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| 00:27.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.14.0 (2017/02/13), 11.25.1 (2016/12/08), Standard: 14.3.0 (2017/02/13); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:21.08 | siraskalot | Hello i've been reading the asterisk definitive guide book, has the freenum.org site moved? i can't seem to connect to it |
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| 08:38.05 | drmessano | Freenum is dead |
| 08:48.40 | siraskalot | oh! ok can yo point me to another solution similar to freenum? |
| 09:04.27 | drmessano | There's nothing else |
| 09:04.44 | drmessano | It was a solution without a problem |
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| 09:36.15 | everestt | Hi, I couldn't find how to set sip headers using ARI (originate: POST /channels) to make an outbound call. Specifically I need to set the a number in the sip INVITE request. If anyone has any idea, please help. |
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| 13:17.42 | sotoz | When creating a channel with ARI (originate: POST /channels) is it possible to use a different callerID for a second channel that will use the first channel as source and will be on the same bridge? |
| 13:18.58 | sotoz | I do a channelsOriginate() and I set a callerID for this channel which works properly, but when I'm creating a second channel and dialing this channel with the first channel as "caller" I get the same callerID as the first channel. |
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| 13:31.26 | sotoz | anyone? ^^ |
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| 13:35.05 | igcewieling | If you provided your Asterisk version and exactly how you set the CallerID, someone might be interested enough to reply. I don't use ARI, and I suspect few people here use it. |
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| 14:57.04 | sotoz | When using the ARI to dial a channel do we need to specify the "caller" from the query parameter in order to set a callerid for this channel? |
| 14:57.54 | sotoz | I'm trying to use the SetChannelVar to assign a CALLERID(all) variable for this channel but I get an "unknown" caller id on the phone |
| 14:59.30 | sotoz | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Channels+REST+API#Asterisk14ChannelsRESTAPI-dial |
| 14:59.52 | sotoz | It has as a query parameter: "caller: string - Channel ID of caller" |
| 15:00.28 | sotoz | but what if I want to dial a newly created channel that has no other channel related to it so I can declare it as caller. |
| 15:03.22 | Samot | So you want to use a local channel? |
| 15:05.21 | Samot | What what are you setting as CALLERID(all)? |
| 15:05.31 | Samot | What is the value you are setting? |
| 15:06.36 | igcewieling | If you provided your Asterisk version and exactly how you set the CallerID, someone might be interested enough to reply. I don't use ARI, and I suspect few people here use it. |
| 15:06.46 | sotoz | While I couldn't set a "caller" channel ID for the newly created channel I thought that I could set a CallerID with a SetChannelVar, so I'm doing a setting CALLERID(all) with a specific number which I want to be the caller id. |
| 15:06.51 | sotoz | It's Asterisk 14. |
| 15:07.26 | igcewieling | CALLERID(all)="The Name" <thenumber> |
| 15:07.58 | igcewieling | if you are just settng the number, then set the number. |
| 15:07.58 | sotoz | igcewieling: I also tried CALLERID(number)="mynumber" |
| 15:08.19 | igcewieling | sotoz: quotes are not valid. |
| 15:08.34 | sotoz | I know. |
| 15:08.41 | igcewieling | then why are you using them? |
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| 15:09.01 | sotoz | cause I didn't want to enter my number in a public chat |
| 15:09.22 | igcewieling | hiding stuff will prevent help. |
| 15:09.30 | sotoz | sure. |
| 15:09.38 | igcewieling | Try CALLERID(num)=12345 no quotes |
| 15:10.23 | sotoz | I'm using a go ari library that I created, and setting channel vars is working great. |
| 15:10.26 | igcewieling | I don't see "number" in "core show function CALLERID", but I'm running Asterisk 11 |
