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| 00:45.53 | CoolCanuck | Hi. This is a stupid question -- do I *need* external hardware? My desktop has a phone jack. Is that suffice? |
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| 00:52.44 | [TK]D-Fender | ? |
| 00:52.57 | [TK]D-Fender | What do you mean "your desktop" has a phone jack? |
| 00:53.25 | [TK]D-Fender | And no, you do not NEED external hardware. You might WANT it if you have physical devices to plug in. |
| 00:53.30 | [TK]D-Fender | And intend on using |
| 00:53.33 | CoolCanuck | Well |
| 00:54.09 | CoolCanuck | I would just like it to see the number calling into the line, and pickup and disconnect it if it matches a blacklist. That's all :) |
| 00:55.33 | [TK]D-Fender | Unless that is a very specific class of analog interface supported by DAHDI (the driver interface layer for *), then that device won't be usable |
| 00:55.43 | [TK]D-Fender | And you'll have to get another device |
| 01:00.12 | CoolCanuck | I'm not sure what you mean. I have no experience with that lingo :( |
| 01:02.58 | [TK]D-Fender | what is this "jack" you say you have on your desktop? |
| 01:03.28 | CoolCanuck | a phone port |
| 01:03.31 | CoolCanuck | rj-45 or whatever |
| 01:03.40 | CoolCanuck | used for fax/dial up mostly |
| 01:04.11 | CoolCanuck | (I don't use dial up, but I could) |
| 01:06.34 | CoolCanuck | Conexant PCI Modem |
| 01:07.35 | [TK]D-Fender | unusable |
| 01:08.21 | CoolCanuck | dang. |
| 01:08.23 | CoolCanuck | oh well |
| 01:09.05 | *** join/#asterisk fornax (~fornax@85.183.43.221) |
| 01:10.27 | [TK]D-Fender | You'd need another interface that * can support for this. |
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| 03:24.08 | igcewieling | the answers you seek are within you, grasshopper |
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| 09:07.36 | Mulder | if i'm wanting my asterisk server to connect to another asterisk server as a peer via tls (with media encrypted as well), how should i set that up? the guides i can find online seem to discuss when my asterisk server is letting other peers connect to me with tls. |
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| 11:05.38 | ruied | Hello. Is there a way so I can make a call to ext "21" and notify (without ringing) the ext "22" displaying the number ringing at ext 22, so I can pickup the call knowing ho is calling at ext 20 ? |
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| 11:07.08 | ruied | I am for example at extension 21 and would like to pickup the ringing extension (20) but knowing ho is calling that extension... |
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| 11:12.05 | pa | is there by chance a limitation to max 1 gtalk account with chan motif (or in xmpp)? |
| 11:14.21 | *** part/#asterisk aandrew (foobar@gromit.mixdown.ca) |
| 11:19.32 | pa | hm.. i found this, but it's not exactly my problem, and also i donno if asterisk in ubuntu 14.04 has the fix.. dial_exec_full: Unable to create channel of type 'Motif' (cause 42 - Switching equipment congestion) |
| 11:19.35 | pa | ops |
| 11:19.41 | pa | i meant http://asteriskfaqs.org/2012/11/05/asterisk-users/google-voice-and-back-chan_motif.html |
| 11:21.32 | pa | thing is : i get all the debug info from the second account |
| 11:21.37 | pa | so it registers and everything i think |
| 11:31.52 | pa | ok it seems that the difference is made by the undocumented option "(default)" |
| 11:32.01 | pa | without that, it doesnt work |
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| 12:10.48 | [TK]D-Fender | ruied: "notifycid=yes" under [general] if your phone supports it |
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| 12:52.12 | dajaz27 | What is the correct way to ask an asterisk question? |
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| 12:58.05 | dadrc | asking helps ;) |
| 12:59.12 | [TK]D-Fender | ~ask |
| 12:59.12 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:09.44 | dajaz27 | I am trying to use Asterisk ARI to create an outbound survey menu. I need to 1. originate a phone call, 2. play menu 3. pass the data back to the database table. |
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| 13:17.50 | asteriskmonkey | anyone know how to setup sip debugging to go to a file? |
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| 14:12.49 | jknox | I would use wireshark to capture the signaling and media, if you are trying to debug calls. |
| 14:14.17 | Chainsaw | Yes, you can capture with tshark on the server locally, and then use the wireshark frontend on your local computer. It's got detailed SIP & RTP analysis that will help you make sense of the flow. |
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| 14:30.27 | dan_j | When using pjsip, how can I tell if an endpoint is registered within the dialplan? |
| 14:31.02 | dan_j | For chan_sip, I was using ${SIPPEER(company_201,status)} |
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| 15:59.40 | liquidempire | Are there any PJSIP TLS experts out there? |
