentication not registration. Do you still want me to pastebin that for you? |
| 18:02.22 | [TK]D-Fender | Acidshock: Provide what I have requested |
| 18:02.39 | grinder13 | will "make uninstall-all && make distclean && make" again... |
| 18:03.06 | Acidshock | just making sure :) |
| 18:04.02 | t_dot_zilla | Chainsaw: can you tell me more about how you've set up MOH? |
| 18:04.09 | Acidshock | Fender, http://pastebin.com/VDn6ZVgh |
| 18:04.19 | Chainsaw | t_dot_zilla: I use the DUMB suite of software to play sequenced music in S3M format. |
| 18:04.27 | Chainsaw | t_dot_zilla: As I asked, do you have such software in your package manager? |
| 18:04.41 | t_dot_zilla | no, i don't |
| 18:04.42 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 18:04.51 | t_dot_zilla | i'm running centos, i don't see it in yum search |
| 18:05.12 | Chainsaw | t_dot_zilla: Right, you may have to compile that from source then. You can listen to my hold music if you call me. |
| 18:05.26 | Chainsaw | t_dot_zilla: Perhaps tell me what the volume difference is like as well. |
| 18:05.34 | t_dot_zilla | ok |
| 18:06.20 | *** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net) |
| 18:07.48 | iratik | How can I make it so that calls cannot be made from outside our network and invite requests from our ITSPs? My initial thought is to make hosts.deny ALL and hosts.allow selectively permit local ip's and itsp ip's .. is there a setting in asterisk i can use? |
| 18:08.20 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:09.37 | southtel_ | Acidshock: can you access that ip from your machine? |
| 18:09.44 | Acidshock | yup |
| 18:10.26 | southtel_ | Can you put on the pastebin the output from the cli when you try to dial using it? |
| 18:11.18 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 18:11.37 | Acidshock | sure 1 sec |
| 18:12.50 | Acidshock | southtel_, http://pastebin.com/B4Gsw1Qu |
| 18:13.33 | Draiven | hi, i want install a cdr and realtime with my postgres db but it show me the error: found to engine 'pgsql', but the engine is not available |
| 18:13.34 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-77-81.home.otenet.gr) |
| 18:14.07 | [TK]D-Fender | Acidshock: Addr->IP : (Unspecified) Port 0 <--- You did nOT specify the HOST to call |
| 18:14.19 | [TK]D-Fender | Acidshock: Fix your peer. Asterisk has nowhere to call now |
| 18:14.39 | Draiven | somebody can help me? |
| 18:14.49 | Draiven | please |
| 18:16.11 | [TK]D-Fender | Draiven: PASTEBIn your configs, show us the module attempting to load. Show that you're able to conenct locally to the DB, etc. |
| 18:16.13 | [TK]D-Fender | ~pb |
| 18:16.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:16.15 | [TK]D-Fender | ^^^^^ |
| 18:17.00 | southtel_ | D-Fender: how many times a day do you trigger that pastebin infobot? |
| 18:18.04 | *** join/#asterisk dkirker-openmo-1 (~dkirker@openmobl/ceo/dkirker) |
| 18:19.13 | kaldemar | Draiven: res_config_pgsql.so is the module you're missing. you can also use a postgresql db via ODBC, as in the example in the book. |
| 18:19.30 | Draiven | I am conected to postrges localy with pgadmin |
| 18:19.32 | [TK]D-Fender | southtel_: Every time someone new needs to show us something and likely has no clue |
| 18:19.45 | [TK]D-Fender | Draiven: PASTEBIN <------------------------- |
| 18:19.50 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 18:20.22 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:20.58 | southtel_ | Good times. |
| 18:21.03 | Draiven | i dont have de res_config_pgsql module |
| 18:21.06 | Draiven | :( |
| 18:21.41 | southtel_ | D-Fender: I'm looking elsewhere as well, but do you have any thoughts on tracking down a CID problem in a rural area? |
| 18:21.57 | Acidshock | Fender, what would I use to specify that address because I am using host=<IP ADDRESS> |
| 18:22.22 | kaldemar | Draiven: how did you install asterisk? |
| 18:23.14 | *** join/#asterisk hfb (~hfb@96.247.65.56) |
| 18:23.34 | Draiven | I download the repositories from http://downloads.digium.com |
| 18:24.08 | Draiven | asterisk 1.6.2.9 and asterisk-addons 1.6.1.4 |
| 18:24.26 | Qwell | Draiven: repositories? downloads.digium.com? one of those is wrong |
| 18:24.42 | kaldemar | Draiven: you compiled from source? |
| 18:25.56 | Draiven | i install runing ./configure |
| 18:26.00 | Draiven | make |
| 18:26.04 | Draiven | make install |
