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| 00:21.21 | x86 | anyone else having issues with outbound calls via google voice? |
| 00:21.39 | x86 | inbound calls seem to work fine, and outbound calls were working up until today... |
| 00:21.48 | x86 | no config changes on my side |
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| 00:33.15 | CrossWired_ | [TK]D-Fender: you still around? |
| 00:33.44 | [TK]D-Fender | yup |
| 00:33.49 | [TK]D-Fender | for 5 more min.. |
| 00:33.58 | CrossWired_ | fantastic |
| 00:34.38 | CrossWired_ | for my local -> outbound context , i haven't the slightest clue where to start, is that just another context with a DIAL(...) in it? |
| 00:36.00 | [TK]D-Fender | it'll have an extension to do the outward dial for you and you check the status, see if you want to loop, try another number, wait X, or whatever. |
| 00:36.02 | [TK]D-Fender | Logging, etc |
| 00:36.56 | [TK]D-Fender | Channel: <- local channel exten to do actual dialing. Context: Exten: Priority: <- where you dump the caller once you get an answer. (AGI_ |
| 00:37.43 | CrossWired_ | so i have a simple context, that has dial(...) next is dialstatus, thne push to my agi |
| 00:38.25 | CrossWired_ | where do i setup the local extension? |
| 00:38.45 | [TK]D-Fender | ...? |
| 00:38.49 | [TK]D-Fender | there are 2 extension in play. |
| 00:39.10 | [TK]D-Fender | your Channel: points to one who's sole purpose is dialing and logging. Your AGI end remains completely unchanged. |
| 00:39.19 | [TK]D-Fender | we are changing the dialing log, not the answering logic |
| 00:40.04 | [TK]D-Fender | 2 completely separate ends |
| 00:40.09 | CrossWired_ | so the first exten kicks to the 2nd? |
| 00:41.04 | [TK]D-Fender | CrossWiredwhen the Dial gets answered then it dumps the callee into wherever you pointed it to |
| 00:41.31 | CrossWired_ | i think my problem is the local exten isn't set up properly, i'm not sure how they play together i guess |
| 00:41.38 | CrossWired_ | Heres the one i have: http://pastebin.com/7ByzNv6V |
| 00:41.56 | [TK]D-Fender | ok, that is ONE end |
| 00:42.05 | [TK]D-Fender | this takes 2 ends |
| 00:42.11 | [TK]D-Fender | completely different extensions. |
| 00:42.20 | [TK]D-Fender | this is where you send the call after it answer.s |
| 00:42.23 | CrossWired_ | so that one works for the logic piece |
| 00:42.33 | [TK]D-Fender | the context,exten,priority you already set |
| 00:42.36 | [TK]D-Fender | do not change this. |
| 00:42.45 | [TK]D-Fender | that is the 2nd half of the equation |
| 00:42.49 | CrossWired_ | but i do not dial that one |
| 00:42.56 | CrossWired_ | i dial the missing first part |
| 00:43.07 | [TK]D-Fender | before you probably had something like "channel: SIP/provider/number". |
| 00:43.09 | [TK]D-Fender | THIS is what you change |
| 00:43.22 | [TK]D-Fender | point that to a local channel exten in your dialplan and do Dial in there |
| 00:43.44 | CrossWired_ | right i have it to LOCAL/100, but i don't think i have a local channel exten in my dialplan |
| 00:43.57 | [TK]D-Fender | Perhaps you should consider making it :) |
| 00:43.59 | CrossWired_ | what i showed you is my entire dialplan :) |
| 00:44.23 | [TK]D-Fender | make more |
| 00:44.42 | CrossWired_ | ok, and that would be very simple as well right? |
| 00:45.09 | [TK]D-Fender | Make an exten that does the actual dial for you. Check the dialstatus. Decide if you want multiple retries. Decide when you want to give up. Decide if you want to make some kind of log entry when you give up. |
| 00:45.26 | [TK]D-Fender | Its as simple as the decisions you want to make around it. |
| 00:45.59 | [TK]D-Fender | And I jsut looked at the PB.. you don't put a DIAL on the processing end. |
| 00:46.00 | [TK]D-Fender | jsut the AGI |
| 00:46.04 | CrossWired_ | i get the logic inside the piece, its how the fit together, bu t i'm getting there |
| 00:46.06 | CrossWired_ | ok |
| 00:46.15 | *** join/#asterisk coppice (~chatzilla@m121-203-203-74.smartone-vodafone.com) |
| 00:46.20 | [TK]D-Fender | It looks like you tried to mash 2 halves into one |
| 00:46.37 | CrossWired_ | so split those, one sec another attempt |
| 00:48.08 | CrossWired_ | how close is this? http://pastebin.com/khepGgg3 |
| 00:48.28 | CrossWired_ | shit, how do i make the ffirstpart, call the star part? |
| 00:49.38 | [TK]D-Fender | ok, it doesn't seem to be registering, so here it is : http://pastebin.com/gux7ktXv |
| 00:50.48 | CrossWired_ | ahhh @OrignateOut is the provier wher aretta is my orignal SIP, i get it |
| 00:50.54 | [TK]D-Fender | Channel = left half. When that gets answered then it goes on the the right half. Do not put any kind of "answer" in the left half or it will dump to your AGI immediately |
| 00:51.06 | CrossWired_ | ok |
| 00:51.08 | [TK]D-Fender | No, it is not a provider. |
| 00:51.16 | [TK]D-Fender | Local points to your dialplan |
| 00:51.43 | [TK]D-Fender | Dialplan on left, dialplan on right |
| 00:51.53 | [TK]D-Fender | the fact that is uses SIP to dial out is secondary |
| 00:52.01 | CrossWired_ | got it |
| 00:52.27 | CrossWired_ | i would have literally pulled my hair out over this tonight |
| 00:52.39 | CrossWired_ | let me give it a shot now that it makes sense |
| 00:54.01 | [TK]D-Fender | Ok, I'm off for a while... |
| 00:54.22 | CrossWired_ | thank you very much |
| 00:54.31 | CrossWired_ | btw do you consult? |
| 00:54.44 | [TK]D-Fender | I do |
| 00:54.46 | sawgood | If core set verbose 5 is the 'highest' level most Digium staff have told me to use (core set verbose 5) ... what else could show on the console with core set verbose 10? |
| 00:54.58 | [TK]D-Fender | the number "10" |
| 00:55.28 | sawgood | what is a reason someone might use something higher than a 5 setting? |
| 00:56.08 | [TK]D-Fender | Because they make or mod an app to output something higher. Or use Verbose in their dialplan for special notices. |
| 00:56.29 | sawgood | very nice answer, sir ... thank you |
| 00:57.05 | sawgood | If I put a verbose command in my dialplan, what level is the 'lowest' number it would show up using? |
| 00:57.28 | [TK]D-Fender | sawgood: Simple math |
| 00:57.47 | [TK]D-Fender | if verbose is 3 and you do Verbose(5, blah) you won't see it |
| 00:57.51 | sawgood | [TK]D-Fender: are you in the USA? |
| 00:57.55 | sawgood | I'm in California |
| 00:58.02 | [TK]D-Fender | Montreal, QC |
| 00:58.22 | sawgood | nice! I've wanted to get to Canada (near Nigraga falls_ |
| 00:58.25 | sawgood | sorry (sp |
| 00:59.08 | [TK]D-Fender | ok... out the door... BBL |
| 00:59.14 | sawgood | peace! |
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| 05:54.25 | joobie | hey guys.. is there a simple way to realtime monitor another call? |
| 05:54.45 | joobie | when a user receives an inbound call from their queue, i want it to be able to listen in on the call in realtime |
| 05:55.00 | joobie | can have it dial out to anotehr extension or alternatively allow dial in to tap into it.. |
| 05:55.54 | kaldemar | app ChanSpy |
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| 05:58.41 | joobie | thanks |
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| 06:08.37 | schmidts | good morning |
| 06:09.05 | joobie | kaldemar, that seems to work with agents only |
| 06:09.14 | joobie | how can i do the same just on a standard sip call? |
| 06:09.21 | joobie | .. this user is not setup as an agent |
| 06:10.08 | kaldemar | it works with any channel |
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| 06:12.11 | ChannelZ | ...so long as the audio is running through * |
| 06:12.13 | joobie | kaldemar, what do you specify? |
| 06:12.22 | joobie | i tried ChanSpy(Agent) |
| 06:12.37 | ChannelZ | ChanSpy(SIP) |
| 06:12.47 | joobie | and then tried typing 1234# (where 1234 is the extension) |