| 16:07.44 | [TK]D-Fender | ~ask |
| 16:07.44 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:17.26 | dan_j | What options do I have if I want to share sounds files between multiple (more than 2) asterisk servers? |
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| 16:18.12 | igcewieling | dan_j: rsync seems best. |
| 16:19.00 | dan_j | can rsync sync between multiple servers at the same time? the files should change on any of 4 different servers and rsync will need to be able to sync in all directions. |
| 16:19.42 | dan_j | should=could |
| 16:19.42 | seanbright | is there maybe an #rsync channel? |
| 16:19.55 | dan_j | yep. will give it a go. |
| 16:19.56 | seanbright | or a website with useful information about rsync? |
| 16:20.44 | *** join/#asterisk casdude (57ed466d@gateway/web/freenode/ip.87.237.70.109) |
| 16:20.52 | casdude | hi |
| 16:21.53 | casdude | I am running a dell server with asterisk 1.8. however am unable to authenticate with a SIP trunk |
| 16:22.24 | casdude | I am being told that this is due to me using incompatible hardware, i think i am being lied to... has anyone been told that the reason you |
| 16:22.39 | dan_j | hardware shouldnt make any difference. |
| 16:22.49 | casdude | no i didn't think so either |
| 16:23.01 | dan_j | pastebin a sip debug of the registration or call attempt |
| 16:23.15 | [TK]D-Fender | casdudeI am being told that this is due to me using incompatible hardware, i think i am being lied to... has anyone been told that the reason you <- that is ridiculous. |
| 16:23.29 | casdude | it wont register, keeps telling me that my password is incorrect |
| 16:23.56 | [TK]D-Fender | probably means what it says... password is not correct for the username it is trying to auth as |
| 16:24.10 | casdude | yeh |
| 16:24.14 | casdude | ok thanks guys |
| 16:24.15 | [TK]D-Fender | Doesn't mean the USERNAME is right either. Or if it's counting as a realm name they aren't expecting |
| 16:24.27 | tompaw | Hey guys, can you please help me with this situation here: |
| 16:24.29 | tompaw | http://hastebin.com/ikodaguvay.avrasm |
| 16:24.44 | casdude | i think i am good now, catch you later |
| 16:24.48 | tompaw | I've been banging my head against the freaking wall for the past 3 days |
| 16:24.50 | casdude | have a good weekend all |
| 16:25.50 | [TK]D-Fender | tompaw: you show what looks like an inbound register... and we see no answer to it... |
| 16:26.12 | tompaw | ah, sorry |
| 16:26.13 | [TK]D-Fender | tompaw: So I'll just go right ahead and assume that there is a blatant "wrong password" response you aren't showing us |
| 16:26.44 | tompaw | nope |
| 16:26.45 | tompaw | http://hastebin.com/ivipilacut.avrasm |
| 16:26.58 | tompaw | it works great as a Local extension on that particular server |
| 16:27.16 | igcewieling | <-- Waiting for Asterisk 15 before trying PJSIP |
| 16:27.23 | tompaw | but I need to scale up on inbound calls and need to do cross-server calls to extensions, which means I *MUST* have valid contacts on file |
| 16:27.51 | tompaw | for some reason the contact information is simply removed from db after around 1 minute |
| 16:28.02 | tompaw | no matter what I configure as minimum_expiration |
| 16:28.17 | tompaw | and all the log says is this: Contact 2/sip:2@10.57.13.171:5060 has been deleted |
| 16:28.31 | file | noload res_pjsip_registrar_expire? |
| 16:29.25 | tompaw | file: hm... this + limit number of contacts to "1" so they can be updated |
| 16:29.39 | tompaw | (we have roaming agents) |
| 16:30.41 | tompaw | I'd really like to know why they are removed from db, tho :-( |
| 16:31.03 | file | are you using multiple servers against the same database? |
| 16:31.07 | tompaw | yes |
| 16:31.29 | file | then you should noload the res_pjsip_registrar_expire module, otherwise if a client moves from one server to another another server may remove the contact |
| 16:32.36 | tompaw | file: I will noload it anyway, but I'm 100% sure it's not the only reason, as agents do not move between the servers (REGISTER-wise) without my explicit command to do so |
| 16:32.56 | file | then you'd have to add logging to determine exactly what is happening |
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| 16:34.59 | tompaw | file: what kind of logging? I have debug 4 and verbose 4 |
| 16:35.08 | file | you'd have to add logging to the code |
| 16:35.13 | file | and diagnose the flow |
| 16:35.24 | tompaw | :o |
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| 16:39.39 | liquidempire | Is there a way force the transport in PJSIP to use tls without specifying sips? I only have a tls transport specified. However, if I use a contact URI like sip:1234567890@sipprovider.com:5061;transport=tls the call fails with a transport error. If I set the uri with a sips: the invite is sent. Unfortuantely, my carrier does not like sips. |
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| 16:46.22 | file | liquidempire, no - PJSIP internally forces it |