| 18:26.39 | kaldemar | you need to install the dependency for res_config_pgsql, re-run configure and recompile. |
| 18:27.22 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:29.17 | [TK]D-Fender | Draiven: Asterisk add-ons 1.6.1 is NOT meant to be used with Asterisk 1.6.2 |
| 18:29.39 | [TK]D-Fender | Draiven: Stop putting Ford Mustang parts in your Toyota Prius |
| 18:29.54 | [TK]D-Fender | Draiven: Asterisk-Addons 1.6.2.1 <--------------------- |
| 18:30.07 | [TK]D-Fender | CRAZY PEOPLE |
| 18:30.27 | Acidshock | cheerful group lol |
| 18:31.02 | [TK]D-Fender | [14:21]<Acidshock>Fender, what would I use to specify that address because I am using host=<IP ADDRESS> <--- ummm.. |
| 18:31.12 | [TK]D-Fender | [14:14]<[TK]D-Fender>Acidshock: Addr->IP : (Unspecified) Port 0 <--- You did nOT specify the HOST to call |
| 18:31.24 | [TK]D-Fender | Acidshock: NO. Or you are not showing me what I asked for. |
| 18:31.56 | [TK]D-Fender | Acidshock: Whatever you think you've done does not matcht eh evidence. |
| 18:31.58 | Acidshock | let me PASTEBIN it for you :P |
| 18:32.53 | grinder13 | i cannot take a backtrace. Asterisk crashes immediately after I issue the command "asterisk -r" (or "gdb asterisk" or "safe_asterisk" or whatever) and exits, so absolutely nothing is captured by gdb. any ideas? |
| 18:33.16 | [TK]D-Fender | grinder13: start it MANUALLY |
| 18:33.45 | grinder13 | but that's what I am doing |
| 18:34.19 | Acidshock | Fender, http://pastebin.com/ZgQHSpyR |
| 18:35.10 | [TK]D-Fender | grinder13: Nothing you have shown me is "manually". "asterisk -gvvvvvvvvvvvvvc" is "manually" |
| 18:35.46 | [TK]D-Fender | Acidshock: type=friend <- should be "peer". |
| 18:35.51 | grinder13 | but that's what I am telling you Fender |
| 18:36.14 | [TK]D-Fender | Acidshock: update. Apply. then PB "sip show peers", the dump for that specific peer again, and your actual failed call at verbose 10. |
| 18:36.42 | [TK]D-Fender | [14:35]<grinder13>but that's what I am telling you Fender <_ you did NOT do it properly, MANUALLY ilke I showed. so go DO it, and then SHOW US |
| 18:37.22 | grinder13 | how many times do I have to say that I did so??? |
| 18:37.48 | Draiven | [TK]D-Fender, hehe, can i use asterisk-1.6.2.9 with addons-1.6.2.1? |
| 18:38.06 | [TK]D-Fender | [14:29]<[TK]D-Fender>Draiven: Asterisk-Addons 1.6.2.1 <--------------------- |
| 18:38.09 | [TK]D-Fender | ^^^^^ NOT OBVIOUS ENOUGH? |
| 18:38.21 | [TK]D-Fender | looks for the giant flashing neon sign again |
| 18:39.02 | kaldemar | what's the "<---------------------" in the version? |
| 18:39.18 | grinder13 | http://pastebin.com/Cx5GUAfk |
| 18:39.19 | kaldemar | almost .1 but not quite? |
| 18:39.20 | wcselby | it's the awesome arrow version |
| 18:39.32 | grinder13 | at the end of the above link you 'll see a segmentation fault |
| 18:39.40 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:39.41 | wcselby | grinder13 - have you tried starting asterisk with "asterisk -gvvvvvvvvc" from the command line? |
| 18:39.53 | wcselby | grinder13 - nvm |
| 18:40.02 | grinder13 | yeap, that's exactly what I did wcselby |
| 18:40.28 | grinder13 | asterisk exits with a segmentation fault |
| 18:40.32 | [TK]D-Fender | pulls another arrow from his quiver, nocks, draws, releases, and watches kaldemar drop like a sack of potatoes |
| 18:42.21 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:42.23 | [sr] | howdy |
| 18:42.53 | WIMPy | [sr]! |
| 18:43.07 | [sr] | is there something that can send the information do the SIP phone's to learn the SIP server, a bit at the image of dhcp boot |
| 18:43.10 | [sr] | hi WIMPy |
| 18:43.24 | [sr] | dont know if this exists, but its interesting if yes |
| 18:43.29 | Acidshock | Fender, http://pastebin.com/QARaZrFJ |
| 18:43.31 | Bladerunner05 | wich is the dahdi command to make a call from cli ? |
| 18:43.40 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 18:44.31 | *** part/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:44.49 | Acidshock | Fender, slightly updated one with error from CLI http://pastebin.com/8m5N3mS5 |
| 18:45.10 | WIMPy | [sr]: Depends on the phone, but most have some way of mass provisioning. |
| 18:45.17 | kaldemar | Bladerunner05: there is no such dahdi command. either "console dial" or "channel originate". |
| 18:45.40 | tzafrir | Bladerunner05, originate / channel originate |
| 18:45.41 | Bladerunner05 | kaldemar: of course.. |