| 15:10.59 | igcewieling | which might mean you are not setting the CallerID number. |
| 15:11.02 | sotoz | I think the CALLERID(num) is not the problem |
| 15:11.25 | sotoz | When dialing a channel with ARI |
| 15:11.30 | sotoz | you can provide the caller |
| 15:11.40 | sotoz | which is the ChannelID of the caller |
| 15:11.41 | sotoz | https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Channels+REST+API#Asterisk14ChannelsRESTAPI-dial |
| 15:11.57 | sotoz | but what if NO channel exists that could be the caller |
| 15:12.22 | igcewieling | You'd need the "all" format of "thename" <thenumber> |
| 15:12.47 | igcewieling | not sure what you mean. |
| 15:12.56 | Samot | What does the channel id have to do with callerid? |
| 15:13.08 | igcewieling | I am ONLY commenting on how you are setting CallerID. |
| 15:13.11 | sotoz | "caller: string - Channel ID of caller" |
| 15:13.16 | Samot | OK |
| 15:13.16 | sotoz | It's on the official documentation |
| 15:13.20 | Samot | That's the CHANNEL ID |
| 15:13.24 | Samot | You're talking about the CALLERID |
| 15:13.46 | sotoz | what the caller means then |
| 15:13.59 | sotoz | where is this being used? |
| 15:14.05 | igcewieling | I think he means callerId: string - CallerID to use when dialing the endpoint or extension. |
| 15:14.45 | sotoz | hmm now I got more confused... |
| 15:14.48 | sotoz | sorry |
| 15:15.51 | sotoz | When dialing with ARI and while providing this: "caller: string - Channel ID of caller" where is this "caller" being used from asterisk??? |
| 15:16.30 | sotoz | I thought it was being used to provide the caller id for this channel. |
| 15:16.32 | igcewieling | sotoz: A channel ID is an internal Asterisk ID, not CallER id. |
| 15:17.07 | sotoz | igcewieling: Is this caller being used for anything in asterisk? Or I can ommit it? |
| 15:17.40 | igcewieling | It is used to link different CHANNELS by their ID. |
| 15:17.50 | igcewieling | Assuming the docs are correct. |
| 15:18.47 | igcewieling | Body parametervariables: containers - The "variables" key in the body object holds variable key/value pairs to set on the channel on creation. Other keys in the body object are interpreted as query parameters. Ex. { "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" } } |
| 15:18.54 | igcewieling | now you see where to set the CallerID |
| 15:19.42 | sotoz | So should I send JSON representation on the body? |
| 15:19.45 | sotoz | like the example? |
| 15:19.51 | igcewieling | no idea. |
| 15:20.01 | igcewieling | I'm just looking at the docs. |
| 15:20.08 | sotoz | ok |
| 15:21.23 | sotoz | the ARI documentation needs a little bit of more clarification on some parts |
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| 16:13.30 | wasanzy | am getting this when I tried to register my simbox with asterisk https://paste.linux.community/view/d68cecf5 |
| 16:13.52 | wasanzy | zenith/dwg 192.168.10.91 D No No 5060 Unmonitored |
| 16:14.05 | wasanzy | seems registered but call are not getting to asterisk |
| 16:15.28 | Samot | Show a call that fails.. |
| 16:16.05 | wasanzy | Samot: the big problem is call doesn't get to asterisk. |
| 16:16.28 | Samot | So there's no INVITE at all? |
| 16:16.43 | wasanzy | looking at my sip register log, can you say the client registered? |
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| 16:16.57 | wasanzy | Samot: yes nothing at all |
| 16:17.21 | Samot | Well first, you should qualify your device. |
| 16:17.31 | Samot | So it doesn't show up as Unmonitored |
| 16:17.44 | Samot | And Asterisk knows that it still is where it says it is. |
| 16:18.17 | igcewieling | sounds like a nat issue |
| 16:18.33 | wasanzy | Samot: qualified? how do I set that ? |
| 16:18.37 | Samot | There's no NAT |
| 16:18.44 | wasanzy | igcewieling: I have nat=no |
| 16:18.45 | Samot | qualify=yes |
| 16:18.48 | Samot | In the peer. |
| 16:19.00 | wasanzy | type=peer |
| 16:19.04 | igcewieling | At first glance nothing seem wrong with the pastebin. |
| 16:19.48 | Samot | The REGISTER wasn't overly in question. |
| 16:19.50 | igcewieling | the initial register request is ALWAYS rejected. The 2nd register request uses the NONCE from the rejected back as a "key" |