| 16:49.16 | liquidempire | So, we can't run sip over tls without specifying sips? |
| 16:49.40 | file | not without hacking up the PJSIP library itself |
| 16:50.48 | liquidempire | Got it. That certainly explains why it doesn't work. Thank you! |
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| 17:09.32 | *** join/#asterisk flujan (~flujan@191.255.230.73) |
| 17:10.19 | flujan | hello guys. Is it possible to detect audio on a SIP call before it is aswered? Does asterisk raise a event if it detect early media? |
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| 18:10.29 | *** join/#asterisk infobot (ibot@69-58-76-73.ut.vivintwireless.net) |
| 18:10.29 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 18:19.41 | liquidempire | This error is cosmetic, correct? " ERROR[9371]: pjsip:0 <?>: tlsc0x26f1778 RFC 5922 (section 7.2) does not allow TLS wildcard certificates. Advise your SIP provider, ple |
| 18:19.41 | liquidempire | ase!" |
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| 18:25.19 | tompaw | file: unloading res_pjsip_registrar_expire seems to have done the trick, thanks! |
| 18:25.30 | *** join/#asterisk NephFL (~NephFL@107.144.5.174) |
| 18:26.26 | NephFL | hello, I am working on agi...just getting started with it actually. When I enable the agi debug, I can see the request and response, is there a way to enable the normal command output in the log the way you would see with any other command? |
| 18:26.42 | NephFL | for example, so that when I send a noop, it goes into the log and the cli |
| 18:27.04 | WIMPy | No |
| 18:27.29 | WIMPy | You could enable CEL, but that might be more than you want. |
| 18:27.58 | NephFL | so, I need to catch the return values when I issue a command and send my own debug output? |
| 18:28.31 | WIMPy | What return values? |
| 18:28.57 | NephFL | lol, good question, I'm just pulling up the php-agi ref docs to see what data return types, etc I have available |
| 18:29.22 | *** join/#asterisk atna99 (~atna99@S01067824afed7d58.vc.shawcable.net) |
| 18:29.29 | WIMPy | If you want to debug PHP, you have to do it there. |
| 18:29.45 | NephFL | ok, cool |
| 18:29.57 | NephFL | I need to create a seperate log for my project anwho |
| 18:30.17 | atna99 | is there a asterisk JIRA Admin available? I need an issue deleted |
| 18:30.41 | WIMPy | You can send it back to Asterisk for logging with Verbose or Log, but you still need to write somthing in there yourself. |
| 18:31.17 | WIMPy | atna99: You might also try in #asterisk-bugs |
| 18:31.47 | atna99 | kk thanks |
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| 18:40.16 | ghost75 | i tried possibly everything to get rid of this problem with t.38: sendfax_t38_init: error reading frame while generating CNG tone |
| 18:40.25 | ghost75 | what can it be? |
| 18:41.46 | ghost75 | according to source code i think the provider doesnt answer to CNG tone |
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| 20:19.08 | ewook | gaha. anyone knows if chan_dongle got it's own little support corner or AA-like meeting centre somewhere? getting the SMS to work just kills me :p. |
| 20:28.19 | *** join/#asterisk avb (~avb@37.44.103.184) |
| 20:28.22 | avb | hey guys |
| 20:29.00 | avb | any ways to persist channel variables between channels? for example on call transfer, joining person into conference, etc |
| 20:30.28 | avb | i have found https://wiki.asterisk.org/wiki/display/AST/Function_MASTER_CHANNEL |
| 20:36.16 | jvhester | avb: Sounds like you are describing variable inheritance https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance |
| 21:00.49 | avb | jvhester: no, not exactly |
| 21:01.28 | avb | this is working, but in case of the call transfer/call from the queue/conference/etc channel variables are getting lost |
| 21:01.30 | avb | :-/ |
| 21:05.24 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:34.26 | *** join/#asterisk Dovid (~dovid@ool-45738930.dyn.optonline.net) |
| 21:44.09 | NephFL | can I add <path1>:<path2> in asterisk.conf to add an agi directory? |
| 21:44.13 | *** join/#asterisk Chotaire (chotaire@vegetarian.cannibal.club) |
| 21:48.21 | NephFL | nope |
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| 23:56.52 | vitorbaptista | Hello! I'm having some problems when connecting two Asterisk boxes through IAX2. I'm getting "No authority found". I googled around and found that this usually was because the passwords of the servers weren't correct, but I checked and they are. I can call from server A to server B, but not from server B to server A. They are both connected when I try with "iax2 show peers". I tried debugging, and the log |
| 23:56.54 | vitorbaptista | is at http://pastebin.com/NTmwwSTw |
| 23:57.22 | vitorbaptista | I'm poking around randomly now, but I'm getting out of ideas on where to look. Any suggestions? |
| 23:57.58 | robmal | fromuser should match type=user on the other end |
| 23:58.44 | robmal | https://www.youtube.com/watch?v=gx4jn77VKlQ |
| 23:59.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |