| 00:09.15 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 00:13.44 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
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| 01:11.07 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 01:11.29 | volga629 | Is any body have experience with res ldap ? |
| 01:14.03 | volga629 | ERROR[25003]: res_config_ldap.c:1328 update_ldap: Couldn't modify 'name'='', dn:uid=,ou=,dc=,dc= because Object class violation |
| 01:33.56 | *** join/#asterisk hfb (~hfb@cpe-98-151-249-95.socal.res.rr.com) |
| 01:34.31 | volga629 | by default which conttext use for mailbox ? |
| 02:36.24 | *** join/#asterisk adeel (~adeel@72.53.65.233) |
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| 04:08.29 | volga629 | I am getting http://fpaste.org/QntK/ |
| 04:08.41 | volga629 | this warning ? |
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| 06:06.02 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:06.38 | ziz212 | hi there, I am trying to connect google voice to my asteisk server. I have done config according to "https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google" and I need to know how to check the google account registration status in asterisk box? Is there any way in asterisk prompt ? Pls let help me for this. |
| 06:17.03 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 06:17.27 | siva4080 | Hi all, I'm trying to implement automated phone calls to few phone numbers (Given some text, using Text to Speech). Will Asterisk helps me for this? |
| 06:18.38 | ziz212 | Friends, can anybody help me for my question pls. |
| 06:19.24 | kaldemar | siva4080: it will help you with making the calls, but you'll most likely need some third party application to do the TTS. |
| 06:20.40 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 06:27.19 | ziz212 | Hi friends, any one can help me for my problem? |
| 06:27.32 | ChannelZ | ~ask |
| 06:27.32 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 06:29.27 | *** join/#asterisk oyugik (~oyugik@41.212.110.90) |
| 06:31.57 | ziz212 | Sorry for being problematic, I know you all good people and doing this voluntarily work with your busy time while doing all other important work schedule. I have asked my question before. Here is it again.. --> I am trying to connect google voice to my asteisk server. I have done config according to "https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google" and I need to know how to... |
| 06:31.58 | ziz212 | ...check the google talk registration status in asterisk box? Is there any way in asterisk prompt ? Pls let help me for this. |
| 06:34.07 | ChannelZ | jabber show connections |
| 06:35.50 | ziz212 | Thanks for the information. |
| 06:38.09 | oyugik | Hey guys does anyone know how to configure sip to make outbound calls |
| 06:38.22 | oyugik | using and fxo module ... |
| 06:38.56 | ChannelZ | those are two different things |
| 06:41.14 | ChannelZ | (and it's also a vague question) |
| 06:45.25 | *** join/#asterisk sarthor (~shabakah@unaffiliated/sarthor) |
| 06:46.13 | sarthor | Hi, Can we use a normal analog home telephone set with asterisk server or pbx? |
| 06:46.38 | ChannelZ | With proper equipment, yes |
| 06:47.36 | sarthor | ChannelZ, what equipment it will need? Is the equipment costly? can you please name that equipment and is that available easily? |
| 06:49.26 | ChannelZ | Depends on how many lines. The cheapest ATA if you just have one phone line and want to start playing around iis the SPA3102 |
| 06:50.17 | oyugik | @ChannelZ Hi, what I mean is how do you configure a softphone to make outbound calls.Sorry for the unclear question. |
| 06:51.12 | ChannelZ | oyugik: Have you configured a sip peer for the softphone in sip.conf? |
| 06:51.34 | kaldemar | oyugik: so you have a soft phone that you want to connect to asterisk and make outbound calls using the FXO? |
| 06:53.50 | LiuYan | sarthor: you may need a digium or compatible analog card with at least 1 FXO port |
| 06:55.59 | sarthor | LiuYan, for example , in a building where there are about 50 rooms and each room have analog telephone, for example i have a pbx server configured and I want to use that all analog 50 telephones with this pbx server as extentions. So Is that possible?? |
| 06:56.57 | oyugik | correct. I have a sip phone, that is already connected to asterisk. I have an outbound gsm line that I want to use to make calls from the softphone. |
| 06:56.59 | sarthor | LiuYan, because I think that the extention will need an IP and I do not think that analog telephone support IP settings. |
| 06:58.22 | kaldemar | oyugik: what is the "gsm line"? |
| 06:59.20 | *** join/#asterisk siva4080 (~siva@27.97.196.145) |
| 06:59.28 | ChannelZ | sarthor: so you want to use existing analog phones... in which case you need something with lots of FXS ports |
| 06:59.52 | LiuYan | sarthor: if you want use your analog phones, you need 50 FXS ports on analog card; and if you want to dial out, you need some(maybe less than 50) FXO port(s) too |
| 07:00.55 | *** join/#asterisk shadebob (~shadebob@41.143.13.26) |
| 07:01.21 | ChannelZ | unless as you say you already have them on an old PBX, you might be able to interface the two (it and Asterisk) via a T1 card or something, and essentially use the old PBX as a giant ATA (which is most of the expensive part) |
| 07:02.25 | oyugik | gsm line = fxo port |
| 07:03.42 | kaldemar | oyugik: FXO has really nothing to do with GSM. what is the FXO? an internal card? some external device? |
| 07:09.42 | oyugik | an analog interface card has both fxo and fxo ports right? |
| 07:10.17 | *** join/#asterisk siva4080 (~siva@27.97.196.145) |
| 07:11.02 | *** join/#asterisk siva4080 (~siva@27.97.196.145) |
| 07:11.50 | oyugik | the fxo can be used as internal extensions, and the fxs are for connecting to POTS line from PSTN |
| 07:12.14 | LiuYan | reverse |
| 07:13.33 | LiuYan | http://www1.digium.com/en/products/telephony-cards/analog |
| 07:15.54 | kaldemar | oyugik: an interface card does not necessarily have both FXO and FXS. what do you have? |
| 07:17.26 | oyugik | I have a tdm400p with 2fxs and 2fxo |
| 07:18.06 | kaldemar | finally. |
| 07:18.19 | kaldemar | have you configured the channels in DAHDI and asterisk? |
| 07:19.35 | oyugik | yes, I have configured the card in DADHI and it works well :-) |
| 07:20.55 | oyugik | I have also created a softphone using 3cx, the softphone extensions are communication very well ... but I cant make outbound calls outside the network |
| 07:21.00 | kaldemar | do you have an extension that dials with an FXO? |
| 07:21.46 | kaldemar | extension as in an exten in asterisk's dialplan... |
| 07:21.47 | oyugik | yes .. |
| 07:21.57 | kaldemar | so what is the problem? |
| 07:23.45 | oyugik | I want to configure softphones to make outbound calls, right now I can only dial extensions, I need to make outbound pstn calls, what should I configure to achieve this |
| 07:24.57 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:25.41 | kaldemar | oyugik: dialplan. |
| 07:26.54 | kaldemar | oyugik: show what you have now in sip.conf and extensions.conf. |
| 07:27.19 | oyugik | thank caldemar, how should I configure the dialplan? |
| 07:27.30 | oyugik | okay letme paste them in abit |
| 07:30.22 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 07:30.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:35.09 | *** join/#asterisk angryuser_laptop (~angryuser@187-244-135-95.pool.ukrtel.net) |
| 07:51.17 | *** join/#asterisk siva4080 (~siva@27.97.196.145) |
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| 07:53.22 | ziz212 | hi there, my google voice and asterisk is done according to the wiki side and status says " [asterisk] xxxyyyzzz@gmail.com - Disconnected" |
| 07:53.38 | ziz212 | What will be the problem |
| 07:53.49 | ziz212 | Pls help me for this. |
| 07:58.08 | ziz212 | my conf files are http://pastebin.ca/2155086 |
| 07:58.27 | ziz212 | pls help |
| 08:14.02 | *** join/#asterisk shadebob (~shadebob@adsl196-45-235-206-196.adsl196-8.iam.net.ma) |
| 08:21.02 | *** join/#asterisk arekm (matrix157@pld-linux/arekm) |
| 08:21.23 | arekm | hi, how to make asterisk execute next dialplan command regardless if Dial() succeeded or failed? |
| 08:22.09 | kaldemar | arekm: see options g and F in "core show application Dial" |
| 08:26.11 | arekm | kaldemar: seems to not work if caller hangs up before called party answers |
| 08:27.51 | arekm | what I'm actually trying to do is to set some action based on which device will be selected when using round-robin calling Dial(Zap/r3) for example |
| 08:28.54 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 08:31.05 | *** join/#asterisk Vince-0 (c4d7bcbd@gateway/web/freenode/ip.196.215.188.189) |
| 08:32.29 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 08:33.13 | arekm | looks like not possible to do - no such dial flag :/ |
| 08:38.33 | arekm | other question then. Two asterisk connected via SIP. One calls via the second one, second one dials to outside. Now second one gets: " status is 'CHANUNAVAIL'" but first gets "channel... is circuir busy" and "status is 'CONGESTION'". Why first doesn't get CHANUNAVAIL? |
| 08:39.14 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yborjafhohkmfulo) |
| 08:40.52 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
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| 08:52.53 | *** join/#asterisk din3sh (~din3sh@41.212.201.8) |
| 08:53.02 | din3sh | mrning all |
| 08:53.17 | oyugik | morning dinsh .. |
