| 00:00.13 | JonathanD | I've not found this documented anywhere and I'm currently working through the source, but wondered if anyone knew specifically what it was. |
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| 11:03.38 | mangiafazula_ | Hello ! Q: in /var/lib/asterisk/agi-bin/ ive got a little php-script . I cant see anything inside my asterisk -rvvvvvv about it. I expect "$agi->verbose("hello")" there for example. Ideas ? |
| 11:04.58 | mangiafazula_ | also, every $agi-> directive I have to manually process hitting the "Enter" Button . That seems wrong to me. Right ? |
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| 11:06.01 | mangiafazula_ | ** inside asterisk -rvvvv, i got agi debug on, and core debug 5, but I cant see anything about my agi - script |
| 11:06.32 | mangiafazula_ | I call my agi - script from inside my bash-console. Its owner is asterisk, tho |
| 11:06.55 | mangiafazula_ | and i intend to call it from the console .. |
| 11:08.55 | sibiria | -fvvvvvv not -r |
| 11:09.17 | sibiria | if you want to start asterisk in foreground mode |
| 11:09.51 | sibiria | also make sure you have 'notice' enabled for console output in logger.conf |
| 11:13.42 | sibiria | AGI commands must be terminated with a newline |
| 11:13.52 | sibiria | it's an stdin/stdout kind of deal |
| 11:14.23 | sibiria | your framework should be doing this automatically for you, for every method (such as verbose() etc.) |
| 11:16.04 | mangiafazula_ | sibiria thank you very much for reply !!! |
| 11:20.20 | sibiria | usually AGI scripts are called not manually but within a dial plan during a call, so the behavior you're seeing might be related to that |
| 11:21.41 | mangiafazula_ | sibiria --- Im trying to make a "call-the-customer-from-extension" button for my folks ... |
| 11:22.11 | mangiafazula_ | params shall be customerphonenumber and extensionnumber |
| 11:22.50 | mangiafazula_ | so they push the button, and customerphones ringing and also extension shall be ringing |
| 11:23.38 | mangiafazula_ | i try to do this with an AGI script |
| 11:29.05 | mangiafazula_ | sibiria : I have to press enter to process "$agi = new AGI()" |
| 11:29.18 | mangiafazula_ | ; |
| 11:29.32 | mangiafazula_ | thats odd, right ? |
| 11:29.50 | sibiria | yes. i'm not sure why, i never used any AGI frameworks in PHP, only in other languages |
| 11:30.12 | sibiria | but i would guess that it's related to you executing the script in this way |
| 11:30.28 | sibiria | as said, AGI operates through stdin/stdout |
| 11:31.28 | sibiria | what you want to achieve is possibly best done with AMI/ARI instead |
| 11:31.38 | mangiafazula_ | ahh cool |
| 11:31.41 | sibiria | or perhaps even a simple callfile |
| 11:31.42 | mangiafazula_ | very nice |
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| 13:07.19 | mangiafazula_ | help |
| 13:09.32 | Samot | Be more descriptive. |
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| 13:43.04 | mangiafazula_ | <PROTECTED> |
| 13:46.38 | Samot | That doesn't tell me much. |
| 13:46.52 | Samot | What does the rest of the call file look like? How is this call being done in some other way? |
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| 14:50.31 | overyander | is there a way for a polycom phone to update it's directory provisioning without resetting it to factory defaults? |
| 14:51.12 | overyander | from what i've read, the phone will only pick up the directory file during first boot and has to be reset to factory settings in order for it to get an update made to that file. |
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| 15:00.10 | ChkDigit | You can force a polycom phone to reboot (and reload config) from the Asterisk CLI: sip notify polycom-check-cfg ext |
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| 15:01.16 | ChkDigit | If you find a way for them to do it without the reboot, I'd love to know. |
| 15:03.39 | overyander | ChkDigit, I'm aware of that for the phones config but my understanding was that it didn't apply to updating the directory.xml file |
| 15:04.18 | Samot | Of course it does |
| 15:04.43 | Samot | It would be really dumb to have to reset your phone because a contact was removed/added/edited. |
| 15:06.15 | overyander | that's what i thought as well. lol |
| 15:06.33 | overyander | let me give it a shot. |
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| 15:35.51 | overyander | ChkDigit, btw, this is the config setting you're looking for voIpProt.SIP.specialEvent.checkSync.alwaysReboot="0" |
| 15:36.03 | overyander | it's in my site.cfg file |
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| 15:39.27 | ChkDigit | ty |
| 15:50.36 | overyander | Samot, is the 00..-directory.xml file supposed to be specified in the 000.cfg file? |
