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| 04:50.48 | SpaceInvaders | Help! Suggestions on SIP soft phone clients that work on Droid? I have Samsungs (s2 and 5). Linphone just crashes over and over. |
| 04:56.26 | nix8n82 | Csipsimple worked ok and grandstream client seems to be working ok too |
| 04:56.51 | SpaceInvaders | I check them out. Thanks!!! |
| 04:58.04 | ChannelZ | I use CSipSimple |
| 05:17.04 | SpaceInvaders | CSIPSimple - initial test; no crashes!!! thanks! |
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| 05:26.15 | wyoung | SpaceInvaders: default android client is nice |
| 05:26.27 | wyoung | SpaceInvaders: that is what I use |
| 05:26.48 | SpaceInvaders | hmmm am i missing something? |
| 05:27.55 | SpaceInvaders | I don't see any way to point the default client in my SG2 to my Asterisk server |
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| 12:53.12 | dpecka | hello |
| 12:53.35 | dpecka | does ael (in 11..13) support something like system() or `backticks` ? |
| 13:00.42 | WIMPy | There are things like System and SHELL. |
| 13:03.01 | mub | How many v's is ideal? Whenever I need an asterisk shell I literally just hold the 'v' key down |
| 13:03.06 | mub | asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv |
| 13:04.11 | mub | 999 v's ?! |
| 13:05.59 | WIMPy | I don't thing that more than 4 make any difference. |
| 13:08.09 | mub | my life is a lie |
| 13:10.11 | angryuser | hi, i am launching agi script with AGI(test.php) asterisk complains '/var/lib/asterisk/agi-bin/test.php': No such file or directory" 1 asterisk is launched as root so its not a permissions problem |
| 13:10.16 | angryuser | What else ? |
| 13:10.28 | angryuser | the script is there |
| 13:11.14 | WIMPy | Is it executable? Does it contain the path to a valid interpreter? |
| 13:11.28 | angryuser | WIMPy: yes i did chmod +x |
| 13:11.41 | angryuser | WIMPy: i can execute the script locally no problem |
| 13:12.13 | Samot | How are you executing it locally? |
| 13:12.17 | angryuser | php interpreter is in place and working, i am using php-cgi standalont |
| 13:12.56 | angryuser | ah i see Samot got it |
| 13:13.04 | Samot | ? |
| 13:13.16 | angryuser | ./script is failing but php script is not |
| 13:13.41 | angryuser | path to php inside script is wrong |
| 13:13.42 | WIMPy | did ask if it contained a valid interpreter. |
| 13:14.33 | Samot | You did. |
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| 13:15.36 | albatrosss | hi all |
| 13:16.46 | albatrosss | i have a problem with 2 sip trunks registered to one provide, the incoming calls all comes to the context of the first sip trunk, i have different contexts setup for them, the outbound works fine |
| 13:16.53 | mub | and here I thought that all agi scripts had to be perl |
| 13:17.02 | mub | I should read the manual |
| 13:18.48 | mub | albatrosss: and you're matching a DID from the second trunk under the proper context? |
| 13:19.00 | mub | It must be matching in the first trunk's context |
| 13:21.50 | angryuser | it was not written by me, anyway thanks |
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| 13:26.51 | albatrosss | mub: i have the trunk in 2 different context |
| 13:37.31 | mub | albatrosss: let's see that dialplan! I can't explain things well, so I just want to see it |
| 13:37.43 | mub | Blank out any sensitive info |
| 13:37.59 | mub | :c |
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| 13:40.47 | dan_j | Hi. Using asterisk 11, I've got a Local channel as a member of a queue. It can accept one call, but after that the status changes to Invalid. Any ideas what might be causing that? |
| 13:41.38 | sikun | question... I've been tasked with separating one of the three companies off of our current virtualized PBX to a physical server that has a line card so they can utilize hard wired phones, so far I have successfully been able to configure the new PBX running AsterisksNOW to be able to make and receive calls. So, I'm good to go there. |
| 13:42.13 | sikun | So now.. what I want to do is be able to link the two together to still provide access to transferring/calling the extensions of the other companies like they can now |