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| 08:59.20 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 08:59.21 | mirelab | gm |
| 09:01.11 | *** join/#asterisk shadebob (~shadebob@196.206.235.45) |
| 09:15.03 | *** part/#asterisk oyugik (~oyugik@41.212.110.90) |
| 09:17.42 | *** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com) |
| 09:17.53 | Ice_Strike | Hello |
| 09:19.22 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 09:21.17 | din3sh | Ice_Strike: have you been able to fix your dinstar prob? |
| 09:23.50 | Ice_Strike | din3sh Yes I think, help from a user here. |
| 09:32.57 | din3sh | what was actually the problem? |
| 09:37.54 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 09:46.08 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 09:47.42 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85f8.bcn.adamo.es) |
| 09:49.02 | *** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell) |
| 09:49.18 | nickfennell | hi all. trying to 'channel request hangup <CHANNEL>' but it's not taking effect |
| 09:49.27 | nickfennell | channel remains in Up state |
| 09:54.22 | nickfennell | Call is stale I think |
| 10:01.19 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 10:08.28 | Ice_Strike | din3sh kinda complicated, I need to define prefix to assign spefic sim port |
| 10:26.55 | din3sh | uhu |
| 10:41.27 | *** join/#asterisk wonderworld (~ww@dsdf-4db55b7f.pool.mediaWays.net) |
| 10:41.36 | *** join/#asterisk RZero (~RZero@gateway-hq-uk.oxygen8.com) |
| 10:42.23 | RZero | Hi Guys quick Dahdi question, does a single call use 2 channels audio in/audio out ? |
| 10:44.12 | kaldemar | RZero: no. |
| 10:44.34 | RZero | it should be single channel then ? |
| 10:47.12 | kaldemar | what kind of a call? |
| 10:48.15 | kaldemar | a single channel is used between asterisk and an endpoint in the call. if you get the call in via a DAHDI channel and it goes out of asterisk using another DAHDI channel, then the call uses two. |
| 10:48.44 | RZero | ahh so if asterisk is just handling the call then it will use single channel, thanks |
| 10:49.27 | RZero | I just thought it was two channels like SIP calls :D |
| 10:50.13 | *** join/#asterisk lokk911 (a14834f1@gateway/web/freenode/ip.161.72.52.241) |
| 10:50.18 | lokk911 | hi all |
| 10:51.15 | lokk911 | i have a problema im using asterisknow and i configure one sip telephone, all was oke, all works, but i have interferences on my phone |
| 10:51.25 | lokk911 | just like echo sound |
| 10:51.37 | lokk911 | it disturb my message |
| 10:51.43 | lokk911 | how can i solved? |
| 10:51.59 | kaldemar | RZero: if by channel you mean a channel in asterisk, SIP works in the same way. |
| 10:53.32 | RZero | but the audio is pasted over two ports ? |
| 10:53.44 | kaldemar | not to a single endpoint. |
| 10:53.47 | lokk911 | My phone is heard with interference, use AsteriskNOW, how I can fix it? |
| 10:53.48 | *** join/#asterisk angryuser_laptop (~angryuser@187-244-135-95.pool.ukrtel.net) |
| 10:55.16 | lokk911 | I tried changing the codecs but does not work, dont know what else to try, help please |
| 10:58.11 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 10:58.25 | *** part/#asterisk lokk911 (a14834f1@gateway/web/freenode/ip.161.72.52.241) |
| 11:03.19 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 11:03.21 | cusco | hello folks |
| 11:03.52 | cusco | I may have issue and my not sure of its origins, may be asterisk, or network |
| 11:04.07 | cusco | basically I want to register a sip account at our ISP |
| 11:04.59 | cusco | our ISP provided in our router, port2 with a new wan connection, just for this SIP communication |
| 11:05.37 | cusco | now, I have a router (mikrotik) where I set a mangle rule that states all traffic with src ip 10.100.101.7 (asterisk) will have as a gateway that 2nd WAN |
| 11:05.48 | cusco | and asterisk will not register.. (request sent) |
| 11:06.08 | cusco | but, if using the same rule but on the IP of my local windows machine |
| 11:06.14 | cusco | x-lite registers just fine |
| 11:06.33 | cusco | now, if I connect asterisk directly to that port2 to use only wan2, it registers |
| 11:06.50 | cusco | I would like to understand the difference from x-lite and asterisk regarding this registation |
| 11:23.58 | cusco | nevermind fixed that |
| 11:24.04 | cusco | I have another question tho |
| 11:28.57 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 11:33.57 | Ice_Strike | DWG2000-8G support RFC2833 and SIGNAL two ways. |
| 11:34.01 | Ice_Strike | What is SIGNAL? |
| 12:05.45 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:09.28 | CommaCrazy | a two way communication |
| 12:10.40 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:11.59 | CommaCrazy | if it is trough gsw gateway it can be two ways byt calling back the number and using dtmf for further features |