| 15:51.42 | overyander | I've created my directory.xml file but I don't see my phone grabbing it when tailing my ftp xfer log. |
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| 16:16.53 | tmoore | morning everyone, yesterday i asked about how to manually add pjsip registrations to asterisk from a forward facing registrar and a static contact was suggested. This doesn't seem to work in my case because we have multiple realms the customers would be coming from, this is all handled by the registrar. We would like to manually add the pjsip registration (asterisk is using pjsip realtime) bcuz we would otherwise have to either |
| 16:16.53 | tmoore | add an endpoint w/ a different ps_auth for each realm (which is unrealistic) or implement realm translation on the registrar (seems more complicated than needed). Is there not an AGI or ARI func we can call to manually add a pjsip registration? |
| 16:20.52 | file | no, Asterisk isn't really written with such a use case in mind |
| 16:22.25 | Samot | It's a database. |
| 16:22.29 | file | Throwing it in a real-time database of the correct schema may or may not work depending on usage |
| 16:22.33 | Samot | Insert the record. |
| 16:22.36 | file | options probably wouldnt |
| 16:23.44 | tmoore | i know ps_registrations and sippeers is usually updated after I register an endpoint, are there any other tables I would need to update? |
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| 16:24.27 | Samot | What proxy are you using? |
| 16:24.45 | tmoore | kamailio |
| 16:24.52 | Samot | Use the outbound |
| 16:25.20 | file | Most people just let the registrar do the lookup on outbound calls |
| 16:25.26 | Samot | When a contact registers to Kamailio trigger an outbound auth to Asterisk with the creds/realm for Asterisk. |
| 16:25.59 | Samot | You'll still need to route things properly |
| 16:26.13 | Samot | the registration would be Kamailio -> Asterisk |
| 16:26.33 | file | Yeah, that too |
| 16:27.13 | Samot | I just have it static. |
| 16:27.51 | tmoore | so your suggestion is that crafting the REGISTER for asterisk would the simplest solution? |
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| 16:28.11 | Samot | It's 6 and a half dozen really. |
| 16:28.26 | Samot | If you really need to them register... |
| 16:28.49 | Samot | I mean where is all the auth really done? |
| 16:28.53 | Samot | Kamailio? |
| 16:29.39 | tmoore | auth is really done at kamailio, ideally asterisk would recognize kamailio as a permitted IP, but I am unsure how to do this in pjsip |
| 16:30.56 | Samot | You do static contacts to kamailio |
| 16:31.19 | tmoore | contacts are synced to kamailio from realtime db |
| 16:31.24 | Samot | And set up the endpoints to auth with custom headers sent by Kamailio. |
| 16:31.35 | Samot | Wait..what? |
| 16:31.50 | Samot | Oh you're not using Kamailio for the locations. |
| 16:31.57 | tmoore | kamailio is acting as the registrar |
| 16:33.12 | Samot | So the only thing it auths is REGISTERs? |
| 16:33.56 | tmoore | it does auth INVITE's as well |
| 16:35.35 | Samot | OK so the device auth's at Kamailio and on the 200 OK ACK you write it to sippeers? |
| 16:36.20 | file | perhaps phrased this way - what purpose does Kamailio serve for REGISTER? |
| 16:36.46 | Samot | OK, that's better. |
| 16:37.11 | tmoore | all auth is done by kamailio, if kamailio says its good asterisk should allow the REGISTER |
| 16:37.19 | tmoore | thats the overview |
| 16:37.50 | Samot | Which means? |
| 16:38.01 | Samot | Asterisk doesn't challenge? |
| 16:38.17 | Samot | How does Asterisk allow/accept this REGISTER? |
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| 16:40.18 | tmoore | yes it does have to challenge (according to SIP), unless there is a way to tell asterisk to allow all reg from kamailio |
| 16:40.56 | Samot | And what location in Asterisk? |
| 16:41.14 | Samot | Kamailio's IP or the device's IP? |
| 16:42.52 | Samot | tmoore: Let me ask you this. Are you using the UAC module? |
| 16:43.02 | Samot | Writing out the REGISTER to Asterisk? |
| 16:44.47 | tmoore | kamailio's ip |
| 16:46.47 | Samot | tmoore: Is this based on the Kamailio/Asterisk RealTime setup documents? |
| 16:56.27 | tmoore | its not based on vo1pbx, its based on https://github.com/dOpensource/dsiprouter/blob/master/kamailio/kamailio_dsiprouter.cfg, but we are adapting for pjsip |
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| 17:07.08 | AL13N_lappy | anyone know if for inbound calls, i can just do: Answer(); Dial(SIP/201, 10); Dial(SIP/201&SIP/202, 30) ? |
| 17:07.45 | AL13N_lappy | do the others get dials even if the call succeeds? or how can i just ring 1 ext first for 10s and then do the others as well? |