| 13:42.32 | sikun | I'm wondering, how do I go about setting up an IAX2 trunk to do this. |
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| 17:16.00 | raspberrypifan | is there a spanish version of the asterisk book |
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| 17:23.30 | mike3 | Afternoon, what would be some of the best ios softphones to use? |
| 17:26.43 | [TK]D-Fender | raspberrypifan, no |
| 17:33.56 | raspberrypifan | that sucks |
| 17:34.02 | raspberrypifan | im going to try and teach my grandmother asterisk |
| 17:34.04 | raspberrypifan | just because |
| 17:38.40 | TandyUK | so, im less than impressed with the current UK sound packs for asterisk. |
| 17:39.03 | TandyUK | Anyone think there would be a market for selling decent quality, COMPLETE, ausio packs? |
| 17:39.05 | TandyUK | audio* |
| 17:40.18 | TandyUK | when i say complete, i mean all snippets, |
| 17:40.32 | TandyUK | eg atm the uk voice pack we have doesnt have every file converted |
| 17:40.39 | Samot | Yes but you're going to pay for it. |
| 17:41.03 | TandyUK | if we go into phrase management (*55), most messages are in a uk voice, but occasionally we get one thats still american |
| 17:41.13 | TandyUK | Samot: what do you mean I'll have to pay for it? |
| 17:41.20 | TandyUK | I'd be the one selling the sound pack |
| 17:41.55 | TandyUK | We're having this done for our own system, im just toying with the idea of bundling it up for selling it to others |
| 17:41.58 | Samot | Well then, there's already a market and those meeting the demand of the market. |
| 17:42.00 | Samot | So yeah. |
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| 17:42.35 | TandyUK | well id disagree there, unless theres other packs im not aware of |
| 17:42.45 | TandyUK | we bought 3 different packs, none of whih are actually complete |
| 17:42.48 | Samot | That's what I meant by you'd have to pay for it. I misread your comment and through you were looking for a sound pack. |
| 17:43.28 | TandyUK | and the people selling them clealry just employed a voice artist on a temporary basis, so cant even get the missing bits done in the same voice |
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| 18:43.59 | rp_ | Hello to all in this channel. Does anyone know if it's possible to create a catch all context in a real time multi tenant setup? I would like to create a context for each customer without the need to modify my extensions.conf file and reload the dial plan. |
| 18:49.41 | [TK]D-Fender | No |
| 18:50.57 | cresl1n | rp_ if you set one context to be the default context for everyone, wouldn't that accomplish what you desire? |
| 18:55.39 | rp_ | cresl1n, I woul like to match every tenant to each own specific context. I was hoping that I could create a [realtime-%] context where % would be the catch all identify. If I sent a sip peer to a contexc there is'nt created the peer will land in [default]? |
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| 18:57.33 | [TK]D-Fender | no |
| 18:57.39 | [TK]D-Fender | Thre is no fallback |
| 18:57.50 | rp_ | okay |
| 18:58.06 | rp_ | So there is no smart way to achieve what I would like to do? |
| 19:05.10 | rp_ | Have anyone looked into the source code for the context handler? Would it make sense to develop a wild card function or would it break anything else? |
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| 19:12.54 | K0HAX | I'm trying to get pjsip_wizard.conf working with my provider, and the require "insecure=port,invite" in sip.conf, what is the equivalent for that in pjsip? |
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| 19:36.40 | newtonr | K0HAX, I believe you would use an "identify" section to match by IP address, additionally you would not configure an auth for the endpoint if you don't want authentication. |
| 19:38.04 | newtonr | also depending on what you are doing, the line option may help: http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/ |
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| 20:08.05 | K0HAX | That worked. :) Now I've just got to get SRTP to work on my Polycom phones again and I'm all set |
