| 03:30.58 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 03:30.58 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 03:31.07 | fulcan | ChannelZ where would there be a sample or guide to this? |
| 03:31.12 | ChannelZ | I guess you could use the FILE function and write a text file |
| 03:31.25 | fulcan | ChannelZ that would be perfect! |
| 03:31.29 | ChannelZ | core show function FILE |
| 03:32.06 | volga629 | Hello Everyone, How I can set that traffic going into SIP trunk will no use srtp, but local branch srtp stay in use |
| 04:10.33 | *** join/#asterisk infobot (~infobot@207135154175.dsl.xmission.com) |
| 04:10.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 04:11.12 | fulcan | exten => s,n,Set(FILE(/var/spool/asterisk/sms/.(${STRFTIME(${EPOCH},,%F-%T)}-${CALLERID(num)}-${EXTEN}.txt),a)=${SMSINRAW}) |
| 04:11.29 | fulcan | ChannelZ is that what you mean? |
| 04:12.12 | ChannelZ | ?? |
| 04:12.26 | Captain_Proton | ChannelZ, Well I guess I will go home and they will have to deal. thanks for your help |
| 04:12.37 | ChannelZ | does /var/spool/asterisk/sms exist, and does whatever user your Asterisk runs as have write access to it? |
| 04:13.03 | fulcan | exten => s,n,Set(FILE(/var/spool/asterisk/sms/.(${STRFTIME(${EPOCH},,%F-%T)}-${CALLERID(num)}-${EXTEN}.txt),a)=${MESSAGE(body)}) |
| 04:13.23 | fulcan | ChannelZ ${MESSAGE(body)} and ${SMSINRAW} appear to have the same value |
| 04:13.32 | ChannelZ | Your syntaxis a little odd |
| 04:14.33 | *** join/#asterisk infobot (~infobot@207135154175.dsl.xmission.com) |
| 04:14.33 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 04:14.39 | ChannelZ | well you can write whatever it is you want to write |
| 04:15.23 | ChannelZ | my original answer to "I am not sure what to assign it to?" was generically "presumably you want to write the text" |
| 04:15.48 | ChannelZ | Captain_Proton: You said this is AsteriskNOW or something? |
| 04:15.57 | fulcan | ChannelZ yes sir! :) |
| 04:21.39 | *** join/#asterisk infobot (~infobot@207135154175.dsl.xmission.com) |
| 04:21.39 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 04:22.42 | *** join/#asterisk infobot (~infobot@207135154175.dsl.xmission.com) |
| 04:22.43 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.5.1 (2012/06/14), 1.8.13.0 (2012/06/04), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 04:22.47 | ChannelZ | then you're not getting any callerid |
| 04:23.09 | ChannelZ | well.. actually your NoOp is just wrong |
| 04:23.18 | ChannelZ | it's ${CALLERID(num)} not $(CALLERIDNUM} |
| 04:23.44 | ChannelZ | CALLERID is a function |
| 04:24.25 | ChannelZ | the Set statement is using the right syntax, if that isn't becoming part of the filename then you aren't getting any CID |
| 04:24.40 | volga629 | is the way asterisk set srtp optional for local extension ? |
| 04:25.27 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-ebdedeeknrhlonmr) |
| 04:26.06 | fulcan | ChannelZ it's not a show stopper, thank you very much. |
| 04:26.18 | *** join/#asterisk blindhog (~josh@d5-201.rb3.clm.centurytel.net) |
| 04:26.44 | ChannelZ | volga629: does encryption=xxx not work per-peer? |
| 04:27.18 | ChannelZ | (although, does 'no' reject if they try to use it? I dunno, not using SRTP myself..) |
| 04:28.35 | volga629 | Yes, the all problem Outbound SIP trunk not support srtp and branches use srtp |
| 04:29.21 | volga629 | can't dial out of trunk, looking for some way is it possible to do some kind of bridging |
| 04:31.45 | volga629 | If set no and phone set yes I did not tried, but when opposite way get rejected and this expected |
| 04:37.49 | fulcan | [Syntax] MessageSend(to[,from]) so I -> MessageSend (0742510@50.16.59.123:5060[,9175039894]) and get -> No such command 'MessageSend (0742510@50.16.59.123:5060[,9175039894])' (type 'core show help MessageSend (0742510424@50.116.59.123:5060[,9175039892])' for other possible commands)????? |
| 04:42.50 | fulcan | MessageSend how do I use MessageSend? |
| 04:47.46 | kaldemar | do you have a space between d and (? |
| 04:48.15 | fulcan | kaldemar yes |
| 04:48.33 | kaldemar | well, don't. |
| 04:48.36 | fulcan | kaldemar tried both |
| 04:49.02 | fulcan | I have never sent an sms from asterisk so I am a little lost. |
| 04:50.35 | *** join/#asterisk twanny796 (~twanny@46.11.8.155) |
| 04:50.41 | [TK]D-Fender | fulcan: Where do you get the impression that that is a CLI command? |
| 04:52.39 | *** part/#asterisk jmordica (~jmordica@c-76-123-170-101.hsd1.ms.comcast.net) |
| 04:52.40 | fulcan | [TK]D-Fender ummm, monkey and a lightbulb? |
| 04:53.33 | volga629 | ChannelZ thank you for opinion, I have strange behavior of snom 370 I dial from phone and don't here any dial ring, but dial plan is processing and my cell dialing, tcpdump I see RTP flying. |
| 04:54.03 | volga629 | cell is ringing |
| 04:54.06 | fulcan | I found this -> http://stackoverflow.com/questions/7755245/in-asterisk-10-how-can-i-use-messagesend-in-dialplan-to-send-messages-outside but I am scratching my head trying to figure out how text will enter this context? |
| 04:55.21 | fulcan | or at least the simplest way. |
| 04:56.36 | fulcan | is there a straight cli function for sending sms? |
| 04:56.46 | fulcan | over sip? |
| 04:58.23 | fulcan | maybe a .call file with the MessageSend function in it? is this possible? |
| 05:07.07 | fulcan | anyone? |
| 05:14.12 | *** join/#asterisk sun28 (~light@cafe.zesdid.com) |
| 05:18.10 | *** join/#asterisk twanny796 (~twanny@46.11.6.241) |
| 05:46.29 | *** join/#asterisk angryuser_laptop (~angryuser@ps444-1-78-224-152-133.fbx.proxad.net) |
| 05:47.01 | *** part/#asterisk Marquel (~Marquel@unaffiliated/marquel) |
| 05:53.43 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 06:00.26 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:06.54 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:26.56 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
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| 06:33.35 | *** join/#asterisk shadebob (~shadebob@41.250.121.96) |
| 06:44.15 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:54.29 | bulkorok | hello |
| 07:18.51 | *** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net) |
| 07:19.03 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
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| 07:41.17 | Get_The_Fish | anyone in here? |
| 07:41.40 | Get_The_Fish | command line sip clients that dont force you to write the entire setup by hand? |
| 07:42.41 | *** join/#asterisk aberrios (~aberrios@77.107.183.144) |
| 07:43.39 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:44.24 | Samuel_CTX | Get_The_Fish: command line sip clients? What do you want to do with them???? (My end-users all like a gui) |
| 07:45.20 | Get_The_Fish | test phone systems. As in use my laptop and my VM and register 100 clients at a time, set up and tear down calls, etc. |
| 07:45.30 | Get_The_Fish | to make sure it works. |
| 07:45.53 | Samuel_CTX | google 'sipp'. I think that is what you want |
| 07:46.04 | Samuel_CTX | http://sipp.sourceforge.net/ |
| 07:46.37 | Get_The_Fish | ive used SIPP, but I was looking for something a little more "automated", if you will. |
| 07:46.46 | *** join/#asterisk shadebob (~shadebob@41.250.121.96) |
| 07:47.00 | Get_The_Fish | as in, I want a client to "react" to the server instead of following my scripts. |
| 07:47.25 | Get_The_Fish | I appreciate the suggestion, I use sipp for load testing machines and making sure Asterisk is still alive. |
| 07:47.47 | *** join/#asterisk shadebob (~shadebob@41.250.121.96) |
| 07:47.50 | Samuel_CTX | well it that case, I can't help you (sorry). The only thing I can think about is write some bash scripts that use sipp? |
| 07:48.30 | *** join/#asterisk shadebob (~shadebob@41.250.121.96) |
| 07:48.56 | Get_The_Fish | ima gona try me some pjsip an see what shakes out |
| 07:49.00 | *** join/#asterisk fulcan (~brads@li345-191.members.linode.com) |
| 07:49.25 | Get_The_Fish | some really robust ones sure, I could see that. |
| 07:49.33 | fulcan | How do I generate a message using messagesend? |
| 07:49.41 | Get_The_Fish | I just dont have the time to figure out every interaction. |
| 07:50.13 | *** join/#asterisk shadebob (~shadebob@41.250.121.96) |
| 07:51.36 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 07:51.47 | v0lZy | lo |
| 07:51.47 | Get_The_Fish | asterisk version? |
| 07:53.25 | Get_The_Fish | here https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend |
| 08:01.21 | *** join/#asterisk fling (~fling@unaffiliated/fling) |
| 08:01.43 | fling | Hello! Don't I need multilib? |
| 08:02.12 | kaldemar | fling: for what? |
| 08:02.39 | fling | kaldemar: for something asterisk may need |
| 08:03.06 | fling | kaldemar: I have a box for my pbx and I'm about disabling multilib on it |
| 08:04.09 | kaldemar | vanilla asterisk does not depend on it afaik. |
| 08:06.17 | fling | kaldemar: ok, thanks. |
| 08:10.57 | v0lZy | uh, guys, i get a problem.. |
| 08:11.34 | v0lZy | Jun 29 10:06:24 asterisk[1507]: WARNING[1529]: chan_sip.c:3552 in retrans_pkt: Retransmission timeout reached on transmission 47a65ccc03ca09da2c26a93d75a4b84d@192.168.1.2:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/dis |
| 08:11.36 | v0lZy | Jun 29 10:06:24 asterisk[1507]: WARNING[1529]: chan_sip.c:3581 in retrans_pkt: Hanging up call 47a65ccc03ca09da2c26a93d75a4b84d@192.168.1.2:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). |
| 08:12.10 | v0lZy | i have no idea why this is happening, but could it be powe routs effecting the phones/switches? |
| 08:14.05 | kaldemar | v0lZy: did you read the page behind the link? |
| 08:16.08 | v0lZy | page doesnt show up, but i searched the site for siP retransmissions |
| 08:16.53 | v0lZy | my PBX and my phones connect to the same switch |
| 08:17.01 | v0lZy | and that switch connects to my router |
| 08:17.14 | ChannelZ | The packets either aren't making it off your network, or are and aren't being replied to (or making it back in) |
| 08:17.33 | v0lZy | they all have static IP's configured so even if i turn off the router, it should all work nicely (router is just so i can access the gui's from wherever |
| 08:17.39 | v0lZy | and so that the pbx has a line out |
| 08:17.45 | v0lZy | i don thtink its a firewall thing |
| 08:17.53 | v0lZy | because some of my phones kind of like hang. |
| 08:18.00 | v0lZy | could it be the network? |
| 08:18.08 | v0lZy | llike... we get a lot of powerouts. |
| 08:19.27 | v0lZy | packet loss? |
| 08:19.42 | ChannelZ | If the routing is screwed up, or there is a firewall or NAT involved anywhere not configured right, and yes packet loss |
| 08:24.05 | v0lZy | not between the phones and the pbx |
| 08:24.25 | v0lZy | router and firewall is only between the switch and the outside world |
| 08:24.36 | v0lZy | pbx and phone reside on the same switch which resides behind a router. |
| 08:24.57 | v0lZy | and with static IP's etc, you'd figure their traffic doesnt reach the router when calls are internal |
| 08:33.40 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:44.50 | fulcan | How do I generate an sms message using messagesend? |
| 08:45.45 | v0lZy | sudo send me an sms :D |
| 08:46.17 | ChannelZ | v0lZy: well I guess you'd have to do some network sluething and try to figure out if the packets are making it to their intended destinations, or if their replies aren't making it back or are otherwise being misdirected |
| 08:46.33 | ChannelZ | It could literally be 50 different things |
| 08:50.51 | v0lZy | thing is |
| 08:50.53 | v0lZy | i restart the phones |
| 08:50.55 | v0lZy | and its ok |
| 08:52.57 | ChannelZ | fulcan: Guessing Set(MESSAGE(body)="Fart on my head") then MessageSend(SIP:someone@somewhere) |
| 08:53.39 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:53.44 | ChannelZ | Maybe the phones are losing their minds. Dunno. |
| 08:55.20 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:03.35 | fulcan | ChannelZ MessageSend(SIP:someone@somewhere) at the cli? |
| 09:11.41 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 09:11.54 | Get_The_Fish | fulcan you have been given the link to do this by me an several others. You have to read it. |
| 09:13.06 | Get_The_Fish | If you are unsure about it, or the directions in the link are unclear, you need to read the ~book |
| 09:13.35 | fulcan | Get_The_Fish I must be missing this link somehow because I have read everthing sent. Can you kindly resent so I can verify? |
| 09:13.57 | Get_The_Fish | https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend |
| 09:15.12 | fulcan | Get_The_Fish yup, been there all night long. Where does that syntax go? not the cli I can assure you. :( |
| 09:15.17 | Get_The_Fish | ~book |
| 09:15.17 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:15.42 | fulcan | Get_The_Fish I've got that one too already. |
| 09:16.05 | Get_The_Fish | MessageSend is a DIALPLAN application. The book covers dialplan applications in detail. |
| 09:16.36 | fulcan | Get_The_Fish how is raw text injected into body? |
| 09:16.47 | Get_The_Fish | " |
| 09:16.59 | fulcan | Get_The_Fish your assuming it is coming from another comm device. |
| 09:17.23 | Get_The_Fish | "The body of the message that will be sent is what is currently set to MESSAGE(body)." |
| 09:18.33 | fulcan | Get_The_Fish "body of the message" is not coming from a comm device. It has to come from * cli or a script on the server. |
| 09:19.17 | Get_The_Fish | fulcn |
| 09:19.31 | Get_The_Fish | fulcan http://bit.ly/LkGkOD |
| 09:19.47 | fulcan | Get_The_Fish can this syntax be placed into a .call file and digested into asterisk? |
| 09:21.00 | fulcan | Get_The_Fish so you are saying I have to use the AGI? |
| 09:21.15 | Get_The_Fish | I never said a thing about AGI. |
| 09:21.36 | fulcan | Get_The_Fish we are not connecting here. |
| 09:21.48 | Get_The_Fish | You have every tool and piece of information you need to accomplish what you seek. |
| 09:22.11 | Get_The_Fish | You need to read. I cannot, and will not, do it for you. Nor will anyone else here. |
| 09:22.31 | fulcan | Get_The_Fish I don't even know where you are going with this |
| 09:23.02 | Get_The_Fish | I'm saying that you need to do some reading. |
| 09:23.42 | fulcan | if I have a string of text sitting on the tip of my tounge "xyzpdq". by what manner will that text enter a 'body of message'? |
| 09:23.53 | fulcan | I already rtfd. |
| 09:24.08 | fulcan | this is a new feature very much NOT documented. |
| 09:27.26 | Get_The_Fish | Any user agent that you use that is capable of SIP messaging will have it's own interface to create, send, display, and store messages. See the documentation for each UA for specific instructions. |
| 09:27.57 | fulcan | Get_The_Fish there is not supposed to be a 3rd party agent. |
| 09:28.13 | fulcan | Get_The_Fish no outside device, nothing. just asterisk. |
| 09:28.17 | Get_The_Fish | According to whom? |
| 09:28.27 | fulcan | Get_The_Fish to me, the designer. |
| 09:28.34 | Get_The_Fish | that is for you to design fulcan |
| 09:28.44 | Get_The_Fish | ~book |
| 09:28.44 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:29.10 | fulcan | Get_The_Fish how about the definative guide to messagesend, you got that one? |
| 09:31.08 | *** join/#asterisk pithagorians (~pithagori@217.12.122.18) |
| 09:31.34 | Get_The_Fish | it's covered. I just created a short dialplan that sent a SIP message to a UA on my network. |
| 09:31.47 | Get_The_Fish | so yeah, I got that one. |
| 09:32.06 | fulcan | Get_The_Fish how did you include custom text? |
| 09:32.36 | Get_The_Fish | by reading what was here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_MESSAGE |
| 09:32.47 | Get_The_Fish | and using the knowledge that I gained from here: |
| 09:32.48 | Get_The_Fish | ~book |
| 09:32.49 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:33.43 | fulcan | Get_The_Fish can you point to a line number specifically allocated to 'body of message'? |
| 09:33.46 | pithagorians | anybody had issues integrating asterisk and openfire with asterisk-im plugin that is sending "On phone" status into the XMPP client? on my side it send the status but it changes back to "Online" even the user is still on phone |
| 09:33.59 | Get_The_Fish | I already did. |
| 09:34.21 | pithagorians | other clients like Spark and Empathy works correctly |
| 09:35.02 | Get_The_Fish | pithagorians, yes, I think that you need use a hint and config asterisk-im to look at that context?? Been years since I messed with it. |
| 09:35.38 | pithagorians | i know it is not related to asterisk, but on pidgin community nobody is claiming such a problem |
| 09:35.40 | fulcan | MESSAGE(body) what are my options for declaring this. Python via uri? HOW cuz static won't work at all (unless static in an outside file). |
| 09:37.01 | fulcan | Get_The_Fish because MESSAGE(body) removed the body from an inbound sms. mine is being generated locally. |
| 09:38.18 | Get_The_Fish | http://ofps.oreilly.com/titles/9780596517342/asterisk-DP-Deeper.html |
| 09:39.24 | Get_The_Fish | pithagorians is asterisk-im still maintained? |
| 09:40.47 | pithagorians | seams like that |
| 09:41.00 | pithagorians | it is in the openfire plugins list |
| 09:41.05 | pithagorians | in the web ui |
| 09:41.16 | fulcan | Get_The_Fish the link has nothing about generating text from an outside source. I looked there before too. |
| 09:41.52 | Get_The_Fish | yes it does. |
| 09:42.08 | Get_The_Fish | You seriously need to read. I cant tell you this any other way. |
| 09:42.11 | fulcan | Get_The_Fish line number or section? |
| 09:42.14 | fulcan | I HAVE!!!! |
| 09:42.21 | Get_The_Fish | that link has information on ASTERISK FUNCTIONS |
| 09:42.41 | Get_The_Fish | OF WHICH MESSAGE() IS A FUNCTION. |
| 09:43.12 | Get_The_Fish | If you had read the book you would already understand what an asterisk function is, what is does, and how it's used. |
| 09:43.39 | Get_The_Fish | But you don't. So you either read the entire book and didn't understand a thing or you didnt read a thing. |
| 09:45.07 | Samuel_CTX | fulcan: from where does your body come? How do you want Asterisk to know it? |
| 09:46.12 | fulcan | Get_The_Fish no sir, it discusses ${VoicemailMessage} but never message(). |
| 09:46.54 | Get_The_Fish | doing a find on a page will not give you the information you seek. |
| 09:46.57 | Get_The_Fish | You have to READ |
| 09:47.02 | *** join/#asterisk Jo_Bell (sasishiju@115.249.39.106) |
| 09:47.40 | fulcan | Samuel_CTX that is the MAJOR question at hand. I am trying to find any local method of doing this without out third party devices like a script injecting the message or a simple cli stroke of the keyboard. |
| 09:47.49 | fulcan | Get_The_Fish I HAVE!!!!!!! |
| 09:48.10 | Get_The_Fish | AGAIN, THIS MESSAGE() IS AN ASTERISK FUNCTION. |
| 09:48.40 | Jo_Bell | Hi, how to get hold of media data (such as buffer pointer or some structure handle of the media data) when a call is done from a SIP phone to another through Asterisk? |
| 09:48.52 | Get_The_Fish | THAT CHAPTER COVERS ASTERISK FUNCTIONS. I HAVE GIVEN YOU THE COMPLETE BREAKDOWN OF THE FUNCTION, AND A LINK THAT HAD GOOGLE RESULTS THAT HAD EXAMPLES. |
| 09:49.12 | fulcan | Get_The_Fish the word message occusrs 12 time on that link. 4 of which are directly related to voicemail. the other aren't even close. |
| 09:49.30 | Get_The_Fish | STOP DOING A FIND ON THE PAGE |
| 09:49.32 | fulcan | Get_The_Fish I am looking for specifics on 1 (one). |
| 09:49.37 | fulcan | Get_The_Fish not all! |
| 09:51.01 | Samuel_CTX | fulcan: you can make a call to Local/destionation-message@sendsmscontext, and then in endsmscontext you can use CUT ( http://www.voip-info.org/wiki/view/Asterisk+func+cut ) to CUT ${EXTEN} and get the message, but I think thiw will not work with spaces, and will sure not work when there are - in your message |
| 09:51.53 | Samuel_CTX | Or you can use this to read it from a file: http://www.voip-info.org/wiki/view/Asterisk+cmd+ReadFile |
| 09:52.06 | Get_The_Fish | this is going to hurt.... |
| 09:52.23 | Get_The_Fish | sorry Jo_Bell what specifically are you looking for> |
| 09:52.24 | Get_The_Fish | ? |
| 09:53.20 | fulcan | Samuel_CTX readfile can be concatinated with sendmessage? |
| 09:53.47 | Jo_Bell | Get_The_Fish: I want to get hold of the audio data from the SIP phones when it reaches Asterisk server and want to do some processing on it and pass it to an already running application server |
| 09:55.25 | Get_The_Fish | define "audio data" - as in RTP payload? Perhaps some transcoding on an audio stream? |
| 09:55.33 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 09:56.17 | Samuel_CTX | fulcan: with ReadFile you read a file on your harddrive, and put the content in an variable. I never used MessageSend, but As I understand it needs a variable MESSAGE(body) |
| 09:57.34 | Samuel_CTX | so theortically. You do: exten => _X.,n,ReadFile(lMESSAGE(body)=/tmp/SMSbody.txt,140) |
| 09:58.17 | Samuel_CTX | And hten the first 140 characters of the file /tmp/SMSbody.txt are in the MESSAGE(body) |
| 09:58.34 | Samuel_CTX | and sorry type, its exten => _X.,n,ReadFile(MESSAGE(body)=/tmp/SMSbody.txt,140) |
| 09:58.42 | Get_The_Fish | let the copy and paste begin... |
| 10:01.30 | kaldemar | app ReadFile is deprecated, Set(MESSAGE(body)=${FILE(/tmp/SMSbody.txt,0,140)}) would also do. |
| 10:02.16 | fulcan | Samuel_CTX like this? -> s,n,MessageSend(2125551212,2124441212).,ReadFile(blah=/home/myText.txt,20); |
| 10:03.02 | Jo_Bell | Requirement is to get the data coming from Usual SIP phones and convert it to speex format and send it to the application server. DO we have any generic format for all the available SIP hard/soft phones? |
| 10:03.18 | Jo_Bell | If there are multiple, i have to parse the data and convert to speex and send. |
| 10:03.40 | Get_The_Fish | Jo_Bell, Asterisk will do it all for you. |
| 10:03.58 | Samuel_CTX | eum????? 1 action on 1 line. Asterisk will remember what it did before. First you get the variable, then on the next line you use it. |
| 10:05.36 | Get_The_Fish | you could register with the remote server, then dial application and bridge the call the remote server |
| 10:05.53 | fulcan | Samuel_CTX line 1 -> s,n,Set(MESSAGE(body)=${FILE(/tmp/SMSbody.txt,0,140)}) then next -> s,n,MessageSend(SIP:2125551212,2124441212); ? |
| 10:06.39 | Get_The_Fish | oops Jo_Bell you could register with the remote server then dial it and asterisk will bridge the call (and handle the codec negotiation for you). |
| 10:06.41 | Samuel_CTX | fulcan, line 1 always has to be number 1 (al least asteriusk 1.4-1.8). so like 1 is s,1,.... |
| 10:07.12 | fulcan | ? |
| 10:07.45 | fulcan | my screen just went crazy |
| 10:08.48 | fulcan | Samuel_CTX right now I am mainly worried about this one function line or 2. They are the core of my issue my friend. |
| 10:08.59 | *** join/#asterisk wonderworld (~ww@dsdf-4db5cdef.pool.mediaWays.net) |
| 10:10.19 | kaldemar | fulcan: if you ask "like this?", show something concrete. "-> s,n,MessageSend(2125551212,2124441212).,ReadFile(blah=/home/myText.txt,20);" is garbage by syntax. |
| 10:11.18 | Samuel_CTX | fulcan: I never used MessageSend, and we don't run Asterisk 10, So I can't check that. The SIP part looks wrong to me. (you want to do it over SIP right?) |
| 10:11.21 | fulcan | Samuel_CTX http://pastie.org/4170617 |
| 10:12.27 | Samuel_CTX | fulcan: so wat does ${MESSAGE(from)} and ${MESSAGE(to)} contain? |
| 10:13.47 | fulcan | Samuel_CTX oops http://pastie.org/4170617 |
| 10:14.22 | Samuel_CTX | try the last line exten => s,n,MessageSend(SIP/peer,2124441212) OR exten => s,n,MessageSend(SIP/trunk/destination,2124441212) |
| 10:14.35 | Jo_Bell | Get_The_Fish: didnt get fully. Do you mean to say that the Remote Application Server also needs to be registered as a SIP user of the Asterisk? |
| 10:14.53 | Get_The_Fish | that would be the easiest way |
| 10:15.11 | Samuel_CTX | for the rest execute originate local/123@smsdial application echo on your CLI and see what happens |
| 10:17.11 | Samuel_CTX | and also on your cli execute core show application MessageSend and see what that tells you |
| 10:18.20 | fulcan | Samuel_CTX this looks like exactly what I was looking for http://pastie.org/4170636 |
| 10:18.34 | fulcan | Samuel_CTX give me a second to clean some stuff up. |
| 10:19.37 | Samuel_CTX | fulcan: an originate needs 2 parts, thats why I added application echo to it. that way it will run. |
| 10:19.56 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:20.50 | Jo_Bell | Get_The_Fish: Then to route the calls to the remote server, Asterisk works similar to dialing an extension. Is this correct understanding? |
| 10:21.05 | Get_The_Fish | yes, |
| 10:22.53 | kaldemar | Jo_Bell: from asterisk's point of view, it does not matter if the remote end is a phone, some server or whatever. |
| 10:23.32 | fulcan | Samuel_CTX http://pastie.org/4170657 |
| 10:23.54 | Jo_Bell | Get_The_Fish: Fine. What about the codec negotiation? Is there any thing like, users can mention the preferred codec as part of registering the extension? |
| 10:24.11 | Jo_Bell | kaldemar: thanks |
| 10:24.18 | Get_The_Fish | exactly that Jo_Bell, it's in the configuration file |
| 10:24.58 | Jo_Bell | Get_The_Fish: oh. good. Is Speex supported? :-) |
| 10:25.25 | kaldemar | Jo_Bell: it's not a part of registration. it's a part of call setup. registration is just a way to let the other end know where you are. |
| 10:25.42 | Get_The_Fish | I do believe do |
| 10:25.49 | Get_The_Fish | *so |
| 10:25.50 | Samuel_CTX | fulcan, now you are executing some sms() command??? what is it? did you reload? |
| 10:26.03 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:27.20 | fulcan | Samuel_CTX I guess not. no errors now! :) http://pastie.org/4170673 a text didn't arrive either but this is huge! |
| 10:28.00 | kaldemar | fulcan: did you read your output? |
| 10:28.29 | Jo_Bell | kaldemar: Ok. Is there any uniformity in the the format of the media data from SIP phones ( soft / hard ) ? |
| 10:28.41 | kaldemar | that's far from huge, a notice that you don't have an extension in [smsdial] that would match 123. |
| 10:28.49 | kaldemar | Jo_Bell: what do you mean? |
| 10:29.11 | fulcan | kaldemar yes, no extension. it should be the DID. |
| 10:29.16 | kaldemar | Jo_Bell: they implement standard protocols and codecs if that's what you mean. |
| 10:30.48 | Samuel_CTX | fulcan: no sutch context or extention. So or you don't have the context smsdial, or it doesn't accept the extention 123 |
| 10:30.50 | Jo_Bell | kaldemar: yes, my questions was just to make sure that the media data payload from SIP phones does not go beyond the support of Asterisk. |
| 10:31.19 | kaldemar | Jo_Bell: you'd have to elaborate on "get the data coming from Usual SIP phones" and "send it to the application server". |
| 10:31.35 | kaldemar | Jo_Bell: extremely unlikely. |
| 10:32.16 | kaldemar | i have yet to see a SIP phone that only speaks codecs not supported by asterisk. |
| 10:32.29 | fulcan | Samuel_CTX http://pastie.org/4170690 |
| 10:33.37 | Jo_Bell | kaldemar: that is what i wanted. if you havent seen, it gives me a good statistics. I will elaborate. |
| 10:33.38 | fulcan | Samuel_CTX I have a context smsdial but not an extension. I should just need to create the DID extension in the smsdial context and that would fix it? |
| 10:33.39 | kaldemar | Jo_Bell: in case of G.729a you need codec licence(s) to be able to transcode though. |
| 10:34.38 | kaldemar | fulcan: looks like you do have an extension. it is called "s". |
| 10:35.57 | Samuel_CTX | fulcan: s should accept everything |
| 10:36.35 | Samuel_CTX | in what file did you put this code? |
| 10:37.21 | fulcan | Samuel_CTX :) so close http://pastie.org/4170718 |
| 10:37.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:a138:f7f4:4ec7:ede5) |
| 10:37.58 | fulcan | No message technology 'SIP/peer' found. |
| 10:38.26 | Samuel_CTX | yes, you need to replace peer by the name of your sip peer where you want to send the sms to |
| 10:42.18 | fulcan | sip show peers -> anveo/0742510424 72.9.19.9 N 5010 Unmonitored but I tried MessageSend(anveo/0742510424,2124442996) and MessageSend(SIP/anveo,2124442996) and get the same technology error?? |
| 10:42.56 | fulcan | No message technology 'anveo/0742510424' found. |
| 10:43.36 | Jo_Bell | kaldemar: As of now I have a server based conferencing solution with video/audio facilities, all through computers. Now we want to enhance it and include phone / soft phone customers also in to the meeting. In case of phones/soft phones, they will connect to an Asterisk Server and Asterisk has to transmit the data in Speex format ( as done by other computer clients) to the application server. |
| 10:43.37 | Jo_Bell | The application server will transmit data from computer clients to the Asterisk server and asterisk server has to give the data back to the phones/soft phones. |
| 10:44.02 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 10:44.19 | Jo_Bell | I am trying to learn Asterisk these days and design a solution for the above mentioned draft requirement statement. |
| 10:44.25 | Samuel_CTX | fulcan: can you call SIP/anveo ? |
| 10:44.50 | fulcan | Samuel_CTX yes, same tech error. :) |
| 10:44.53 | fulcan | :( |
| 10:45.29 | Samuel_CTX | so I think your anveo thing is not configured correctly? |
| 10:45.44 | fulcan | Samuel_CTX No message technology 'SIP/anveo' found. |
| 10:45.45 | fling | fulcan: hey :p |
| 10:46.02 | kaldemar | Jo_Bell: how were you going to connect asterisk to the existing conferencing solution? |
| 10:46.07 | fulcan | fling long time no see fling, how you been? |
| 10:46.32 | fling | fulcan: fine, installed asterisk :p |
| 10:46.46 | fling | fulcan: how are your doing? |
| 10:48.24 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:48.59 | Jo_Bell | kaldemar: i havent reached that far. I was trying out elementary conf changes in extensions.conf and sip.conf. Today I understood that the server can be treated as an extension in the dial plan. But I have a query here, for very custom requirements, is it a common practice to add code snippets into the Asterisk source or usually ppl can live with dial plans alone? |
| 10:49.23 | fulcan | Samuel_CTX that is looking for a context in sip.conf, correct? |
| 10:49.56 | fulcan | fling I am building an sms server using Asterisk 10. new toy. :) |
| 10:50.30 | fling | fulcan: oh! idk much about sms |
| 10:51.11 | *** join/#asterisk scubes13 (~scubes13@cpe-098-025-009-117.sc.res.rr.com) |
| 10:52.24 | fulcan | Samuel_CTX here is my sip.conf, I am confused. http://pastie.org/4170761 |
| 10:52.56 | fulcan | fling I'm getting a working mans degree in it today. :P |
| 10:53.13 | fling | wooo cool! |
| 10:54.30 | kaldemar | Jo_Bell: you can do many things with the defalts nowadays, but of course depending on requirements something custom may be needed. |
| 10:55.40 | fling | My ISP is providing me voip over ethernet cable. 4 phone numbers and separate lines with passwords. I have asterisk installed and started. What to do next? What should I read? wiki.asterisk.org ? |
| 10:55.42 | kaldemar | Jo_Bell: the main part to tackle in your scenario is to have a way for asterisk and the conferencing solution to communicate. if that is doable, everything else will work. |
| 10:55.51 | fulcan | fling asterisk is the neatest thing since peanut butter and jelly. I just always get stuck doing something weird to it. :) |
| 10:56.15 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:56.27 | fulcan | fling voip-info.org has a ton in it too. |
| 10:56.50 | fling | I want a fine step-by-step how-to |
| 10:57.18 | kaldemar | voip-info is probably the last place you should look for documentation. |
| 10:58.40 | fulcan | kaldemar No message technology 'SIP/anveo' found. |
| 10:58.55 | fulcan | kaldemar here is my sip.conf, I am confused. http://pastie.org/4170761 |
| 10:59.50 | kaldemar | fulcan: and your dialplan? |
| 11:02.19 | fulcan | kaldemar that was my sip.conf, here is extensions. that error is because of context in the sip.conf is it not? http://pastie.org/4170804 |
| 11:03.09 | fulcan | kaldemar cannot find the SIP context called avneo in order to dial out? |
| 11:07.49 | fulcan | . |
| 11:08.08 | kaldemar | change type=friend to type=peer and do a sip reload |
| 11:08.55 | fling | what config file do I need to use for my phone numbers and passwords? |
| 11:09.09 | kaldemar | fling: what protocol do your phones use? |
| 11:10.10 | fulcan | kaldemar there are no phone, only the sms carrier. I am launching the sms from the * cli. |
| 11:10.11 | fling | kaldemar: I do not have any phones, I have 4 alaw voip phone numbers from my ISP. |
| 11:10.26 | fulcan | fling sip.conf |
| 11:10.38 | fulcan | fling if they are sip phones. |
| 11:11.00 | fulcan | fling cisco uses skinny and sip. |
| 11:11.10 | fling | fulcan: phones you mean physical phone devices? |
| 11:11.31 | fulcan | fling yup, like physical handsets. |
| 11:11.53 | fling | I do not have these |
| 11:12.29 | fling | I want my users to connect to my pbx over internets with voip client |
| 11:13.24 | fulcan | fling if you are working with sip phones and sip carrier providers, authenticating to them and what protocols are managed goes in sip.conf. What happens to the call goes in extensions.conf (your dialplan). |
| 11:13.55 | kaldemar | fling: what protocol are you supposed to use with the ISP? |
| 11:13.56 | fling | so they will be able to talk with each other. And I also have phone numbers and passwords provided by ISP |
| 11:14.31 | fling | kaldemar: they support alaw only |
| 11:14.52 | kaldemar | fling: alaw is not a protocol, it is a codec. |
| 11:14.59 | fling | oh! |
| 11:15.52 | fling | voip? :p |
| 11:16.13 | kaldemar | voip is not a protocol either, but an umbrella term. |
| 11:16.15 | fulcan | fling compression actually. |
| 11:16.37 | kaldemar | SIP would be a protocol, but there are others. |
| 11:16.55 | fling | they say SIP |
| 11:17.12 | kaldemar | okay. then sip.conf & extensions.conf like stated earlier. |
| 11:17.35 | fling | thanks! |
| 11:17.51 | fulcan | fling http://www.voip-info.org/wiki/view/Codecs |
| 11:18.12 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 11:18.15 | fulcan | kaldemar any thoughts on my little technology error my friend? |
| 11:18.19 | fling | so I put ISP provided creds and my users creds to sip.conf… |
| 11:19.38 | fulcan | fling don't forget your register statement. and as I was reminded earlier, it must be in the [default] context of sip.conf |
| 11:20.00 | fling | fulcan: ITU G.711 < it is |
| 11:20.29 | fulcan | fling register to IVSP register => statement |
| 11:21.04 | fling | fulcan: ok, thanks |
| 11:21.13 | fulcan | fling typically g.711 |
| 11:22.03 | kaldemar | fulcan: did you try what i suggested last? |
| 11:24.20 | fulcan | kaldemar yes sir, same error. :( |
| 11:25.34 | fulcan | kaldemar peer is same error. |
| 11:26.05 | fling | gtg, cya |
| 11:28.34 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 11:28.52 | kaldemar | fulcan: try dropping the "SIP/" part |
| 11:30.06 | fulcan | kaldemar No message technology 'anveo' found |
| 11:34.15 | kaldemar | how about sip:anvil since it wants a URI? |
| 11:35.43 | kaldemar | you also may need to change the numeric from part to a URI. |
| 11:37.26 | fulcan | kaldemar no error but not sms has hit my phone yet ? http://pastie.org/4170925 |
| 11:38.36 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 11:38.44 | *** join/#asterisk Jo_Bell (sasishiju@115.249.39.106) |
| 11:39.04 | kaldemar | is your provider supposed to convert sip MESSAGEs to SMS's? |
| 11:39.46 | fulcan | kaldemar yes |
| 11:40.12 | kaldemar | ask them what they require for it to work. |
| 11:40.28 | fulcan | kaldemar this is now an issue with them though. the sip debug clearly states 'successful transmission'. |
| 11:40.35 | Jo_Bell | kaldemar: Transcoding ( changing the SIP data to Speex ) and then forwarding to conferencing solution is readily supported in Asterisk right? |
| 11:40.58 | fulcan | kaldemar thank you soooo much! I have to post with them and they take a couple of hrs. |
| 11:42.11 | fulcan | kaldemar http://www.anveo.com/api.asp?code=apihelp_sms_receive_sip |
| 11:43.31 | kaldemar | Jo_Bell: yes. but the conferencing solution and asterisk must have a common protocol to communicate. |
| 11:44.10 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 11:44.26 | kaldemar | fulcan: you missed the header parts. |
| 11:45.19 | kaldemar | fulcan: SIPAddHeader(X-ANVEO-SMS-FROM: yournumbergoeshere) |
| 11:45.28 | fulcan | kaldemar oops, your right. How do I change those? |
| 11:45.38 | kaldemar | they may well drop all messages without a valid from number. |
| 11:45.55 | fulcan | kaldemar where do I add that header statement? |
| 11:46.16 | kaldemar | ugh. that may not work as the SIPAddHeader is supposed to work with the Dial app only. |
| 11:46.52 | kaldemar | however, as the page states you can also use INVITE instead of MESSAGE. |
| 11:47.00 | Jo_Bell | Kaldemar : As per my understanding, the difficult part is to send the computer clients' data to Asterisk and from there to the phone clients. Theoretically, is this flow similar to any extension trying to connect any other extension ? |
| 11:47.47 | kaldemar | fulcan: add the from-, to- and message headers and just dial them if the header is not added to messages created by MessageSend. |
| 11:48.40 | kaldemar | Jo_Bell: no, that's not difficult at all. you can set up a conference in asterisk which the phones connect to. |
| 11:49.40 | kaldemar | Jo_Bell: or just make the phones dial via asterisk to the conferencing solution and it will just work. start with the connection between asterisk and the conferencing solution. |
| 11:49.49 | kaldemar | Jo_Bell: everything else is a peace of cake. |
| 11:52.21 | Jo_Bell | kaldemar - Oh good. you said about the difficulty in implementing a common protocol for both Asterisk and the conferencing solution? Which other area do you think will be difficult to implement? Sorry for trivial queries, I need to collect some expert voice before digging in to get going. |
| 11:53.15 | kaldemar | Jo_Bell: i didn't say anything about implementing anything. check if they have a common protocol already. |
| 11:53.20 | *** part/#asterisk k-man (~jason@unaffiliated/k-man) |
| 11:58.00 | fulcan | kaldemar like this? http://pastie.org/4171011 |
| 11:58.22 | Jo_Bell | kaldemar: Fine. Let me check the stuff. One more query - I didnt quite get your suggestion to check the 'common protocol'. The conferencing server is a Java based Server using SSL sockets. By 'common protocol' which abstraction do you refer to? |
| 11:58.43 | fulcan | kaldemar Got SIP MESSAGE response 404 "Not here" back from host '72.9.149.69:5010' |
| 11:58.53 | *** part/#asterisk TrevorH1 (~trevor@cpc3-brig16-2-0-cust913.3-3.cable.virginmedia.com) |
| 11:58.56 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-0-168.bois.qwest.net) |
| 12:01.23 | kaldemar | fulcan: "WTF" describes the last pastebin... |
| 12:01.29 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 12:01.41 | kaldemar | where did you come up with this: exten => 19175039892,n,NoOp(MESSAGE(from)'X-ANVEO-SMS-FROM SIP') ? |
| 12:03.23 | kaldemar | first of all, app NoOp just prints stuff to CLI, it does not do anything else. second, MESSAGE(from) is not even writable and "X-ANVEO-SMS-FROM SIP" makes me think you did not understand the sentence on the anveo page at all. |
| 12:03.52 | kaldemar | and i already gave you an example on how to add the particular SIP header. |
| 12:06.49 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 12:07.39 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:09.21 | mirelab | Did anyone have problem with RT QueueMembers, i need help , --->http://forums.asterisk.org/viewtopic.php?f=1&t=83208 |
| 12:09.41 | fulcan | kaldemar I'm just tired :( http://pastie.org/4171063 http://pastie.org/4171066 |
| 12:13.44 | kaldemar | fulcan: change the from URI for MessageSend. "07..." <07...@sip.anveo.com> |
| 12:14.57 | *** join/#asterisk CommaCrazy (1000@nikola.office.bvcom.net) |
| 12:15.31 | [TK]D-Fender | fulcan, They are talking about SIP headers, not SMS. |
| 12:15.49 | [TK]D-Fender | fulcan, You need to set the FROMUSER in your peer to the domain as they showed you. |
| 12:16.10 | [TK]D-Fender | fulcan, DROMDOMAIN rather |
| 12:16.15 | [TK]D-Fender | FROMDOMAIN |
| 12:16.16 | fulcan | kaldemar http://pastie.org/4171106 same |
| 12:17.13 | kaldemar | fulcan: MESSAGE(to) is read-only. you can't set it. and i told you to modify the URI in MessageSend. |
| 12:17.33 | fulcan | [TK]D-Fender add fromuser if it is not there? |
| 12:19.03 | fulcan | [TK]D-Fender in sip.con FROMDOMAIN=sip.anveo.com:5060 I get the same 404 |
| 12:19.04 | [TK]D-Fender | fulcan, fromdomain |
| 12:19.25 | [TK]D-Fender | fromuser might be needed as well. |
| 12:19.33 | [TK]D-Fender | fulcan, pastebin everything. |
| 12:19.45 | [TK]D-Fender | fulcan, never just say "it didn't work". |
| 12:19.55 | [TK]D-Fender | fulcan, call with SIP DEBUG and the configs |
| 12:21.22 | fulcan | [TK]D-Fender my apologies http://pastie.org/4171123 |
| 12:21.26 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
| 12:21.54 | [TK]D-Fender | facepalms |
| 12:22.01 | fulcan | dial plan http://pastie.org/4171106 |
| 12:22.10 | [TK]D-Fender | SIP DEBUG <- |
| 12:22.45 | [TK]D-Fender | Wait... is that even using the peer? |
| 12:23.05 | [TK]D-Fender | could be... |
| 12:23.23 | [TK]D-Fender | and remove the 5060 from the fromdomain |
| 12:24.15 | fulcan | kaldemar I modify the URI in MessageSend isn't clear to me when looking at the syntax requirements. :( |
| 12:25.50 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:26.20 | fulcan | [TK]D-Fender http://pastie.org/4171145 |
| 12:28.08 | fulcan | from anveo http://pastie.org/4171154 |
| 12:28.32 | [TK]D-Fender | fulcan, Last chance on this. I've asked repeatedly. in your CLI output you had better have SIP DEBUG enabled. AND I told you to remove the 6050 from the FROMDOMAIN |
| 12:28.44 | [TK]D-Fender | * CLI SIP DEBUG |
| 12:28.50 | [TK]D-Fender | We've gone through this before. |
| 12:30.54 | Captain_Proton | anyone know a work around for wcte12xp 0000:05:00.0: Failed to load VPMADT032 firmware. I submited a bug, but wanted to see if anyone has had this problem and know a fix |
| 12:31.24 | fulcan | full debug http://pastie.org/4171170 |
| 12:32.43 | kaldemar | [TK]D-Fender: it is using the peer. |
| 12:33.31 | fulcan | My to: is fine but my from: looks incorrect. |
| 12:33.44 | kaldemar | fulcan: why are you still trying to set values to MESSAGE(to) and MESSAGE(from)? THEY ARE READ-ONLY. |
| 12:34.29 | kaldemar | fulcan: "12124442996" in MessageSend is the from URI. |
| 12:35.13 | kaldemar | fulcan: if you configure the from settings in the SIP peer, remove it. if not, modify it to a proper uri. |
| 12:37.29 | fulcan | kaldemar like this? http://pastie.org/4171194 |
| 12:38.33 | fulcan | I think it worked...? http://pastie.org/4171201 |
| 12:41.05 | kaldemar | fulcan: no. what do you think "X-ANVEO-SMS-FROM SIP" is? |
| 12:42.50 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 12:42.53 | fulcan | kaldemar "For MESSAGE SIP Method Anveo uses X-ANVEO-SMS-FROM SIP header to pass sender's phone number" http://www.anveo.com/api.asp?code=apihelp_sms_receive_sip |
| 12:42.59 | kaldemar | fulcan: that is a snippet from a sentence where X-ANVEO-SMS-FROM was a name for a header (which you still are not even trying to add with SIPAddHeader despite me telling you twice) and SIP is a word that does not belong there. |
| 12:43.33 | fulcan | http://pastie.org/4171194 |
| 12:43.46 | kaldemar | why are you pasting that again? |
| 12:44.08 | fulcan | kaldemar that line doesn't exist anymore. |
| 12:44.41 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 12:44.44 | fulcan | kaldemar and -> http://pastie.org/4171201 |
| 12:44.52 | kaldemar | neither X-ANVEO-SMS-FROM nor SIP belongs there. |
| 12:45.47 | kaldemar | X-ANVEO-SMS-FROM is a name for a header just like From, To, Contact or Via. |
| 12:47.39 | fulcan | kaldemar exten => 19175039892,n,MessageSend(sip:19175039892@sip.anveo.com:5010,19175039892) |
| 12:50.18 | kaldemar | fulcan: what does "core show application MessageSend" say about the from URI? |
| 12:52.00 | fulcan | kaldemar well, in the asterisk manual it says https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend MessageSend(to[,from]) hence MessageSend(sip:19175039892@sip.anveo.com:5010(to),19175039892(from). |
| 12:53.08 | kaldemar | fulcan: it says a bunch of other things beyond MessageSend(to[,from]) as well. |
| 12:53.25 | fulcan | kaldemar 1 paragraph? |
| 12:53.30 | kaldemar | which you did not read. |
| 12:53.50 | *** join/#asterisk Neptu (~Neptu@41.141.91.6) |
| 12:54.07 | fulcan | kaldemar looking at it right now. |
| 12:56.15 | fulcan | kaldemar your trying to tell me the outbound context needs to refered to in the from uri? that doesn't make sense. |
| 12:56.42 | fulcan | I am guessing you are looking at this -> For SIP the from parameter can be a configured peer name or in the form of "display-name" . |
| 12:56.52 | kaldemar | "For SIP the from parameter can be a configured peer name or in the form of "display-name" <URI>." |
| 12:57.30 | kaldemar | "in the form of "display-name" <URI>." |
| 12:57.41 | fulcan | kaldemar it wants my outbound context name? because my peers are anveo and the anveo context. |
| 12:58.56 | fulcan | kaldemar -> http://www.anveo.com/api.asp?code=apihelp_sms_receive_sip "pass sender's phone number" |
| 13:00.05 | fulcan | kaldemar ""display-name" <URI>."" does not make sense. |
| 13:00.40 | fulcan | you say 'name' anveo says number. I have no clue. |
| 13:01.33 | kaldemar | i don't say 'name'. i say '"display-name" <URI>'. |
| 13:03.17 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 13:04.21 | fulcan | there is no RFC on "display-name URI", can you define this for me? I know what a uri is, I play with _GET enough in php for a complete tour of the world. How asterisk is imply this I have no clue. |
| 13:06.16 | kaldemar | "yourdisplayname" <user@domain> |
| 13:06.48 | fulcan | because if they are not looking for the origination phone number or user account for 'display-name', I got no ef'fin clue what the author means... |
| 13:08.59 | kaldemar | http://www.ietf.org/rfc/rfc3261.txt gives you a definition of those. |
| 13:10.36 | fulcan | exten => 19175039892,n,MessageSend(sip:19175039892@sip.anveo.com:5010,"Joe Schmuckatelie" 0742510424@sip.anveo.com) |
| 13:12.11 | fulcan | http://pastie.org/4171325 |
| 13:13.36 | kaldemar | fulcan: your command lacks <> |
| 13:14.08 | kaldemar | and possibly sip: |
| 13:15.55 | *** join/#asterisk philfine (~tino@a95-92-251-229.cpe.netcabo.pt) |
| 13:16.07 | philfine | Hi everyone |
| 13:17.06 | fulcan | kaldemar they just wrote back, the only thing hurting me is the return proxy auth failure that is going on. http://pastie.org/4171338 |
| 13:17.15 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gpufsrnnznlqyfhe) |
| 13:17.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:17.42 | fulcan | SIP/2.0 407 Proxy Authentication Required |
| 13:17.44 | *** join/#asterisk blindhog (~josh@173.219.55.70) |
| 13:17.49 | philfine | I am getting the famous Symbol not found: ___ast_adsi_available on mac os x 10.7.4 after compiling 10.5.1 using GCC or LLVM (makes no difference). |
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| 13:18.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:18.14 | philfine | From all I googled I could not find a definite answer for why this happens |
| 13:18.27 | fulcan | http://pastie.org/4171346 |
| 13:18.46 | philfine | By looking at the modules I can see res_adsi.so there and when doing nm of it the function is available |
| 13:19.09 | fulcan | http://pastie.org/4171350 |
| 13:19.36 | philfine | Any hints why this happens ? |
| 13:20.08 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:22.37 | philfine | Online I see lots of people where able to install it correctly, so I should be able to do the same |
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| 13:23.32 | fulcan | philfine I would use macports personally. |
| 13:24.43 | philfine | Well, I would like to be able to compile it :D |
| 13:25.08 | philfine | Not just because I want to … :) |
| 13:26.57 | *** join/#asterisk Tuju (~tuju@176.75.219.213.sta.estpak.ee) |
| 13:27.40 | fulcan | philfine well then I would install xcode and macports in order to get the developmental libraries up2date (mac gui sucks for this) and then chase down a source install. |
| 13:30.20 | *** join/#asterisk serafie (~erin@nat/digium/x-fgwdntouyftbvjsn) |
| 13:32.40 | kaldemar | fulcan: add auth=<user>:<secret>@anveo.com under [authentication] in sip.conf |
| 13:32.46 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:33.11 | bulkorok | anyone knows how I can send the CED tone when receiving a fax!? |
| 13:33.51 | fulcan | kaldemar on it! |
| 13:37.05 | philfine | fulcan: I have Xcode and macports installed |
| 13:37.16 | philfine | This looks like a linking problem |
| 13:37.36 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:38.00 | philfine | It install the modules to /Library/Application Support/Asterisk/Modules but the binary of asterisk is not "dynamic linking" to that particular directory |
| 13:38.44 | fulcan | kaldemar Got SIP MESSAGE response 404 "Not here" back from host '72.9.149.69:5010' <--I made auth=dummy:dummy@anveo.com the first heading under the context [authentication] --> http://pastie.org/4171437 |
| 13:44.41 | fulcan | philfine then cheat, install * from ports, pull in all the dependancies and then simply remove it then compile asterisk by hand. |
| 13:48.50 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
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| 13:50.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:53.52 | fulcan | philfine sudo port -v selfupdate |
| 13:55.11 | philfine | Clearly there is a problem on asterisk regarding linkage on Lion. I would rather understand what it is and submit some patch |
| 13:57.18 | *** join/#asterisk shadebob (~shadebob@41.141.162.46) |
| 14:00.15 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 14:00.39 | shadebob | Hi, is there a way to cancel a reload in progress ? |
| 14:02.57 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-dnwvqeqfayvaexaz) |
| 14:03.54 | [TK]D-Fender | shadebob, No |
| 14:05.23 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
| 14:06.36 | shadebob | [TK]D-Fender : :( thanks |
| 14:07.31 | bulkorok | anyone can help ith T38 and generating the CED tone!? |
| 14:10.02 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 14:13.12 | pzn | I have a call that is made in this "sequence": SIPPhone -> asterisk server (my own PBX) -> asterisk server (sip provider) -> targetphone. I'd like to insert "music" in the call, while the person still didn't answer the call. is it possible do to this in my own server? |
| 14:15.10 | [TK]D-Fender | pzn, "core show application dial" <- |
| 14:15.28 | pzn | [TK]D-Fender, tks, I'll check that |
| 14:16.13 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
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| 14:24.44 | jeffspeff | ok, so i can make calls to and from exten 1501, but if i setup the same phone as exten 1500 it will only make calls. here's the sip debug of trying to call 1500 from 1605 http://pastebin.com/uPtck0px |
| 14:25.01 | *** join/#asterisk shadebob (~shadebob@41.141.162.46) |
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| 14:37.05 | [TK]D-Fender | jeffspeff, Looking for 1500 in FL (domain 10.200.100.238) SIP/2.0 484 Address Incomplete |
| 14:37.09 | [TK]D-Fender | jeffspeff, Dialplan error |
| 14:37.47 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:37.51 | [TK]D-Fender | eek |
| 14:41.27 | *** join/#asterisk timahvo1 (~rogue@196.200.32.36) |
| 14:42.18 | Captain_Proton | is vpmadt032 firmware in the dahdi-linux-complete source? if I complete it willl it also create the firmware |
| 14:43.04 | Captain_Proton | compile* |
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| 14:52.07 | Qwell | Captain_Proton: It's not in the source, but it gets downloaded as part of the build process. |
| 14:54.04 | Captain_Proton | Qwell, one more ? I have dahdi and the firmware install now ut there a problem with it bug # DAHLIN-296 if I build it from source can I just install over top of it ? |
| 14:55.13 | Qwell | Captain_Proton: Installing will overwrite stuff that's already there. |
| 14:56.11 | jeffspeff | [TK]D-Fender, i have 1500 in the same context as 1501 |
| 14:56.29 | jeffspeff | and i searched through, no pattern matching on 15XX or anything |
| 14:56.32 | [TK]D-Fender | jeffspeff, PASTEBIN |
| 14:56.39 | jeffspeff | k, just a sec |
| 14:56.42 | Captain_Proton | Qwell, you think I can build it NOT installing with asterisk w/ dahdi running this is a production system or should I wait till tonight |
| 14:57.23 | jeffspeff | [TK]D-Fender, http://pastebin.com/8dkjgQFr |
| 14:57.57 | Katty | hello my asterisk does not work at all |
| 14:57.58 | Katty | how to fix plz |
| 14:58.11 | newtonr | Katty: re-install windows |
| 14:58.19 | Katty | i've heard that one before |
| 14:58.22 | Katty | i want an ORIGINAL SIN |
| 14:58.28 | Katty | s/sin/suggestion |
| 14:58.35 | Qwell | Captain_Proton: If it's running, then it's already in memory. Installing won't hurt anything until you restart. |
| 15:03.06 | chuckf | Katty: plug in the network cable |
| 15:03.42 | Katty | :> |
| 15:03.45 | Katty | FIXED! |
| 15:03.54 | Katty | i should start keeping a list |
| 15:04.14 | chuckf | yay |
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| 15:09.15 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 15:09.35 | Captain_Proton | Qwell, well it going to have the same problem :( WARNING: could not find /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32.o.cmd for /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o |
| 15:09.57 | Qwell | That's a warning. |
| 15:10.03 | Qwell | You can ignore it. |
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| 15:17.28 | jeffspeff | [TK]D-Fender, do you anything that would cause 1500 not to work? |
| 15:18.54 | [TK]D-Fender | jeffspeff, not yet... |
| 15:19.02 | jeffspeff | ok |
| 15:19.41 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:19.53 | jeffspeff | [TK]D-Fender, by the way, when i try to do a console dial to 1500 from the * cli, it says that 1500 doesn't exist in AR-EXTENSIONS, but console dialing 1501 works |
| 15:23.09 | [TK]D-Fender | "dialplan show ar-extensions" |
| 15:24.01 | Katty | arrrrrr extensions. |
| 15:24.05 | Katty | now with FEATHERS! |
| 15:24.07 | *** join/#asterisk GameGamer43 (users.5533@gateway/web/irccloud.com/x-bqfrygdycvoqmnov) |
| 15:24.56 | jeffspeff | [TK]D-Fender, this is too weird |
| 15:25.07 | jeffspeff | for giggles, i did yet another dialplan reload, and now they work |
| 15:25.14 | jeffspeff | this makes now sense |
| 15:25.16 | jeffspeff | *no |
| 15:25.27 | jeffspeff | i even restarted the * service yesterday trying to resolve this |
| 15:25.36 | [TK]D-Fender | jeffspeff, Is something generating that dialplan that might not have finished during a load process? |
| 15:25.50 | jeffspeff | the dialplan is a flat file |
| 15:25.56 | jeffspeff | i'm generating it by had |
| 15:25.58 | jeffspeff | *hand |
| 15:28.02 | [TK]D-Fender | jeffspeff, At which point : you desperately need to learn to start using macros |
| 15:28.04 | [TK]D-Fender | :p |
| 15:28.09 | jeffspeff | i know |
| 15:28.15 | jeffspeff | this is stage 1 of the DP |
| 15:28.42 | jeffspeff | stage 2 will make heavy use of macros and store in realtime mysql |
| 15:29.17 | jeffspeff | stage 3 will have a windows forms gui to manage all the realtime databases and integrate quite a bit with the AMI |
| 15:29.59 | darrenlooby | I need to get a unique and persistent id for a call flow... rather than uniqueid - which is different per leg of the call - I'd like an id that follows the call around no matter what it does from the point that it first starts in the server... any ideas? |
| 15:35.18 | darrenlooby | Much like "CallAPIID" in VoIPNow... http://wiki.4psa.com/display/VSOAP/CallAPI+Events+Default+Request+Parameters |
| 15:46.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 16:05.34 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 16:06.09 | ThinkGNU- | Is there any way to make the outgoing call always show the caller id specified in the outgoing SIP profile other than setting it in the dialplan? |
| 16:06.58 | [TK]D-Fender | ThinkGNU-, "FROMUSER" |
| 16:07.28 | ThinkGNU- | Can that take a string as well as numbers? |
| 16:07.43 | ThinkGNU- | I've got the number in there so that we authenticate with our SIP provider |
| 16:22.38 | [TK]D-Fender | ThinkGNU-, then make sure you don't send rpid |
| 16:29.08 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-yupgxozwbdkhcxvk) |
| 16:32.42 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 16:34.05 | *** join/#asterisk ziz212 (~chatzilla@61.245.172.53) |
| 16:34.48 | [TK]D-Fender | correct |
| 16:34.53 | ziz212 | Hi firends, Is there any way to temp disable sip account in asterisk ? if it is what will be the parameter? |
| 16:35.26 | [TK]D-Fender | ziz212, comment it out |
| 16:36.13 | [TK]D-Fender | or use permit/deny to block it. Or point it's context to somewhere non-existent |
| 16:36.38 | ziz212 | if it is whole lines related to that need to be comment out in sip.conf. It is not practical, is there any specific parameter need to be consider? |
| 16:37.02 | ThinkGNU- | [TK]D-Fender: That parameter has had no affect. |
| 16:37.03 | [TK]D-Fender | I just told you THREE |
| 16:37.11 | [TK]D-Fender | ThinkGNU-, show us |
| 16:39.23 | ziz212 | Thanks for the comment |
| 16:41.40 | *** join/#asterisk zoell (~zoell@analsexlover.net) |
| 16:48.53 | JamKo | ziz212: you could aslo change the name of the UA to something alternate, like 101-disabled, and of course not update the client. As long as you are using registration. |
| 16:52.07 | ziz212 | thanks for your comment |
| 16:52.29 | *** join/#asterisk viro (~viro101@cpe-71-72-174-83.woh.res.rr.com) |
| 16:53.19 | JamKo | ziz212: I would go with the non-existent context though, and just leave yourself a comment about why you did it. This way you don't see a constant stream of failed reg attempts. |
| 16:55.19 | ziz212 | yes your right, if i use authentication option then it getting fail if the uname or passwd change. Thanks |
| 16:58.19 | ziz212 | ok let say if a asterisk pbx sip account controlled with time, last call wont get cut if we use authentication option in the end of the allowed time period? |
| 16:59.30 | [TK]D-Fender | ziz212, there is no "time" factor |
| 16:59.32 | ziz212 | if it is the option is context, so both neeeded |
| 16:59.54 | [TK]D-Fender | And auth will take effect any time a call has reason to. |
| 17:00.01 | ziz212 | let say we use some time limited account with the help of external script |
| 17:00.31 | [TK]D-Fender | ziz212, You're already using FREEPBX for this. This isn't the channel you should be asking this in |
| 17:00.53 | [TK]D-Fender | ziz212, it has rules that DIY-ers don't have to deal with |
| 17:01.19 | ziz212 | no I just told ... any asterisk based pbx is same for me..... |
| 17:01.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:01.45 | [TK]D-Fender | Well if you're modding the text configs, no it isn't |
| 17:01.45 | ziz212 | and i feel it is a A S T E R I S K pbx |
| 17:01.55 | [TK]D-Fender | because FreePBX drives right over them any time you make a change |
| 17:02.10 | ziz212 | yup I know that |
| 17:02.16 | [TK]D-Fender | What you "feel" is irrelevant. What matters is how it works. |
| 17:02.44 | [TK]D-Fender | So script whatever you want, but be aware of the other processes that can walk right over that |
| 17:03.14 | [TK]D-Fender | And you're already bound to one of the bigger ones |
| 17:04.11 | [TK]D-Fender | <ziz212> hi,Is it database contain those settings as like sip_additional.conf? if it is in certain time db settings can be again past to the sip_additional.conf file? <- there are no time settings in either |
| 17:05.41 | [TK]D-Fender | ThinkGNU-, stop using "/notice" to talk to me. And I asked you to SHOW ME. Where's the configs? Where's the failed cal with debug so we can SEE what's happening? |
| 17:06.32 | ziz212 | ok thanks for help see you bye |
| 17:06.49 | ThinkGNU- | Sorry [TK]D-Fender, didn't mean to irritate you. Let me get that info for you. I do appreciate the help you are offering. |
| 17:07.47 | *** part/#asterisk ziz212 (~chatzilla@61.245.172.53) |
| 17:10.04 | *** join/#asterisk grutz (~grutz@grutzv6-1-pt.tunnel.tserv8.dal1.ipv6.he.net) |
| 17:11.07 | grutz | it looks as if outbound SIP via DNS can only lookup/do UDP communication - at least it only searches for _sip._udp.domain.com and never _sip._tcp.domain.com. Is there any reason for this? |
| 17:16.12 | pzn | what does it mean musiconhold class? how can I create a musiconhold class and assing some audio files to it? |
| 17:20.22 | pzn | don't care about my previous question... I found I was changing the configuration in the wrong asterisk server... I have 2 servers and I was changing the configs in the wrong server... |
| 17:20.49 | ThinkGNU- | [TK]D-Fender, do you want me to send you the text file containing the debug info for this call? |
| 17:22.55 | [TK]D-Fender | ~pb |
| 17:22.55 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 17:24.44 | ThinkGNU- | http://pastebin.com/sC2viMPc |
| 17:24.53 | ThinkGNU- | I hope I have what you need in there |
| 17:32.12 | philfine | Does anyone here knows exactly how are traditional phone numbers regulated ? I am from Portugal if that matters ? |
| 17:32.25 | [TK]D-Fender | No, we aren;t looking at the SIP DEBUG to see what is really being sent. We don't see basic verbose to see the dialplan processing as this isn't even the WHOLE call. And that core debug is of no use. |
| 17:32.40 | Qwell | philfine: it's done so by your local numbering authority or government. |
| 17:32.51 | Qwell | In North America (Canada, US, Mexico), that's NANP |
| 17:33.05 | philfine | Can you buy numbers directly to them ? |
| 17:33.06 | Qwell | err, NANPA. |
| 17:33.09 | Qwell | no |
| 17:33.26 | Qwell | Talk to your telco, or find an ITSP. |
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| 17:35.17 | philfine | I guess there is a pretty big monopoly in that respect. Phone number should be more like DNS servers, don't you think ? Or there is some serious problem with that ? |
| 17:35.54 | [TK]D-Fender | phiProblem being : the world is big. Area have rules. Go learn your own |
| 17:36.00 | [TK]D-Fender | philfine, ^ |
| 17:36.12 | WIMPy | Move to a country with an open number plan, get a line with DDI and do it yourself. |
| 17:37.37 | philfine | Sure, so do DNS :D |
| 17:38.06 | philfine | but you pay for the domain and not for the many times the page is accessed |
| 17:38.18 | ThinkGNU- | [TK]D-Fender, I'm not sure how exactly to collect what you're looking for. I have my verbose level at 3 and my verbose log isn't collecting much of anything when the call is made. I appreciate all your help but I can't get the logs so I don't want to waste your time. |
| 17:38.20 | philfine | In case of phones you pay for everything |
| 17:40.48 | ThinkGNU- | [TK]D-Fender I'm still learning and don't expect the folks here that know what they're doing to step me through every little part of running a system. I'll keep reading and see what I can figure out. Thanks again. |
| 17:44.36 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 17:44.55 | [TK]D-Fender | ThinkGNU-, "sip set debug on", and live CLI only, no logs |
| 17:45.04 | slav3_kitten | i hate how it takes a minute for nickserv to identify me, and this is reg only chan |
| 17:45.18 | Qwell | slav3_kitten: Use a server password |
| 17:45.21 | Qwell | it's all magic |
| 17:45.32 | slav3_kitten | ? |
| 17:45.47 | Qwell | The servers accept your NickServ password. It will register you as you connect. |
| 17:45.50 | Qwell | err, auth |
| 17:46.06 | Qwell | Check your client settings. If it can't use server passwords, get a new client. :) |
| 17:46.21 | slav3_kitten | interesting.. most the servers i've connected to in the past didn't do that |
| 17:46.30 | Qwell | Yes, freenode is very nice in that regard. |
| 17:46.40 | ThinkGNU- | I'll try that [TK] |
| 17:47.04 | slav3_kitten | it is |
| 17:48.14 | *** join/#asterisk UnregisteredSnow (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 17:48.32 | Qwell | yeah, it helps if your client also knows your nick |
| 17:48.56 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 17:49.09 | slav3_kitten | what about clients knowing your nick? |
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| 17:59.23 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-lxciieqtvqcqqwrj) |
| 17:59.45 | pzn | in dial(x,timeout,y) command, the timeout is in seconds or miliseconds? |
| 18:00.48 | [TK]D-Fender | seconds |
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| 18:22.34 | paule32 | hello |
| 18:22.35 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 18:23.18 | paule32 | i have played with asterisk (trixbox) pc to pc calls are gone |
| 18:23.30 | paule32 | but how can i call from intern to extern? |
| 18:23.37 | paule32 | and vice versa |
| 18:23.48 | *** join/#asterisk classix (~salven@64.16.220.132) |
| 18:23.56 | paule32 | but intern to extern is ok at the moment |
| 18:23.58 | paule32 | ? |
| 18:24.30 | paule32 | extern to intern i mean |
| 18:24.58 | WIMPy | Asterisk doesn't have a concept of internal and external. |
| 18:25.32 | paule32 | when i type in in cli |
| 18:25.46 | paule32 | sip show registry |
| 18:25.50 | paule32 | i get |
| 18:26.05 | pzn | [TK]D-Fender, tks... "core show application dial" does not say that timeout is in seconds. I'm personally not familiar with asterisk helps... do you think that should be told, or all asterisks things are in seconds and this should not be mentioned? |
| 18:26.23 | paule32 | sip.1und1.de:5060 telnio 120 No Auth... |
| 18:26.56 | WIMPy | paule32: So you failed to register. |
| 18:27.35 | paule32 | if someone from germany here? and can give me a tipp to register |
| 18:27.36 | paule32 | ? |
| 18:27.58 | WIMPy | #asterisk-de |
| 18:30.35 | [TK]D-Fender | pzn, I'm not aware of anything in the dialplan that is in MS |
| 18:30.58 | WIMPy | Dial option L is in ms. |
| 18:31.18 | pzn | [TK]D-Fender, ok, then that should be OK for most users... not saying anything in helps -> means seconds... |
| 18:31.45 | [TK]D-Fender | pzn, Yeah, I'm all for clearer instructions. Go post it on the tracker :) |
| 18:32.09 | pzn | [TK]D-Fender, which URL is the bug submit of asterisk? |
| 18:32.22 | [TK]D-Fender | bugs.asterisk.org IIRC |
| 18:34.06 | pabelanger | issues.asterisk.org/jira |
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| 18:45.01 | *** join/#asterisk Alex_Bkash (b4eac34d@gateway/web/freenode/ip.180.234.195.77) |
| 18:45.36 | Alex_Bkash | need billing for asterisk with reseller levels |
| 18:45.39 | pzn | done! https://issues.asterisk.org/jira/browse/ASTERISK-20078 |
| 18:47.00 | Alex_Bkash | which opensource billing is most flexible for this? |
| 18:48.06 | Get_The_Fish | pzn thank you I think that you are right on that. |
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| 18:58.16 | newtonr | pzn: thanks for that submit, i'll ack it and probably fix that myself |
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| 19:29.40 | Alex_Bkash | hi oej |
| 19:31.08 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 19:32.19 | CrossWired | within my DialPlan i am calling DIAL, how can i get the SIP CALL-ID of the call being dialed |
| 19:32.42 | CrossWired | i can do it once that call is answered but i haven't found a way to do it, or am overlooking it, before the call is answered |
| 19:33.05 | WIMPy | Where do you want it? |
| 19:33.27 | WIMPy | And the SIP id or the channel name? |
| 19:34.08 | CrossWired | into a variable, or any where i can get it from AMI, i can get the channel and the uniqueid of that channel,not the CALL-ID |
| 19:35.09 | CrossWired | i might be able to grab it on the sip channel event |
| 19:39.24 | CrossWired | and thats a negative |
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| 19:46.50 | CrossWired | what i'm looknig for shows up in the BRIDGEPVTCALLID variable |
| 19:55.10 | CrossWired | looks like its not available before the calls are bridge |
| 19:55.41 | CrossWired | there seems to be a patch going through the works for something of the sorts, but its not ready yet so i'll just have to deal with it |
| 19:57.05 | ThinkGNU- | [TK]D-Fender, sorry for the delay. Some things came up. Here's my sip debug for the call(I think) http://pastebin.com/vZJistZn |
| 20:01.17 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:02.04 | ThinkGNU- | [TK]D-Fender, I just send this to the channel, didn't realize you were gone. Resending: http://pastebin.com/vZJistZn |
| 20:02.28 | ThinkGNU- | Sorry for the delay on getting that and I hope that's the info you were asking about |
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| 20:27.07 | *** join/#asterisk gusto (~gusto@82.113.121.106) |
| 20:27.15 | gusto | hey |
| 20:28.46 | gusto | how is it going? what's new? |
| 20:29.17 | gusto | somehow, it's silent here ... hmm, may change soon :-D |
| 20:29.55 | WIMPy | Do we need popcorn? |
| 20:31.16 | gusto | that may be true |
| 20:31.30 | gusto | ESM is through the slovak and now also german parliament |
| 20:32.37 | gusto | the good message is that asterisk will still prevail also in a world where ESM is in place |
| 20:34.12 | WIMPy | So we get some nice court cases again. |
| 20:34.22 | gusto | yes |
| 20:35.06 | gusto | but not in slovakia yet, only in germany it's announced, but maybe sulik takes it to constitutional court also in slovakia, will see |
| 20:57.48 | *** join/#asterisk ShaunC (44b1806a@gateway/web/freenode/ip.68.177.128.106) |
| 20:58.11 | ShaunC | Hi! |
| 20:58.24 | ShaunC | is anyone at Google IO this year from the Asterisk team? |
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| 20:59.49 | gusto | what is google IO? |
| 21:05.03 | carrar | Google ID == Google account |
| 21:05.13 | carrar | Google ID == Google account username |
| 21:05.21 | gusto | ??? |
| 21:05.34 | carrar | go create a google account |
| 21:05.55 | carrar | www.google.com/accounts/ |
| 21:06.24 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 21:06.36 | carrar | come over with a rack of beer and I will set you up with a google account |
| 21:06.47 | carrar | the GOOD STUFF mind you |
| 21:08.38 | autofsckk | hello, is yealink t20p a good phone? |
| 21:08.56 | carrar | never used |
| 21:09.09 | carrar | go with polycom |
| 21:10.53 | Qwell | carrar: ...IO, not ID. |
| 21:11.14 | pzn | newtonr, tks |
| 21:12.43 | carrar | oh |
| 21:12.47 | carrar | the developer conference |
| 21:12.49 | carrar | it's over |
| 21:13.11 | carrar | https://developers.google.com/events/io/ |
| 21:13.28 | carrar | I think they are doing replays now |
| 21:14.02 | autofsckk | ok carrar thanks |
| 21:14.03 | carrar | That said, you can still bring over a rack of beer |
| 21:14.19 | autofsckk | and yealink vs aastra? |
| 21:20.48 | ThinkGNU- | So when I'm making a call from an internal phone through my * box to a SIP provider, the CID that gets passed along is from the internal phone and I can't find a way to force CID from * through to the SIP providerl. Any thoughts? |
| 21:22.07 | WIMPy | Either Set(CALLERID(... or fromuser/defaultuser in your ITSP account. |
| 21:22.58 | ThinkGNU- | I have fromuser set to the phone number the SIP provider gives. I probably can put a name in there too then right? |
| 21:23.28 | WIMPy | I don't think so. |
| 21:24.13 | ThinkGNU- | Okay, simple enough. I'll just set it in the dial plan then. |
| 21:26.40 | ThinkGNU- | Thank you for the input WIMPy. [TK]D-Fender was trying to help earlier but I got pulled away for something. |
| 21:26.51 | ThinkGNU- | As always, thanks to everyone in here for being so willing to help |
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| 22:55.47 | tomwish | hopefully someone has used the iax encryption. I have bot sides set up with auth=md5 and encryption=aes128. I have the foreencryption=yes. When the call cpomes in I get the following error. chan_iax2.c:10598 socket_process: rejected connect attempt. No secret present while force encrypt enabled. there is a secret set. what am I missing? |
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