| 12:12.20 | CommaCrazy | I'm guessing that that's what they meant |
| 12:12.46 | CommaCrazy | or if you show me where it is written like that I could interpret it |
| 12:14.28 | CommaCrazy | yup for that device it's what' I first said |
| 12:18.35 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
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| 13:29.18 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 13:42.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 13:58.32 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:02.20 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:02.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:11.09 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 14:11.26 | eXcAliBuR | am i crazying for having the loud speaker of my phone play hold music from asterisk |
| 14:11.34 | eXcAliBuR | it's rather soothing |
| 14:11.46 | eXcAliBuR | an no one can call me |
| 14:11.46 | eXcAliBuR | :] |
| 14:12.07 | *** join/#asterisk bencc (~user@bzq-84-111-74-191.red.bezeqint.net) |
| 14:12.10 | *** part/#asterisk bencc (~user@bzq-84-111-74-191.red.bezeqint.net) |
| 14:12.10 | eXcAliBuR | i've been playing it forr over 2 hours |
| 14:12.38 | *** join/#asterisk shadebob (~shadebob@adsl196-45-235-206-196.adsl196-8.iam.net.ma) |
| 14:13.55 | WIMPy | is getting historic now. |
| 14:14.18 | WIMPy | That was a time when you just put your handset aside and noone could clal you... |
| 14:17.50 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 14:18.11 | eXcAliBuR | :P |
| 14:18.20 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 14:18.30 | eXcAliBuR | i could just goto sirius and listen to music there |
| 14:18.32 | eXcAliBuR | but meh |
| 14:18.42 | eXcAliBuR | i wanna see how long before asterisk dies |
| 14:18.47 | eXcAliBuR | cuts me off |
| 14:18.51 | eXcAliBuR | i don't think it will |
| 14:18.51 | eXcAliBuR | :[ |
| 14:23.43 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 14:24.12 | coppice | sirius? that's a bloody long way to go for a concert |
| 14:24.16 | *** join/#asterisk samba35 (~shrikant@unaffiliated/samba35) |
| 14:24.51 | samba35 | i am new to asterisk i have installed asterisk on ubuntu now how do i start voip call |
| 14:25.02 | WIMPy | I guess that's somewhere near the Kissy Klub. |
| 14:25.18 | WIMPy | samba35: Start with the ... |
| 14:25.23 | WIMPy | ~book |
| 14:25.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:26.34 | samba35 | ok thanks |
| 14:26.40 | volga629 | I am getting this error on res ldap and voice mail http://fpaste.org/7Bbk/ |
| 14:26.48 | volga629 | any help thank you |
| 14:27.10 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 14:29.59 | volga629 | and this error http://fpaste.org/Tdqe/ that asterisk trying modify ldap |
| 14:33.58 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 14:37.31 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 14:37.40 | kami | Hello #asterisk. |
| 14:37.59 | WIMPy | Hi kami |
| 14:38.01 | *** join/#asterisk angryuser_laptop (~angryuser@252-95-135-95.pool.ukrtel.net) |
| 14:38.42 | eXcAliBuR | hi kaii |
| 14:38.45 | eXcAliBuR | and kami |
| 14:38.48 | eXcAliBuR | :P |
| 14:39.33 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 14:41.47 | kami | WIMPy: I compiled a stock 3.1.10 kernel and have complete silence when calling from ISDN or from SIP. |
| 14:41.56 | kami | So, things got worse for me. |
| 14:43.22 | kami | WIMPy: I'll write a message to the isdn4linux mailing list and wait for help. |
| 14:44.32 | WIMPy | Strange. |
| 14:45.18 | WIMPy | That's the version from kernel.org then patched for the x-tensions thing? |
| 14:46.00 | samba35 | WIMPy, i am in india ,to make voip call from my ubuntu what i suppose to do |
| 14:47.23 | WIMPy | samba35: That question is far too general. Start with the book. |
| 14:47.30 | kami | WIMPy: yes. Patched with the vendor and device id lines for X-tension .. |
| 14:47.40 | samba35 | yes i am reading that book |
| 14:47.56 | samba35 | ok i will come back |
| 14:47.58 | kami | The device is recognised without specific driver loading. |
| 14:48.00 | samba35 | thanks |
| 14:48.32 | WIMPy | Doesn;t seem likely that it's about 3.1.10 when 3.1.7 and 3.4 are ok. |
| 14:50.25 | WIMPy | Hmm. The DSP module is loaded? |
| 14:50.32 | WIMPy | aJust to make sure. |
| 14:50.44 | *** join/#asterisk kami` (~user@unaffiliated/kami-) |
| 14:51.57 | WIMPy | Hmm. The DSP module is loaded? |
| 14:51.59 | WIMPy | Just to make sure. |
| 14:52.36 | kami` | Wait, I'll check. |
| 14:53.15 | kami | Oh, I rebooted into 2.6.32. Will reboot and check. |
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| 15:06.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 17:02.10 | Sjors | hmm |
| 17:02.50 | Sjors | when I call from Telephone (a Mac OS X SIP client) to SIPdroid (an Android client) over my Asterisk server, the receiver (Sipdroid) already plays my voice even before it picks up |
| 17:03.30 | Sjors | as in, I hear a ringing tone and talk at the same time, the receiver plays its ringtone and my voice simultaneously |
| 17:03.32 | Sjors | does anybody know why this is? |
| 17:12.22 | leifmadsen | Sjors: sounds like early media |
| 17:13.44 | leifmadsen | Sjors: try adding 'r' to the Dial() command |
| 17:13.59 | leifmadsen | which from what I've seen on a bug report, will force asterisk to not setup early media |
| 17:14.29 | leifmadsen | but I've never seen that actually happen before so... umm... no idea :) |
| 17:15.10 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 17:15.12 | WIMPy | IIRC I had two simultaneous ring back tones some long time ago. |
| 17:16.16 | Sjors | leifmadsen: thanks, I'll try that |
| 18:08.36 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 18:13.38 | *** join/#asterisk dandate2 (~dan@180.190.232.184) |
| 18:13.54 | dandate2 | does high verbosity use CPU even if i'm not in the CLI with terminal? |
| 18:14.02 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 18:18.57 | *** join/#asterisk ccesario (~ccesario@187.17.166.162) |
| 18:19.32 | ChannelZ | probably not any worth calculating |
| 18:20.27 | leifmadsen | dandate2: technically, yes, amount? I would suspect trivial |
| 18:20.54 | leifmadsen | you'd mostly notice it via logging if you enabled verbose logging in logger.conf |
| 18:41.27 | *** join/#asterisk eicto (~eicto@eicto.broker.freenet6.net) |
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| 18:46.45 | *** join/#asterisk Keisuke (~Keisuke@modemcable201.185-200-24.mc.videotron.ca) |
| 18:47.21 | *** join/#asterisk reber (~reber@tsm83-4-78-232-65-13.fbx.proxad.net) |
| 18:47.58 | Keisuke | hi all, I want to create a route in asterisk, to join dahdi/1 when i call with dahdi/4, how can I do that ? |
| 18:48.09 | Keisuke | for every number |
| 18:49.48 | pabelanger | place exten => 12345,1,Dial(DAHDI/4/9876) in the context for DAHDI/1 |
| 18:50.45 | Keisuke | pabelanger: what is the 9876 after DAHDI/4/ ? |
| 18:51.19 | pabelanger | an example number to dial, could you just do same => _X.,1,Dial(${EXTEN}) |
| 18:51.29 | pabelanger | I don't know what dial patterns you want |
| 18:51.38 | pabelanger | s/same/exten/ |
| 18:51.43 | Keisuke | ok, every number for the pattern |
| 18:53.01 | [TK]D-Fender | Keisuke, Your description is vague as to what signalling is on those channels, and what you mean by "join". |
| 18:53.40 | Keisuke | ok, i'm french, so I have some trouble to find my words... |
| 18:55.10 | Keisuke | when i call a number from the dahdi/4 (phone connected to channel 4), i want to translate/join/forward to the dahdi/1 (channel 1, the land line) |
| 18:55.32 | pabelanger | FXS to FX0 |
| 18:55.38 | Keisuke | yes |
| 18:55.49 | pabelanger | do what I said |
| 18:55.58 | pabelanger | this is basic functionality for asterisk |
| 18:56.00 | pabelanger | ~book |
| 18:56.00 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:58.43 | Keisuke | ok, pabelanger, can you tell me, approximatly, where (which section) can I found on this book ? |
| 18:59.37 | Keisuke | on the section, Dialplan ? or Dahdi section ? |
| 19:00.00 | pabelanger | Keisuke: 6. Dialplan Basics |
| 19:00.06 | Keisuke | thx |
| 19:01.24 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 19:02.54 | [TK]D-Fender | Keisuke, Set the context for DAHDI/4. In that context make your exten patterns for the kinds of numbers you want it to handle and call the Dial app as appropriate to call out DAHDI/4 |
| 19:03.05 | [TK]D-Fender | errr 1 |
| 19:03.08 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 19:06.18 | *** join/#asterisk phanohanover (phanohanov@modemcable096.110-80-70.mc.videotron.ca) |
| 19:06.45 | phanohanover | I need a little help setting up by asterisk outbound call. Can someone help me? |
| 19:07.15 | phanohanover | anyone there? |
| 19:07.27 | pabelanger | ~ask |
| 19:07.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:08.04 | leifmadsen | phanohanover: use call files, originate on the CLI, Originate() dialplan application, or the Asterisk Manager Interface (AMI) |
| 19:08.12 | phanohanover | how can I fix this: "Forbidden" from '"asterisk" |
| 19:08.25 | leifmadsen | more information required |
| 19:08.34 | pabelanger | ~collectdebug |
| 19:08.35 | infobot | extra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 19:08.38 | pabelanger | phanohanover: ^ |
| 19:08.39 | leifmadsen | Forbidden usually mean incorrect username/password or somethign else |