| 20:08.58 | K0HAX | I'm getting "SRTP unprotect failed with: authentication failure 10" and "SRTP unprotect failed with: authentication failure 110" |
| 20:13.09 | newtonr | K0HAX, great. I'm not sure what is going on with the SRTP deal off-hand |
| 20:13.52 | K0HAX | Yeah.. I'm going to double-check the phone works without SRTP and TLS quick.. :P It could be NAT or something |
| 20:28.09 | K0HAX | weird... 'rtp set debug on' doesn't show anything when I place a call from that phone |
| 20:50.53 | K0HAX | I got clear audio to work, but non-tls SRTP doesn't work. I think it has something to do with SHA1 if I had to guess. :( "a=crypto:72 AES_CM_128_HMAC_SHA1_80 inline:i7COCCwt6xy0Fba0STahkR+AxftiG4sIi4DX3ZNI" |
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| 21:26.35 | Alblasco1702 | .quit |
| 21:26.41 | Alblasco1702 | .quit |
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| 23:19.48 | _TWS | anyone know why asterisk will not play VM envelope information when using *97, but it will play the envelope when using *98 from another extension? |
| 23:21.26 | SpaceInvaders | what are reasons I'd get the error - |
| 23:21.28 | SpaceInvaders | Received SIP subscribe for peer without mailbox: |
| 23:21.36 | SpaceInvaders | when voicemail.conf has the mailbox |
| 23:21.45 | SpaceInvaders | *and* voicemail show users shows the mailbox? |
| 23:24.09 | *** join/#asterisk KaliLinuxGR (~alexandro@188.117.250.184) |
| 23:24.54 | SpaceInvaders | ok adding username= in sip.conf seems to have resolved the problem |
| 23:34.02 | SpaceInvaders | or not :/ |
| 23:34.53 | SpaceInvaders | so same question--how can i get the "peer without mailbox" error when there appears to be a mailbox? What am I missing? |
| 23:35.14 | [TK]D-Fender | Specifying that that peer has a box |
| 23:35.31 | SpaceInvaders | in the dialplan? |
| 23:35.39 | [TK]D-Fender | Obviously not |
| 23:35.46 | [TK]D-Fender | that isn't where you tell a peer anything |
| 23:36.18 | SpaceInvaders | ok so I already said I defined the mailbox in voicemail.conf.... what else is there?? |
| 23:36.43 | [TK]D-Fender | <[TK]D-Fender> that isn't where you tell a PEER anything |
| 23:37.01 | SpaceInvaders | i knew there was a clue in that.... so im reading...... |
| 23:37.32 | wyoung | Mornin' |
| 23:37.38 | SpaceInvaders | Mornin' |
| 23:38.51 | wyoung | hey [TK]D-Fender! I figured out the issue I was having the other week regarding weird volume changes, cutting out of sound etc.. The solution was to set all IP handsets and asterisk to prefer alaw (as I am using alaw for my ISTP) |
| 23:39.25 | SpaceInvaders | ok so mailbox= in the sip.conf? and "default" is assumed (according to the Definitive Guide) so if the context is default I can leave it off? |
| 23:40.01 | [TK]D-Fender | Don't half-ass your configs |
| 23:40.25 | SpaceInvaders | ok so as-a-rule don't leave out defaults. got it |
| 23:41.11 | SpaceInvaders | Thank you [TK]D-Fender :-) |
| 23:43.01 | SpaceInvaders | Next question--where do I start researching how to build a menu? Like you hear on business lines (e.g. Press 1 for sales). I'm hoping theres a best-practices document so I don't re-invent the wheel with scripting |
| 23:43.23 | SpaceInvaders | and is there a name for that kinda menu? |
| 23:53.17 | [TK]D-Fender | IVR |
| 23:53.23 | [TK]D-Fender | And it's not a "thing" |
| 23:53.29 | [TK]D-Fender | it's an idea. |
| 23:53.41 | [TK]D-Fender | "core show application waitexten" |
| 23:53.48 | [TK]D-Fender | "core show application background" |
| 23:53.58 | [TK]D-Fender | "core show function TIMEOUT" |
| 23:54.00 | [TK]D-Fender | read up |
| 23:54.11 | SpaceInvaders | will do thank you!!!!!!! |
| 23:54.13 | *** join/#asterisk gtjoseph (~gtjoseph@unaffiliated/gtj) |
| 23:54.19 | [TK]D-Fender | and you should know your Asterisk Standard Extensions back & forth by now |
| 23:55.46 | SpaceInvaders | hmmm I'm just getting familiar with Asterisk. Do you mean this - http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 23:56.15 | SpaceInvaders | if that's it I'm set |