| 16:19.57 | Samot | It's the fact that he's not seeing INVITES from the device. |
| 16:20.35 | Samot | Which appears to see some GSM gateway. |
| 16:20.48 | Samot | What do the logs/debugs on the GSM gateway say? |
| 16:20.59 | Samot | When it tries to send a call to Asterisk... |
| 16:25.40 | wasanzy | Samot: am checking on the logs |
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| 18:12.00 | wabbits | linphone client sends subscribe and gets SIP/2.0 489 Bad Event |
| 18:12.06 | wabbits | anyone seen this? |
| 18:15.26 | Samot | What are the subscribers for? |
| 18:15.41 | Samot | s/subscribers/subscribes/ |
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| 18:17.02 | wabbits | presence |
| 18:17.38 | wabbits | Samot |
| 18:17.58 | Samot | Do you have the proper hints created for it? |
| 18:18.38 | [TK]D-Fender | That should 404 if there is no match |
| 18:18.51 | wabbits | have not seen the asterisk configuration or console and don't know what a hint is or anything about asterisk |
| 18:19.38 | [TK]D-Fender | Since you're asking about Asteriskyou need togetyourhandson it |
| 18:19.38 | wabbits | looking for traces |
| 18:20.19 | wabbits | my mistake |
| 18:21.17 | wabbits | thanks |
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| 18:43.46 | qakhan | hi all, |
| 18:44.23 | qakhan | i setup sip realtime and made some extensions in table "sip_extensions" |
| 18:46.41 | qakhan | when i run sip show peers. i can see all the registered extension |
| 18:47.08 | qakhan | but after sometime there are no extension when i run sip show peers |
| 18:47.37 | qakhan | i have to restart the phone or re-register the extensions on the phone |
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| 19:01.24 | tuxd00d | qakhan: firewall between phone and server? |
| 19:03.56 | qakhan | tuxd00d the only firewall i have on server, which allows port 5060 and rtp 10000-20000 |
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| 19:05.29 | qakhan | i also double checked that rtcachefriends=yes in sip.conf |
| 19:05.44 | tuxd00d | Youâve checked your PCAP |
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| 19:06.22 | qakhan | no i did not |
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| 19:09.57 | tuxd00d | You can quickly check live traffic using âsngrepâ |
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| 19:16.54 | qakhan | thanks tuxd00d i will check with this |
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| 22:52.21 | jfindley | is there a reliable way to get the Sip Call-ID used during a DIAL? |
| 22:52.34 | jfindley | PJSIP_HEADER(read,Call-ID) only seems to work for the inbound side |
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| 23:00.42 | rmudgett | jfindley: You could try getting it in the Dial's U pre-bridge routine. That would be about the only place you could get it on the outgoing channel. The call-id isn't created until the INVITE goes out. |
| 23:01.42 | jfindley | oh neat! |
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| 23:36.34 | jfindley | rmudgett: It's possible to use both pre-dial and pre-bridge right? |
| 23:36.48 | jfindley | in the same dial command** |
| 23:36.54 | rmudgett | yes |
| 23:37.33 | jfindley | the documentation isn't super clear, but I assume with U you can do U(context^exten^priority(args)) right? |
| 23:37.42 | rmudgett | yes |
| 23:38.18 | rmudgett | correction. In this case no. |
| 23:39.56 | rmudgett | Dial(PJSIP/100,,U(context^args) |
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| 23:40.35 | rmudgett | The format tried to make a gosub look more like a macro. Newer things like pre-dial allow more freedom. |
| 23:41.26 | jfindley | interesting. Does the order in the dial string matter do you think? I've tried placing the U(pre-bridge-hook) in various places but Asterisk seems to not care about it at all |
| 23:42.05 | jfindley | Asterisk 13.1 is what I'm testing with |
| 23:42.39 | rmudgett | The gosub in dialplan would be [gosub-name] exten s priority 1 |
| 23:43.02 | rmudgett | The order of the flags doesn't matter. |
| 23:50.54 | jfindley | Hmm... I'll have to play around with it more later. Thanks for the suggestion! |