| 06:14.28 | kaldemar | joobie: why did you try Agent if you have no agents? |
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| 07:17.47 | devyll | hello. any ideeas regarding "audiohook.c: Failed to get 160 samples from read factory" ? (many debug messages during an active call) |
| 07:25.18 | dym | Good morning, telephony world! |
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| 09:48.48 | jacc0 | Hmm, I have a serious problem with dahdi-linux-complete-2.5.0.1+2.5.0.1 + debian squeeze + asterisk 1.8.7.0 |
| 09:49.09 | jacc0 | Attempting to test a timer with 50 ticks per second. |
| 09:49.09 | jacc0 | Failed to open timing fd |
| 09:49.09 | jacc0 | Command 'timing test' failed. |
| 09:49.53 | jacc0 | will post some more details about my installation procedure on pastebin; one moment |
| 09:51.51 | *** join/#asterisk gaetronik (~gaetan@ks370400.kimsufi.com) |
| 09:51.57 | gaetronik | Hi! |
| 09:52.08 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 09:52.42 | gaetronik | i know i'm a bit off-topic but anyone has a good SIP provider to recommand in Poland? |
| 09:55.37 | FlashDeluxe | hi @all! i am using asterisk 1.8.7 with dahdi 2.5.0.1, libpri 1.4.12 and dahdi-zaphfc (i got two cheap hfc cards installed and connceted to bri_cpe). But i cannot dial out, i got a few errors, i pasted them here http://paste.debian.net/137296/ Can somebody help me please? |
| 09:56.02 | jacc0 | hi FlashDeluxe; it might be te same problem I just reported |
| 09:56.17 | jacc0 | try : timing test |
| 09:56.21 | jacc0 | in asterisk CLI |
| 09:56.41 | FlashDeluxe | jacc0: It has been 1019 milliseconds, and we got 51 timer ticks |
| 09:56.42 | *** join/#asterisk iggy_work (~iggy@2001:67c:21c4:1800:21c:23ff:fe18:a995) |
| 09:57.00 | jacc0 | okay, then we have a differend problem I guess; tho it could be related somehow |
| 09:57.17 | iggy_work | hi there. Does anyone know if it's possible to VLAN a cisco 7940 into a voice vlan and still use the port on the back of the phone for untagged workstation traffic? |
| 09:57.19 | FlashDeluxe | jacc0: mhh and is there a solution yet? |
| 09:57.24 | jacc0 | here is my installation precedure: has been 1019 milliseconds, and we got 51 timer ticks |
| 09:57.32 | jacc0 | sorry , wrong past |
| 09:57.42 | jacc0 | http://pastebin.com/xv1LsRwP |
| 09:58.55 | FlashDeluxe | jacc0: but you are not using zaphfc, right? |
| 10:01.52 | jacc0 | nope |
| 10:02.40 | FlashDeluxe | mhh i just looked into my syslog and found this ->"vzaphfc: card 0: chan B1 opened as ZTHFC1/0/1." But it didn`t install vzapfc and only zaphfc is loaded, how could this be related? |
| 10:06.21 | jacc0 | no clue |
| 10:09.56 | kaldemar | jacc0: why did you reboot? you probably don't have the dahdi module loaded. |
| 10:10.11 | *** join/#asterisk Falcon_1 (5354586b@gateway/web/freenode/ip.83.84.88.107) |
| 10:10.53 | Falcon_1 | question any one with experience with chan_mobile |
| 10:12.15 | kaldemar | ~ask |
| 10:12.16 | 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. |
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| 10:16.16 | Falcon_1 | Sorry, Trying to use chan_mobile. Search shows the telephones, but asterisk does not connect. Altough sometimes when comming back from sleep mode it sometimes connect. No errors... no debug info... nothing |
| 10:17.21 | *** join/#asterisk kladze (~kladze@4706ds2-kj.0.fullrate.dk) |
| 10:19.14 | kaldemar | Falcon_1: not much to say without any configs if you don't get any debug output. |
| 10:27.06 | *** join/#asterisk Da-Geek (~Da-Geek@11.74.155.90.in-addr.arpa) |
| 10:28.57 | kladze | Hello, im having an issue with my asterisk... Im running Asterisk 1.6.2.9-1~bpo50+3. I want to change the language sound files to Danih... I have placed my downloaded gsm files into /var/lib/asterisk/sounds/da |
| 10:29.28 | kladze | changed /etc/asterisk/asetisk.conf with languageprefix=yes |
| 10:29.53 | kladze | and inside sip.conf i have language=da |
| 10:30.17 | kladze | but it still keeps playing the english voice.. |
| 10:30.39 | kladze | my output from asterisk says this... |
| 10:30.41 | kladze | [Oct 17 11:26:02] -- <IAX2/iaxdundi-2624> Playing 'vm-isunavail.gsm' (language 'en') |
| 10:30.41 | kladze | [Oct 17 11:26:04] -- <IAX2/iaxdundi-2624> Playing 'vm-intro.gsm' (language 'en') |
| 10:30.41 | kladze | [Oct 17 11:26:09] -- <IAX2/iaxdundi-2624> Playing 'beep.gsm' (language 'en') |
| 10:30.48 | kladze | Anyone have a clue? |
| 10:31.49 | jacc0 | kladze: did you restart asterisk after changing language? |
| 10:32.27 | kladze | i did a core restart |
| 10:32.31 | kladze | so yes |
| 10:32.57 | jacc0 | did you see sip.conf was loaded all right? maybe there is a typ0 in there |
| 10:35.50 | kladze | it gives no error's other than a few places where i did a comment with # in front of a view lines |
| 10:36.52 | jacc0 | you should replace the # with ; |
| 10:37.10 | jacc0 | the entire file won't load if there is an error |
| 10:37.50 | jacc0 | # is not the correct way to comment out a line in sip.conf, use ; |
| 10:41.24 | kladze | done, corrected it |
| 10:44.39 | kladze | Still using english voice artist |
| 10:45.40 | kladze | and still gives me the output |
| 10:45.40 | kladze | [Oct 17 11:26:02] -- <IAX2/iaxdundi-2624> Playing 'vm-isunavail.gsm' (language 'en') |
| 10:45.40 | kladze | [Oct 17 11:26:04] -- <IAX2/iaxdundi-2624> Playing 'vm-intro.gsm' (language 'en') |
| 10:45.40 | kladze | [Oct 17 11:26:09] -- <IAX2/iaxdundi-2624> Playing 'beep.gsm' (language 'en') |
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| 10:51.34 | jacc0 | Kladze: are you using asterisk realtime? |
| 10:52.55 | jacc0 | if so; did you check the values in the langage collumn? |
| 10:53.10 | jacc0 | could you post your sip.conf in pastebin.com ? |
| 11:01.04 | kladze | here is my sip.conf |
| 11:01.05 | kladze | http://pastebin.com/gcBUTafa |
| 11:12.17 | jacc0 | kladze: I don't see any sip clients configured. are they in the database? or in users.conf? make sure they don't hav a client spesific language configuration |
| 11:13.41 | kladze | they are in users.conf |
| 11:15.06 | jacc0 | I fixed my timmer problem |
| 11:15.30 | jacc0 | did a reinstall without adding the following lines to sources.list: |
| 11:15.31 | jacc0 | deb http://packages.asterisk.org/deb squeeze main |
| 11:15.31 | jacc0 | deb-src http://packages.asterisk.org/deb squeeze main |
| 11:15.47 | jacc0 | then everything was okay |
| 11:17.38 | kaldemar | kladze: and no language defined in users.conf? |
| 11:19.11 | kladze | kaldemar yes there is a language in users.conf |
| 11:19.30 | kladze | sry |
| 11:19.32 | kladze | in sip.conf |
| 11:19.49 | kladze | but also in users.conf |
| 11:20.37 | kladze | http://pastebin.com/ePbD8FLC users.conf |
| 11:20.39 | kladze | part of it |
| 11:21.27 | jacc0 | somehow, using the asterisk.org sources, breaks the timmer in asterisk 1.8.7 + dahdi 2.5.0.1: maybe it installes some dependencies that ar incompatible with the new dahdi? |
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| 11:33.13 | e-fon_patrick | Hi Guys, quick question. I'm trying to implement app_fax in Asterisk 1.6.1.20 with spandsp0.0.0.6pre17 |
| 11:33.31 | e-fon_patrick | are there any known bugs? |
| 11:33.57 | e-fon_patrick | because after compiling spandsp there is still no app_fax in astrisk menu select |
| 11:34.14 | e-fon_patrick | any hint is welcome |
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| 12:33.47 | *** join/#asterisk doolittlework (~doolittle@41-134-22-10.dsl.mweb.co.za) |
| 12:33.56 | doolittlework | hi all |
| 12:35.17 | doolittlework | i have 6 remote asterisk servers i want to monitor, can you guys suggest a monitoring tool where i can see remote system status(uptime, load, sip trunks(registered or not) any help welcome? |
| 12:36.42 | e-fon_patrick | hi doolittlework, we're using Nagios |
| 12:37.07 | e-fon_patrick | in combination with cacti |
| 12:38.30 | doolittlework | e-fon_patrick: is it easy to setup? |
| 12:38.43 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:39.00 | doolittlework | hi [TK]D-Fender |
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| 12:46.13 | kladze | zabbix maybe ? |
| 12:46.20 | kladze | thats what we are using |
| 12:48.50 | doolittlework | thanks |
| 12:48.57 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
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| 13:25.20 | pabelanger | doolittlework: opennms is another option |
| 13:25.23 | *** join/#asterisk djb_ (~je_moer@62.41.172.10) |
| 13:25.36 | djb_ | Hi |
| 13:28.40 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:29.10 | like_a_horse | hi, anyone have a funky way of listening to recordings made by "automon" in the features.conf ? |
| 13:31.59 | [TK]D-Fender | I normally use my ears... but I can't speak for anyone else... |
| 13:40.31 | dym | [TK]D-Fender: agreed. I tend to use similar means |
| 13:40.32 | FlashDeluxe | hi! i got a problem with asterisk 1.8 and dahdi 2.4. if i want to fax multiple sites via a handytone adaptor from grandstream, not all sites are transmitted or some sites are incomplete. Does anybody got an idea how i can debug it? i just get a few errors but i don`t think they are helpful: http://paste.debian.net/137354/ |
| 13:41.44 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 13:42.24 | [TK]D-Fender | FlashDeluxe, intern,4yyyyy6,1 <----------- doesn't exist as it says |
| 13:42.35 | [TK]D-Fender | FlashDeluxe, You are sending your DAHDI calls to a place that doesn't exist |
| 13:43.42 | CrossWired | morning all, how big is the upgrade from 1.6 to 1.8? big deal? |
| 13:44.25 | kaldemar | CrossWired: http://svn.digium.com/svn/asterisk/tags/1.8.7.0/UPGRADE.txt |
| 13:46.06 | CrossWired | kaldemar: gracias |
| 13:47.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:49.02 | CrossWired | in the first note it indicates that SIP_CAUSE is not set by default as 1.6.2, http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause page seems to indicate this function is only available in 1.8, hence my reason to upgrade |
| 13:49.33 | CrossWired | if that is the case, what is the proper usage of HASH(SIP_CAUSE,<chan name>) in dialplan? |
| 13:49.39 | FlashDeluxe | [TK]D-Fender ohhh now that you are saying it.. :) thanks! |
| 13:49.57 | CrossWired | nevermind, that example I gave gives the example |
| 13:49.58 | [TK]D-Fender | FlashDeluxe, chan_bigprint FTW :) |
| 13:50.01 | CrossWired | thanks for listening :) |
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| 13:51.44 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 13:58.15 | SuperNull | hey TK .. have you heard of any reason why ast 1.8.6 would have an increasing number of unix sockets open to the point it hits the system limit ? |
| 14:00.19 | jacc0 | @SuperNull: failing iax trunk? |
| 14:00.41 | SuperNull | hmmm... i dont believe so .. but maybe i have some old config |
| 14:01.31 | SuperNull | the only real hint i have to it .. which makes no sense to me .. is that it started after i created a context with MYSQL calls.. everything appears to be getting freed tho.. so and i use tcp not unix socks |
| 14:03.04 | SuperNull | i will put it in pastebin .. its a rollover style dial using mysql tables for lookup |
| 14:03.12 | *** join/#asterisk lcat (~lcat@187.45.254.221) |
| 14:06.01 | SuperNull | http://pastebin.com/zSRiF3Vm |
| 14:07.15 | [TK]D-Fender | SuperNull, No idea... except that you are a version ebhind |
| 14:07.18 | [TK]D-Fender | behind* |
| 14:07.52 | SuperNull | :-( |
| 14:07.58 | SuperNull | but but .. i just installed ! DAR!BMADFAS! |
| 14:08.09 | *** join/#asterisk master_of_master (~master_of@p57B55667.dip.t-dialin.net) |
| 14:09.59 | SuperNull | searches for release notes |
| 14:12.25 | FlashDeluxe | [TK]D-Fender hi! i still got the problem, this time without errors :( do you have any suggestions? http://paste.debian.net/137360/ |
| 14:13.13 | jacc0 | @superNull: you have an error extension, so I asume you have been experiancing errors (that will stop the execution of the current dialplan) |
| 14:14.08 | jacc0 | @superNull: you don't close the sql connection in case of an error |
| 14:14.11 | SuperNull | ehhh i can probably remove it and see what happens.. no errors i know of .. honestly.. but it could be incase mysql is dead or something |
| 14:14.15 | jacc0 | I guess there is your problem |
| 14:14.28 | SuperNull | alright.. let me add the close.. |
| 14:14.34 | jacc0 | ;) |
| 14:14.34 | SuperNull | but .. im using mysql tcp not sock .. so i dunno |
| 14:14.53 | [TK]D-Fender | FlashDeluxe, Add "/nj" to your Dial (No Jitterbuffer) |
| 14:14.57 | StaRetji | folks, I know how to forward the call, but how to forward call after 3 rings. Call 9999 rings 9999, if not picked after 3 rings, forward to 5555. Any help would be highly appreciated, thx ;) |
| 14:15.03 | jacc0 | that is why it is taking up maore and more tcp sockets |
| 14:15.21 | [TK]D-Fender | FlashDeluxe, You can't buffer faxes (and survive). Also I don't think that your HT supports T.38 and well.. lets not go there... |
| 14:15.54 | jacc0 | where did you tell asterisk to use tcp sockets? |
| 14:16.00 | jacc0 | what config file? |
| 14:16.48 | SuperNull | jacc0 its using unix sockets tho it doesn't show them as tcp .. |
| 14:17.14 | jacc0 | what config file did you set it to use unix sockets? |
| 14:17.17 | SuperNull | i didnt.. the mysql server isn't even local its remote. im saying that the problem is unix sockets open .. but the only change i made dealt with ip based sockets |
| 14:18.01 | jacc0 | in what config file did you make the change? I think you did it in te wrong place; there are 2 place where you can set socket |
| 14:18.14 | kaldemar | [TK]D-Fender: actually, /j means to use the generic jitterbuffer. https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers |
| 14:18.28 | SuperNull | uhhhm. i didnt make any changes to existing functional config other than extensions |
| 14:18.34 | [TK]D-Fender | kaldemar, Ahh, also only local |
| 14:18.44 | [TK]D-Fender | FlashDeluxe, ok, strike that... |
| 14:19.18 | Dovid | hello all |
| 14:19.22 | [TK]D-Fender | FlashDeluxe, Chalk it up to FoS & GS |
| 14:19.26 | jacc0 | superNull: ' but the only change i made dealt with ip based sockets' where did you make the change so that it uses tcp sockets? please answer |
| 14:19.42 | jacc0 | did you set it in extensions.conf? |
| 14:20.03 | *** join/#asterisk ruied (~ruied@dial-b2-180-152.telepac.pt) |
| 14:20.04 | SuperNull | MYSQL(Connect) my friend.. |
| 14:20.13 | jacc0 | okay |
| 14:20.16 | SuperNull | no actual configuration change.. |
| 14:20.19 | SuperNull | just extensions |
| 14:20.20 | jacc0 | one moment |
| 14:21.38 | FlashDeluxe | [TK]D-Fender i sended some faxes to me an got a trace now http://paste.debian.net/137361/ What do you mean by "chalk it up to FoS & GS"? |
| 14:21.55 | [TK]D-Fender | Fax Over SIP and ... I'm sure you know what GS means... |
| 14:22.13 | [TK]D-Fender | The smallest hiccup on your lan, etc and *poof* |
| 14:22.40 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 14:22.45 | FlashDeluxe | i didn`t know what chalk it up means :D but i looked it up ;) |
| 14:23.03 | Dovid | anyone here use voipmonitor? |
| 14:24.36 | x86 | anyone else having issues with outbound calls via google voice? |
| 14:24.45 | x86 | inbound calls seem to work fine, and outbound calls were working up until today... |
| 14:24.53 | x86 | no config changes on my side |
| 14:25.48 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 14:26.11 | [TK]D-Fender | FlashDeluxe, "write it off", "grab a drink and sulk", etc |