| 19:09.14 | phanohanover | ok, leifmadsen, can you pm me for a minute? i am no good at IRC! |
| 19:09.21 | leifmadsen | phanohanover: I can not |
| 19:09.39 | leifmadsen | you can use this room as tool -- do not msg me directly please |
| 19:10.41 | phanohanover | ok well, here: I have to android setup has stations on my asterisk 1.8 using GUI and cli in debian. I can receive calls but when I dial out, I get this forbiden. But I can reach the other internal phones by extensions like 6001... |
| 19:10.55 | leifmadsen | sounds like you've not setup a route |
| 19:11.14 | leifmadsen | or you've not enabled the ability for an endpoint to access the context for outbound dialing |
| 19:11.19 | [TK]D-Fender | phanohanover, Show us the failed call at CLI with SIP DEBUG enabled. |
| 19:11.19 | [TK]D-Fender | ~pb |
| 19:11.20 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 19:11.21 | leifmadsen | you mentioned GUI as well, which can't be supported here |
| 19:11.21 | [TK]D-Fender | ^^^ |
| 19:12.00 | pabelanger | ~gui |
| 19:12.00 | infobot | gui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png. Of course Real Programmers use the command line interface. See cli |
| 19:12.26 | phanohanover | well, I have 3 SIP trunk with Iristel and they register ok. But it seems that when I dial out, it is not taking the right route...Has you say...but I am lost. I have 2 out outbound rules setup... |
| 19:12.55 | leifmadsen | registration doesn't get used for outbound calls -- it's a separate authentication when you place calls outbound |
| 19:13.04 | leifmadsen | registration is just to tell the other end where you are on the internet/network |
| 19:13.12 | phanohanover | ok, How can i send you a log? |
| 19:13.20 | leifmadsen | you can use a pastebin |
| 19:13.29 | [TK]D-Fender | ~pb |
| 19:13.29 | infobot | 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/; or install pastebinit with yum or aptitude. |
| 19:13.30 | [TK]D-Fender | ^^^^^ |
| 19:14.03 | phanohanover | ok, one second...i'll put this in pastebin |
| 19:19.55 | phanohanover | ok, I have submitted has phanohanover !!! this is great! |
| 19:20.51 | phanohanover | can you read it? |
| 19:21.32 | phanohanover | sorry: here is the link: http://pastebin.com/JaxbuWA6 |
| 19:22.49 | [TK]D-Fender | phanohanover, Improper NAT setup |
| 19:23.07 | [TK]D-Fender | ontact: <sip:asterisk@192.168.1.2:5060> |
| 19:23.09 | [TK]D-Fender | line 52 |
| 19:23.17 | [TK]D-Fender | ~sipnat |
| 19:23.18 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 19:23.48 | [TK]D-Fender | phanohanover, in 1.6+ canreinvite has been replaced by directmedia. |
| 19:24.35 | phanohanover | Sorry? how and where do I change that? in the users? I am on 1.8 |
| 19:24.58 | phanohanover | does that apply to me? sip.conf: nat, directmedia, externhost or externaddr, and localnet |
| 19:25.27 | phanohanover | ok...well hang on...I'll test it thanks guys... you are way too fast for me!!!! |
| 19:28.32 | phanohanover | tell me if I am right? should I use directmedia=yes |
| 19:28.45 | [TK]D-Fender | several seetings to make here |
| 19:28.51 | [TK]D-Fender | in your peers, in [general] etc |
| 19:29.02 | [TK]D-Fender | and for your next pastebin make sure to set verbose to 10 |
| 19:29.57 | phanohanover | can you be more explicite for me please? |
| 19:31.29 | *** join/#asterisk eicto (~eicto@144-71.dsl.aichyna.com) |
| 19:31.45 | [TK]D-Fender | "core set verbose 10" <--- |
| 19:32.46 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 19:33.57 | *** part/#asterisk Keisuke (~Keisuke@modemcable201.185-200-24.mc.videotron.ca) |
| 19:35.01 | phanohanover | http://pastebin.com/YLDDKy7F |
| 19:36.03 | *** join/#asterisk asilva (~asilva@gandalf.ai.unesp.br) |
| 19:36.34 | asilva | hello , does anyone know or have a good tutorial of Asterisk HA with heartbeat and peacemaker ? |
| 19:37.48 | phanohanover | i have change the directmedia to no... |
| 19:38.18 | WIMPy | asilva: I don't think those conditions go aling. |
| 19:38.42 | [TK]D-Fender | phanohanover, I did not say to disable SIP debug. |
| 19:38.53 | [TK]D-Fender | phanohanover, have both enabled and PB a new call |
| 19:38.56 | phanohanover | sorry!!!my mistake |
| 19:38.58 | asilva | WIMPy: what u mean ? |
| 19:39.31 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 19:41.18 | phanohanover | ok, it is up to date now, same url |
| 19:42.49 | phanohanover | try this one: http://pastebin.com/gxfLFSV6 |
| 19:42.59 | phanohanover | sorry for the previous, it did not went through |
| 19:45.59 | [TK]D-Fender | Contact: <sip:asterisk@192.168.1.2:5060> |
| 19:46.00 | pabelanger | phanohanover: you don't have your NAT setup properly, even if you fix the 403 forbidden you'll likely have audio issues |
| 19:46.07 | [TK]D-Fender | You stil have not fixed your trunk config |