| 18:45.59 | [TK]D-Fender | Acidshock: [Jun 23 11:43:48] WARNING[3954]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory <-- you calling channel is already broken. |
| 18:46.06 | [TK]D-Fender | Acidshock: Try again with a sane call |
| 18:46.10 | [sr] | WIMPy: thats the name for this protocol/method/techology? |
| 18:46.22 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:46.53 | WIMPy | [sr]: can be anything. tftp, ftp, http or eben https. Check your phones manual. |
| 18:47.01 | Acidshock | Fender, thats because I am dialing from CLI though correct? I can dial from a sip device and not receive that message |
| 18:47.06 | Acidshock | Fender, trying again |
| 18:47.08 | [sr] | WIMPy: hum.. going to check :) |
| 18:47.20 | [sr] | WIMPy: for now i just have one yealink |
| 18:47.39 | WIMPy | Never heard of. |
| 18:47.50 | [sr] | yealink.com |
| 18:48.29 | Acidshock | Fender, same problem http://pastebin.com/TLXespUL |
| 18:48.37 | WIMPy | Looks important |
| 18:49.15 | [sr] | WIMPy: they are accessible, i have the T28P |
| 18:50.18 | *** join/#asterisk smooth_penguin (~smoove@59.95.54.237) |
| 18:52.46 | WIMPy | Looks interesting. |
| 18:53.13 | [sr] | 112â¬+VAT |
| 18:55.05 | ruben23 | hi guys can i set the directory manually of my recordings..? the dedault is /var/spool/asterisk/monitor/ can i put it like /var/spool/asterisk/monitor/archive |
| 18:55.17 | WIMPy | [sr]: Seems ok. I'm looking forward to your experiences with it. |
| 18:56.01 | idespinner | [sr], for autconfiguration of phones, it depends on the phones. polycoms can use DHCP + FTP to be provisioned |
| 18:57.29 | grinder13 | i managed to get a backtrace. the problem is the libsrtp. damn it's the thing I need for my project!!!! will start hitting my had on the wall!!! |
| 18:58.35 | *** join/#asterisk fish-bulb (~cstewart@nat/digium/x-gamfhoashrmsohrk) |
| 19:01.37 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 19:02.41 | [TK]D-Fender | Acidshock: Where are your configs? check your routing on your server |
| 19:04.47 | *** join/#asterisk neurosys (~neurosys@69.199.204.33) |
| 19:05.46 | *** join/#asterisk xxiao (~xxiao@140.242.26.81) |
| 19:05.51 | *** join/#asterisk neurosys (~neurosys@69.199.204.33) |
| 19:06.12 | xxiao | new to asterisk, what's the sip stack used by asterisk? |
| 19:06.21 | Qwell | Asterisk |
| 19:06.40 | xxiao | grep the source code and did not find any specifics |
| 19:06.53 | [TK]D-Fender | Qwell: QUICK : What colour was Napoleon's white horse?!?! |
| 19:06.53 | Qwell | it's just Asterisk. |
| 19:06.55 | xxiao | do you mean asterisk use its own sip stack |
| 19:07.03 | [TK]D-Fender | xxiao: Clearly |
| 19:07.34 | xxiao | thanks. i was looking at osip2, pjsip and opensip and wondering if asterisk used them directly |
| 19:07.57 | *** join/#asterisk radic (~radic@178.2.208.89) |
| 19:08.13 | xxiao | need find a small footprint sip stack on a portable device to work with asterisk |
| 19:09.17 | [TK]D-Fender | xxiao: Perhaps you should look at osip2, pjsip and opensip |
| 19:10.01 | xxiao | [TK]D-Fender, thanks! i looked at openh323 and vovida 6 years ago and today with a new voip project, i found they all gone |
| 19:10.26 | xxiao | and asterisk is just getting stronger |
| 19:11.02 | [TK]D-Fender | xxiao: sofiasip <- |
| 19:11.09 | [TK]D-Fender | xxiao: sofia-sip <- |
| 19:11.21 | [TK]D-Fender | http://sofia-sip.sourceforge.net/ |
| 19:11.51 | xxiao | thanks again, another new project |
| 19:12.09 | xxiao | will compare them soon |
| 19:12.14 | radic | I've a dynamic IP. after a reconnect I'm not reachable until I did "sip reload" |
| 19:12.34 | [TK]D-Fender | radic: READ <- |
| 19:12.38 | [TK]D-Fender | ~sipnat |
| 19:12.38 | infobot | [~sipnat] Quick guide on configuring * + 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 |
| 19:12.40 | [TK]D-Fender | ^^^^ |
| 19:14.47 | [sr] | WIMPy: i'll tell, only have it for less than a day :p |
| 19:15.02 | [sr] | idespinner: hum... will check that |
| 19:15.34 | WIMPy | just noticed, that the website only says copyright 2001-2009 and the firmware link is dead. |
| 19:15.46 | radic | [TK]D-Fender: that dosn't help me |
| 19:16.25 | radic | externhost= is set in sip.conf |
| 19:16.34 | [TK]D-Fender | radic: Fix your REFRESH |
| 19:16.52 | [TK]D-Fender | radic: And verify your forwarding <- |
| 19:17.39 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 19:18.12 | *** join/#asterisk sekil (~sekil@78.24.111.218) |