| 14:27.29 | *** part/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 14:27.38 | jacc0 | <PROTECTED> |
| 14:27.44 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:27.52 | SuperNull | i gave it an ip address... not a hostname .. or anything else. |
| 14:27.59 | jacc0 | @superNull: because it can not be done in extensions.conf |
| 14:28.24 | SuperNull | how does MYSQL(Connect) work ? |
| 14:28.33 | SuperNull | im fairly certain only on tcp. |
| 14:28.53 | jacc0 | correction: I'm not sure if it can beset in extensions.conf |
| 14:28.57 | jacc0 | look at : res_config_mysql.conf |
| 14:28.59 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
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| 14:29.35 | SuperNull | ehh i only have the older res_mysql.conf |
| 14:29.58 | SuperNull | but it references tcp info not a socket .. the dbsock line is commented |
| 14:29.59 | Dovid | SuperNull: Look in ur srcs folder |
| 14:30.02 | Dovid | for the file |
| 14:30.38 | SuperNull | Dovid what do you want me to look for ... anything unix socket creation ? that might take a bit |
| 14:30.48 | x86 | no one is using google voice as an outbound trunk? |
| 14:30.50 | SuperNull | oh oh.. you mean the res_config_mysql.conf lol |
| 14:32.14 | SuperNull | nearly identical.. if not. |
| 14:35.10 | jacc0 | take a look at your mysql server and see how many open connections it has |
| 14:35.24 | like_a_horse | [TK]D-Fender, " I normally use my ears... but I can't speak for anyone else..." |
| 14:35.25 | like_a_horse | pfft |
| 14:35.27 | like_a_horse | smart ass :P |
| 14:36.13 | like_a_horse | how do i get to the point to use my ears to listen to the recordings? ;) |
| 14:36.13 | n3hxs | [TK]D-Fender, is a bit eerie. |
| 14:36.48 | like_a_horse | is there a feature code of sorts that someone has written |
| 14:36.49 | like_a_horse | :) |
| 14:37.28 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:37.39 | SuperNull | jacc0 .. 1 single socket per server.. but it would require us to see what happens when that extension gets fired. |
| 14:39.12 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:41.03 | jacc0 | @SuperNull: increase allowed open sockets in mysql and try again? :) |
| 14:42.26 | SuperNull | one second.. oddly.. my clearing/disconnected of the mysql connect has reverted to before i changed it |
| 14:42.33 | SuperNull | mysql doesn't lock the extension file right? |
| 14:42.54 | gaetronik | hi again |
| 14:43.04 | gaetronik | is there any chan dedicated to sip providers? |
| 14:43.37 | [TK]D-Fender | like_a_horse, "core show application playback" |
| 14:44.41 | [TK]D-Fender | gaetronik, Doubt it highly.. |
| 14:45.41 | gaetronik | not good |
| 14:46.03 | gaetronik | finding a good sip provider for european countries is quite hard |
| 14:46.28 | gaetronik | not for, in |
| 14:47.20 | Faustov | can I have a suggestion that asterisk reports errors with permissions to devices? Currently wrong permissions in /dev/dahdi devices give no error output at all, verbosity set to 10 |
| 14:47.31 | SunTsu | gaetronik: sipgate? |
| 14:47.49 | gaetronik | SunTsu: why not |
| 14:47.51 | SuperNull | jacc0 sorry for the delay i gotta type up an 'excuse' email to customers :-( .. why my shit is breaking. |
| 14:48.32 | SunTsu | gaetronik: you can try them out for free, rates are OK, they offer iax if you want to.. |
| 14:48.45 | gaetronik | SunTsu: hmm the idea is to have a sip provider for a sip trunk in the country |
| 14:48.45 | wdoekes2 | Faustov: when opening which file/device exactly? |
| 14:48.57 | gaetronik | my company is setting up an office in Poland |
| 14:49.27 | gaetronik | so i need 10 did in the country and a sip trunk |
| 14:49.32 | Faustov | wdoekes2: /dev/dahdi/pseudo |
| 14:49.36 | like_a_horse | [TK]D-Fender, an example file would be /var/spool/asterisk/monitor/auto-1318856696-1000-2002.wav |
| 14:50.12 | SunTsu | gaetronik: OK, maybe some polish people can help you there. I'm not from poland, so I'm out |
| 14:50.29 | like_a_horse | I can setup a playback of that file but is there not a web tool or well known dialplan entry that can let you listen to/delete your recordings? |
| 14:51.03 | gaetronik | SunTsu: it's the issue with small countries it's hard to find information from outside |
| 14:53.17 | *** join/#asterisk devil_evoxxx (~d3v1l@host79-37-dynamic.4-87-r.retail.telecomitalia.it) |
| 14:54.25 | [TK]D-Fender | like_a_horse, GUI's, plenty. Diallpan app? No, that is for you to script. * is a toolkit, not a "ready-to-fly" model. |
| 14:54.35 | devil_evoxxx | hi all guys |
| 14:55.19 | like_a_horse | [TK]D-Fender, ok. Any recommended GUI's that I should look out for ? |
| 14:56.12 | [TK]D-Fender | ARI |
| 14:57.04 | wdoekes2 | Faustov: oddness.. in trunk I see lots of ast_log's wherever /pseudo is opened |
| 14:58.48 | devil_evoxxx | when you want to make a call in "unknown" mode |
| 14:59.04 | Faustov | wdoekes2: please feel free to try - make the device permissions unaccessible and try to use meetme - nothing. |
| 14:59.16 | devil_evoxxx | in wichm mode you configure your callerid(num) and callerid(name)? |
| 14:59.16 | Faustov | we had to use strace to find it |
| 14:59.31 | wdoekes2 | which asterisk version? |
| 15:00.55 | devil_evoxxx | 1.8.7 |
| 15:01.37 | wdoekes2 | sorry, my question was directed at Faustov |
| 15:02.04 | Faustov | wdoekes2: 1.8.6 |
| 15:02.44 | kaldemar | devil_evoxxx: core show function CALLERID |
| 15:02.46 | Faustov | wdoekes2: sorry, 1.9.7 |
| 15:02.51 | Faustov | erm |
| 15:02.53 | wdoekes2 | hehe |
| 15:02.57 | Faustov | sigh |
| 15:03.03 | Faustov | I fial with typing today ;) |
| 15:03.26 | kaldemar | devil_evoxxx: core show function CALLERPRES |
| 15:03.30 | devil_evoxxx | kaldemar: thankyou for reply, but i'm having strange problem. Because if i leave it blank the call exit to my provider with unknown cid |
| 15:03.48 | devil_evoxxx | but when call go out to a mobile phone... |
| 15:04.19 | devil_evoxxx | my provider say's that when i leave cid num/name blank in sip packet there is specified asterisk as num or name.. |
| 15:04.43 | kaldemar | devil_evoxxx: set presentation with CALLERID function. |
| 15:04.57 | kaldemar | devil_evoxxx: if that doesn't work, keep asking your provider. |
| 15:06.24 | devil_evoxxx | my actual dialplan for testin this issue with my provider is this http://pastebin.com/cCvCNKKd |
| 15:06.43 | devil_evoxxx | if i leave callerdi(num) and name as i specified in pastebin |
| 15:06.52 | devil_evoxxx | the call to mobile network does not go out |
| 15:07.35 | devil_evoxxx | so, now i try to specify the presentation |
| 15:08.41 | wdoekes2 | Faustov: which syscall failed then? the open()? |
| 15:09.27 | Faustov | wdoekes2: correct |
| 15:11.26 | wdoekes2 | Faustov: well.. I'm still of the opinion that you should've gotten a LOG_WARNING |
| 15:12.27 | devil_evoxxx | kaldemar: i try to specifu name / num presentation, leaving callerdi(num) and name blank |
| 15:12.30 | Faustov | <PROTECTED> |
| 15:12.30 | Faustov | <PROTECTED> |
| 15:12.30 | Faustov | <PROTECTED> |
| 15:12.34 | Faustov | wdoekes2: ^ |
| 15:12.44 | devil_evoxxx | and the call still not exit on mobile network |
| 15:13.33 | wdoekes2 | and warnings do go to the console? |
| 15:14.13 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:14.35 | kaldemar | devil_evoxxx: the idea is not to leave anything blank. |
| 15:15.42 | Faustov | wdoekes2: this is the CLI output |
| 15:16.59 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:17.03 | wdoekes2 | I meant: console => ... warning ... in logger.conf? |
| 15:18.05 | kaldemar | devil_evoxxx: also, feel free to show a CLI output of a call. |
| 15:18.29 | devil_evoxxx | wait a moment and i post it on pastebin |
| 15:18.58 | navaismo | morning! |
