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| 01:08.34 | UncleKiwi | hi, im just wondering if a call is forwarded out to a mobile phone over the pstn network ( ie comes in over sip then is forwared out to a mobile phone on pstn via sip ) if the person on the mobile can then initiate a xfer of the caller back to an internal extention |
| 01:12.37 | UncleKiwi | to to clarify, PBX i has sip connection to PSTN, caller-a calls DID asterisk answers the call and dials the mobile and connects the two partys |
| 01:13.35 | UncleKiwi | then the mobile caller ( caller-b ) wants to then transfer the call back to an internat extention caller-c |
| 01:14.21 | UncleKiwi | i know if she were using a softphone on her smartphone none of this would be an issue |
| 01:15.08 | [TK]D-Fender | "core show application dial" <- |
| 01:15.59 | UncleKiwi | cheers |
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| 01:17.26 | UncleKiwi | can i pipe that output to more or less |
| 01:17.45 | UncleKiwi | i cant seem to read it from tha start |
| 01:17.50 | [TK]D-Fender | scroll |
| 01:17.57 | [TK]D-Fender | or read the instructions on the WIKI |
| 01:18.02 | UncleKiwi | ok |
| 01:18.25 | [TK]D-Fender | I've never seen a terminal you can't scroll... |
| 01:18.45 | UncleKiwi | i can scroll but its buffers dont get me to the start |
| 01:20.15 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial |
| 01:20.19 | UncleKiwi | sorted 'lines of scrollback |
| 01:22.20 | UncleKiwi | i like the colors |
| 01:22.23 | UncleKiwi | ;) |
| 01:49.45 | UncleKiwi | [TK]D-fender: this sounds like what i need , g: Proceed with dialplan execution at the next priority in the current |
| 01:49.46 | UncleKiwi | <PROTECTED> |
| 01:50.25 | UncleKiwi | would that mean that the person on the mobile could just end the call an it would put the caller-a back to an IVR for ex |
| 01:50.32 | UncleKiwi | *for eg |
| 01:50.50 | UncleKiwi | and they could then choose their own extn |
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| 01:52.36 | [TK]D-Fender | No. |
| 01:52.53 | [TK]D-Fender | You should look at the very clear parameters saying who can TRANSFER the call. |
| 01:52.54 | UncleKiwi | ok so maybe i should use t: |
| 01:53.24 | UncleKiwi | ok |
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| 01:53.52 | [TK]D-Fender | :) |
| 01:54.05 | [TK]D-Fender | Read very carefully on those 2 parameters.... |
| 01:54.18 | UncleKiwi | cheers |
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| 02:26.25 | UncleKiwi | [TK]Defender: thanks it works perfect ! |
| 02:27.28 | UncleKiwi | mobile person can blind xfer for attended-xfer also |
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| 03:17.23 | UncleKiwi | what does this actually mean ? 'This setting does not perform |
| 03:17.23 | UncleKiwi | <PROTECTED> |
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| 12:52.17 | samersg | hello! |
| 12:52.25 | samersg | any good soul for online support? |
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| 13:08.45 | samersg | hello! |
| 13:08.46 | samersg | any good soul for online support? |
| 13:09.27 | samersg | i need some assistance, some of my phones aren't working, only one side hears the voice |
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| 13:36.47 | Belliash | hello |
| 13:37.34 | Belliash | Im brand new to asterisk and I have one (stupid) question... Will asterisk work with USB phone? I would like to install asterisk on my HTPC and connect USB phone to it. What do you think about this solution? |
| 13:37.34 | Cyford33 | hi |
| 13:38.46 | Cyford33 | unless the usb phone uses a supportedprotical i would o with no |
| 13:42.14 | Belliash | you mean i would need usb sip phone? |
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| 14:28.45 | Greenlight | Anyone every had "crosstalk" on SIP lines. It's the strangest thing I've ever come accros, but the call recordings confirm the issue the customer reported, mid call they're getting "ringing" sounds over their calls |
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| 16:09.17 | samersg | hello! some of my phone are off (only one side hears the voice), any suggestions? we can't find the problem |
