| 00:04.46 | alunca | man, after I reboot, it not saved my setting |
| 00:05.24 | Psychobilly | reboot what? |
| 00:05.33 | Psychobilly | ur pc/server? |
| 00:05.52 | Psychobilly | how is ur network configured? u use dhcp? |
| 00:05.55 | alunca | I saved, then type "reboot" |
| 00:06.09 | Psychobilly | there was no need for reboot |
| 00:06.14 | alunca | static local ip, but something wrong with dns |
| 00:07.41 | alunca | Psychobilly if I don't reboot, which command should I use to load the new setting of dns? |
| 00:08.39 | Psychobilly | thers no need for any comamnd, since u got resolv.conf right name resolution should work |
| 00:08.59 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 00:09.11 | alunca | not working for me, cannot ping google.com nor nslookup |
| 00:15.34 | jaytee | firewall? |
| 00:18.47 | alunca | not sure, it works off and on hehe |
| 00:19.11 | alunca | anyway, which software voIP client should I use to test out my asterisk + gv ? |
| 00:19.33 | alunca | 3cx voip client always no reponding ... |
| 00:21.40 | Psychobilly | ekiga, xlite , kphone, zoiper |
| 00:21.59 | Psychobilly | there are lots of soft phone (all bad quality) |
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| 01:07.04 | manxpower | All softphones suck. xlite seems to suck less. |
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| 01:20.32 | drfreeze | Hello |
| 01:20.53 | drfreeze | I'm building an 1.4.26 version of asterisk using a Digium TE121 card |
| 01:21.10 | drfreeze | Do I need to install asterisk with the TE card installed in the system, or can I add that later? |
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| 01:38.09 | manxpower | as long as you install Zaptel/DAHDI and (if using a PRI) libpri before you build Asterisk, you should fine. |
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| 02:06.52 | drfreeze | manxpower: thanks |
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| 02:10.35 | drfreeze | manxpower: I'm using 1.4.26 |
| 02:10.47 | drfreeze | should I install zaptel 1.4 or dahdi 2.0.0 |
| 02:12.36 | thansen | how can I send multiple options to the EXEC command in AGI? instructions from here don't work.. http://www.voip-info.org/wiki/view/exec |
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| 02:33.42 | manxpower | drfreeze: that is totally up to you |
| 02:34.42 | lmadsen | use DAHDI |
| 02:34.50 | lmadsen | zaptel isn't updated anymore, and you might as well just do it now |
| 02:34.56 | lmadsen | instead of having to learn how to convert later |
| 02:44.31 | drfreeze | lmadsen: I assume it is stable |
| 02:45.03 | drfreeze | I don't understand why they had to rewrite zaptel. Couldn't they have just changed the name? |
| 02:46.02 | lmadsen | drfreeze: it's not entirely re-written. The configuration files look almost the same. |
| 02:48.17 | lmadsen | svn co http://svn.asterisk.org/svn/dahdi/tags/2.2.0+2.2.0.2 ; cd 2.2.0+2.2.0.2 ; make all ; make install ; make config |
| 02:56.49 | drfreeze | lmadsen: was the make config meant to be literal or is it a function of getting dahdi from svn: make: *** No rule to make target `config'. Stop. |
| 02:57.20 | lmadsen | make config is just the same as asterisk -- installs the init script if you are running a system that supports that |
| 02:57.54 | lmadsen | hmmm.... thought dahdi had a make config as well... has been a while |
| 02:58.09 | lmadsen | my svn link may be wrong too -- it could be dahdi-linux-complete -- I can't remember. Checking |
| 02:58.16 | lmadsen | (I just did all that from memory) |
| 02:58.24 | drfreeze | I pulled the complete down |
| 02:58.31 | drfreeze | wget http://downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz |
| 02:58.35 | drfreeze | current, I mean |
| 02:58.48 | drfreeze | wget http://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-current.tar.gz |
| 02:58.56 | lmadsen | working link: http://svn.asterisk.org/svn/dahdi/linux-complete/tags/2.2.0.2+2.2.0/ |
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| 02:59.24 | drfreeze | lmadsen: make config works with the dahdi tools |
| 02:59.42 | lmadsen | drfreeze: http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz |
| 03:00.05 | lmadsen | get both at the same time with dahdi-linux-complete |
| 03:00.14 | drfreeze | oh |
| 03:00.42 | lmadsen | anyways, I'm done working now. I was just killing time here while I was waiting for things to install and complete. Peas out. |
| 03:07.46 | *** join/#asterisk M-I-A (n=chacha@207.35.50.210) |
| 03:08.38 | M-I-A | Anyone know of any conflicts with a Rhino R1T1 and a Sangoma A102 in the same system? |
| 03:10.54 | jaytee | no, but it wouldn't be surprising since they are bitter enemies |
| 03:12.03 | M-I-A | :) |
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| 03:17.15 | M-I-A | The thing I thought was weird was that I couldn't find any error messages and ztcfg configured all the spans |
| 03:18.02 | shinao1 | hi i was wondering.. i want to be able to setup dundi between several xorcom units, all running elastix.. is it possible to setup dundi between several elastix/asterisk boxes all having the same extension type number plan? i have one particular site that will have 300+ extensions, and i have about 12 sites to setup. i'm hoping to have the same extension plan (1XX) at all sites to simplify things. how can i go about it? |
| 03:20.54 | shinao1 | is it even possible? |
| 03:23.28 | M-I-A | shinao1 your light years beyond my scope :) |
| 03:24.08 | shinao1 | more fool me then.. dabbling into things "beyond" my ken |
| 03:24.15 | shinao1 | :) |
| 03:31.05 | *** join/#asterisk geneticx (n=ethicx@adsl-10-114-195.mia.bellsouth.net) |
| 03:31.27 | geneticx | sup you all. |
| 03:32.46 | M-I-A | geneticx not much just trying to get a rhino and sangoma to play nice together |
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| 03:33.40 | geneticx | M-I-A: hummm...interesting. |
| 03:34.47 | drfreeze | What's the make command to install asterisk as a service? |
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| 03:40.03 | alunca | whenever I start up my AsteriskNow, it won't access to online until I typed "system-config-network" and change something, and saved it. Is there anyway to fix this issue? |
| 03:40.37 | spiri | Im going to ask a weak question.. maybe someone can tell me what I should be searching for in google.. Im trying to accept incoming calls to extension@mydomain.org however Its not finding any extensions.. Im sure its something basic. any hints? |
| 03:41.03 | spiri | ok |
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| 04:29.59 | drfreeze | <PROTECTED> |
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| 04:30.23 | drfreeze | Does Asterisk work well with an AMD chip/mb? |
| 04:30.45 | drfreeze | I usually buy Intel, but am looking to save a bit of money on a new system |
| 04:42.42 | manxpower | alunca: ask on #AsteriskNOW |
| 04:42.56 | manxpower | drfreeze: Modern AMD should work just fine |
| 04:45.06 | alunca | my /etc/resolv.conf << alwasys reset when network restart ... please help |
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| 05:55.59 | shinao1 | hi i was wondering.. i want to be able to setup dundi between several xorcom units, all running elastix.. is it possible to setup dundi between several elastix/asterisk boxes all having the same extension type number plan? i have one particular site that will have 300+ extensions, and i have about 12 sites to setup. i'm hoping to have the same extension plan (1XX) at all sites to simplify things. how can i go about it? is it even posssible? |
| 06:05.38 | alunca | anyone using 3cx voip client? |
| 06:05.47 | alunca | somehow, my incoming call stop show up ... |
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| 07:27.39 | alunca | which voip software is good to use with computer to test out/in calling? thank you! |
| 07:30.53 | kron4eg | any softphone? |
| 07:33.07 | alunca | yeah, i failed to use 3cx voip, only can dial out, however, when I have a call in, it not show oup (or pop-up) ... but the asterisk GUI show incoming call active |
| 07:33.19 | alunca | so I want to try other sip softphone |
| 07:34.58 | TommyBotten | Which operating system are you using=? |
| 07:35.05 | alunca | winxp pro |
| 07:35.22 | TommyBotten | For linux Ekiga, sflphone and wengophone are good alternatives |
| 07:35.24 | TommyBotten | ah |
| 07:35.34 | WindowsUser | ekiga, x-lite? |
| 07:36.16 | alunca | just un-install x-lite. |
| 07:37.01 | alunca | somehow, I cannot received incoming call even the asterisk web GUI shows that incoming is calling in ... any help please? |
| 07:38.17 | WindowsUser | does asterisk web gui mean theres a bunch of messy includes in your configuration? |
| 07:38.34 | alunca | no |
| 07:38.59 | kaldemar | alunca: your soft phone cannot reveice a call or asterisk cannot receive a call? |
| 07:39.16 | WindowsUser | maybe you didn't set up incoming calls correctly? asterisk has to know to send calls to your softphone or wherever |
| 07:42.23 | *** join/#asterisk Psychobilly (n=moi@adsl282-123.kln.forthnet.gr) |
| 07:43.38 | alunca | I don't know what is going on; it was working hehehe then I lost the incoming call |
| 07:45.26 | dandre | Hello, |
| 07:45.52 | dandre | I have a problem with queues, see here please: http://pastebin.fr/5268 |
| 07:49.31 | *** join/#asterisk Hatrix (n=Hatrix@213.201.24.127.static.user.ono.com) |
| 07:51.45 | alunca | reboot |
| 07:54.01 | kaldemar | dandre: you don't have a single line in default context. macro calls in from-internal have erroneous syntax. those are your first problems. please be more specific on the "steps" you refer to. |
| 07:54.41 | Hatrix | hi guys, i have lot's of troubles with a medium (or rather small) sized call-center (around 50-70 agents). The asterisk system is either dead-locking or segfaulting on a regular base and my client is not happy about this. We used 1.4.21*, had problems, swiched hardware, then used 1.4.26-rc*, 1.4.26 to no avail, segfaults and dead-locks ... we kill asterisk once in the night, we use caches to reduce manager access from scripts, i think |
| 07:55.05 | Hatrix | (i forgot to mention that I try to reproduce those problems in the lab but it's rather difficult) |
| 07:56.18 | dandre | kaldemar: sorry for the erroneous syntax, |
| 07:56.31 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:56.44 | kaldemar | dandre: and show a failed call show we know what really happens |
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| 07:58.57 | MACscr1 | wow, I swear, all the softphones out there are junk |
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| 08:00.12 | dandre | the problem is that the Dial(Local/950@from-internal/n) seems to be answered even if the queue timesout so the next step isn't reached |
| 08:00.31 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:02.40 | xrmx__ | Hatrix, not specific sorry, but as a general rule if you backtraces of segfaults looks for similar issues in the bug tracker |
| 08:03.04 | kaldemar | dandre: well.. "exten = 950,n,Answer()" |
| 08:04.05 | Hatrix | xrmx__: my problem is, i'l tried this, but at times tha crashes are to random to relate, the only one which is actually being discussed in 1.4.26 right now is the crash with app_queue, which versions of asterisk are in production use out there for call-centers, anybody knows? |
| 08:06.37 | dandre | ok kaldemar I will try without it but I am afraid the queue won't be played. |
| 08:07.35 | kaldemar | dandre: you could replace the whole [app-queues] with three lines that are called with GoSub from [kwin-100007] |
| 08:10.02 | kaldemar | dandre: or a single three line extension that matches all your queues if you want to use dial |
| 08:10.09 | *** part/#asterisk [gnubie] (n=bintut@bb219-74-72-154.singnet.com.sg) |
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| 08:11.15 | dandre | I don't understand all what you mean |
| 08:14.48 | dandre | the problem if I don't answer the call before the queue cmd is that if called from a Zap line the queue MOH isn't played |
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| 08:49.41 | TSM | IAX2 from what i can see possably suffers from one thing, because it runs all over one port it cant utilise muliple paths over the ISP to increase bandwidth when running over long hauls US-UK, but on the otherhand as its all UDP i guess thats not an issue whereas TCP traffic gets realy realy slow |
| 08:50.28 | *** join/#asterisk redax (i=redax@r6.hu) |
| 08:50.30 | redax | hi |
| 08:50.51 | *** join/#asterisk Pouet78 (i=56dcdc7b@gateway/web/freenode/x-e57c7b145133217a) |
| 08:51.09 | Pouet78 | Hi! |
| 08:51.34 | redax | how can I specify the reregistration time on a sip trunk (type=peer) ? registerseconds and defaultexpirey seems to be not working |
| 08:51.51 | Pouet78 | I have a problem with MGCP. |
| 08:52.16 | Pouet78 | I have a box registering as MGCP client |
| 08:53.12 | *** join/#asterisk MACscr (n=Mark@98.214.100.212) |
| 08:53.12 | Pouet78 | Asterisk is replying a 200 OK but I have no tone :( |
| 08:53.20 | MACscr | how do I empty my voicemail box? |
| 08:53.33 | Pouet78 | any suggestion? |
| 08:53.44 | TSM | Pouet79: over WAN or LAN? |
| 08:54.02 | Pouet78 | WAN |
| 08:54.11 | Pouet78 | I have to test this box |
| 08:54.13 | TSM | have you opened the RTP ports |
| 08:54.15 | WindowsUser | MACscr: delete the files in it? rm /var/spool/asterisk/voicemail/context/mailbox/*/* |
| 08:54.23 | Pouet78 | I have a DSLAM and an asterisk behind |
| 08:54.40 | TSM | you mean you have a firewall and asterisk behind? |
| 08:55.14 | Pouet78 | no I don't have any firewall |
| 08:56.15 | TSM | ok so asterisk is directly on the WAN, are you running IPTables? |
| 08:56.27 | Pouet78 | I have : Phone -> FXS BOX FXO -> DSLAM -> Asterisk |
| 08:57.07 | TSM | so where is the WAN Internet Cloud in this? |
| 08:57.40 | TSM | anyway have you checked that IPTables is not blocking the RTP ports, usualy 10000-20000 |
| 08:58.21 | Pouet78 | When I do tcpdump, I have : "MGCPRSIP 804289384 aaln/*@NDITEST199.IZP.DGP.NEUF.COM MGCP 1.0" |
| 08:58.32 | Pouet78 | and the answer : "200 804289384 OK" |
| 08:58.41 | Pouet78 | But then nothing else. |
| 08:59.13 | TSM | im not to up on MGCP over ASterisk, im all SIP, easier |
| 08:59.13 | Pouet78 | the "VoIP" LED on the BOX is on but I have no tone |
| 08:59.38 | Pouet78 | I have no choice, I have to test this box doing MGCP |
| 09:00.20 | Pouet78 | My Asterisk is already ok with SIP and H323 (through gnugk) |
| 09:01.13 | Pouet78 | It could help me if I can have some traces of a correct MGCP traffic... |
| 09:02.28 | *** join/#asterisk stephan_n (n=nachtshe@77-21-104-180-dynip.superkabel.de) |
| 09:03.12 | stephan_n | Hallo |
| 09:03.53 | stephan_n | ich suche Hilfe bei einem Problem mit Asterisk. Es geht um die Anbindung einer MySQL-Datenbank zur Speicherung der sip.conf Einträge |
| 09:04.46 | stephan_n | ich habe in der extconfig.conf folgende Zeile eingetragen: |
| 09:04.50 | stephan_n | sip.conf => mysql,asterisk,ast_config |
| 09:05.17 | stephan_n | beim Einloggen in der Konsole steht dort auch: == Binding sip.conf to mysql/asterisk/ast_config |
| 09:05.41 | stephan_n | ich habe die Einträge bindaddr und port in die Tabelle ast_config eingetragen |
| 09:06.01 | TSM | english |
| 09:06.13 | stephan_n | oh, i'm sorry |
| 09:06.18 | stephan_n | sure |
| 09:06.43 | stephan_n | i tried to use mysql instead of sip.conf |
| 09:07.28 | stephan_n | but asterisk does not load the entries from my table |
| 09:08.24 | stephan_n | where can i find the table schema? |
| 09:08.35 | stephan_n | for the configuration tables |
| 09:09.11 | stephan_n | in the docs i only found examples for sip_users and the cdr-stuff |
| 09:09.37 | stephan_n | but i want to store the [general]-stuff from sip.conf in the database |
| 09:10.09 | stephan_n | my table looks like |
| 09:10.10 | stephan_n | id, cat_metric, var_metric, commented, filename, category, var_name, var_val |
| 09:10.10 | stephan_n | right now |
| 09:10.54 | TSM | did you setup cdr_mysql.conf? |
| 09:11.02 | stephan_n | and my entries are |
| 09:11.02 | stephan_n | 1, 0, 0, 0, sip.conf, general, port, 5060 |
| 09:11.02 | stephan_n | 2, 0, 1, 0, sip.conf, general, bindaddr, 0.0.0.0 |
| 09:11.19 | *** join/#asterisk Breaking_Pitt (n=pgarcia@193.147.51.104) |
| 09:11.23 | TSM | did you follow http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql |
| 09:11.25 | stephan_n | i setup cdr_mysql_conf and res_mysql.conf |
| 09:11.31 | stephan_n | cdr works fine |
| 09:11.44 | stephan_n | and also the sip users are loaded from database |
| 09:12.35 | stephan_n | extconfig.conf: "sipusers => mysql,asterisk,sip_users" works fine |
| 09:12.54 | stephan_n | extconfig.conf: "extensions => mysql,asterisk,extensions" works fine |
| 09:13.11 | stephan_n | extconfig.conf: "sip.conf => mysql,asterisk,ast_config" doesnt work |
| 09:13.28 | TSM | mabey it does not do it |
| 09:13.29 | kaldemar | ~book |
| 09:13.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 09:13.30 | stephan_n | but i get "== Binding sip.conf to mysql/asterisk/ast_config" on console |
| 09:13.32 | kaldemar | stephan_n: ^^ |
| 09:14.25 | TSM | i gave wrong link. look at http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime |
| 09:14.27 | stephan_n | i have this book already opened ;-) |
| 09:14.49 | stephan_n | yes i read this link on voip-info too |
| 09:15.05 | TSM | did you read this one specificly for sip http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip i guess you did |
| 09:15.10 | stephan_n | especcially http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static |
| 09:15.24 | stephan_n | sip works |
| 09:15.37 | stephan_n | but the stuff here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Static doesnt |
| 09:16.04 | stephan_n | i created my table with the table-structure listed there |
| 09:16.48 | stephan_n | sadly the link to the ast2sql.pl doesn't work |
| 09:16.58 | TSM | "NOTE: If you store sip.conf in the RealTime database, you need to rename/remove the text file otherwise the text file will superceed RealTime. " |
| 09:17.07 | stephan_n | so i inserted the rows for sip.conf at my own |
| 09:17.21 | stephan_n | i renamed the sip.conf to sip.conf.bak already |
| 09:17.29 | TSM | good boy :) |
| 09:17.42 | kaldemar | TSM: that note is inaccurate |
| 09:17.50 | TSM | good girl :) |
| 09:17.56 | kaldemar | or even plain wrong |
| 09:18.15 | stephan_n | ^^ |
| 09:19.17 | kaldemar | at least with 1.6 and odbc it doesn't apply. if sip.conf is defined in extconfig.conf, the text file is always ignored. |
| 09:19.20 | stephan_n | mh, i think my configuration should be ok, because of the " == Binding sip.conf to mysql/asterisk/ast_config" at console |
| 09:19.34 | stephan_n | i use 1.4 from debian lenny |
| 09:19.57 | stephan_n | but maybe the table-structure isn't right |
| 09:20.09 | stephan_n | but i didn't find anything in the official docs |
| 09:20.26 | stephan_n | from the packet asterisk-doc from the debian repository |
| 09:20.53 | stephan_n | there are only examples for the sip-users and extensions-table |
| 09:21.44 | kaldemar | i use http://pastebin.com/m2fea402e and it works fine. |
| 09:22.18 | stephan_n | ok, thanks. i will try this right now |
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| 09:26.29 | stephan_n | ok, now i'm using your table structure with the following inserts |
| 09:26.29 | stephan_n | INSERT INTO `asterisk`.`ast_config` ( `id` , |
| 09:26.29 | stephan_n | <PROTECTED> |
| 09:26.29 | stephan_n | <PROTECTED> |
| 09:26.29 | stephan_n | <PROTECTED> |
| 09:26.30 | stephan_n | <PROTECTED> |
| 09:26.32 | stephan_n | <PROTECTED> |
| 09:26.34 | stephan_n | <PROTECTED> |
| 09:26.36 | stephan_n | <PROTECTED> |
| 09:26.38 | stephan_n | VALUES ( NULL , '0', '0', 'sip.conf', 'general', 'bindaddr', '0.0.0.0', '0'), ( NULL , '0', '1', 'sip.conf', 'general', 'port', '5060', '0'); |
| 09:26.41 | stephan_n | but it doesn't work |
| 09:26.57 | stephan_n | but i switched on mysql-logging |
| 09:27.24 | stephan_n | and looked at "tail -f /var/log/mysql/mysql.log" while i'm restarting the asterisk |
| 09:27.58 | stephan_n | an there are only two connections withour any queries! |
| 09:28.25 | stephan_n | i would expact, that asterisk should load the ast_config after restart |
| 09:28.35 | stephan_n | something like "SELECT * FROM ast_config" |
| 09:28.39 | stephan_n | but nothing! |
| 09:30.00 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:30.04 | stephan_n | so, why did asterisk no single query to the database table "ast_config" at startup? |
| 09:30.16 | stephan_n | is there any configuration-line missing anywhere? ;-) |
| 09:31.05 | stephan_n | i'm still wondering about the console, because it says: " == Binding sip.conf to mysql/asterisk/ast_config" |
| 09:33.20 | stephan_n | can i get more debug messages from res_config_mysql.c anyhow? |
| 09:36.20 | stephan_n | @kaldemar: what asterisk-version are you using? where did you change the configuation for mysql? I only changed extconfig.conf, res_mysql.conf and cdr_mysql.conf (and of course the rename of sip.conf) |
| 09:36.55 | dandre | how can I know, from my dialplan, if a particular extension exists in a given context? |
| 09:42.46 | dandre | I have found this: http://www.voip-info.org/wiki/view/Asterisk+func+dialplan_exists |
| 09:43.09 | dandre | but is there any way to do the same in asterisk 1.4? |
| 09:43.19 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) |
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| 09:49.08 | HenrikBe | Hi, I have some problems with AJAM/Rawman, I can login via an http-request but cannot do any commands after that. The response is "Authentication required". There is a valid cookie created from the login so it shouldn't be any problem. Anyone have any ideas? |
| 09:50.06 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 09:52.48 | stephan_n | when i execute "realtime load sip.conf filename sip.conf" at console, asterisk does "SELECT * FROM ast_config WHERE filename = 'sip.conf'" |
| 09:55.09 | czindy | Hello. I have a problem with 2 asterisks interconnection via iax. Here is a detailed desc: http://pastebin.com/m9365a45 (when dial server 2 from 1 it is ok but when dial 1 from 2, 1 writes out: Host 192.168.5.8 failed to authenticate as 230) Could you help please. thank you. |
| 09:58.25 | WindowsUser | trunk=yes? |
| 09:59.35 | *** join/#asterisk mchou_ (n=quassel@unaffiliated/mchou) |
| 10:05.07 | czindy | Nos :( I removed the line and the result is the same |
| 10:05.13 | czindy | Host 192.168.5.8 failed to authenticate as 233 |
| 10:06.01 | czindy | on server1: |
| 10:06.01 | czindy | SENDOUT 192.168.5.8 (S) 255.255.255.255 4569 OK (5 ms) |
| 10:06.01 | czindy | on server2: |
| 10:06.01 | czindy | SENDOUT 192.168.5.6 (S) 255.255.255.255 4569 (T) OK (24 ms) |
| 10:19.39 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 10:24.18 | WindowsUser | (T) means it still sees itself as a trunk |
| 10:25.58 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 10:26.28 | czindy | hmm I restated the servers |
| 10:26.51 | czindy | now they are nottrunk and the authentication is still failed |
| 10:28.57 | czindy | Now I tried an other way (user and peer) and with the same result: |
| 10:28.59 | czindy | http://pastebin.com/m44ef9c2 |
| 10:32.08 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
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| 10:47.16 | zeeesh | can we make gtalk realtime like sip and extensions and voicemail already in a state of realtime ? |
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| 12:08.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:11.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:24.21 | jkroon | what's the chances that SendFAX and ReceiveFAX can detect rotated pages and handle them? |
| 12:24.40 | *** join/#asterisk Alfio (n=amunoz@75.112.88.200.m.sta.codetel.net.do) |
| 12:29.37 | [TK]D-Fender | jkroon: Zero |
| 12:30.14 | [TK]D-Fender | jkroon: What are you going to do to accomplish this? You'd need something like OCR. That that comes included in that puny receive app? |
| 12:30.32 | jkroon | returns to trying to detect it beforehand in the tiff and pre-rotating. |
| 12:32.06 | jkroon | [TK]D-Fender, no, page resultion. |
| 12:32.06 | jkroon | identify picks out the per-page resolution out of a tiff file. |
| 12:32.06 | jkroon | if x > y then it's landscape, so rotate. |
| 12:32.09 | dandre | is there any way to know if the stack used by gosub/return is empty? |
| 12:32.20 | jkroon | what typically happens at the moment is that I use openoffice to convert documents to pdf, and impress stuff ends up being landscape. the output resolution from gs command is correct, but it's rotated. |
| 12:33.00 | [TK]D-Fender | dandre: highly doubt it. why would you need to? |
| 12:33.50 | dandre | ok |
| 12:34.13 | dandre | still with my queue problem I told you yesterday |
| 12:34.33 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 12:34.44 | [TK]D-Fender | dandre: Which? |
| 12:36.09 | dandre | http://pastebin.fr/5270 |
| 12:39.24 | [TK]D-Fender | dandre: Where do I see your macro? Or the CLI output? Testing the var? |
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| 13:14.31 | dandre | [TK]D-Fender: ok this was some cleanup of my whole testing conf. I am posting more complete informations |
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| 13:26.50 | dandre | [TK]D-Fender: http://pastebin.fr/5271 |
| 13:27.10 | *** join/#asterisk czindy (n=czindy@91.120.28.147) |
| 13:27.45 | [TK]D-Fender | dandre: Where is MyDial? |
| 13:27.56 | [TK]D-Fender | dandre: You just showed me 2 completely different things |
| 13:28.03 | [TK]D-Fender | dandre: And took 3/4h to do it |
| 13:28.31 | ceegee | hello there |
| 13:28.53 | dandre | the first was some sumary with things that didn't work |
| 13:29.00 | [TK]D-Fender | dandre: AND... exten = 0979949719,1,Set(__NOHANGUP="1") |
| 13:29.10 | [TK]D-Fender | darYou don't put quotes on vars like that. |
| 13:29.16 | [TK]D-Fender | dandre: You don't put quotes on vars like that. |
| 13:29.36 | dandre | the second is the full working version on my sys with stuff that is not directly related to my problem |
| 13:29.48 | [TK]D-Fender | dandre: exten => s,n(next) ,GotoIf($["${NOHANGUP}"="1"]?exit) <--- if the quotes are PART of it like that you get DOUBLE QUOTEs |
| 13:30.07 | [TK]D-Fender | dandre: DON'T PUT QUOTES WHEN SETTING YOUR VARIABLES |
| 13:30.25 | [TK]D-Fender | dandre: It becomes part of the content |
| 13:31.08 | ceegee | we have a problem with call transfer after pickup. we user asterisk 1.6.0.10 and snom320 with firmware 7.3.23. key is programmed as blf, e.g. 21|*8. if I do a call pickup and then want to transfer the call to another phone, the call hangs on hold after hook up, if I do the transfer without a pickup first this works. I dont know why it doesnt work with a pickup first. |
| 13:32.28 | dandre | ok I'll removes quotes, do you think this can hangup the call prematurally? |
| 13:32.40 | *** join/#asterisk dymaxion (n=dymaxion@78-86-174-224.zone2.bethere.co.uk) |
| 13:34.16 | czindy | Hello again, I still have problem with connect 2 asterisk (http://pastebin.com/m448ce460) if I dial Server 2 from 1 it is ok, but reverse there is an error: Host 192.168.5.8 failed to authenticate as 231. Could you help please, you can find the config in the pastebin. thank you |
| 13:35.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:35.26 | czindy | Why it is trying authenticating as 231? |
| 13:35.35 | [TK]D-Fender | dandre: and you should have a single underscore, not a double |
| 13:35.44 | [TK]D-Fender | dandre: "_" not "__" |
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| 13:36.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:38.07 | [TK]D-Fender | czindy: When you call from 1 to the other, your peer names & usernames don't match <- |
| 13:38.20 | [TK]D-Fender | czindy: You use 6 the 6 peer and there is no 6 on the receiving side |
| 13:40.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 13:41.41 | czindy | I register Server2 As Asterisk8 on 192.168.5.6: register => Asterisk8:Sattaratta@192.168.5.6 |
| 13:41.53 | czindy | and there is an Asterisk8 context |
| 13:42.12 | [TK]D-Fender | czindy: But the PEer you call with is important. |
| 13:42.16 | czindy | Could you explain please a little more detailed. |
| 13:42.36 | [TK]D-Fender | czindy: iJust because you register properly is irrelevent. Registration does not AUTH your calls |
| 13:42.48 | dandre | if I don't put two underscore, the variables are not set in stdexten |
| 13:43.04 | [TK]D-Fender | czindy: Use a peer named [fred] and there had BETTER be a[fred] on the other side to match it |
| 13:43.23 | dandre | I still have my call being prematurally hungup |
| 13:44.35 | [TK]D-Fender | dandre: Executing [s@macro-stdexten:1] NoOp("Local/50@from-internal-1f05,2", "1") in new stack |
| 13:44.46 | [TK]D-Fender | dandre: exten => s,1,Noop(${FROMQUEUE}) |
| 13:44.54 | [TK]D-Fender | dandre: Ok, seems to match |
| 13:44.54 | czindy | May I ask you to correct please my config in pastebin please. It would be a great help. |
| 13:45.15 | [TK]D-Fender | czindy: You need the same user on both systems <---- |
| 13:46.23 | *** part/#asterisk rseste (i=c93f9ac2@gateway/web/freenode/x-iwmephmpgvmcrwzj) |
| 13:47.30 | dandre | yes that match but there is a double undercore in exten = 950,1,Set(__FROMQUEUE=1) |
| 13:48.36 | jkroon | __ indicates inheritance to spawned channels. |
| 13:49.04 | jkroon | read core show application Set |
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| 13:49.48 | dandre | my problem is that I must answer the chanel for the music on hold in queue to be eared by the caller and if so, the dial(Local/queue/n) doesnt go further |
| 13:50.37 | dandre | yes but I think I am spawning channels as I use the Local channel |
| 13:54.28 | *** join/#asterisk KavanS (n=KavanS@71.117.242.28) |
| 13:55.11 | [TK]D-Fender | dandre: -- Executing [950@from-internal:2] Answer("Local/950@from-internal-0743,2", "") in new stack |
| 13:55.59 | dandre | yes |
| 13:55.59 | [TK]D-Fender | dandre: You're wondering why "exten = 0979949719,n,Dial(Local/42@from-internal/n" <- this never gets called? |
| 13:56.03 | dandre | that's true |
| 13:56.41 | [TK]D-Fender | dandre: You are ANSERING the damn channel |
| 13:56.55 | [TK]D-Fender | dandre: You are never getting out of that Dial. |
| 13:57.09 | czindy | D-Fender: Okay I think I understand. when I call 2 from 1 it is a SoftPhone registered in server1. But when I call server1 from 2 I accept calls from a PRI card (PRI GSM gateway), what are arriving to my dialplan. |
| 13:57.30 | dandre | ok but if I don't answer it the music is never heard by the caller |
| 13:57.39 | dandre | so how could I do? |
| 13:57.57 | czindy | this is my problem, that this PRI calls has no registered user? |
| 13:58.30 | [TK]D-Fender | dandre: http://pastebin.com/m31e9bc5b |
| 13:59.00 | [TK]D-Fender | [09:55]<[TK]D-Fender>dandre: -- Executing [950@from-internal:2] Answer("Local/950@from-internal-0743,2", "") in new stack <-- this is an EXPLICIT answer. You have jsut shot yourself in the head. And Queue alon answers the call. |
| 13:59.26 | [TK]D-Fender | czindy: taht made no sense. |
| 13:59.44 | dandre | so there isn't any way? |
| 14:00.07 | [TK]D-Fender | czindy: You have 1 peer named [asterisk6] and on the other side an [asterisk8] the names are not the same. When you use [asterisk6] to call the other side there is no [asterisk6] there to MATCH IT |
| 14:00.20 | [TK]D-Fender | dandre: Yuo FUCKING ANSWERED the call. ****DEAD*** |
| 14:00.36 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 14:00.46 | [TK]D-Fender | dandre: And Queue alon answers the call <------ |
| 14:01.22 | [TK]D-Fender | dandre: your outer local channel call will fail no matter what. |
| 14:01.36 | dandre | ok I'll try without answering the call |
| 14:01.43 | [TK]D-Fender | dandre: you need to fix the outer dial to continue if it has been answered <------ |
| 14:01.58 | [TK]D-Fender | dandre: Which is what I told you yesterday |
| 14:02.06 | [TK]D-Fender | dandre: "core show application dial" <- |
| 14:05.09 | _Raptor_ | what happens when there is no priority for an extension any more? in 1.2 asterisk was waiting for some input but in 1.6 i have so add an waitexten(123) extension otherwies asterisk hangs up and says Auto fallthrough, channel 'SIP/cipphone-082d8dc8' status is 'UNKNOWN. is there anything so set for the whole channel (at the beginning) which tells asterisk to wait for input? Set(TIMEOUT(response)=123) didn't work TIMEOUT(digit) ne |
| 14:05.16 | _Raptor_ | igther |
| 14:05.28 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 14:06.30 | timeshell_atwork | [TK]D-Fender Hey, did you read what happened yesterday? |
| 14:06.40 | [TK]D-Fender | timeshell_atwork: LOTS of things happened yesterday... |
| 14:06.53 | timeshell_atwork | I discovered if I called INTO the line first it would init the channel and afterward I could make outgoing |
| 14:07.14 | coppice | I lived what happened yesterday |
| 14:07.21 | timeshell_atwork | When I restart my server, no outgoing calls work until I call into the lines first. |
| 14:07.22 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:08.10 | [TK]D-Fender | _Raptor_: unless you set "autofallthrough=no" in [general] or set it in the dialplan the "running out of 's' to start IVR" has not worked since 1.4 <--- |
| 14:08.31 | [TK]D-Fender | timeshell_atwork: what "line"? |
| 14:08.42 | [TK]D-Fender | timeshell_atwork: and I have heard of bugs sounding like this before... |
| 14:08.45 | timeshell_atwork | Analog channels on my TDM4 |
| 14:08.59 | [TK]D-Fender | _Raptor_: time to read the upgrade docs <- |
| 14:09.18 | timeshell_atwork | A while back I remember reading about this too. I thought it would have been fixed by now. |
| 14:09.21 | dandre | I have read core "show application dial" but don't see how to fix the outer dial to continue if it has been answered |
| 14:09.21 | czindy | D-Fender Could you check it please: http://pastebin.com/m128b2c8c there are users created for each register. |
| 14:09.28 | _Raptor_ | [TK]D-Fender: thx. i think you won't believe me, but i did :-) |
| 14:10.05 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:10.13 | czindy | And the dial from server 1 to 2 is working, but reverse is not |
| 14:12.28 | [TK]D-Fender | czindy: there is no [asterisk8] on server 2 <---- |
| 14:13.23 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-17.cst.dccnet.com) |
| 14:13.45 | rue_mohr | is it suggested to turn echo suppression off on a polycom phone when using it witha digium card that also has echo suppression? |
| 14:14.07 | [TK]D-Fender | rue_mohr: No. handset EC is AEC, not far-end |
| 14:14.16 | rue_mohr | hmm k |
| 14:14.20 | [TK]D-Fender | rue_mohr: All handsets should be doing this |
| 14:14.40 | coppice | unfortunately a lot of them don't |
| 14:14.44 | rue_mohr | still fighting a volume control |
| 14:15.08 | [TK]D-Fender | rue_mohr: 3/4 of a year and going strong! |
| 14:15.14 | rue_mohr | we got the pots calls up enough, but calls from the aastra phones almost blow their ear off |
| 14:15.40 | rue_mohr | yea, I'm still beating the polycom tech support trying to get any support |
| 14:16.18 | rue_mohr | I give them 1/10 for answering atleast every 5th email with more than 20 words |
| 14:16.38 | rue_mohr | 0/10 for technical knowledge |
| 14:17.21 | rue_mohr | aastra support gets 9/10, only cause I'm reserved about saying anything perfect |
| 14:17.33 | rue_mohr | digiums responce is always regarding a new version |
| 14:18.00 | rue_mohr | even when I had the newest versions they seems to have asked my to try an upgrade |
| 14:18.33 | rue_mohr | they still get atleast 7/10 or so |
| 14:19.38 | [TK]D-Fender | rue_mohr: So far all the "answering" hasn't led to "solving" |
| 14:19.55 | [TK]D-Fender | rue_mohr: So your scores are little more than "feel good" ratios |
| 14:20.11 | [TK]D-Fender | rue_mohr: Keep fighting that card Ahab..... |
| 14:21.01 | jaytee | sounds like Sisyphus |
| 14:21.13 | coppice | being nice but giving no useful info seems to score better marks in CS than being a little sharp but solving the problem |
| 14:21.39 | [TK]D-Fender | is well trained in things 'sharp' |
| 14:21.51 | jaytee | and pointy |
| 14:22.33 | [TK]D-Fender | jaytee: YUP! |
| 14:22.53 | eppigy | TRABAJO |
| 14:23.11 | czindy | D-Fender yes there is no user Asterisk8, but on server 1 there is no user Asterisk6 also.... so why it is working in this direction? |
| 14:23.36 | jaytee | hi dave |
| 14:24.17 | dandre | ok I have found the g option in dial |
| 14:24.46 | [TK]D-Fender | dandre: WOW |
| 14:25.03 | eppigy | hello |
| 14:25.18 | dandre | is there any parameter in queue definition in queue.conf that is equivalent to the timeout parameter in queue cmd? |
| 14:25.31 | [TK]D-Fender | dandre: Now if the Queue side is the one that hangs up, then you're still in trouble. |
| 14:25.51 | dandre | yes |
| 14:26.06 | [TK]D-Fender | dandre: And I told you before that you should not be using a local channel to call Queue |
| 14:26.22 | dandre | i remember |
| 14:26.32 | [TK]D-Fender | dandre: [10:25]<dandre>is there any parameter in queue definition in queue.conf that is equivalent to the timeout parameter in queue cmd? <-- no, that would be redundant |
| 14:28.07 | dandre | with my use of local channel, i had setup the queue timeout once. If i call queue ... everywher I must call that queue I must set that timeout everywhere |
| 14:28.41 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 14:28.52 | [TK]D-Fender | dandre: You are causing yourself a ton of pain with this approach |
| 14:29.50 | leifmadsen | dandre: timeout in queue.conf is typically different than the timeout argument to Queue() anyways... timeout in queue.conf is how long to ring a member for, and timeout in queue is total time to be in the queue I believe |
| 14:30.04 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 14:30.23 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:30.34 | dandre | yes the timeout in queue.conf is 'round trip' period |
| 14:31.15 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:31.55 | [TK]D-Fender | dandre: No, its the AGENT ring. |
| 14:31.59 | [TK]D-Fender | dandre: nothing round about it |
| 14:32.37 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:32.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:32.44 | dandre | ok |
| 14:32.54 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 14:32.55 | ariel_ | Hello everyone |
| 14:36.03 | *** join/#asterisk mweichert (n=mweicher@216.13.154.21) |
| 14:36.17 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:37.50 | mweichert | I have one pbx connecting to another as a trunk. After 10 simultaneous calls, I can no longer route calls over the trunk as I get congestion. Is there a setting that I can change to allow more than 10 open channels to a trunk? |
| 14:38.44 | [TK]D-Fender | mweichert: * has no such restriction. |
| 14:38.55 | [TK]D-Fender | mweichert: Something else is wrong. |
| 14:40.52 | TSM | is there a way to make asterisk order the codecs |
| 14:41.16 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:41.39 | [TK]D-Fender | TSM: Yes, the order you mention them in your peer config <- |
| 14:41.51 | *** join/#asterisk moy (n=moy@74.12.123.137) |
| 14:43.46 | TSM | ahh so i cant do it globaly |
| 14:44.09 | [TK]D-Fender | TSM: [general] might propagate the order.... but that is sloppy |
| 14:45.43 | *** join/#asterisk lizone (n=zenst@user-0ccejib.cable.mindspring.com) |
| 14:53.43 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 14:54.27 | TSM | grr ive set g729 and g722 as passthru, but now my phones wont go to voicemail, is there a special setting to make voicemail work |
| 14:55.42 | [TK]D-Fender | TSM: If the call is accepted as one of those, then VOICEMAIL is NOT passthrough unless you are recording in those codecs and your prompts are already in them as well <- |
| 14:55.50 | *** join/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 14:57.14 | TSM | is there no way to make voicemail tell the phone it will only accept ulaw? |
| 14:57.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 14:58.02 | TSM | i set the phone to allow g729&ulaw but when it goes to vm it selects g729, i would have thought there was a way to deny g729 for vm |
| 14:58.27 | grandpapadot | TSM: You can't change codecs mid-call. |
| 14:58.36 | grandpapadot | TSM: The call has to be established with ulaw. |
| 14:58.42 | [TK]D-Fender | TSM: problem is the codec has ALREADY been agreed upon |
| 14:58.46 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 14:58.49 | grandpapadot | TSM: or make sure you have prompts for all your codecs |
| 15:00.59 | TSM | thats just plain annoying that you cant set ulaw before the Answer command |
| 15:01.27 | grandpapadot | TSM: No it's not, that just doesn't make sense. The codec is established by the channel driver. |
| 15:02.14 | grandpapadot | TSM: You "set" ulaw by establishing a call with the ulaw codec. |
| 15:02.24 | [TK]D-Fender | TSM: Depends when it has been set. |
| 15:02.36 | grandpapadot | TSM: or just download all of the other prompts or record your custom prompt in all supported formats with Record() |
| 15:02.37 | [TK]D-Fender | TSM: go read the CHANNEL VARIABLES doc. |
| 15:02.37 | TSM | grandpapadot: as with extentions, its possable to set allow and deny codecs, it would be good if the voicemail could do this, because if it did then it could correctly establish what the phone offers and what vm can do |
| 15:02.53 | czindy | D-Fender please help me... I created a user [Asterisk8] on server2, then I tried to call this user as well, but still does not work. |
| 15:03.07 | grandpapadot | TSM: No it wouldn't, VoiceMail() is an application just like any other. |
| 15:03.19 | grandpapadot | TSM: And you don't set codecs for extensions, ever. |
| 15:03.24 | grandpapadot | TSM: You set codecs for SIP peers |
| 15:03.26 | *** join/#asterisk shido6 (n=shido6@67.204.25.64) |
| 15:03.40 | grandpapadot | TSM: Rather, you *can't* set codecs for extensions, not possible. |
| 15:04.55 | czindy | Could you correct my settings please. http://pastebin.com/m128b2c8c |
| 15:05.19 | TSM | grandpapadot: co currently this is a small limitation when using phones which have HD codecs and the voicemail system unless you convert all the audio files |
| 15:05.39 | grandpapadot | TSM: It's not a limitation, it's part of the very logical design of Asterisk. |
| 15:06.16 | leifmadsen | extensions are assigned to devices, which have certain configuration options, such as the codecs that device supports |
| 15:06.32 | leifmadsen | users->extensions->devices |
| 15:06.45 | grandpapadot | TSM: Asterisk either has to transcode the audio ("core show translation" to see what transcoding is supported in your deployment) or play a compatable format. |
| 15:07.09 | TSM | grandpapadot: it may be logical but its not good, you have a phone that can do g722,ulaw,alaw it should be able to negotiate down to the correct codec thats allowed before the application starts |
| 15:07.38 | TSM | if phones have to negotiate their codec then so should the apps if we want them to |
| 15:07.45 | leifmadsen | eh? codec negotiation is a channel thing -- not a dialplan layer thing |
| 15:07.57 | leifmadsen | sounds like you want a SIP proxy, not a B2BUA |
| 15:08.07 | grandpapadot | TSM: It's great, I would recommend learning a bit more about the Asterisk architecture and SIP in general. |
| 15:08.25 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:08.35 | leifmadsen | codec negotiation is done before the call gets to the dialplan |
| 15:08.42 | grandpapadot | TSM: So in your design, codecs would just change mid stream, not very optimal and impossible to manage the related SIP traffic. |
| 15:08.53 | grandpapadot | s/optimal/practical |
| 15:09.42 | grandpapadot | TSM: Just use codecs that will transcode, problem solved. Or get prompts in the correct formats, problem also solved. |
| 15:10.44 | *** join/#asterisk gramulhao (n=gramulha@c-76-110-248-244.hsd1.fl.comcast.net) |
| 15:10.47 | gramulhao | hey Guys |
| 15:10.53 | gramulhao | anybody having trouble with Vonage today ? |
| 15:12.57 | TSM | no not midstream, i make call from my phone, its g722/g729/ulaw/alaw capable, it gets to the application and sees that it will only take ulaw, i cant see it being much diffrent from calling peer to peer, but mabey my understanding of the internals of asterisk is off |
| 15:13.04 | *** join/#asterisk NickRios05 (n=unicall2@static-200-105-140-237.acelerate.net) |
| 15:13.33 | grandpapadot | TSM: But the codec is *already* established before the dialplan step and the VoiceMail application are executed. |
| 15:14.01 | TSM | ahh well thats a bugger |
| 15:14.18 | grandpapadot | TSM: You're not calling an application, you're calling the sip channel driver and then the call is passed to the dial plan and then the application VoiceMail() is called. |
| 15:14.41 | grandpapadot | TSM: So in computer time, it's been years since the codec has been negotiated and forgot about. |
| 15:14.42 | [TK]D-Fender | grandpapadot: Not entirely accurate |
| 15:15.07 | grandpapadot | TK: I was going for concept, but please ellaborate if I botched it. |
| 15:15.12 | [TK]D-Fender | TSM: Look throgh the CHANNELVARIABLES doc like I told you and look VERY closely at the actions your call is taking along with the SIP negotiation |
| 15:16.02 | TSM | is there a simple way i can convert all my wavs to g722 and g729? |
| 15:16.30 | shido6 | :) |
| 15:17.30 | grandpapadot | TSM: Not sure about the transcoding capabilities with g722, but if you get g729 codec licensing you wouldn't have to convert anything. |
| 15:18.03 | grandpapadot | TSM: What version of Asterisk? Looks like no g722 transcoding in 1.4, but in 1.6 it may be there. |
| 15:18.15 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 15:18.22 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:18.33 | [TK]D-Fender | swears nobody reads the docs... |
| 15:18.39 | TSM | i have enough WAN bandwidth for g711u for loads of channels anyway |
| 15:18.51 | TSM | g722 is just nice for the polycom HD codecs |
| 15:19.15 | grandpapadot | TSM: Frankly, your users won't care or notice a big difference if you just use ulaw vs the HD codecs. |
| 15:19.55 | leifmadsen | [TK]D-Fender: amen |
| 15:20.21 | coppice | well, the nearly deaf ones might not notice the difference :-) |
| 15:20.38 | TSM | grandpapadot: this is true, it sounds good enough, i played with g722 and it did sound better at the higher freqs but nothing thats an issue |
| 15:20.39 | [TK]D-Fender | coppice: WHAT YOU SAY? |
| 15:24.41 | Maxxed | you guys know of a way to log an agent in via the CLI or manager api? |
| 15:24.50 | Maxxed | i see agentlogoff |
| 15:24.54 | Maxxed | but no, agentlogon :/ |
| 15:25.02 | Maxxed | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+AgentLogoff |
| 15:28.04 | [TK]D-Fender | Maxxed: I answered this the other day. Make an exten to do the logoff using channel variables, and Originate a call against it |
| 15:29.01 | Maxxed | oh yeah |
| 15:29.07 | Maxxed | not sure what that means :p |
| 15:29.19 | Maxxed | il have to spend a lil time foolin with it i guess |
| 15:29.22 | Maxxed | thx again ;) |
| 15:29.48 | [TK]D-Fender | Maxxed: What part don't you understand? |
| 15:30.56 | Maxxed | well, im not trying to logoff an agent, im trying to login/logon an agent, the chan variable i dont quite follow in relation to sourcing a call from it |
| 15:31.16 | [TK]D-Fender | Maxxed: AMI Originate <--- Make * call out to itself with 2 local channels. "Channel:" side answers and waits 5 seconds. the "Exten:" side logs off the device using VARIABLES you set in your originate |
| 15:31.37 | [TK]D-Fender | s/off/on/ |
| 15:31.42 | [TK]D-Fender | same thing |
| 15:32.40 | Maxxed | sounds like what im after |
| 15:33.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:33.11 | [TK]D-Fender | Maxxed: Yeah, its almost like I answered this... twice :) |
| 15:33.19 | Maxxed | hey now ;) |
| 15:33.19 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 15:33.22 | *** part/#asterisk pittmodal (n=chatzill@multimodal-fw0.cust.expedient.net) |
| 15:34.23 | Maxxed | ooh! i get it now! |
| 15:34.33 | Maxxed | damn thats easy :p |
| 15:36.38 | Maxxed | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 15:36.47 | Maxxed | after checking that out, it makes a lil more sense now |
| 15:37.24 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:42.34 | NickRios05 | hello can anyone give me a hand I have a E1 PRI comming to my digium te120p card i see the calls comming in but i get this message http://www.pastebin.org/7264... |
| 15:42.39 | *** join/#asterisk s14ck (n=s14ck@190-76-99-118.dyn.movilnet.com.ve) |
| 15:44.33 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 15:47.58 | citywok | <PROTECTED> |
| 15:48.07 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 15:48.35 | NickRios05 | ok but i ve setup the extension on extensions.conf |
| 15:49.52 | NickRios05 | http://www.pastebin.org/7269 |
| 15:52.46 | leifmadsen | NickRios05: missing an _ |
| 15:52.53 | leifmadsen | pattern matches need _ in the front |
| 15:53.00 | leifmadsen | right now you have literal extension 34XX |
| 15:53.04 | citywok | yea, either do 3400, or _34XX |
| 15:53.13 | leifmadsen | and don't use priority numbers! |
| 15:53.20 | citywok | 1,n,n :) |
| 15:53.20 | NickRios05 | ok let me try that |
| 15:53.25 | leifmadsen | please... |
| 15:53.29 | leifmadsen | citywok: amen |
| 15:54.22 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:57.25 | NickRios05 | hey another question what would be the best context to work with, right now am using vicidial to answer those calls and i want them to be redirected to an DID with a default context, cuz this configuration worked before with sip but this is the first time i work with a digium card |
| 15:58.04 | NickRios05 | i mean that in the DID tool of vicidial i have the default DID to redirect all calls to my in_group |
| 16:03.38 | *** join/#asterisk korcan (n=korcan@99.23.50.73) |
| 16:04.40 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:05.11 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:06.47 | *** join/#asterisk JD (n=david@elmer.catnip.org.uk) |
| 16:06.55 | JD | hi |
| 16:07.18 | JD | anyone had much experience of Cisco ATA 186 devices? |
| 16:07.45 | JD | I've got a device set up to receive calls, but I get an engaged tone whenever I try to dial a number |
| 16:08.06 | Carlos_PHX | What do you see on the CLI? |
| 16:08.18 | JD | -- MGCP mgcp_new(MGCP/1@715-1) created in state: Down |
| 16:12.26 | *** join/#asterisk afink (n=chatzill@204.26.87.226) |
| 16:13.17 | JD | Carlos_PHX: okay, I think I worked out what was wrong. my mgcp.conf had the wrong context set in it |
| 16:13.23 | JD | sorry for the noise :) |
| 16:16.01 | NickRios05 | ok never mind ive already fixed it worked very well thnx so much citywork and leifmadsen |
| 16:25.27 | *** join/#asterisk afink (n=chatzill@204.26.87.226) |
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| 16:34.16 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:34.40 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 16:34.48 | *** join/#asterisk xa0z (n=Interex@75-129-230-28.dhcp.mtvr.il.charter.com) |
| 16:35.20 | xa0z | Can anyone tell me if using a Cisco 79xx series phone if there is a way to handle call-waiting without using another line? |
| 16:35.52 | xa0z | For instance the 7940, a 2 line phone... could I receive call-waiting on line 1, if using line 1, and not disturb line 2 ? |
| 16:36.29 | thansen | is there any way that I can limit how long Dial() tries a specific number? |
| 16:36.42 | WindowsUser | Dial(somebody,45) |
| 16:36.46 | thansen | I'm trying to prevent a cellphone vm from picking up |
| 16:37.00 | *** join/#asterisk Tim_Toady (n=moi@adsl282-123.kln.forthnet.gr) |
| 16:37.05 | thansen | WindowsUser: yeah, but I have mutliple endpoints getting hit :( |
| 16:37.14 | WindowsUser | ? |
| 16:37.31 | thansen | ie, somebody&somebody2 |
| 16:37.38 | WindowsUser | so go via a local channel |
| 16:38.11 | thansen | WindowsUser: I'm a little noob at this stuff, can you give me a litter more explanation? |
| 16:38.37 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 16:39.54 | WindowsUser | hrm my use of it isn't the best |
| 16:40.48 | WindowsUser | but you tell your multiple Dial to Dial(somebody&local/cellphone) |
| 16:41.10 | WindowsUser | exten => cellphone,1,Dial(cellphone,45) |
| 16:43.13 | NickRios05 | hey another question am trying to get calls out from my E1 line but am getting this error on asterisk http://www.pastebin.org/7274 |
| 16:44.49 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 16:46.06 | thansen | WindowsUser: http://www.voip-info.org/wiki/view/Asterisk+Local+channels in this example would it ever reach .. exten => 200,102,VoiceMail(${EXTEN}@default) |
| 16:46.41 | NickRios05 | sorry ive already corrected the extensions.conf dial plan, but now am getting this http://www.pastebin.org/7275 |
| 16:46.46 | NickRios05 | any ideas?' |
| 16:48.16 | *** join/#asterisk profxavier (n=chatzill@unaffiliated/neverblue) |
| 16:48.50 | profxavier | can I release a phone, from the * console? two IPs are being used on one account... |
| 16:49.05 | profxavier | so when I dial that extension, both phones will ring |
| 16:51.45 | profxavier | s/so// |
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| 17:32.05 | jplank | wow everyone is pretty quite today |
| 17:32.24 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 17:32.27 | b14ck | sup |
| 17:32.39 | jplank | heh |
| 17:33.10 | jplank | seems like everyone is recovering from a netsplit |
| 17:33.24 | b14ck | my auto-identify failed :( |
| 17:33.31 | b14ck | so i couldnt join, just realized this channel was dead |
| 17:33.32 | b14ck | heh |
| 17:34.09 | jplank | very possible that we are the only ones effected by the netsplit, and we are actually in here alone |
| 17:34.32 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:35.27 | [TK]D-Fender | very few people split |
| 17:35.58 | jplank | it looked like everyone split to me, thats why I thought I was on the losing end |
| 17:36.18 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 17:37.50 | jplank | i remember a time where netsplits were unheard of on freenode, only on the evil efnet |
| 17:41.58 | errr | jplank: how long ago was that? Its been a pretty regular thing for the last 4 or so years |
| 17:43.33 | jplank | 4 years sounds long |
| 17:43.52 | coppice | I remember a time when there were no netsplits on freenode. No internet, either |
| 17:44.15 | errr | heh |
| 17:44.36 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
| 17:46.23 | TSM | whats the dig string to check an ENUM number? |
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| 18:09.36 | darkdrgn2k3 | Hey, i have a VOIP provider who's backend is a NORTEL MCS |
| 18:09.43 | darkdrgn2k3 | how can i figure out what settings i need to use it? |
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| 18:11.51 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 18:11.52 | WindowsUser | erm |
| 18:12.24 | WindowsUser | any VOIP provider should willingly tell if you SIP credentials if they want you to connect to them |
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| 18:24.10 | b14ck | use flowroute :) |
| 18:24.43 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:26.15 | lesouvage | A call file geerated call is going fine but I have this message in the cli "Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?" after a recorded message is played. |
| 18:28.25 | lesouvage | So I just have a kind of worrying error message while the sip conection with my sip provider keeps working without problem. |
| 18:28.41 | shido6 | look away |
| 18:28.47 | shido6 | and make calls - |
| 18:29.03 | jplank | lol |
| 18:30.04 | b14ck | crontab -e | echo "* * * * * rm -f /var/log/asterisk/full*" |
| 18:30.06 | b14ck | heh |
| 18:30.14 | b14ck | err |
| 18:30.21 | b14ck | w/e |
| 18:30.25 | b14ck | just echo it to the crontab urself! |
| 18:30.44 | lesouvage | It is just kind of strange and perhaps someone has a suggestion how to fix this. |
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| 18:37.45 | [TK]D-Fender | Timmeh! |
| 18:40.25 | eppigy | livin a lie |
| 18:40.30 | eppigy | LIVIN A LIE |
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| 19:18.11 | K3rN3L | Helloo |
| 19:18.46 | K3rN3L | somebody know how i can know the number that i am dial from a .call file? |
| 19:19.18 | K3rN3L | i need pass this to a agi script |
| 19:19.52 | WindowsUser | try asking for the EXTENSION variable |
| 19:20.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:21.34 | leifmadsen | EXTEN? |
| 19:23.29 | [TK]D-Fender | lol, not implicit... |
| 19:23.36 | [TK]D-Fender | silly wabbits |
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| 19:25.18 | *** join/#asterisk czindy (n=czindy@gprs5e1bdfbf.pool.t-umts.hu) |
| 19:27.18 | czindy | Hello again, D-Fender, are you there? I think I not understand you. I connected the two asterisk box exacty that the 2nd edition asterisk book writes. |
| 19:27.38 | czindy | Could you help me how can I connect those 2 asterisk servers please. |