| 15:21.57 | Faustov | wdoekes2: my bad! someone left it directed at syslog |
| 15:21.58 | Faustov | sigh |
| 15:22.00 | Faustov | nevermind! |
| 15:24.10 | SuperNull | omg jacc0 i think .. you da man. |
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| 15:26.43 | Katty | ohai |
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| 15:34.18 | devil_evoxxx | kaldemar: http://pastebin.com/uLUg43nZ |
| 15:34.37 | devil_evoxxx | kaldemar: here is the cli output of a call..that does not go out |
| 15:34.43 | devil_evoxxx | still dial but never go to ringing.. |
| 15:35.38 | [TK]D-Fender | devil_evoxxx, Enable SIP debug. You aren't really looking at the call yet |
| 15:37.01 | kaldemar | devil_evoxxx: and CALLERID(name-pres)=0458538897 is invalid. |
| 15:37.04 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:41.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 15:42.16 | devil_evoxxx | here is the complete call flow with sip debug on |
| 15:42.33 | devil_evoxxx | http://pastebin.com/3CESAag7 |
| 15:43.30 | devil_evoxxx | kaldemar: with the setting shown i precedent pastebin log, the call to a mobile phone operator is ok, but to another operator still Dial but never ring |
| 15:43.44 | [TK]D-Fender | devil_evoxxx, global SIP debug. You only got part of the call |
| 15:43.51 | [TK]D-Fender | devil_evoxxx, do not restrit to IP/peer |
| 15:44.30 | devil_evoxxx | [TK]D-Fender: is a little bit difficult, in this asterisk box there are 30/40 cocncurrent call |
| 15:44.45 | kaldemar | devil_evoxxx: irrelevant, it is still invalid. |
| 15:45.04 | devil_evoxxx | name-pres is character? |
| 15:47.22 | kaldemar | devil_evoxxx: does not compute. did you read the documentation and valid values from "core show function CALLERPRES"? |
| 15:47.51 | [TK]D-Fender | kaldemar, he isn't even using that function. |
| 15:48.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v014-093.mobile.uci.edu) |
| 15:48.14 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 15:48.41 | kaldemar | [TK]D-Fender: well he's not supposed to since it is deprecated and CALLERID is favored, but it happens to be the only one that lists the valid values for the presentation. |
| 15:48.50 | devil_evoxxx | oh..shit |
| 15:49.06 | devil_evoxxx | ...right kaldemar sorry |
| 15:49.22 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 15:49.42 | devil_evoxxx | i've read that was deprecated..and i not proceed with reading |
| 15:49.44 | devil_evoxxx | sorry |
| 15:49.47 | kaldemar | why the CALLERID documentation does not list those values, i don't know. |
| 15:50.20 | EmleyMoor | Thank you for spotting another deprecated thing in my dialplan |
| 15:52.46 | devil_evoxxx | now i correct the name-pres |
| 15:53.23 | devil_evoxxx | ..same problem , i'm really thinking that is my provider |
| 15:54.21 | devil_evoxxx | i try with allowed_not_screened |
| 15:54.24 | *** join/#asterisk hfb (~hfb@pool-98-119-37-20.lsanca.dsl-w.verizon.net) |
| 15:54.30 | [TK]D-Fender | devil_evoxxx, I don't see a real complete call yet... |
| 15:56.13 | devil_evoxxx | [TK]D-Fender: i know, i'm still n00b, but filtering sip call trough 30 concurrent call |
| 15:56.26 | devil_evoxxx | if you have an idea how to filter this call |
| 15:58.04 | albro | Can anybody explain this error message: WARNING[5680]: chan_iax2.c:1071 iax_error_output: Information element length exceeds message size |
| 15:58.27 | albro | I have searched thru forums and such but no luck! |
| 15:59.08 | albro | And I only see this error message when the asterisk box is behind a 3G data connection. |
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| 16:06.44 | *** join/#asterisk Diffen (~diffen@c-f477e555.042-17-73746f11.cust.bredbandsbolaget.se) |
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| 16:09.36 | SuperNull | [TK]D-Fender my open sock problem .. seems to have ben resolved by autoclear in the app_mysql conf.. monitoring for new fails. |
| 16:22.11 | devil_evoxxx | kaldemar: ok, i've read the callingpres function, and trying those values..but still having same problem |
| 16:22.37 | devil_evoxxx | what's your hints about this? |
| 16:23.08 | *** join/#asterisk pdtpatrick (~pdtpdt@12.249.4.226) |
| 16:32.42 | dym | devil_evoxxx: what are you trying to achieve? |
| 16:33.13 | p3nguin | I noticed that the CALLERID func says the syntax is CALLERID(datatype[,CID]). It says that CID is an optional caller ID. What does optional caller ID mean in this case? |
| 16:35.24 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-xnvzkqmecbtktsnh) |
| 16:35.33 | *** join/#asterisk irroot (~irroot@41.54.211.109) |
| 16:36.30 | EmleyMoor | p3nguin: Where are you getting that from? |
| 16:36.52 | p3nguin | core show function CALLERID |
| 16:37.51 | EmleyMoor | Hmmm... I'm puzzled by it too... |
| 16:38.10 | p3nguin | There's no explanation for it, and it isn't something I've ever seen anyone use. |
| 16:38.27 | dym | mhh |
| 16:39.01 | dym | isnt that for CLIR? |
| 16:39.04 | EmleyMoor | I notice the CALLERID function is due more changes in 1.8 |
| 16:39.10 | *** join/#asterisk cerberus_za (~coert@8ta-151-18-226.telkomadsl.co.za) |
| 16:39.24 | devil_evoxxx | dym: i'm try to understand wich callepres work in 1.8 |
| 16:39.33 | p3nguin | Typical uses of CALLERID often include Set(CALLERID(num)=123) and Verbose(${CALLERID(num)}). I've never seen anything using the ,CID part. |
| 16:39.36 | dym | devil_evoxxx: depends on your uses |
| 16:39.59 | devil_evoxxx | i'm having problem to send a call with unknown callerid |
| 16:40.03 | devil_evoxxx | trough my provider |
| 16:40.08 | dym | p3nguin: looks like a syntax change |
| 16:40.14 | dym | devil_evoxxx: SIP? |
| 16:40.26 | devil_evoxxx | specifically to a mobile network work, to another operator no.. |
| 16:40.33 | devil_evoxxx | dym, yes, SIP |
| 16:40.34 | EmleyMoor | devil_evoxxx: Try to set CALLERID(num-pres) to prohib |
| 16:40.40 | dym | devil_evoxxx: maybe this needs to be done via a SIP header setting. |
| 16:40.48 | dym | You'll have to check with your provider |
| 16:40.52 | p3nguin | The fact of it being a syntax change was never in question. |
| 16:40.55 | devil_evoxxx | already done, num-pres and name-pres to prohib |
| 16:41.00 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-129-69.chyn.qwest.net) |
| 16:41.09 | dym | devil_evoxxx: where tho? |
| 16:41.10 | devil_evoxxx | to a mobile operator work, to another operator no.. |
| 16:41.18 | p3nguin | The question was regarding the usage of the syntax, and why there is no explanation as to the usage. |
| 16:41.26 | dym | devil_evoxxx: no. where the setting needs to be done. |
| 16:41.44 | devil_evoxxx | dym, what? |
| 16:41.49 | devil_evoxxx | dym: what? |
| 16:42.05 | dym | Well - As I said - maybe this needs to be set in the header - "SIPAddHeader" wise. |
| 16:42.51 | dym | SIPAddHeader(Privacy: id) |
| 16:43.00 | dym | is a common thing |
| 16:43.07 | [TK]D-Fender | SETCALLERPRES |
| 16:43.25 | dym | ? |
| 16:43.32 | dym | Is that even valid within the header? |
| 16:43.38 | devil_evoxxx | so, using callerpress add this in sip header? |
| 16:43.49 | dym | devil_evoxxx: well - try it? |
| 16:45.11 | devil_evoxxx | ok.. |
| 16:45.12 | EmleyMoor | Is there an easy way to see the full headers of an incoming call? (SIP or IAX2) |
| 16:45.19 | p3nguin | devil_evoxxx: What happens if you simply set your CALLERID(num) to nothing? Set(CALLERID(num)=) |
| 16:45.21 | dym | sip set debug on |
| 16:45.24 | dym | @ EmleyMoor |
| 16:45.39 | dym | EmleyMoor: on the CLI that is |
| 16:46.43 | devil_evoxxx | dym..it's working if i call a Vodafone mobile cell.. |
| 16:46.43 | EmleyMoor | dym: Right - well, familiar with that anyway but may have a tinker and see what's what - to help me understand |
| 16:46.43 | dym | (: |
| 16:46.47 | dym | there. |