| 19:46.22 | [TK]D-Fender | And I don't see a proper username being passed to them either |
| 19:46.35 | [TK]D-Fender | and for the next call, please show the comlpete call, not just partway through |
| 19:46.57 | [TK]D-Fender | Reliably Transmitting (NAT) to 209.167.234.109:5060: <- your provider is not behind NAT.... |
| 19:47.15 | [TK]D-Fender | Please read the guide again. |
| 19:47.59 | asilva | WIMPy: ? |
| 19:52.19 | phanohanover | here is a more complete log: http://pastebin.com/Tik02Exz |
| 19:53.04 | phanohanover | please tell me what and where to fix? |
| 19:54.02 | phanohanover | I am behind a router usinf dyndns for domain on a dynamic ip with ISP. My sip provider is Iristel. |
| 19:58.16 | phanohanover | I amm using asterisk-gui but I can edit with nano directly if it is simplier... |
| 20:01.51 | pabelanger | you have been told, start with ~sipnat |
| 20:02.18 | phanohanover | thank you so much for helping me D-fender...It is my work phone. We are switching over to VOIP from an old Norstar analog system! |
| 20:02.43 | [TK]D-Fender | phanohanover, Reliably Transmitting (NAT) to 209.167.234.109:5060: <- you need to set your trunk peer to NAT=NO |
| 20:02.55 | phanohanover | in what file? |
| 20:03.31 | [TK]D-Fender | Contact: <sip:asterisk@192.168.1.2:5060> <- and you still haven't fixed your other [general] settings. Your server is handing out its PRIVATE IP instead of the public one you are behind |
| 20:03.34 | [TK]D-Fender | SIP.CONF |
| 20:03.39 | [TK]D-Fender | this is all in the guide... |
| 20:03.55 | [TK]D-Fender | an I'm not seeing you showing me configs to match so that we can see if there is more that you have skipped. |
| 20:05.52 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 20:07.12 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 20:08.01 | phanohanover | here is an update after changer NAT=no http://pastebin.com/vEMSnJ9N |
| 20:10.37 | [TK]D-Fender | Reliably Transmitting (NAT) to 209.167.234.109:5060: <- still wrong |
| 20:10.56 | phanohanover | here is my sip.conf http://pastebin.com/gQLYM1vc |
| 20:11.52 | [TK]D-Fender | phanohanover, There is a ton of garbage in there. permanently remove all the commented out stuff and repaste |
| 20:12.09 | phanohanover | ok hangon |
| 20:13.11 | pabelanger | ;! Automatically generated configuration file |
| 20:13.13 | pabelanger | heh |
| 20:14.24 | [TK]D-Fender | phanohanover, and a side-line note : AsteriskGUI is a poor business decision for you to implement... |
| 20:19.48 | carrar | I second that |
| 20:20.01 | carrar | Motion passes |
| 20:22.43 | phanohanover | reviewed http://pastebin.com/gQLYM1vc |
| 20:23.26 | [TK]D-Fender | externhost = genest.homelinux.com |
| 20:23.31 | phanohanover | <PROTECTED> |
| 20:23.33 | [TK]D-Fender | Should be externaddr now IIRC |
| 20:23.49 | [TK]D-Fender | also you ned nat=yes under general... |
| 20:24.38 | [TK]D-Fender | And it doesn't matter if you're "a small business". AsteriskGUI hasn't had a real official maintainer in over 2 years now and you can DOUBLE to population of their chanell.. just by joining it. |
| 20:24.42 | [TK]D-Fender | #asterisk-gui |
| 20:24.54 | [TK]D-Fender | Anyway, it's checkout time here... hopefully you'll hit progress soon... |
| 20:24.58 | [TK]D-Fender | I'm off |
| 20:36.26 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
| 20:38.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:42.41 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 20:44.02 | phanohanover | can someone relay D-Fender? |
| 20:44.10 | phanohanover | I am stuck there??? |
| 20:46.46 | phanohanover | anyone? |
| 20:53.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:08.05 | *** join/#asterisk HyperNerdV2_ (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 21:09.08 | *** join/#asterisk powerunits (b6b1d653@gateway/web/freenode/ip.182.177.214.83) |
| 21:09.10 | phanohanover | exit |
| 21:09.18 | powerunits | hello team, |
| 21:09.35 | powerunits | please could any one can guide me regarding gtalk with asterisk. |
| 21:09.46 | powerunits | i have configure gtalk with asterisk. |
| 21:09.55 | powerunits | now i have 1 question. |
| 21:10.22 | powerunits | in gtalk configration i ve used 1 gmail account. |
| 21:11.06 | powerunits | now if some one know that plz let me know how many concurnt SIP calls i can make to gtalk? |
| 21:39.18 | *** join/#asterisk last1 (~last1@bas2-hull20-845448106.dsl.bell.ca) |
| 21:47.03 | last1 | when is dahdi supposed to be loaded ? at boot time ? |
| 21:47.07 | last1 | the kernel module |
| 21:49.54 | powerunits | ? |
| 21:50.17 | last1 | I can't seem to be able to start dahdi |
| 21:50.21 | last1 | I see the module in here: /lib/modules/2.6.18-308.4.1.el5/dahdi/dahdi.ko |