| 19:28.27 | czindy | Your advice would be very appreciated |
| 19:33.10 | [TK]D-Fender | czindy: Go read the WIKI's page on "asterisk dual servers" |
| 19:34.14 | czindy | ok thank you |
| 19:34.21 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:35.17 | czindy | do you think I need the Peer-User configuration? |
| 19:36.02 | [TK]D-Fender | czindy: you never had matching peers int he first place. You seem to be missing some basic common logic. Maybe reading that WIKI page a few times will let it sink in. |
| 19:36.18 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:38.18 | eppigy | DONDE |
| 19:38.23 | czindy | Ok I trust you. Do you think that this page is good: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 19:38.24 | czindy | I configured exactly my servers what is the 2nd example say |
| 19:39.05 | [TK]D-Fender | czindy: PB your configs |
| 19:39.16 | [TK]D-Fender | eppESTA |
| 19:39.31 | [TK]D-Fender | auto-complete FAIL |
| 19:40.54 | *** part/#asterisk MACscr (n=Mark@98.214.100.212) |
| 19:41.21 | *** join/#asterisk ethicx (n=chatzill@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 19:42.36 | ethicx | Im trying to stream an online radio station for my music on hold but I get this error on Asterisk's CLI: [Aug 7 15:38:35] NOTICE[2112]: res_musiconhold.c:556 monmp3thread: Request to schedule in the past?!?! this is my musiconhold.conf http://pastebin.com/m268caea2 what could be going on? |
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| 19:43.17 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:43.39 | czindy | Here it is http://pastebin.com/m1aec1b40 |
| 19:45.48 | Corydon76-dig | ethicx: is this a virtual box or are you running other services on your machine? |
| 19:46.11 | ethicx | nope its an asterisk dedi server i got running at home |
| 19:46.57 | Corydon76-dig | ethicx: if you're getting that message, then something else is eating CPU |
| 19:47.38 | ethicx | you think overheating would spit out these messages cause I do admit the server runs a bit hot |
| 19:47.56 | ethicx | working on better ventilation at the moment =D |
| 19:48.16 | Corydon76-dig | Only if the OS is sending the CPU into low-power mode in order to allow the CPU to cool |
| 19:48.26 | *** join/#asterisk speedy (n=speedy@78.154.212.183) |
| 19:48.26 | [TK]D-Fender | czindy: You have not done it accoring to that guide |
| 19:48.44 | eppigy | eppESTA |
| 19:49.03 | ethicx | yeah, but otherwise it shouldnt be a config error on my end hu? |
| 19:49.11 | *** part/#asterisk LarsN (n=lars@unaffiliated/w9zeb-lars) |
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| 19:49.57 | jaytee | Estoy intentando muy difÃcilmente no perder mi genio. |
| 19:50.00 | czindy | Ok I check it again, here is my config files more detailed and the errors |
| 19:50.05 | czindy | http://pastebin.com/m4d4aebea |
| 19:50.07 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:52.04 | czindy | Do you mean the declaration of iax2 user and peer? |
| 19:55.06 | [TK]D-Fender | czindy: You did not specify a USERNAME |
| 19:55.22 | [TK]D-Fender | czindy: so the peername is what thigns fall back to. |
| 19:56.02 | eppigy | jaytee: you have exceeded my linguistics ability |
| 19:56.55 | jaytee | eppigy, it should translate as: I am trying very hard not to lose my temper. |
| 19:57.46 | eppigy | 8[] |
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| 20:00.53 | czindy | I added this on each side user=MyCrossAs but the same error |
| 20:00.54 | czindy | Why server 2 recognize server 1 as 123: |
| 20:00.54 | czindy | <PROTECTED> |
| 20:01.18 | [TK]D-Fender | czindy: yOUR peer DOES NOT SPECIFY THE remote ACCONT TO AUTH AS. |
| 20:01.26 | *** join/#asterisk shazaum (n=aaaaa@unaffiliated/shazaum) |
| 20:01.56 | darksmurf | When I 'reload' from the asterisk CLI it tells me about contexts it is loading, files it is parsing, etc. I need to verify that a context I have created is actually getting loaded. I have tried to save the output from the reload command and grep it, but I do not find what I am looking for. |
| 20:02.32 | [TK]D-Fender | darksmurf: "dialplan sho thecontextname" |
| 20:02.36 | [TK]D-Fender | darksmurf: "dialplan show thecontextname" |
| 20:02.41 | darksmurf | ahh, thanks |
| 20:02.54 | NickRios05 | hey one question on the extensions.conf file if I have a zap trunk how can i select the channel the call will go out?? |
| 20:04.46 | [TK]D-Fender | NickRios05: Dial(Zap/channelorgrouphere/numbertodial) |
| 20:07.46 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:08.33 | czindy | ok I tried to dial: Dial(IAX2/MyCrossAs:Sattaratta@Asterisk8/${EXTEN:1},60,Tr); but the same result. |
| 20:08.34 | czindy | Do you think I'm very blond? |
| 20:10.09 | czindy | I stuck on this... really |
| 20:10.20 | NickRios05 | thnx [TK]D-Fender , but for instance with the zap trunk defined as 1 works but when i define it like this 3 http://www.pastebin.org/7307, i get the following message http://www.pastebin.org/7308 |
| 20:11.06 | [TK]D-Fender | czindy: EW |
| 20:11.28 | [TK]D-Fender | czindy: and I don't see a user specified in your peers |
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| 20:13.29 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 20:13.31 | czindy | Sorry I'm trying to concentrate but.... Could you please make some correction in my DP? |
| 20:13.35 | czindy | or in my brain :) |
| 20:14.02 | eppigy | hans czindy some lsd |
| 20:14.06 | eppigy | there you go |
| 20:14.33 | czindy | Thanks it was great ... |
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| 20:16.54 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 20:17.27 | [TK]D-Fender | NickRios05: "zap show status", "zap show channels" |
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| 20:17.55 | czindy | USER do you mean an iax user what I use to connect to asterisk with my softphone? |
| 20:18.23 | *** part/#asterisk Shotygun (n=thorn@DSL212-235-83-213.bb.netvision.net.il) |
| 20:18.59 | ruben23 | hi im using asterisk Asterisk 1.2.30.2 and set all calls be recorded..problem is suddenly my recordings are getting 0 bytes. |
| 20:19.17 | ruben23 | almost all records are 0 bytes. |
| 20:19.29 | NickRios05 | [TK]D-Fender, http://www.pastebin.org/7309 |
| 20:19.29 | [TK]D-Fender | czindy: http://pastebin.com/m7721d875 |
| 20:21.14 | czindy | Hmm... so may I need [Asterisk8] type=user? |
| 20:22.33 | [TK]D-Fender | czindy: So you read the WIKi example 2 a few times, right? |
| 20:22.38 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 20:22.56 | [TK]D-Fender | czindy: And that's what you based yours on? |
| 20:25.32 | NickRios05 | [TK]D-Fender, ok I managed to use the rest of the channels, where can i put the caller id i wanna send through my E1? |
| 20:26.04 | [TK]D-Fender | NickRios05: "core show function CALLERID" |
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| 20:31.32 | [TK]D-Fender | checkout time, BBL |
| 20:33.49 | *** join/#asterisk f0ner00t (i=f0ner00t@99-21-148-71.lightspeed.frokca.sbcglobal.net) |
| 20:33.51 | f0ner00t | Hello. |
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| 20:52.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 21:04.48 | darksmurf | I have a script that needs access to a MySQL database. It does not seem to be connecting, and there is not much for diag information. how can I tell if a mysql(connect...) command worked or not? |
| 21:05.23 | leifmadsen | darksmurf: which language? (although this doesn't seem like an asterisk question...) |
| 21:05.37 | darksmurf | in an asterisk dialplan |
| 21:05.40 | leifmadsen | I'd enabled the logging in mysql and check there |
| 21:05.43 | leifmadsen | using what? |
| 21:06.04 | leifmadsen | use res_odbc instead of the MYSQL() dialplan command |
| 21:06.11 | leifmadsen | way better to use func_odbc instead |
| 21:06.15 | darksmurf | exten => _s,1,MYSQL(Connect connid localhost user password timeclock |
| 21:07.06 | leifmadsen | 1) you don't need the _ in that scenario, 2) is that the whole thing?, you're missing a closing ) |
| 21:07.14 | darksmurf | leifmadsen, if I were developing this myself I would consider that, but this is supposed to work already. It dies on a MYSQL(query line |
| 21:07.16 | leifmadsen | sorry, can't help much more now, working on dinner |
| 21:07.50 | darksmurf | I omited the last ) so that my screwy copy/paste thing would not send the line before I changed the user/pass. |
| 21:08.08 | darksmurf | thanks anyway. Have a good dinner. |
| 21:08.26 | darksmurf | I will look at the _ bit... |
| 21:11.17 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 21:11.53 | nextime | hello all. I need to do a dialout from an external daemon, i use a callfile with Channel: Zap/g1/number... |
| 21:12.09 | nextime | and i redirect the call to a context/extension where i have a playback() |
| 21:12.35 | nextime | is there a way to intercept the ${DIALSTATUS} of the call after is end? |
| 21:13.54 | nextime | ok, found the way to do it |
| 21:13.54 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:13.56 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 21:15.13 | *** join/#asterisk darksmurf (n=asdf@166.134.135.107) |
| 21:21.20 | darksmurf | should arguments in MYSQL() be |
| 21:21.28 | darksmurf | "quoted" ? |
| 21:23.22 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:25.36 | ruben23 | hi |
| 21:26.17 | ruben23 | anyone have idea--my asterisk are set for all calls record but its can see threcorded file in 0 bytes. |
| 21:29.36 | NickRios05 | [TK]D-Fender, Thx it worked really well |
| 21:32.27 | [TK]D-Fender | NickRios05: You're welcome |