| 16:46.58 | devil_evoxxx | but if i call a H3G mobile network does not work |
| 16:47.02 | devil_evoxxx | i think is my provider too |
| 16:47.03 | dym | well |
| 16:47.07 | dym | should work anyways |
| 16:47.11 | p3nguin | devil_evoxxx: What happens if you simply set your CALLERID(num) to nothing? Set(CALLERID(num)=) |
| 16:47.31 | dym | EmleyMoor: excuse me? |
| 16:48.25 | EmleyMoor | dym: I'm interested in seeing what information might be lurking in the headers for various calls... so I will see what I get sometime with the debug on |
| 16:48.36 | dym | right! |
| 16:48.37 | dym | :) |
| 16:48.40 | dym | that i understood |
| 16:49.01 | devil_evoxxx | p3nguin: ..same behavior..if i call a vodafone mobile tel work with unknown cid |
| 16:49.08 | dym | oh |
| 16:49.17 | dym | then obviously you need to have a chat with your provider |
| 16:49.20 | p3nguin | Didn't you want to restrict your number? |
| 16:49.21 | dym | and dont need the extra header setting |
| 16:49.31 | devil_evoxxx | but the calls to h3g mobile network does not work |
| 16:49.52 | EmleyMoor | p3nguin: I think he's saying that his number gets presented regardless when calling H3G |
| 16:49.59 | devil_evoxxx | dym: my provider say that it's all ok |
| 16:50.10 | dym | devil_evoxxx: obviously cant be (: |
| 16:50.25 | dym | they shoudl be able to give you some in-depth information, since its their network... |
| 16:50.29 | dym | should* |
| 16:50.31 | [TK]D-Fender | I think it's now been an entire hour and still no proper complete call with SIP debug |
| 16:50.38 | EmleyMoor | devil_evoxxx: Which country are you in? |
| 16:50.55 | devil_evoxxx | i think that my provider as finished the minutes over a gsmbox |
| 16:51.01 | devil_evoxxx | EmleyMoor: it |
| 16:51.40 | devil_evoxxx | [TK]D-Fender: it's coming..i'm setup a machine only for making this test |
| 16:52.03 | [TK]D-Fender | devil_evoxxx, If you are using another machine then you are polluting the test and wasting time |
| 16:52.04 | dym | [TK]D-Fender: did you request one of him? |
| 16:52.10 | [TK]D-Fender | dym, repeatedly |
| 16:52.13 | dym | :D |
| 16:52.15 | dym | lovely |
| 16:52.30 | dym | devil_evoxxx: why do you not provide the desired output? |
| 16:52.57 | devil_evoxxx | because if i set sip debug on this machine, where is present 30/40 concurrent call |
| 16:53.07 | dym | well |
| 16:53.09 | devil_evoxxx | i can not filter only the desidered call.. |
| 16:53.13 | dym | its gonne be some effort on your side. |
| 16:53.24 | dym | us guessing in the dark wont make things better. |
| 16:53.32 | dym | so man up! |
| 16:53.33 | dym | (: |
| 16:55.41 | [TK]D-Fender | devil_evoxxx, just because a call is in progress doesn't mean it is spewing out SIP debug constantly. |
| 16:55.52 | [TK]D-Fender | devil_evoxxx, And we're more than competant to grab the part we need |
| 16:56.03 | [TK]D-Fender | devil_evoxxx, This whoe process should ahve taken < 1 minute |
| 16:56.05 | dym | [TK]D-Fender: im not gonna ask again (: |
| 16:56.35 | [TK]D-Fender | Enable debug. Place call. Stop debug. 10 seconds. that's 50 left to pastebin. |
| 16:57.08 | devil_evoxxx | ok |
| 16:57.33 | devil_evoxxx | [TK]D-Fender: in wich mode i've to set num-pres and name-pres? |
| 16:57.38 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:58.37 | p3nguin | Where can I read about the changes in CALLERID()? |
| 16:58.47 | dym | p3nguin: changelog i'd assume |
| 17:02.39 | devil_evoxxx | [TK]D-Fender: http://pastebin.com/QfiNupU4 |
| 17:02.54 | devil_evoxxx | the number i've try to call is 3926997438 |
| 17:03.08 | [TK]D-Fender | devil_evoxxx, Why don't we also have full verbose in there? |
| 17:03.40 | devil_evoxxx | core set debug 1? |
| 17:05.08 | p3nguin | I can't find any information pertinent to the "CID" part of the syntax. |
| 17:05.31 | p3nguin | "[,CID]" to be more exact. |
| 17:09.03 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 17:12.26 | irroot | p3nguin you can use svn blame perhaps and work back from then |
| 17:15.56 | p3nguin | I don't know what that means. |
| 17:18.42 | irroot | p3nguin ah its a bit to get used to it |
| 17:21.17 | devil_evoxxx | here is a complete sip debug , core verbose, of the call that go out correctly: http://pastebin.com/iwRqQVxz , and here there is the debug of the call that don't go out http://pastebin.com/kWgrgf7T |
| 17:21.34 | devil_evoxxx | [TK]D-Fender: here is a complete sip debug , core verbose, of the call that go out correctly: http://pastebin.com/iwRqQVxz , and here there is the debug of the call that don't go out http://pastebin.com/kWgrgf7T |
| 17:22.52 | irroot | p3nguin allows you to work back through the code and see how it fits / fitted in |
| 17:23.55 | EmleyMoor | Anyone here heard of yeastar.com? Got spammed twice by them |
| 17:24.36 | navaismo | EmleyMoor they sell crappy analog cards |
| 17:24.39 | [TK]D-Fender | devil_evoxxx, in your peer do "sendrpid=yes" , "trustrpid=yes", "fromuser=yourusernamehere". Apply chanegs. Retest |
| 17:25.59 | devil_evoxxx | on the peer of my phone, or the peer where i connect to my provider |
| 17:26.51 | devil_evoxxx | ? |
| 17:26.53 | [TK]D-Fender | devil_evoxxx, provider |
| 17:26.59 | [TK]D-Fender | devil_evoxxx, Your phone does not matter. |
| 17:27.22 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 17:28.42 | devil_evoxxx | ok. fromuser on my provider.. |
| 17:30.12 | devil_evoxxx | i've never set..trought this provider |
| 17:30.25 | devil_evoxxx | i can send every call with every cid..so i try |
| 17:34.17 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:34.53 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:38.23 | devil_evoxxx | [TK]D-Fender: http://pastebin.com/ZapfE9Ht |
| 17:38.32 | devil_evoxxx | same problem to H3G, work on Vodafone |
| 17:40.02 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 17:40.16 | [TK]D-Fender | devil_evoxxx, Ok, looking at the call that did work VS the ones that failed I'm not seeing any critical difference. Contact your provider ASAP |
| 17:43.32 | devil_evoxxx | thankyou |
| 17:44.40 | *** join/#asterisk Diffen (~diffen@c-f477e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 17:49.20 | *** join/#asterisk Gugge (~gugge@46.30.120.1) |
| 17:55.14 | p3nguin | irroot: I really just want to know what [,CID] means in the current CALLERID function. There's no explanation of what it does and no example of how to use it. |
| 17:55.27 | irroot | yeah |
| 17:55.58 | irroot | sorry been hectic here and am not near my source |
| 17:56.41 | irroot | busy making plan |
| 17:58.04 | [TK]D-Fender | Careful.. life will happen |
| 17:58.32 | *** join/#asterisk timahvo1 (~rogue@41.223.56.17) |
| 17:59.50 | irroot | [TK]D-Fender heard and experiaced the converse of that :P .... hi there |
| 18:03.10 | *** join/#asterisk jkroon (~jkroon@dsl-241-234-78.telkomadsl.co.za) |
| 18:07.44 | SuperNull | sooo the milliwatt app .. are there test tools that work with this ? |
| 18:07.49 | irroot | ok p3nguin the ${CALLERID(....,CID)} will return CID formated as specified |
| 18:09.48 | p3nguin | I still don't get it. |
| 18:09.58 | p3nguin | What do you mean "as specified?" |
| 18:10.13 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:10.39 | p3nguin | If I echo ${CALLERID(num,123)}, it only shows me 123 even if I have valid callerid. |
| 18:12.46 | irroot | http://pastebin.com/8r6kiuvW p3nguin |
| 18:16.07 | p3nguin | What's the purpose of this? |
| 18:16.31 | p3nguin | And why isn't it explained appropriately in core show function CALLERID? |
| 18:16.48 | irroot | p3nguin correct it works on the supplied CID not the actual CID its a function as far as i can tell to process result from a DB query as example to a usable value |
| 18:17.09 | irroot | p3nguin cause programers suck and documenting code !!!! |
