| 00:02.46 | zopsi | so if I compiled with format_mp3, what exactly does that give me in terms of dialplan or voicemail options? I want to have my voicemails sent in mp3 instead of wav and was wondering if this was built in or if I needed to convert with a shell script. |
| 00:05.41 | file | format_mp3 allows MP3s to be read, it does not write them |
| 00:05.55 | zopsi | file: ah that makes sense. |
| 00:06.19 | zopsi | iOS users don't really like the wav format they get hehe |
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| 00:14.03 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:46.34 | Sjors | Hi all |
| 01:47.00 | Sjors | I was wondering about redundancy with SIP and I've got a question for you guys; if this is not the right channel please let me know |
| 01:47.49 | Sjors | I am a customer of a VoIP service where my PBX sends a REGISTER to the service to let it know where the PBX can be found, so calls can be routed to it |
| 01:48.07 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:48.18 | Sjors | I believe this is quite common, where the only other alternative is that the service itself has the PBX address (be it IP, hostname or SRV record location) hardcoded somewhere |
| 01:48.48 | Sjors | now, suppose I want my telephony to become highly available, and I set up a second PBX and configure it exactly the same |
| 01:49.27 | Sjors | now, both PBXes will send REGISTERs to the VoIP service and I believe its correct behaviour would now be to send any incoming calls to the PBX that, by chance, sent its REGISTER last |
| 01:50.39 | Sjors | so for SRV records one can give priorities and weights to individual servers, saying one should always be tried before another, but is there a way to do something like this with REGISTERs too? |
| 01:51.07 | Sjors | or would it be necessary to have a simple proxy do the REGISTER and have the proxy try the two PBXes in order, allowing one to be down? |
| 01:52.53 | Sjors | because if the PBX that sent its REGISTER last goes down, it may take up to the reregister time for a valid registration to reappear, which would defeat the purpose of having two PBXes in the first place |
| 01:53.20 | Sjors | </essay> |
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| 02:17.31 | voxter | When you attempt to load an asterisk module and it complains that it was compiled using different flags - is there a way to determine which flags those were? |
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| 03:02.27 | snadge | so i had a client with registration issues.. probably nat related.. see the register packet, see unauthorized go back.. but they presumably dont receive that |
| 03:03.13 | snadge | put them on a different server with newer asterisk (11.4) .. and they can register.. but show peer says "unreachable".. despite that, they can still place a test call and it works |
| 03:03.32 | TazzNZ | snadge: what is the networks like between the server and client ? |
| 03:03.50 | snadge | well.. this particular client on a university network, apparently behind some kind of F5 load balancer / firewall |
| 03:04.15 | snadge | i guess why im asking is im not exactly sure what "unreachable" means |
| 03:04.36 | TazzNZ | it means that Asterisk can't talk to the client |
| 03:04.40 | snadge | i would've thought if there was a networking issue.. then a test call would not work |
| 03:04.54 | snadge | ahh okay.. so that could explain why they can initiate a call |
| 03:05.06 | TazzNZ | does sound work on the test call ? |
| 03:05.13 | snadge | yeah.. thats the part that confused me |
| 03:05.18 | snadge | they said it worked fine.. and i was like.. huh |
| 03:05.34 | snadge | how can it be both unreachable.. and work :P |
| 03:05.49 | TazzNZ | yeah - so the problem can be that the F5 closes the UDP side quickly |
| 03:06.06 | TazzNZ | does it ever show reachable or a time |
| 03:06.15 | TazzNZ | and what is the nat settings for that peer in sip.conf |
| 03:07.37 | snadge | its not showing a current registration .. im not speaking with them at the moment |
| 03:07.45 | snadge | the status of the peer now is unknown |
| 03:08.15 | snadge | i think im reasonably happy that its an issue at their end anyway.. and configuring / troubleshooting an F5 load balacner / router / firewall / whatever.. is kind of out of scope |
| 03:09.19 | TazzNZ | what is the nat settings of the peer ? |
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| 06:22.26 | zopsi | Does anyone remember the password for the Spectralink Handset Administration Tool? I think it was like plc or plcm or something. |
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| 07:18.16 | Tester | hi there, please i just downloaded asterisk and installed it on my linux system. I also downloaded the zoiper softphone and installed it too. I will like to know how to cinfigure the asterisk and the zoiper phone to work together on an adhoc network that i created. |
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| 07:28.11 | ChannelZ | ~primer |
| 07:28.12 | infobot | well, primer is http://burner.com/asterisk-primer |
| 07:28.22 | ChannelZ | ~book |
| 07:28.22 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 08:14.58 | kafal1 | hello! does any one know what is the substitute for Agents action in 12.3? |
| 08:15.10 | kafal1 | This action is not longer availabe. |
| 08:15.22 | kafal1 | I wanted to get a list of agents from AMI |
| 08:19.12 | *** join/#asterisk TechAdam (c27dc475@gateway/web/freenode/ip.194.125.196.117) |
| 08:19.51 | TechAdam | hey guys im getting a wierd error on one of my asterisk servers |
| 08:20.02 | TechAdam | When you ring an outside number you either get a dial tone for a few seconds or you get through to a strange number you have not called |
| 08:20.13 | WIMPy | <mjordan> kafal1: that action still exists. https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+ManagerAction_Agents |
| 08:21.07 | skrusty | morning |
| 08:21.15 | TechAdam | mornin :) |
| 08:21.49 | WIMPy | TechAdam: First of all you use a wierd space character. |
| 08:22.20 | TechAdam | and have that eat the server? :) |
| 08:22.56 | WIMPy | And how do you call "outside"? some POTS interface? |
| 08:23.25 | TechAdam | not these calls arnt going over POTS |
| 08:23.41 | TechAdam | they are going over the sip servers |
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| 08:26.23 | kafal1 | I checked it and tried using it. but i get an error: Message: Invalid/unknown command: Agents. Use Action: ListCommands to show available commands. |
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| 08:29.17 | kafal1 | Do we need to load any specific module? |
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| 08:31.15 | WIMPy | Sure. Almost everything is in modules. |
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| 08:48.46 | TechAdam | issue was caller id :L |
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| 10:46.26 | Geek-Linux | Hi All. i am facing problem in early media over ss7 channles can any one help |
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| 11:34.27 | udp` | hi, is there a way to create a queue inside the .call file? or in the spool ? |
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| 11:46.04 | whizzi | Hi all |
| 11:48.01 | whizzi | quick question, is it possible to let a connected peer/friend us the asterisk features? I’m talking about an external incoming (or outgoing) party, that this particular external device can use features like blindxfer and such? |
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| 11:51.33 | WIMPy | Asterisk doesn't have a concept of internal/external. |
| 11:51.45 | goddva | udp`, why not use the database integration (realtime) ? You will not be able to create a new queue, but you might create some dynamic logic, so that you can fake it... |
| 11:53.01 | whizzi | WIMPy: ok, tnx |
| 11:59.25 | udp` | goddva: im not that expirienced with asterisk, im doing some penetration testing |
| 11:59.39 | udp` | and im trying to "flood" a system |
| 12:00.00 | udp` | i want the .call files to be executed one by one |
| 12:00.05 | udp` | inside the spool |
| 12:01.10 | goddva | yes, that should be simple... you can create all the files at the same time, just set the time correct, so they will be executed at the right time.. |
| 12:01.39 | goddva | if you want the to wait for each other, then this will not work |
| 12:01.49 | goddva | *want them |
| 12:02.07 | udp` | well the problem with the time |
| 12:02.17 | udp` | is that if the target is answering |
| 12:02.21 | udp` | and if its not |
| 12:02.32 | udp` | makes everything different |
| 12:02.57 | goddva | ah, if you need more logic, you might want to use something else than spool files.. |
| 12:04.32 | goddva | ... or you create a extension that will not stop trying if they are not answering the call... |