| 21:34.24 | darksmurf | is there anyway to see WHY a MYSQL() command failed in an asterisk dialplan? It seems to be failing on the Connect. I am not able to see any connection attempts in my mysql log, but I suspect it is not logging everything I need. |
| 21:45.24 | [TK]D-Fender | darksmurf: I don't see dialplan messages, check CLi for this |
| 21:45.32 | [TK]D-Fender | darksmurf: core debug should highlight it |
| 21:46.19 | darksmurf | core debug? |
| 21:46.24 | darksmurf | 'no such command' |
| 21:47.00 | darksmurf | nor 'help core debug' |
| 21:47.00 | darksmurf | ahh.. |
| 21:47.02 | darksmurf | core set debug |
| 21:47.45 | darksmurf | higher level = more data? |
| 21:48.37 | russellb | yes, but you also have to enable it in /etc/asterisk/logger.conf for it to show up |
| 21:49.31 | dustybin | darksmurf: what happened to lightsmurf? |
| 21:49.43 | darksmurf | I got hungry :) |
| 21:50.46 | darksmurf | russellb, in that file I have a few comments, and a line that starts 'full' that is not commented. I take it that means full logging is enabled? |
| 21:50.59 | darksmurf | I tried core set debug 100 (was 0) and I do not see any more information. |
| 21:52.36 | *** join/#asterisk strehi (n=strehi@andromeda.planb.de) |
| 21:54.30 | strehi | hi |
| 21:55.01 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 21:55.17 | strehi | somebody arround with experiance in connecting interal s2m devices on a Wildcard TE405P/TE410P (1st gen)? |
| 21:56.20 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 21:56.21 | acxty | hi guys, does someone had integrate a meridian pbx with asterisk? |
| 21:57.17 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:57.37 | acxty | is it possible to use does meridian phones, with asterisk |
| 21:59.05 | *** join/#asterisk awkfu (n=awkfu@66.162.90.57) |
| 22:04.42 | [TK]D-Fender | acxty: Only with expensive gateways, and is model-dependent |
| 22:09.57 | lesouvage | I have "app_dial.c:1638 dial_exec_full: Could not stop autoservice on calling channel" and some strange cdr date with dcontext = inbound while I'm only making call file generated outbound calls and inbound isn't part of the dialplan scheme. Can there be a relation between the two? |
| 22:13.31 | acxty | ok |
| 22:13.34 | acxty | thanks |
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| 22:20.04 | acxty | A small problem. I have a centos running asterisk on panama. It have a ip line connected to it. I have a differente centos on el salvador also running asterisk. What I want to is that sip accounts on el salvador call panama phones using the ip line on the panama box |
| 22:21.54 | acxty | My idea is to add a new sip account on panama and register el salvador asterisk with that account, and use that line to make the calls on panama |
| 22:22.28 | acxty | is that the correct way of doing it, or is there a better way to have does 2 asterisk connected even if they are on different countries |
| 22:22.43 | [TK]D-Fender | acxty: huh? |
| 22:22.50 | andresmujica | acxty : iax trunking is the better way |
| 22:23.51 | acxty | huh iax? |
| 22:25.27 | acxty | when using iax, do I need to physically connect something to the asterisk, or will it work similiar as sip? |
| 22:26.09 | andresmujica | similar like sip, but signalling and media use the same udp port, so no hassle with nat/networking at all. |
| 22:26.37 | acxty | will read about that thanks |
| 22:28.11 | [TK]D-Fender | acxty: SIP, IAX. Hardly makes a difference |
| 22:28.24 | [TK]D-Fender | acxty: And you can pass calls between *'s any way you want |
| 22:28.35 | [TK]D-Fender | andresmujica: ALMOST. |
| 22:29.34 | acxty | I make a test connecting on panama to el salvador using a softphone on my computer, and the sound quality was good. I was using a sip account during that test |
| 22:29.49 | andresmujica | yeap. almost no hassle with networking and that... :) |
| 22:30.26 | andresmujica | acxty: with IAX2 you can save a little bit of bandwidth because of trunking. so go explore that route |
| 22:30.53 | acxty | will give it a try |
| 22:31.04 | acxty | I have 1mbps pure on both computers |
| 22:31.15 | acxty | do you think it is enough or need to upgrade it |
| 22:31.41 | andresmujica | it depends on your needs. simultaneous calls, codec used, headers, etc, etc ,etc |
| 22:42.03 | darksmurf | mysql access from an Asterisk Dialplan has to be compiled into asterisk, correct? Is there any way I can verify it is? |
| 22:43.56 | russellb | darksmurf: there are multiple methods for achieving that. You can verify any of them with "module show ..." |
| 22:44.03 | russellb | to verify that the module you expect to be there is there |
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| 22:47.05 | darksmurf | russellb, thanks. That explains why the MYSQL() command does not work. No module installed. Damn |
| 22:47.19 | russellb | that'll do it |
| 22:47.55 | russellb | By the way, the most supported way of doing that is by using func_odbc |
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| 22:48.04 | russellb | but, of course, it's up to you |
| 22:53.25 | darksmurf | I think I am going to have to convert AsteriskPHPTimeClock to func_odbc...joy. |
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| 23:01.18 | [TK]D-Fender | darksmurf: Or you could show us the CLI output of your failed attempt to see if what you think is wrong is actually wrong |
| 23:01.46 | [TK]D-Fender | darksmurf: Because by default I wouldn't trust the diagnosis of a person who doesn't know the other signs |
| 23:02.35 | drmessano | thinks ODBC in Asterisk is the func |
| 23:05.54 | darksmurf | [TK]D-Fender, when the dialplan reaches a MYSQL function it dies. a 'modules show' does not list anything related to MySQL or anything SQL. It does show a few other DB types, including ODBC. |
| 23:06.13 | darksmurf | by 'it dies' I mean it jumps to hanging up the call. |
| 23:08.23 | [TK]D-Fender | darksmurf: "dies" how descriptive.... |
| 23:08.40 | [TK]D-Fender | darksmurf: SHOW ME THE DEAD BODY |
| 23:09.54 | darksmurf | stand by. I will add a few NoOps so it is a little more obvious.. |
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| 23:13.39 | darksmurf | http://pastebin.com/m2110b665 |
| 23:16.02 | [TK]D-Fender | darksmurf: O RLY |
| 23:16.12 | [TK]D-Fender | darksmurf: -- Executing [s@from-internal:1] Macro("SIP/2111-b73cf6b8", "hangupcall") in new stack <-- this is a DIALPLAN APP EXECUTING |
| 23:16.20 | [TK]D-Fender | darksmurf: this is NOT eve YOUR code. |
| 23:16.35 | [TK]D-Fender | darksmurf: INCLUDE fail <- |
| 23:17.00 | darksmurf | correct, that is why my questions have been limited to how to verify MYSQL support in asterisk, not how to fix it. |
| 23:17.09 | [TK]D-Fender | darksmurf: You don't get it... |
| 23:17.31 | [TK]D-Fender | darksmurf: Your DIALPLAN failed <- It isnsn't RUNNING the app because you don't have stuff in the right place |
| 23:17.44 | darksmurf | If I omit the MYSQL lines, the rest of the dialplan works fine. I get asked for an employee ID, it reads it back, etc. |
| 23:17.46 | [TK]D-Fender | DarkIt isn't calling YOUR "s,1" you showed. |
| 23:18.15 | darksmurf | I disagree. I will demonstrate. A few moments please. |
| 23:18.22 | [TK]D-Fender | darksmurf: You have a DIALPLAN problem with your code overlapping with another context |
| 23:18.32 | [TK]D-Fender | darksmurf: nothing to show. You've already shown enough |
| 23:18.48 | [TK]D-Fender | darksmurf: This evidence shoots you clean out of the water |
| 23:20.07 | darksmurf | So, options are...? |
| 23:20.39 | [TK]D-Fender | darksmurf: -- Executing [25625@from-internal:2] Goto("SIP/2111-b73cf6b8", "s,1") in new stack <-- this is calling [from-internal]. Your shit isn't in that context, and if you were expecting it to be acessible, then INCLUDE prioritization KILLS IT |
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| 23:21.29 | rvhi | hi, how do you find if a sip channel is busy or not? |
| 23:21.33 | darksmurf | If I change the GoTo to S,2 (skipping the MYSQL line), it continues the app. So If I were to .. |
| 23:21.35 | darksmurf | sip show inuse |
| 23:22.09 | [TK]D-Fender | Darkbecause your context includes OVERLAP |
| 23:22.11 | darksmurf | Can I use something different than s? |
| 23:22.14 | rvhi | how about in dialplan? |
| 23:22.28 | [TK]D-Fender | darksmurf: s,1 EXISTS in [from-internal] already and screws you |
| 23:22.29 | rvhi | so if it is busy, i will send calls to voicemail directly |
| 23:23.01 | [TK]D-Fender | This is what happens when FreePBX users who have no clue about the dialplan try writing scripts around the GUI |
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| 23:24.24 | [TK]D-Fender | darksmurf: http://pastebin.com/m6ad5908b <--------------- |
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| 23:25.38 | darksmurf | Ah. Very good. |
| 23:30.12 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 23:33.21 | darksmurf | fender, thanks for the help. I do not expect help at all, much less so due to the fact I am using FreePBX. That said, I am still having the same problem. See the update dialplan and output: http://pastebin.com/m709ba68e |
| 23:34.23 | darksmurf | updated* |
| 23:35.26 | darksmurf | I think there have been two problems, my attempt at verifying the MySQL problem hid the context scope problem. Maybe I am still missing something. |
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| 23:48.47 | [TK]D-Fender | darksmurf: Add some more noop's before the MYSQL. |
| 23:48.55 | [TK]D-Fender | Also in CLI do "core show application MYSQL |
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