| 18:17.13 | p3nguin | The description needs to reflect that. |
| 18:17.28 | irroot | edit it and post it as a patch on JIRA ?? |
| 18:17.36 | p3nguin | I might. |
| 18:17.50 | p3nguin | I can do that. |
| 18:17.53 | irroot | sorry i did not help earlier |
| 18:19.35 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 18:20.25 | navaismo | ~ book |
| 18:20.25 | 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://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 18:21.54 | *** join/#asterisk albertoandrade (~albertoan@177.16.86.172) |
| 18:22.11 | *** join/#asterisk m_tadeu (~quassel@89-180-19-49.net.novis.pt) |
| 18:28.42 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:29.29 | anonymouz666 | if we have distributed device state, we also need distributed wrap uptime |
| 18:29.35 | anonymouz666 | :) |
| 18:32.38 | anonymouz666 | shared_lastcall is fine, but when you have the same queue in two machines... then you got a problem |
| 18:33.42 | [TK]D-Fender | anonymouz666, Bring it back a step.. it'd be nice if 2 queues ont he SAME machine shared wrap-up time for the same agent ;) |
| 18:33.47 | irroot | lol yeah can see that been a pain |
| 18:34.10 | irroot | |
| 18:34.11 | irroot | [TK]D-Fender 2 queues on diff machines :P |
| 18:35.00 | irroot | ok same queue on 2 servers ... |
| 18:35.02 | irroot | that is not possible really as the internal bits are not exposed in dB |
| 18:35.39 | anonymouz666 | irroot: yeap no easy solution |
| 18:35.48 | anonymouz666 | but the agents are about to kill me hehe |
| 18:36.11 | anonymouz666 | they don't have time to put pause to go to the bathroom :` |
| 18:36.15 | anonymouz666 | :( |
| 18:36.36 | irroot | the problem is a "Queue" / ACD is made up of various elements members are only one the call list and then the actual call |
| 18:37.23 | irroot | the queue and members are realtime of sorts as they get read only on a event |
| 18:38.00 | anonymouz666 | irroot: so if I put this queue in a realtime enviroment won't help at all |
| 18:38.15 | irroot | you will need a full realtime system where most importantly the calls will be shared to queue it properly |
| 18:38.48 | anonymouz666 | that sounds... a lot of new code |
| 18:39.07 | *** topic/#asterisk by Qwell -> #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 18:39.11 | Qwell | New security release, 1.8.7.1. |
| 18:39.15 | irroot | so the caller last on server 2 could be answerd before someone waiting 5min on server 1 |
| 18:41.00 | anonymouz666 | irroot: can you think on hack or some workaround that can help on this? |
| 18:41.00 | irroot | Qwell +1 a rc2 due with this also perhaps ?? |
| 18:41.07 | Qwell | irroot: soon |
| 18:41.10 | anonymouz666 | you know, with asterisk, you have to use imagination, sometimes... |
| 18:41.11 | irroot | why use 2 servers ?? |
| 18:41.33 | anonymouz666 | there are too much calls into this queue. |
| 18:41.43 | anonymouz666 | a think that 200 calls is a lot for asterisk to handle. |
| 18:41.55 | anonymouz666 | that I can keep things safe if I spread this |
| 18:41.58 | irroot | well strip the transcoding and telco interface ?? |
| 18:42.19 | anonymouz666 | lots of db queries, recording, no transcoding... |
| 18:43.00 | anonymouz666 | it's about 200 calls from 9am to 6pm. |
| 18:43.01 | anonymouz666 | all the time |
| 18:43.10 | anonymouz666 | just in that queue. there are more. |
| 18:43.25 | irroot | well have seperate agents on each server ? |
| 18:43.33 | irroot | pool A / pool B |
| 18:44.07 | irroot | if you controling the calls to them it can be more fair |
| 18:44.31 | anonymouz666 | the telco does a round robin between the e1s |
| 18:45.13 | anonymouz666 | sometimes the member is local registered |
| 18:45.20 | anonymouz666 | sometimes the members is "remote" |
| 18:45.39 | anonymouz666 | then you spread the queue() between the machines and also the main load |
| 18:47.27 | anonymouz666 | but the "distributed wrap uptime" got me. |
| 18:47.36 | irroot | the problem you will have is io load "wait state" this is from annoucenments and recording |
| 18:48.09 | anonymouz666 | yeap |
| 18:48.32 | irroot | the distributed wrapuptime could add lots of procesing / db queries |
| 18:49.46 | irroot | need some form of network IPC built into app_queue |
| 18:50.15 | anonymouz666 | or a device state wrapuptime :D |
| 18:50.38 | anonymouz666 | that could be used exactly the way distributed device state is used |
| 18:51.24 | irroot | or delayed devicestate |
| 18:51.57 | irroot | i haz to run intresting concepts |
| 18:59.02 | *** join/#asterisk cj (~cjac@208.85.208.53) |
| 18:59.05 | cj | moo |
| 18:59.21 | cj | carrar: so, I just bought an ATA |
| 19:00.19 | SuperNull | what ata did ya get ? We have about 6-7k SPA-2102s deployed. |
| 19:04.27 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:14.09 | pigpen | If I were having issues with call parking. What debug should I turn on? |
| 19:14.42 | pigpen | i.e.; call comes in, via a sip trunk, gets parked, immediately call gets disconnected. |
| 19:14.54 | *** join/#asterisk Tim_Toady (~moi@77.49.252.192) |
| 19:14.57 | [TK]D-Fender | pigpen, verbose, core, SIP, all maxxed |
| 19:15.23 | pigpen | So I am planning on stripping the dial plan way back to nothing. Do the test, grab the output, then put the box back in product. |
| 19:15.27 | pigpen | s/product/production |
| 19:15.29 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:15.33 | pigpen | [TK]D-Fender, tks. |
| 19:15.42 | pigpen | I'll get it going. |
| 19:16.31 | pigpen | it is working fine on my 1.8.7.0 box with my shit dial plan. But I have 6 in the field, in production that is not. |
| 19:34.13 | p3nguin | I'd LOVE to know why I get this on certain calls. It happens as soon as the ringing sound starts and ends when the callee answers the phone: http://pastebin.com/7vUeygEQ |
| 19:35.50 | p3nguin | The phone I am calling uses ulaw. The phone I call from is using ulaw. |
| 19:36.07 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 19:36.52 | p3nguin | I see it when I call a SIP phone and I saw it the other day when calling via google. |
| 19:37.15 | anonymouz666 | 1.8? |
| 19:37.23 | p3nguin | 1.8.7.0 |
| 19:38.52 | anonymouz666 | I don't know how to solve this, but I already saw this. I had an idea to set transcode_via_sln = no in asterisk.conf. I don't know if it is a good idea or no. Not even know if that will help at all |
| 19:39.14 | p3nguin | When trying to call otu google, I see a similar message: [Oct 17 14:38:36] WARNING[11208]: chan_gtalk.c:1606 gtalk_write: Asked to transmit frame type slin, while native formats is 0x4 (ulaw) (read/write = ulaw/ulaw) |
| 19:40.22 | p3nguin | I tried calling my mobile via google voice. That floods the console, and my mobile never rings. |
| 19:40.41 | p3nguin | At least when I see it on SIP I can still get an answer on the other end. |
| 19:42.27 | p3nguin | I changed that setting. I'll see what it does in a minute. |
| 19:42.50 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 19:43.42 | p3nguin | Same thing when calling via google. |
| 19:43.55 | p3nguin | Flood, and never reaches the mobile phone. |
| 19:51.08 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 19:57.54 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 20:15.14 | zamba | anyone got a dinstar gsm gateway here? |
| 20:15.29 | pigpen | [TK]D-Fender, Ok, I got the output. I stripped down the dialplan then set the debug levels, tested, grabbed the output and put back into production. |
| 20:15.52 | pigpen | [TK]D-Fender, should I post the issue again here, using pastebin wisely, or go straight to bug? |
| 20:16.13 | pigpen | dunno if you saw any of my previous conversations or not about this issue. |
| 20:16.31 | [TK]D-Fender | Generic parking issue |