| 17:20.47 | mangiafazula_ | Samot thx for reply ! ! Ill try my luck tomorrow again here, so sorry to alert you --- I was coding some totally different stuff in the meanwhile |
| 17:21.14 | mangiafazula_ | all: have a great day and god bless you |
| 17:38.51 | AL13N_lappy | also, how can i check the source extension in the dialplan? |
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| 18:51.58 | AL13N_lappy | "Had to drop call because I couldn't make PJSIP/trunk-00000008 compatible with PJSIP/202-00000009" |
| 18:52.28 | AL13N_lappy | so, the trunk has alaw|ulaw and 202 has opus|alaw|ulaw ... |
| 18:54.01 | AL13N_lappy | opus is first to try, but it doesn't want to switch to alaw... :-( |
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| 18:55.35 | sibiria | have you inspected the SDPs? |
| 18:56.41 | AL13N_lappy | ah, no, i didn't yet, though it should be fine |
| 18:57.48 | AL13N_lappy | i can do internal calls on opus, but now i'm testing a call from the outside |
| 18:58.16 | AL13N_lappy | sibiria: which comes in alaw, but i figured it would negotiate to switch to alaw |
| 19:00.08 | AL13N_lappy | sibiria: you mean this, right: a=rtpmap:8 PCMA/8000/1 |
| 19:00.14 | AL13N_lappy | this was the first one |
| 19:00.42 | sibiria | yes that's a-law |
| 19:02.10 | AL13N_lappy | i've seen renegotiations when calling internally... why does it not work with sip trunk? |
| 19:02.20 | AL13N_lappy | i just tried outside calling, and it's a similar problem |
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| 19:08.56 | AL13N_lappy | sibiria: i don't understand how this goes: basically, if some internal call comes in and it is ulaw; i the dialplan i direct it to a PJSIP/202 which has lotsa codecs, but for some reason it chooses opus? does it not take into account what the sip trunk sends it in? |
| 19:09.18 | AL13N_lappy | my own asterisk server just declines the call... |
| 19:09.32 | AL13N_lappy | maybe it's due to the ringing? isthe ringing in opus? |
| 19:09.58 | sibiria | it doesn't ring, audibly, until the session is established |
| 19:10.18 | sibiria | negotiation happen before the channel is connected |
| 19:10.47 | AL13N_lappy | sibiria: right, but the 202 endpoint has lots of codecs... |
| 19:11.12 | sibiria | you should look at both SDPs |
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| 19:17.01 | sibiria | meaning the one in the INVITE and the one coming back in the ACK |
| 19:19.29 | AL13N_lappy | sibiria: ok, i see the problem |
| 19:20.13 | AL13N_lappy | from outside, it asks for alaw, etc....; but the asterisk asks my linphone for alaw and ulaw, but also opus and speex |
| 19:20.57 | AL13N_lappy | that's kinda weird, if an internal call comes in and it doesn't have opus, then why when dialing my extension is it asking for opus and speex? i cannot transcode those |
| 19:21.54 | AL13N_lappy | linphone replies with opus only, because it's the best codec... |
| 19:21.57 | file | welcome to Asterisk codec negotiation |
| 19:22.28 | AL13N_lappy | is there a way for asterisk to filter out the codecs not in inbound call? |
| 19:23.28 | file | in 18 there are options to tweak the behavior |
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| 19:24.18 | AL13N_lappy | i read something about advanced negotiation |
| 19:24.22 | file | https://github.com/asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample#L821 |
| 19:24.25 | AL13N_lappy | luckily, i do have 18 |
| 19:24.30 | file | "remote" may do as you desire. |
| 19:24.32 | AL13N_lappy | maybe i'm thinking outgoing_offer: |
| 19:24.40 | AL13N_lappy | oh, i'll read that first |
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| 19:32.30 | acovrig | I'd like to be able to have a call land in my system, and it forward to my cell, but for me to be able to know who is calling. It seems most providers I look at don't allow CID spoofing, I'm using Twilio at the moment, is there something I'm missing? |
| 19:34.16 | acovrig | Currently all calls have to come from a verified number, so I can't spoof the CID for the call to my cell, the closets thing I could think of is some dialplan magic to allow the call to land w/my PBX's number, then use some STT to say the caller's CID and allow me to dial 1 or such from my phone to allow the call through |
| 19:36.22 | acovrig | or some AGI to notify me of the caller's ID via some other communication method (something like pushbullet or such) so I can know who it is before I answer |
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| 20:23.59 | sibiria | from what i understand it's getting trickier in the USA |
| 20:24.06 | sibiria | in europe it's still rarely a problem |
| 20:24.51 | sibiria | (operators are clamping down on it bit by bit and requiring registered caller IDs, but it's moving slowly) |
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