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| 16:24.58 | [TK]D-Fender | Im brand new to asterisk and I have one (stupid) question... Will asterisk work with USB phone? I would like to install asterisk on my HTPC and connect USB phone to it. What do you think about this solution? <- No, * won't directly support this |
| 16:25.03 | [TK]D-Fender | Belliash, ^ |
| 16:25.45 | [TK]D-Fender | Belliash, Those are dumb audio devices typically used with GUI OS's and soft-phones. So via a softphone, yes, but not ciretly |
| 16:26.08 | [TK]D-Fender | samersg, one-way audio is usually a SIP NAT issue |
| 16:27.28 | Belliash | [TK]D-Fender: and softphone can connect to asterisk, i guess? |
| 16:28.09 | [TK]D-Fender | yes |
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| 16:52.25 | WIMPy | Greenlight: Yes. It happens if you have two calls on the same port. |
| 16:53.57 | Greenlight | Udp: |
| 16:53.57 | Greenlight | 198449276 packets received |
| 16:53.57 | Greenlight | 28511756 packets to unknown port received. |
| 16:54.10 | Greenlight | Guessing this is related |
| 16:56.04 | Greenlight | So, that's the *how*... now the *why* |
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| 16:59.00 | Greenlight | I'm guessing this is likely more a network issue than some Asterisk problem, right? |
| 17:00.25 | WIMPy | I managed to get the issue several times with calls that came in while Asterisk was still starting up. |
| 17:01.12 | Greenlight | hmm... so perhaps Asterisk is handing out port numbers that are in use |
| 17:03.51 | WIMPy | It's more like not propperly rejecting the calls it doesn't handle yet. |
| 17:04.19 | Greenlight | Well this isn't during startup, it's while it's been running a while. |
| 17:04.28 | Greenlight | Every now and again we get calls with crosstalk |
| 17:04.32 | WIMPy | Or answering too slowly perhaps. |
| 17:05.09 | WIMPy | I haven't had it at other times, yet. |
| 17:05.23 | WIMPy | Just wanted to mention that it is possible. |
| 17:05.58 | Greenlight | The UDP packets to wrong port would suggest there's lots of "crap" coming in. I'd guess the crosstalk occurs when this "crap" hits a in use RTP port |
| 17:06.47 | WIMPy | Yes |
| 17:07.39 | WIMPy | Do you have nat support enabled? |
| 17:07.43 | Greenlight | I was also seeing "exceptionally long queue length" reports in the logs, wonder if they're related |
| 17:07.50 | Greenlight | I do yes |
| 17:08.00 | Greenlight | Although coming to think of it, that box doesn't need it |
| 17:08.13 | WIMPy | Do you have strictrtp enabled? |
| 17:08.30 | WIMPy | That might help filter unwanted rtp. |
| 17:09.09 | Greenlight | Nope, not currently enabled |
| 17:09.19 | Greenlight | Oh, it's enabled by default |
| 17:09.26 | Greenlight | SO, yes |
| 17:09.48 | Greenlight | Although, that just protects by the source address. Which is likely going to be the same |
| 17:10.00 | Greenlight | (eg an upstream gateway) |
| 17:10.01 | WIMPy | Is it all about calls from the same itsp? |
| 17:10.26 | Greenlight | Well, calls from another one of my Asterisk boxes which handles billing and routing |
| 17:10.47 | WIMPy | So Asterisk on both ends? |
| 17:10.50 | Greenlight | Yea |
| 17:11.08 | WIMPy | always uses IAX for that. |
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| 17:12.28 | Greenlight | I did have some issues with IAX channel names not being unique enough the last time I tried using IAX |
| 17:15.49 | Greenlight | Hmm I think you've pointed me to the problem with the nat question |
| 17:16.10 | Greenlight | I've another box set to use nat (same setup as this) but it's missing a localnet |
| 17:16.19 | Greenlight | And set to use the externalip of this box |
| 17:16.49 | WIMPy | Yes, that sounds like it might be the cause. |
| 17:18.28 | Greenlight | Thanks for the help, appriciated |
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| 17:43.06 | *** join/#asterisk woopstar (~woopstar@unaffiliated/woopstar) |
| 17:44.44 | woopstar | Hi there. So i was just trying to migrate from chan_sip to chan_pjsip. So I was setting up a new Asterisk 13.3.2 with ODBC. Following the guide with PJSIP for realtime, and it seems that asterisk is crashing by the line "contact=realtime,ps_contacts" in the sorcery file now, which is in the guide (but not in the sample file) |