| 21:50.41 | last1 | and if I do modprobe /lib/modules/2.6.18-308.4.1.el5/dahdi/dahdi.ko it says: FATAL: Module /lib/modules/2.6.18_308.4.1.el5/dahdi/dahdi.ko not found. |
| 21:51.39 | WIMPy | That's the way modprobe works |
| 21:51.59 | WIMPy | Try 'modprobe dahdi'. |
| 21:52.15 | last1 | FATAL: Module dahdi not found. |
| 21:52.19 | *** join/#asterisk fubada (~fubada@ool-4573155d.dyn.optonline.net) |
| 21:52.22 | fubada | hi |
| 21:52.40 | fubada | can someone take a look at my confbridge menu attempt |
| 21:52.43 | fubada | http://pastie.org/private/yqvzat6fzy9s7tiedrt3fq |
| 21:53.00 | fubada | should there be more error checking? |
| 21:53.26 | WIMPy | last1: The that's not your current running kernel or you didn't install them correctly. 'depmod -a'? |
| 21:54.19 | last1 | WIMPy: that returns nothing |
| 21:54.44 | WIMPy | and modprobe still doesn;t work? |
| 21:54.58 | last1 | not sure if it matters or not... but this is on rackspace cloud virtual server |
| 21:55.12 | last1 | uname -a reports: Linux test 2.6.35.4-rscloud |
| 21:55.40 | WIMPy | So that's obviousely not the same vewrsion as dahdi was installed for. |
| 21:56.10 | last1 | indeed. but I just used yum install dahdi-linux |
| 21:56.17 | WIMPy | But if you don't have any dahdi hardware, you may not need it anyway. |
| 21:56.21 | [TK]D-Fender | fubada: We are looking at your dialplan.. not the attempt |
| 21:56.39 | last1 | I don't. But I recall I needed the dahdi driver for some timing issues |
| 21:57.12 | WIMPy | Someone should really fix that. |
| 21:57.53 | last1 | is that an error on our end ? |
| 21:59.42 | fubada | [TK]D-Fender: well its working, but its my first time writing a menu like that |
| 22:00.14 | fubada | i just had a general question to see if what i was doing made any sense, and how could I check for invalid confbridge room numbers |
| 22:05.25 | [TK]D-Fender | fubada: Stop using Read() for an autoattendant and use WatExten() like normal and use *'s other standard extensions |
| 22:06.17 | fubada | thank you, ill need to do some reading |
| 22:10.42 | *** join/#asterisk danfromuk (~IceChat77@2.27.1.186) |
| 22:11.19 | danfromuk | Hello. I keep seeing this in the asterisk console: utils.c:1211 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 22:11.24 | danfromuk | Should I worry about it? |
| 22:12.09 | danfromuk | Actually, quick google found the answer. |
| 22:12.14 | danfromuk | thanks anyway |
| 22:12.21 | *** join/#asterisk shadebob (~shadebob@41.141.93.18) |
| 22:12.38 | shadebob | hi |
| 22:12.49 | WIMPy | lo |
| 22:19.11 | *** join/#asterisk imcdona (~imcdona@c-71-227-200-25.hsd1.wa.comcast.net) |
| 22:19.20 | leifmadsen | mid |
| 22:22.27 | *** join/#asterisk imcdona (~imcdona@c-71-227-200-25.hsd1.wa.comcast.net) |
| 22:22.55 | *** join/#asterisk imcdona (imcdona@2001:470:e92e:1:84c4:32b0:da1c:8df) |
| 22:23.59 | fubada | [TK]D-Fender: i followed your advice, can you tll me if this looks better http://pastie.org/private/jyczludoyv3xm1a93y6riw |
| 22:35.12 | [TK]D-Fender | exten => t,n.Hangup <- . |
| 22:35.26 | [TK]D-Fender | same => n,Playback(welcome) <- always background |
| 22:35.53 | [TK]D-Fender | the rest isn't too bad |
| 22:36.05 | fubada | can Background chain audio together? |
| 22:36.16 | fubada | file1,fiel2 |
| 22:36.34 | [TK]D-Fender | yes, and you can just do them on separate lines as well |
| 22:36.44 | [TK]D-Fender | Don't froget to set your TIMEOUT's <- |
| 22:36.57 | fubada | where is it best to set that |
| 22:43.05 | fubada | [TK]D-Fender: Can i sshow you my incoming? http://pastie.org/private/gifp37numthcxnrsfyl2wa |
| 22:43.17 | fubada | i added the entry starting with line #2 |
| 22:45.04 | fubada | is that a good way to map my 1888 number to my [conference] from the previous example? |
| 22:49.02 | *** join/#asterisk nix8n82 (~none@24.143.10.144) |
| 22:52.05 | [TK]D-Fender | sure |
| 23:24.05 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 23:33.41 | *** join/#asterisk albertoandrade (~albertoan@186.206.22.114) |
| 23:42.12 | thecardsmith | i've got this nagging nat issue with a variety of softphones with 1.8.11, if you do 15 test calls. 5/15 asterisk will direct nat to the wan ip |
| 23:42.19 | thecardsmith | 10/15, it will direct to the lan ip |
| 23:42.29 | thecardsmith | nat=yes in sip.conf |
| 23:42.48 | thecardsmith | and it happens with jitsi on mac, 3cx softphone on android, and csipsimple on android |
| 23:42.55 | thecardsmith | but, oddly, it never happens with Bria on android |
| 23:43.32 | thecardsmith | registration always works, i can see the sip messages both ways (either with sip debug, or with wireshark) |
| 23:44.03 | thecardsmith | the softphones are behind a nat, but, asterisk is not -- it's a static IP with holes punched through a firewall |