| 20:16.38 | [TK]D-Fender | pastebin. Be complete. |
| 20:16.44 | [TK]D-Fender | Ther previous desciption wasn't |
| 20:16.57 | pigpen | sure. I have typed it so many times, you can imagine. |
| 20:17.02 | pigpen | k. |
| 20:25.15 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 20:28.36 | pigpen | This is my Generic Parking Issue. Asterisk 1.8.7.0 (64bit)(also on 1.8.5) call come in via sip from an Audiocodes FXO, Call is passed directly to a sip client (Polycom 335 HD), Transfer is initiated (using ##, normal xfer attempted, no change), to exten 70 (parking). Parking answers and notes "71", then the call transferee call is dropped. The caller then is dropped shortly after that. Please see config and debug here: http://pastebin.com/ZTmnbJkS |
| 20:30.15 | [TK]D-Fender | Heading home, BBIAB |
| 20:30.23 | [TK]D-Fender | pigpen, save up to PB me in a bit |
| 20:30.43 | pigpen | pb you in a bit? |
| 20:34.37 | *** join/#asterisk mKn0wt (~mtv@190.111.30.11) |
| 20:36.18 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xnvzkqmecbtktsnh) |
| 20:36.51 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
| 20:37.14 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 20:37.25 | dandate2 | this /proc/kcore takes up a whole 1 gigabyte of space, is it ok to delete this much like /var/log/asterisk/full ? |
| 20:38.43 | Tim_Toady | lol |
| 20:39.09 | pigpen | you can. I would stop asterisk, move it, touch a new one, make sure permissions are the same, then start it back up. |
| 20:39.17 | pigpen | then you can gzip the old, just in case. |
| 20:39.27 | Tim_Toady | dandate2 halt ur pc and remove all ram, that will make kcore vanish |
| 20:40.22 | Tim_Toady | you cant move kcore, you cant delete it and anyway its not taking anyspace in ur disk or ram |
| 20:40.36 | Tim_Toady | its an 'alias' for ur systems memory |
| 20:40.44 | pigpen | haha, wait, I was referring to the /var/log/asterisk/full |
| 20:40.48 | pigpen | haha, not kcore |
| 20:40.51 | Tim_Toady | lol |
| 20:40.58 | pigpen | haha, thats' what I get for not reading it all... |
| 20:41.04 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:41.50 | dandate2 | is /var/log/asterisk/queue_log is another massive, ok to delete? |
| 20:42.19 | navaismo | use logrotate |
| 20:42.33 | pigpen | sure, any log fine is ok to move to the side. there may be a easier way than I noted, but it will work, and yes, logrotate. |
| 20:42.55 | dandate2 | ah the log rotate is broken on my flash install, i have to delete it manually every few days |
| 20:43.36 | pigpen | oh then you=logrotate |
| 20:45.58 | Tim_Toady | dandate2 in that case write a cron job that runs asterisk -rx 'logger roate' |
| 20:46.10 | Tim_Toady | poor mans lograotate :P |
| 20:46.32 | Tim_Toady | rotate* |
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| 21:49.30 | pigpen | Quiet day in asterisk land |
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| 22:05.47 | infinity_ | can someone provide a few pointers. i have an asterisk box connecting to a sip phone on the lan. there is a long delay 15-25 seconds before the phone connects. any ideas why that might be? |
| 22:06.43 | dym | what do you mean before the phone connects? |
| 22:06.46 | dym | from bootup? |
| 22:06.55 | infinity_ | no. when it calls another extension. |
| 22:08.07 | infinity_ | after a call is placed, once its answered and the ringing stops, it takes about 15 seconds for it to work |
| 22:08.42 | infinity_ | er 15 seconds for the call to work correctly (two way audio instead of one way) after the call is answered |
| 22:32.32 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-190-50.hsd1.va.comcast.net) |
| 22:34.32 | jstapleton | anybody having problems with google voice outbound? inbound is working fine? did google change the protocol AGAIN? |
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| 22:44.01 | r0m|u | jstapleton, yes they made c ahcnge |
| 22:44.09 | r0m|u | a change* |
| 22:44.36 | r0m|u | oubound works for me. inbound does not. |
| 22:45.11 | jstapleton | r0m|u: any idea when they mad the change? |
| 22:45.27 | jstapleton | r0m|u: as far as you know, is there a patch/fix yet? |
| 22:45.57 | r0m|u | yes there is |
| 22:46.04 | r0m|u | and it has been quite a few days |
| 22:46.08 | r0m|u | one |
| 22:46.10 | r0m|u | sec |
| 22:46.28 | r0m|u | jstapleton, http://www.dslreports.com/forum/r26425026-Asterisk-GV-call-in-asterisk-does-not-work-again |
| 22:47.03 | jstapleton | r0m|u: thanks! |
| 22:47.04 | *** join/#asterisk neurosys (~neurosys@92.61.176.62) |
| 22:47.16 | r0m|u | jstapleton, your welcome |
| 22:47.43 | r0m|u | Thats why I dont use beta services :P |
| 22:48.07 | r0m|u | I use GV just for outgoing with a CID set to my main number. |
| 22:48.32 | r0m|u | if people call me back they call my voipms account :) |
| 22:50.13 | blizzow | Does anyone have recommendations on getting a browser based click to call plugin to work with PBX in a flash (and queuemetrics)? |
| 22:52.20 | r0m|u | blizzow, /join #pbxinaflash maybe they can help better? |
| 22:52.40 | blizzow | I didn't realize there was a pbxinaflash room. |
| 22:53.03 | r0m|u | :) |
| 22:53.16 | blizzow | It's still based on asterisk, though. I figured I could ask in here and see if anyone had some ideas. |
| 22:53.39 | r0m|u | I think this rooms only supports asterisk and asterisk only :) |
| 22:53.39 | blizzow | especially considering #pbxinaflash has 6 people (including me). |
| 22:53.52 | r0m|u | ~FreePBX |
| 22:53.52 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 22:54.12 | r0m|u | I think pbx in a flash goes along those lines |
| 22:54.31 | r0m|u | though I might be wrong |
| 22:54.48 | r0m|u | stick around and see if any of the gurus can help |
| 22:55.42 | r0m|u | ~pbxinaflash |
| 22:55.42 | infobot | hmm... pbxinaflash is Ward Mundy's toy assembled by joe roper visit pbxinaflash.org or #pbxinaflash |
| 22:56.05 | r0m|u | blizzow, ^^ :) |
| 23:06.32 | pabelanger | jstapleton: r0m|u: Confirm the patch in https://issues.asterisk.org/jira/browse/ASTERISK-18301 works and I'll commit it |
| 23:08.26 | r0m|u | pabelanger, I cant I am on an embedded system with ro to the fs. |
| 23:08.46 | r0m|u | as soon as I get a chance ill bring up a vm |
| 23:10.18 | jstapleton | pabelanger: worked for me. |
| 23:12.14 | navaismo | evening people, anyone has used the appkonference module with asterisk10 to swap in video conferences? |
| 23:13.57 | pabelanger | Naikrovek: Have you tried ConfBridge in asterisk 10? It now has support for video conferences |
| 23:14.17 | pabelanger | Naikrovek: sorry |
| 23:14.19 | pabelanger | navaismo: ^ |
| 23:14.30 | jstapleton | pabelanger: i am on 1.8.6.0. Is patch need on Asterisk 10 as well? I assume so. |
| 23:14.37 | navaismo | aah excelent i just dont know that, let me try thx pabelanger |
| 23:15.00 | pabelanger | jstapleton: yes, all branches will need to be patched |
| 23:15.51 | pabelanger | navaismo: Ya, it was basically rewritten so any existing configuration files will not work. But the video support is very cool |
| 23:16.39 | navaismo | \o/ thx again pabelanger |
| 23:20.31 | phix | any suggestions on VoIP / SIP phones? |
| 23:20.54 | phix | I have a snome 300 atm, it works great but it lacks more user defined buttons |
| 23:21.42 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 23:21.53 | phix | and the LCD is too small, doesn't fit everything in when using directory / address book |
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| 23:45.19 | pabelanger | Can't go wrong with Polycoms |
| 23:46.41 | dym | Or Linksys SPA |
| 23:46.42 | Naikrovek | ^^^ |
| 23:48.30 | Naikrovek | how does the video support work in confbridge in 10? does it tile the other participants? are there any screenshots? |
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