| 12:05.18 | udp` | .call files are making me avoid to create extensions |
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| 12:05.51 | udp` | i also understand that there is an option with CDR() ? |
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| 13:26.24 | curfont | Hi guys |
| 13:26.46 | curfont | I put tcpenable and transport=tcp but asterisk still isnt listening to TCP, I am using 10.8 |
| 13:26.52 | curfont | and bindaddr |
| 13:26.59 | curfont | Is it supported? |
| 13:27.19 | [TK]D-Fender | Yes |
| 13:27.27 | [TK]D-Fender | show us your configs |
| 13:33.01 | curfont | http://pastebin.ca/2816043 |
| 13:34.19 | curfont | I see this |
| 13:34.20 | curfont | tcp 0 0 0.0.0.0:2000 0.0.0.0:* LISTEN 20970/asterisk |
| 13:34.29 | curfont | is this what its listening at? |
| 13:34.38 | curfont | or is it something else? |
| 13:34.52 | [TK]D-Fender | that is ONE thing it is listening for. Asterisk listens for all sorts of different things depending on what you set up |
| 13:35.23 | curfont | I dont really have anything else set up, isnt the default for SIP TCP 5060? |
| 13:36.09 | [TK]D-Fender | I asked for configs.... |
| 13:36.55 | curfont | everything else is the default.. the only thing i changed i gave you |
| 13:37.09 | curfont | http://pastebin.ca/2816043 sip.conf |
| 13:37.15 | curfont | default for debian |
| 13:37.28 | curfont | the other configs are massive do you want all the defaults? |
| 13:37.40 | curfont | will other configs affect sip, tcp? |
| 13:37.58 | [TK]D-Fender | you didn't set your bindports |
| 13:38.01 | [TK]D-Fender | read the sample config |
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| 13:38.25 | curfont | oh, doesnt it automatically do that |
| 13:40.04 | [TK]D-Fender | You shouldn't skip settings that are clearly related to what you are trying to do. |
| 13:41.17 | curfont | you put bindport under the specific user right? |
| 13:42.10 | [TK]D-Fender | Is that where you see it in the sample configs? |
| 13:42.54 | jeremy_laine | hi, does anyone know whether it's possible to tell asterisk (11) to exclude some of its local ICE candidates? |
| 13:42.59 | WIMPy | [TK]D-Fender: Don't ask things you don't want to know. |
| 13:43.09 | file | jeremy_laine, not currently possible |
| 13:43.13 | jeremy_laine | file: ouch |
| 13:43.25 | curfont | ok so the problem seems to be I put tcpenabled instead of tcpenable |
| 13:43.31 | [TK]D-Fender | WIMPy: I do (both) :) |
| 13:43.34 | jeremy_laine | file: the downside is that you tell the world about your internal network layout.. |
| 13:44.16 | WIMPy | jeremy_laine: Only if it's simple enough to fit :-) |
| 13:44.47 | jeremy_laine | WIMPy: not sure I understand, but basically you report the IP addresses of internal interfaces |
| 13:45.26 | WIMPy | jeremy_laine: For me it made the messages explode. |
| 13:45.35 | jeremy_laine | WIMPy: oh I see! |
| 13:47.44 | curfont | SIP related question, when TCP mode is enabled, is it only for control and not media? |
| 13:48.09 | WIMPy | SIP is always cvontrol only, never media. |
| 13:48.10 | file | jeremy_laine, I remember one other person bringing up the ability but I don't know of anyone working on it |
| 13:48.40 | curfont | WIMPy: media is always udp? |
| 13:48.46 | jeremy_laine | file: am taking a look at the pjsip API as it would need to support it |
| 13:49.13 | file | jeremy_laine, not really... it's told the local candidates |
| 13:49.18 | curfont | basically, in a scenario when UDP is unreliable, is it possible somehow to pass media through TCP? Would a VPN be the only way? |
| 13:49.30 | curfont | For example, hotel room NAT |
| 13:49.54 | WIMPy | curfont: Usually, but it's RTP, not SIP. |
| 13:50.08 | curfont | WIMPy: Yeah, which is UDP isn't it? |
| 13:50.16 | curfont | In IAX though you get media from the same port |
| 13:50.20 | curfont | But it's still UDP |
| 13:50.23 | jeremy_laine | file: ah ok, I thought the host candidates were figured out by PJSIP as it needs to listen on them |
| 13:50.26 | WIMPy | TCP is usually a very bad idea for anything realtime. |
| 13:50.32 | curfont | I am in a hotel room trying to make a call basically WIMPy |
| 13:50.39 | curfont | In every case, I get no incoming packets |
| 13:50.50 | curfont | Tried IAX, SIP.. SIP with TCP |
| 13:50.59 | curfont | Even with STUN and RPORT |
| 13:51.04 | file | jeremy_laine, it's used to find the local addresses but it doesn't listen, the RTP stack does |
| 13:51.15 | curfont | And in every case the other person can hear me, but I cant hear them |
| 13:51.20 | file | and the act of finding the local addresses is not tied to the ICE stuff at all - it's just a pjlib function |
| 13:51.39 | jeremy_laine | grepping for AST_RTP_ICE_CANDIDATE_TYPE_HOST .. |
| 13:52.07 | file | jeremy_laine, although this isn't going to stop Chrome from giving you it all from other networks |
| 13:52.36 | jeremy_laine | file: I don't mind customers showing me their setup, I mind giving them mine :) |
| 13:54.03 | [TK]D-Fender | [09:50]curfontIn every case, I get no incoming packets <- from whre? |
| 13:54.21 | [TK]D-Fender | curfont: Three are several settings required to accomodate NAT. |
| 13:54.25 | [TK]D-Fender | there* |
| 13:54.46 | [TK]D-Fender | curfont: If your server itself is behind NAT there are several extra ones as well which I didn't see set up on yours |
| 13:55.09 | jeremy_laine | file: looking at rtp_add_candidates_to_ice, looks as though this is where we can influence what we list |
| 13:55.16 | file | jeremy_laine, sure |
| 13:56.25 | file | I'd actually do it in ast_rtp_ice_add_cand, cleaner and applies to all |
| 13:58.04 | jeremy_laine | file: what would a smart setting be, a "black list" of IP addresses to exclude, or a "white list" of IP addresses to include? |
| 13:58.28 | file | I wouldn't do IP addresses - use the ACL logic so you can do specific IP addresses or ranges |
| 13:59.05 | jeremy_laine | file: ok, sounds like I need to browse some more code :) |
| 13:59.15 | file | and probably, oh, blacklist but that's just me |
| 13:59.53 | jeremy_laine | I think I agree, if you are using ICE after all you expect some magic to happen |
| 14:00.06 | jeremy_laine | you just want the ability to exclude some edge cases |
| 14:00.11 | file | I'm trying to think of an example for ACL |
| 14:00.24 | file | the permit/deny logic in chan_sip comes to mind but it's chan_sip |
| 14:01.52 | jeremy_laine | so the setting would be something local_ice_candidate_acl = some_named_acl ? |
| 14:02.54 | file | I'd make it configurable in rtp.conf so it builds a local ACL in res_rtp_asterisk, and then when adding candidates it is checked to see if they should be ignored |
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| 14:08.12 | file | jeremy_laine, seems as though my branch is... good! |
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| 14:10.21 | jeremy_laine | file: judging by sip.conf, settings XXXacl (acl, directmediaacl, contactacl) all seem to also have an XXXpermit and XXXdeny |
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| 14:10.55 | file | you can do a deny all and permit some approach, but it's not explicitly required to treat things like that |
| 14:11.09 | file | in code you can explicitly permit all and then deny specific stuff |
| 14:11.23 | file | (I used that approach for NAT in chan_pjsip) |
| 14:11.37 | file | chan_sip does the same thing for localnet |
| 14:12.04 | file | all depends on how you want it to behave/be configured |
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| 14:21.29 | jeremy_laine | file: I am looking at how acl, contactacl, directmediaacl are parsed and looks as though it's cheap to use acl / permit / deny |
| 14:22.37 | jeremy_laine | not sure why directmedia doesn't have the same "if (!ast_strlen_zero(v->value))" check by the way |
| 14:22.55 | file | Mars. |
| 14:22.58 | file | I blame Mars. |
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| 14:26.13 | curfont | [TK]D-Fender: Ok so trying IAX which is simpler, very simple test case... I have a number which has music on hold |
| 14:26.14 | curfont | http://pastebin.ca/2816063 |
| 14:26.18 | curfont | I have IAX debug on |
| 14:26.40 | curfont | Obviously something gets received by my client otherwise it wouldn't connect to the call.. but I hear no media coming out |
| 14:26.52 | curfont | I know that IAX uses only one port for both control and media |
| 14:27.07 | curfont | and according to the debug, it went "OK" unless I am missing something? |
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| 14:28.17 | [TK]D-Fender | have you tested to make sure you had MoH playing on another device? |
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| 14:30.22 | curfont | MoH works and Echo works (I tried it when I was local to the computer), actually.. let me ask a question.. with IAX do you need to register on both ends? |