| 19:19.05 | ruben23 | hi guys can i set the directory manually of my recordings..? the dedault is /var/spool/asterisk/monitor/ can i put it like /var/spool/asterisk/monitor/archive |
| 19:20.36 | knctrnl | has anyone ever got successfully integrated an avaya phone system with TN747B card with asterisk? |
| 19:20.49 | [TK]D-Fender | ruben23: No |
| 19:21.49 | radic | ruben23: cd /var/spool/asterisk/monitor && ln -s /var/spool/asterisk/monitor archive |
| 19:22.01 | [TK]D-Fender | knctrnl: Loks like a broing FXO interface. Got a specific question pertaining to your goals? |
| 19:22.08 | p3nguin | Why would you need to cd just to ln? |
| 19:22.29 | [TK]D-Fender | radic: NO. Go lookup RECUSION in the dictionary |
| 19:22.33 | [TK]D-Fender | RECURSION |
| 19:22.51 | p3nguin | I bet MixMonitor can use archive/<string>.<format> as the file name. |
| 19:23.46 | p3nguin | For example: MixMonitor(archive/${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.wav,a) |
| 19:24.11 | knctrnl | at one site we have an ancient avaya that we want to slowly transition to asterisk avaya will have pri to telco and asteirsk will have pri to telco. Telephone company will transition did's over as we need but we need to maintain 4 digit dialing. |
| 19:25.07 | ruben23 | radic: but i want to put it on another mounted HDD, since its getting full on the 1st one |
| 19:25.41 | dohd | knctrnl: ancient avaya and mix with anything?! is that possible? |
| 19:25.58 | WIMPy | knctrnl: Maybe a 2nd pri for your asterisk would habe been a better choice than a 2nd pri from your telco. |
| 19:26.13 | *** join/#asterisk Holos (~cosmond@static-pppoe-209-91-139-211.vianet.ca) |
| 19:26.34 | radic | ruben23: mount it to /var/spool/asterisk/monitor |
| 19:29.57 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 19:30.01 | knctrnl | its a gs3i and I think there is a card in there that supports E&M |
| 19:30.52 | knctrnl | i know if i look through the config ISDN is not enabled. |
| 19:33.01 | x-demon | what i need to configure in order to use sip url calls? |
| 19:35.46 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 19:36.14 | [TK]D-Fender | x-demon: exetnsions.conf dial command |
| 19:37.21 | x-demon | [TK]D-Fender, i know how to make dialplans... but i dunno how to configure rule for all SIP URLs |
| 19:37.47 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 19:37.53 | pabelanger | x-demon: DNS SRV records? |
| 19:38.16 | x-demon | pabelanger, enabled and exists in on my server |
| 19:38.44 | [TK]D-Fender | x-demon: You'll have to make a dialplan script to parse out the SIP header for your destination for the inbound call to strip the URI |
| 19:38.59 | [TK]D-Fender | x-demon: "core show function SIP_HEADER |
| 19:39.25 | x-demon | [TK]D-Fender, "Not available" |
| 19:39.43 | [TK]D-Fender | x-demon: show me |
| 19:40.11 | x-demon | in console i also see that xmldoc is not available, something like that |
| 19:40.33 | x-demon | WARNING[20991]: xmldoc.c:1720 xmldoc_build_field: Couldn't find application Park in XML documentation |
| 19:40.49 | pabelanger | x-demon: install libxml2-dev, recompile, install asterisk |
| 19:41.05 | x-demon | pabelanger, i use asterisk from debian repo |
| 19:41.26 | x-demon | backported from sid, but i also saw such messages with lenny and squeeze builds |
| 19:41.29 | [TK]D-Fender | x-demon: Meaningless. get a VERSION NUMBER |
| 19:41.50 | *** join/#asterisk neurosys (~neurosys@69.199.204.34) |
| 19:42.10 | x-demon | [TK]D-Fender, Asterisk 1.6.2.7-1.1 |
| 19:42.33 | [TK]D-Fender | x-demon: show me the check & error |
| 19:42.40 | x-demon | the check? |
| 19:43.25 | [TK]D-Fender | x-demon: I want to see you issuing the CLI request |
| 19:44.08 | x-demon | version number? |
| 19:45.35 | [TK]D-Fender | [15:38]<[TK]D-Fender>x-demon: "core show function SIP_HEADER |
| 19:46.17 | x-demon | [TK]D-Fender, http://pastebin.com/E11UL011 |
| 19:46.59 | [TK]D-Fender | x-demon: Now spell it wrong and see if it looks the same. Also I do not SEE you issuing the command in that PB |
| 19:47.07 | eliel | x-demon: run this please: #dpkg -l |grep xml |
| 19:47.40 | pabelanger | x-demon: ls -la /usr/share/asterisk/documentation/ |
| 19:48.57 | x-demon | eliel, libxml2 and libxml2-dev installed |
| 19:49.19 | eliel | x-demon: asterisk from a debian package or from sources? |
| 19:49.27 | x-demon | eliel, backport from sid |
| 19:49.49 | eliel | and that means? (from sources?) |