| 17:46.46 | file | woopstar, Asterisk shouldn't crash - I'd suggest filing an issue at https://issues.asterisk.org/jira |
| 17:47.16 | woopstar | oh, crash is not the right term. Sorry. But pjsip module cannot load because of sorcery. I'll file an issue |
| 17:48.39 | woopstar | this is the error when contacts is added: http://pastebin.com/HMrXhHKA |
| 17:50.26 | file | what's the complete output? |
| 17:51.07 | woopstar | It's here: http://pastebin.com/mSSHftzZ |
| 17:52.03 | woopstar | only crashes with the contact line in sorcery. When outcommented, everything works |
| 17:52.10 | file | yeah, file an issue |
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| 17:59.35 | woopstar | Issue filed here: https://issues.asterisk.org/jira/browse/ASTERISK-25050 |
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| 18:16.43 | notanoldman | If you have an included context and you dial an extension that is in there, will the context variable be the original context or the included one? |
| 18:17.02 | notanoldman | If it's the included one, how do you get the original context? |
| 18:17.40 | notanoldman | Is there a built-in from-context variable possibly? |
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| 18:24.32 | phix | WIMPy!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!1111111111111111 |
| 18:28.53 | woopstar | file: I think i found the bug. The issue is not Asterisk related, but guide related. ps_contacts is not defined in extconfig.conf |
| 18:30.57 | phix | ok |
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| 18:36.26 | thejourney | hello, if I run a freePBX install and want to do funky things with the speaking clock, is the advice - "don't" as i'm running FreePBX or do many users also do work from the command line to? |
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| 18:52.55 | SunGod | Hello all,I'm looking into compiling chan_dongle with asterisk11.According to https://code.google.com/p/asterisk-chan-dongle/issues/detail?id=167 comment #12 that should be possible and I think he is referring to the original sources not the patched versions.Has anyone tried this? |
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| 19:03.17 | phix | SunGod!!!!! |
| 19:03.26 | phix | Or should I just call you Ra? |
| 19:15.00 | SunGod | SunGod is just fine phix |
| 19:21.26 | phix | k |
| 19:22.00 | phix | 4 characters too long imo |
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| 19:24.05 | SunGod | well I was between that or Muhammad :D |
| 19:31.03 | WIMPy | thejourney: If you want to modify FreePBX, you'd better ask in #freepbx. |
| 19:31.20 | WIMPy | SunGod: It used to work. |
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| 19:34.39 | SunGod | WIMPy: can you give me a bit more explanation? |
| 19:35.08 | WIMPy | What's the question? |
| 19:36.01 | SunGod | Well you said "It used to work" will it not work now? |
| 19:36.22 | WIMPy | I haven't tried lately. |
| 19:37.02 | SunGod | WIMPy: ok thanks anyway. |
| 19:38.31 | thejourney | WIMPy, but by the same token, is FreePBX not just working with Asterisk modules under the bonnet anyway |
| 19:39.32 | WIMPy | thejourney: Yes, but FreePBX uses Asterisk. If you want to use Asterisk, you need to get rid of FreePBX or find out how you can share it. |
| 19:42.58 | thejourney | WIMPy, not to be argumentative - just because I don't understand - in relation to your last comment, why? |
| 19:44.10 | WIMPy | FreePBX won't like it if you change _it's_ Asterisk config files. If you want to do it you have to do it the way they provide. |
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| 23:08.21 | voip | hi |
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| 23:29.20 | notanoldman | <PROTECTED> |
| 23:29.20 | notanoldman | <notanoldman> If it's the included one, how do you get the original context? |
| 23:29.20 | notanoldman | <notanoldman> Is there a built-in from-context variable possibly? |
| 23:33.55 | [TK]D-Fender | There is no such thing as scope with channel variables |
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