| 14:30.43 | curfont | I use Zoiper which registers to my server, but my server needs to register to Zoiper? |
| 14:31.07 | curfont | Is it possible to configure IAX so the server just responds to the guy who registered? |
| 14:31.34 | [TK]D-Fender | [10:30]curfontI use Zoiper which registers to my server, but my server needs to register to Zoiper? <- no. This is no differnt for SIP |
| 14:31.57 | [TK]D-Fender | IAX is responding to the CALL |
| 14:32.09 | curfont | Zoiper uses IAX |
| 14:32.21 | curfont | I am doing IAX only, no SIP involved for now |
| 14:33.19 | curfont | > It should be noted that trunking requires both sides to know each other (i.e. they need to be valid peers). If one side has trunk=yes and the other does not cannot validate the peer, you will get one-way audio. The easiest way to make this work is to use a register line to register to the systems you want to trunk with. |
| 14:33.21 | curfont | This line here |
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| 14:35.18 | curfont | OK that was it |
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| 16:36.57 | pabelanger | Anybody ever use the zoiper web client? |
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| 17:06.43 | Katty | passed my nessus test! |
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| 17:23.26 | mbowie | :-D |
| 17:32.31 | ChannelZ | Congrats! I have no idea what that is! |
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| 17:56.48 | hexanol | is anyone aware of new freeze/deadlock happening in asterisk since version 11.10.2 (from 11.9.0) ? |
| 17:57.07 | hexanol | since we upgraded, we had 2 occurences of a freeze |
| 17:57.26 | hexanol | but we haven't a "core show lock" yet |
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| 21:00.55 | WIMPy | Why can't handsets be built soo thatr they are easy to take apart? |
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| 21:28.11 | SpeedEvil | All handsets are easy to take apart with Mr Hammer. |
| 21:28.56 | WIMPy | True. |
| 21:29.04 | WIMPy | But not reusable then. |
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| 22:54.28 | _0x5eb_ | hi, does asterisk support SIP provider handling more than a single domain? |
| 22:55.15 | _0x5eb_ | (according to http://lists.digium.com/pipermail/asterisk-users/2013-April/278434.html it seems that it is impossible) |
| 22:55.22 | Torenn | Hello I'm having some severe troubles with the motif channel driver after it gets the jingle session initiation request, it creates an associated rtp session and then immediatly destroys but it's a bit cryptic on the reasons --> pastebin: http://pastebin.com/VFUsRn3h , does anyone have a clue on what may be the issue here? (The complete lack of documentation doesn't help either) |
| 22:56.13 | WIMPy | _0x5eb_: What exactely do you need? |
| 22:56.26 | [TK]D-Fender | _0x5eb_: Perhaps it's possible with * 12 using chan_pjsip. |
| 22:56.58 | WIMPy | Is it just the doimain in the to: ? |
| 22:57.13 | _0x5eb_ | WIMPy: I would like to send an invite to someuser@another.domain through a SIP proxy handling @some.domain |
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| 22:57.24 | _0x5eb_ | INVITE + To |
| 22:57.52 | WIMPy | Yes, I'vew been tol that it's possible with chan_pjsip in Asterisk 12. |
| 22:58.10 | Torenn | (That's Asterisk 12 btw) |
| 22:58.56 | _0x5eb_ | WIMPy + [TK]D-Fender: OK I will have a look there (chan_pjsip), thanks! |
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| 23:22.12 | _0x5eb_ | actually, I'm stuck with Asterisk 1.8, so no chan_pjsip. Is it also impossible (with * 1.8) to use a fromdomain != INVITE domain ? |
| 23:22.47 | WIMPy | You can change fromdomain, but not to. |
| 23:23.46 | _0x5eb_ | sorry I meant "host" domain != INVITE domain? |
| 23:24.59 | _0x5eb_ | so if I understand correctly INVITE domain must always == host, but From is customizable (using fromdomain) |
| 23:26.50 | WIMPy | correct |
| 23:40.48 | _0x5eb_ | OK, thanks; I guess that for exotic routing, it is more reasonable to keep asterisk as registrar only and e.g. kamaillo as proxy only |
| 23:42.30 | WIMPy | Using the current Asterisk might help as well. |
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| 23:44.59 | _0x5eb_ | OK I will experiment with Asterisk 12; thanks again! |
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