| 19:49.50 | x-demon | pabelanger, http://pastebin.com/niYwE32W |
| 19:49.59 | x-demon | eliel, debian package. |
| 19:51.11 | pabelanger | x-demon: core-en_US.xml is the file that lists all the commands. For some reason, asterisk is not loading it. |
| 19:51.24 | x-demon | pabelanger, let me check locale... |
| 19:51.38 | x-demon | oh well, en_US |
| 19:51.42 | pabelanger | x-demon: *CLI> core show settings |
| 19:51.45 | x-demon | and no LC_ALL |
| 19:51.51 | eliel | x-demon: cat /etc/asterisk/asterisk.conf |grep documentation_language |
| 19:52.11 | eliel | sorry, already answered that |
| 19:52.19 | x-demon | oh well... now i understand why... |
| 19:52.41 | pabelanger | drum-roll |
| 19:54.03 | eliel | maybe the libxml2 dependency was not met when installing asterisk (??) |
| 19:54.10 | Kobaz | anyone know how to set up lldp on a dell powerconnect switxh |
| 19:54.36 | pabelanger | eliel: core-en_US.xml exists though |
| 19:55.30 | x-demon | eliel, no i installed it before building backport |
| 19:55.55 | x-demon | eliel, btw, in which context i must put documentation_language = en_US |
| 19:56.06 | x-demon | [options]? |
| 19:56.26 | eliel | yes |
| 19:57.03 | x-demon | same error |
| 19:57.18 | x-demon | does it need separate module? i believe no... |
| 19:57.20 | eliel | x-demon: check file permissions |
| 19:57.27 | eliel | x-demon: no, it is in the core |
| 19:58.02 | x-demon | permissions 755 |
| 19:58.16 | eliel | asterisk running as ....? |
| 19:58.21 | x-demon | as asterisk |
| 19:58.27 | x-demon | with group asterisk |
| 19:58.28 | eliel | su asterisk |
| 19:58.36 | grinder13 | regarding my segfault with the libsrtp: i 've compiled libsrtp from source and it seems its fine now |
| 19:58.39 | eliel | then try to read the core-en_US.xml file |
| 19:58.40 | x-demon | i can't, this is no-login user |
| 19:59.03 | x-demon | su -c "cmd" asterisk? |
| 19:59.30 | eliel | ok, stop asterisk and start it with: asterisk -vvvc and copy all the output to pastebin |
| 19:59.59 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 20:02.00 | x-demon | eliel, http://pastebin.com/QsasWwaM |
| 20:03.14 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 20:03.30 | eliel | mmm |
| 20:04.02 | x-demon | i does not see any errors |
| 20:04.48 | x-demon | also i can read file from asterisk user |
| 20:05.05 | pabelanger | x-demon: pb your asterisk.conf file |
| 20:05.14 | eliel | x-demon: ls -al /var/lib/asterisk/documentation/ |
| 20:05.29 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:05.43 | x-demon | eliel, it's not in /var, it's in /usr/share |
| 20:05.48 | pabelanger | eliel: /usr/share/asterisk/documentation/ :) |
| 20:05.52 | pabelanger | under debian |
| 20:06.01 | eliel | ok, so cat /etc/asterisk/asterick.conf |grep astdatadir |
| 20:06.22 | x-demon | _damn_ |
| 20:07.06 | x-demon | one second, i must check.. |
| 20:07.30 | eliel | pabelanger: good, i have never installed asterisk from packages :S |
| 20:08.18 | x-demon | eliel, changed to /usr/share/asterisk |
| 20:08.21 | x-demon | thanks |
| 20:08.28 | pabelanger | eliel: Ya, need to ask tzafrir why Debian like /usr/share for data. |
| 20:08.57 | eliel | x-demon: good :) |
| 20:09.54 | x-demon | by the way, if i have user 1000, and want this user to be reachable also via "me", i just should setup alias=? |
| 20:13.35 | x-demon | or i can just leave user id as "me" ? |
| 20:16.03 | p3nguin | What is this "user id" you're talking about? |
| 20:16.25 | p3nguin | eliel: And don't use cat if you only need to grep. |
| 20:16.56 | p3nguin | "grep astdatadir /etc/asterisk/asterick.conf" is enough. |
| 20:17.05 | x-demon | p3nguin, what i put inside [] |
| 20:17.23 | eliel | p3nguin: i write faster cat /etc/asterisk/asterisk.conf |grep something, i am used to.. :D |
| 20:17.38 | eliel | but you are right |
| 20:17.43 | p3nguin | You can't write more characters is a less amount of time. |
| 20:20.22 | chuckf | but p3nguin the way you typed it won't work anyway |
| 20:20.29 | chuckf | :) |
| 20:21.09 | eliel | lol |
| 20:22.34 | [TK]D-Fender | checout time, BBIAB |
| 20:24.39 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 20:29.07 | *** join/#asterisk cusco (~trilili@213.141.21.122) |
| 20:29.10 | cusco | hi |
| 20:30.00 | cusco | Im trying to use SMS() app, but for some reason it fails to go trough, call is hangup from telco I guess... could ssomeone take a look at the span debug? --> http://paste.debian.net/78555/ |
| 20:34.29 | rustyclarkson | Hello, I want multiple queues to have their own unique MOH and then when the call is answered by an agent, if that agents puts the caller on hold, i would like a secondary MOH class to play dependent on the queue they've come from. I'm thinking the only way to do this is using the macro option in the Queue() command which I can use to Set(CHANNEL(musicclass)=...) when the call is answered. Does this sound about right or is there a better way? |
| 20:35.18 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-67-77-81.home.otenet.gr) |
| 20:35.39 | cusco | <PROTECTED> |
| 20:35.45 | cusco | that should be the cause?! |
| 20:39.35 | pabelanger | rustyclarkson: Sounds right |
| 20:42.14 | paulc | Meetme: I have a SIP phone connected to a Meetme conf, listen only. I drop a call file that connects to that meetme via a Local channel, then connects to another Local channel that uses Playback to insert audio into that conference. DAHDI, no hardware. The audio stops after a while (could be seconds, could be minutes) but the channels are all still up/active. Any ideas what's going on? |
| 20:48.43 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 20:50.28 | x-demon | there is so much to learn about asterisk |
| 20:50.47 | x-demon | at least i now know basics and i have personal pbx |
| 21:00.13 | *** join/#asterisk hfb (~hfb@pool-98-112-239-44.lsanca.dsl-w.verizon.net) |
| 21:00.57 | radic | what's chanmode c and r here? |
| 21:01.25 | t_dot_zilla | what is the default attended transfer key? |
| 21:04.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:09.31 | rustyclarkson | t_dot_zilla: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 21:09.39 | t_dot_zilla | thanks |
| 21:10.56 | [TK]D-Fender | Familiar looking link |
| 21:11.11 | [TK]D-Fender | Almost Like I provided it almost a half a dozen times before... |
| 21:12.37 | *** join/#asterisk jeffrey (~zabbix@unaffiliated/Jeffrey) |
| 21:12.40 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 21:12.50 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:13.09 | [sr] | the XML phonebook the SIP phones use, it's a standard or each trademark has its own format? |
| 21:13.20 | jeffrey | so my transfer buttons on my hardphone do not work when I use IAX2 instead of SIP |
| 21:13.24 | jeffrey | any clues? |
| 21:14.48 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 21:15.57 | [TK]D-Fender | [17:13]<[sr]>the XML phonebook the SIP phones use, it's a standard or each trademark has its own format? <- every phone that even lets you store one, or access it from outside the phone has its own way. No standards, many not even XML-like |
| 21:15.57 | [TK]D-Fender | jeffrey: Your hardphone supports SIP & IAX2? |
| 21:15.57 | jeffrey | yes |
| 21:16.02 | [TK]D-Fender | jeffWhich? |
| 21:16.10 | jeffrey | Citel C-4110 |
| 21:16.13 | [TK]D-Fender | smells ATCOM crap |
| 21:16.36 | [sr] | [TK]D-Fender: i'd like to have some example, i dont see that on my phone manual |
| 21:16.53 | [sr] | [TK]D-Fender: or could i just make a xml file with two fields, on for name and other for number ? |
| 21:17.07 | [TK]D-Fender | jeffrey: It should. Check your manuals |
| 21:17.14 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 21:17.18 | p3nguin | chuckf: I just copied the information that he offered to the other person, but I used proper syntax. Any problem with that, take it up with a more appropriate complaint department. |
| 21:17.30 | [TK]D-Fender | jeffrey: And contact the manufacturer. Noone I know would come within 10' of one |
| 21:17.52 | [TK]D-Fender | [sr]: And you haven't told me what you're USING <------- |
| 21:17.55 | jeffrey | yeah, unfortunately i don't make the purchasing decisions |
| 21:17.56 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 21:18.32 | [sr] | [TK]D-Fender: sorry, will use remote url |
| 21:25.14 | *** join/#asterisk cusco (~trilili@213.141.21.122) |
| 21:25.31 | *** join/#asterisk thevoke (michiel@future.as3322.net) |
| 21:26.20 | thevoke | greetings, i'm working with asterisk manager api using http, and for some reasons i cannot get the variables in asterisk 1.6, do I need a special way of putting the global variables to make them visible? |
| 21:31.24 | [TK]D-Fender | thevoke: Show us how you're setting them, and how you're trying to retrieve them |
| 21:31.27 | [TK]D-Fender | ~pb |
| 21:31.28 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 21:31.29 | [TK]D-Fender | ^^^^ |
| 21:32.16 | *** join/#asterisk miamiseb (~deigo@208.76.35.132) |
| 21:34.36 | thevoke | ok |
| 21:34.45 | thevoke | i'm trying quite a few methods now |
| 21:34.51 | thevoke | exten => 28030450,3,Set(lang=FR,g) |
| 21:35.06 | miamiseb | Having a problem on cisco 7960 that won't register. I double checked the line name/auth name/password and I'm still getting 401 unauthorized. |
| 21:35.14 | miamiseb | sip debugs are at http://pastebin.com/3KKzgLdn |
| 21:35.21 | thevoke | and i'm reading it trough the asterisk AstConMan .net feature |
| 21:36.02 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 21:36.12 | *** part/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 21:36.14 | miamiseb | None cisco devices (SPAs, polycom, aastra) seem to work fine. |
| 21:36.21 | miamiseb | s/None/non/ |
| 21:38.10 | miamiseb | tried toggling qualify on and off. |
| 21:39.30 | miamiseb | no way to actually see what it's sending as the password to match it up? |
| 21:39.31 | [TK]D-Fender | thevoke: Not the right way. GLOBAL() |
| 21:39.47 | *** join/#asterisk sekil (~sekil@78.24.111.218) |
| 21:39.53 | thevoke | Set(GLOBAL(LANG=FR)) |
| 21:39.54 | thevoke | like this ? |
| 21:40.22 | [TK]D-Fender | thevoke: Close, but no |
| 21:40.37 | thevoke | how then? ;) |
| 21:41.02 | [TK]D-Fender | thevoke: you don't have 2 things on either side of an "='. Go read up on your FUNCTION basics |
| 21:41.49 | miamiseb | http://www.the-asterisk-book.com/unstable/funktionen-global.html |
| 21:41.59 | cusco | Im trying to use SMS() app, but for some reason it fails to go trough, call is hangup from telco I guess... could ssomeone take a look at the span debug? --> http://paste.debian.net/78555/ |
| 21:42.05 | cusco | <PROTECTED> |
| 21:42.11 | cusco | that would be the cause? |
| 21:42.17 | miamiseb | according to ^ it would be Set(GLOBAL(LANG)=FR) |
| 21:45.47 | thevoke | thats what i tried, however no variables in the channel or the linked channel |
| 21:46.52 | cusco | another question... while in queue, I hear the MOH if I produce noise on the microphone, if I stop producing noise on the microphone I stop hear MOH also... |
| 21:47.05 | cusco | like asterisk is only sending me audio if he gets audio from me too |
| 21:47.09 | cusco | how can I make it work? |
| 21:52.34 | miamiseb | cusco: 1.4 or 1.6? |
| 21:52.58 | *** join/#asterisk zeeesh (zeeesh@119.154.61.93) |
| 21:53.08 | zeeesh | hello everybody |
| 21:53.27 | miamiseb | https://issues.asterisk.org/view.php?id=5374 |
| 21:53.41 | [TK]D-Fender | [17:42]<miamiseb>according to ^ it would be Set(GLOBAL(LANG)=FR) <- Case errors |
| 21:54.46 | miamiseb | the caps for the variable name or what? A string should be able to be either case, and global is capitalized in the doc I was referring to. |
| 21:55.49 | *** join/#asterisk JAMMAN2110 (~James@unaffiliated/jamman2110) |
| 21:56.57 | cusco | miamiseb: .16 |
| 21:56.59 | cusco | 1.6 |
| 21:57.39 | zeeesh | anybody has any idea about this feature. "call center setup, a person calling to call center, agents attend the call, supervisor is listening both conversation, is there any feature, that if supervisor give some instructions to his agent then only agent able to hear supervisor voice and the person who is calling at support center unable to hear supervisor voice"? |
| 22:00.00 | [TK]D-Fender | zeeesh: chanspy <- |
| 22:00.24 | zeeesh | is that possible with asterisk ? |
| 22:01.05 | spenguin[work] | zeeesh: ofcourse! |
| 22:01.16 | zeeesh | oooh great |
| 22:01.43 | zeeesh | thanx bros... let me check |
| 22:03.22 | *** join/#asterisk MiserySoft (~Lee@nat66.mia.three.co.uk) |
| 22:05.08 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:12.25 | *** join/#asterisk Guest24362 (mw3@88.151.97.220) |
| 22:14.19 | *** join/#asterisk kotp (~vgoff@96.2.187.67) |
| 22:17.54 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 22:18.21 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 22:19.09 | citywok | I frequently have ghost calls in MeetMe that have been running for a long time even though the channel is gone. Any idea why this is happening? |
| 22:19.51 | citywok | Sometimes the next time a person makes a call from their desk phone rather than make that phone call, they get put back in the conference that their prior call was stuck on. Hanging up and redialing repeats the same thing. only way to get rid of it is soft hangup the sip channel. |
| 22:21.50 | [TK]D-Fender | citywok: * begs to differ |
| 22:22.20 | citywok | ? |
| 22:29.25 | *** join/#asterisk Blackgibson (~inconnu@S01060015e912f559.vs.shawcable.net) |
| 22:30.17 | *** join/#asterisk fas3r (~fas3r@90.25.broadband12.iol.cz) |
| 22:30.19 | fas3r | hello |
| 22:30.36 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 22:30.51 | boodu | hello |
| 22:35.46 | fas3r | i try to use my provider sip account services with asterisk. this is my conf file : http://pastebin.com/VDi8y7WR |
| 22:36.51 | fas3r | xlite register on asterisk with no problem but when i try to call outside i got this message : [Jun 24 00:29:02] NOTICE[17645]: chan_sip.c:14441 handle_request_invite: Call from 'fas3r' to extension '9555XXXXX' rejected because extension not found. |
| 22:37.05 | *** join/#asterisk kotp (~vgoff@96.2.187.67) |
| 22:37.06 | fas3r | if you can help please ... |
| 22:37.29 | *** join/#asterisk WintermeW (~clement@78.251.244.244) |
| 22:39.25 | *** join/#asterisk seanjohn (~seanjohn@ns1.sheltoncomputers.com) |
| 22:39.37 | WintermeW | hi guys...well the channel is not very suited for my question but i can't find find any help. i need some people using asterisk-flite to do tts with their SIP servers , in order to know if flite supports SAPI 4/5 voices |
| 22:40.05 | seanjohn | i'm using tos=0x18 in my sip.conf; now asterisk is complaining it's deprecated. What should I be using? |
| 22:43.16 | Chainsaw | seanjohn: 42! |
| 22:43.29 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:43.35 | Chainsaw | seanjohn: On a more serious note: lowdelay, throughput, reliability, mincost or none |
| 22:46.09 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 22:46.44 | *** join/#asterisk fas3r (~fas3r@90.25.broadband12.iol.cz) |
| 22:50.07 | seanjohn | Chainsaw: I don't get what you mean |
| 22:50.24 | seanjohn | i Know I don't just type lowdelay in sip.conf |
| 22:51.22 | Chainsaw | seanjohn: The new values are those textual ones. lowdelay, throughput, realibility, mincost or none. Specifying the bitfield by hand is what is deprecated. |
| 22:51.28 | Chainsaw | seanjohn: And 42 is just a number I like. |
| 22:51.46 | seanjohn | tos=lowdelay ?? |
| 22:53.47 | Blackgibson | I have a Linksys SPA3102 that is trying to register with the wrong extension. If im reading the log correctly, it is trying to use ext 100, which does not exist. how can I change that? |
| 22:54.25 | Chainsaw | seanjohn: That sounds appropriate, yes. |
| 22:54.43 | seanjohn | didn't work, the entire tos= is deprecated |
| 22:55.40 | seanjohn | tos_sip=cs3 ; Sets TOS for SIP packets. |
| 22:55.41 | seanjohn | <PROTECTED> |
| 22:55.41 | seanjohn | <PROTECTED> |
| 22:57.53 | seanjohn | those worked; I don't use h323 or any video |
| 22:58.00 | Chainsaw | seanjohn: Ah, okay. |
| 22:58.37 | seanjohn | i thought tos= was working for the last 4 years |
| 22:58.40 | seanjohn | lol |
| 22:59.10 | seanjohn | i had the firewall marking the packets to be prioritized but asterisk wasn't marking them |
| 23:03.00 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 23:03.17 | Chainsaw | seanjohn: I hope your performance increases :) |
| 23:25.17 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 23:27.24 | sputnick | Hi again. I use Asterisk 1.4.21.2 on Debian Lenny in a remote dedicated server with only SIP protocol, and ekiga3 as SIP client on my box@home with forwarded UDP ports 8000,5000-5100 with asterisk realms. I can call anybody with french phone or mobile phones via "freephonie", I listen to him, but I cannot talk to him. Any hint ? |
| 23:30.52 | idespinner | sputnick, check localnet, externip in sip.conf.... |
| 23:33.04 | sputnick | thanks idespinner, I take a look |
| 23:38.38 | sputnick | idespinner: but I don't have any local non routables ip/vlan on my dedicated server. Only 1 interface with external IP. |
| 23:51.47 | sputnick | idespinner: I've tested with my local IP range ( box@home ) for "localnet=192.168.0.0/255.255.255.0" and my public adress for "externip=xx.xx.xx.xx" and test with "nat=yes" or "nat=no" but now I can't communicate on both sides. |
| 23:58.16 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 23:58.32 | a1fa | hey, whats a good hosted pbx that can take your number and provides iax or sip trunk? |
| 23:59.18 | sputnick | brb |
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