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| 03:09.33 | RudyValencia | So I'm trying to use Asterisk console and instead of letters I get something like \U+5390A\U+5396F... Why is this happening and how do I fix it? |
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| 03:43.14 | ruben23 | hi guys |
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| 04:51.15 | voipmonk | you still have to use letters in the console to "spell" commands |
| 05:17.52 | ChannelZ | Looks like some sort of character set/character encoding problem.. is this SSH? |
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| 06:15.31 | RudyValencia | ChannelZ: Sorry for the late response; yes, it is over SSH |
| 06:15.50 | RudyValencia | I can try a TTY to see if the problem is there too |
| 06:16.45 | RudyValencia | yes, the TTY also exhibits the same issue. |
| 06:18.10 | RudyValencia | I'm using version 13.10 built as an Arch Linux AUR package. I can uninstall it and build from the latest 13.11 source and see if the problem persists. |
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| 07:36.23 | RudyValencia | I've just compiled Asterisk 13.11.0 and I still get something like \U+5390A\U+5396F when I type |
| 07:36.30 | RudyValencia | Why is this happening and how do I fix it? |
| 07:42.55 | ruied | Hello. I have been using extensions.conf up to now. Now I'm looking for a possible change to ael but then I saw also lua. What are the benefits of using lua vs ael? I would like to go to bet on one of them, but do not know where to bet... |
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| 07:54.05 | ruied | RudyValencia, at * cli ? |
| 07:54.28 | RudyValencia | ruied: at the CLI, yes |
| 07:54.40 | RudyValencia | when I run asterisk -r to connect to a running instance |
| 07:57.34 | RudyValencia | or even if I run it with asterisk -c |
| 07:58.10 | RudyValencia | all I can do is exit with Ctrl-C |
| 08:01.05 | ruied | It seems to be writing some kind of unicode instead of the characters... |
| 08:01.52 | RudyValencia | yes |
| 08:01.59 | RudyValencia | I can't figure out why |
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| 08:15.50 | ruied | it seems something related to libedit... |
| 08:16.18 | ruied | What distro are you using? |
| 08:16.34 | RudyValencia | Arch Linux |
| 08:28.21 | ruied | locale charmap |
| 08:28.44 | RudyValencia | it returns UTF-8 |
| 08:32.58 | RudyValencia | ruied: ^ |
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| 08:33.37 | ruied | just to check, try to access asterisk from other pc using ssh... |
| 08:33.45 | RudyValencia | that's what I'm doing |
| 08:34.02 | RudyValencia | and I also texted using the console of the VM it's running on |
| 08:34.05 | RudyValencia | both do the same thing |
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| 08:36.20 | RudyValencia | I also tried changing the system locale to en_US and it still fails the same way |
| 08:36.48 | ruied | It might be something related to the distro... I have asterisk running with debian 13.11.0 on a VM (proxmox) and it is working.... |
| 08:37.20 | RudyValencia | OK |
| 08:37.23 | ruied | something related to libedit or so... but that, I don't know... |
| 08:38.20 | RudyValencia | I guess I can't use Arch then. |
| 08:38.24 | RudyValencia | eh. |
| 08:38.42 | ruied | are you trying with Arch for the first time with asterisk? (never tryed arch) |
| 08:40.10 | RudyValencia | I had an older version running at one time |
| 08:40.15 | RudyValencia | but that was a few years ago |
| 08:41.15 | ruied | I like debian stable for that... never had a problem with asterisk... |
| 08:42.09 | RudyValencia | how do you use the latest version with Debian anyway? |
| 08:42.30 | RudyValencia | (when they like to hold back quite a bit) |
| 08:42.33 | ruied | had some things with debian that ocassionaly occur when instaling bri cards, but not much o a poblem, minor issues with wrong/missing folders... |
| 08:43.44 | ruied | Debian testing - I use it for desktop environment (latest). For asterisk, I use debian stable. |
| 08:44.45 | RudyValencia | well, I want something with systemd |
| 08:44.53 | RudyValencia | because I'm used to systemd from Arch |
| 08:45.14 | ruied | debian stable already has it |
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| 08:47.44 | RudyValencia | ah |
| 08:47.53 | RudyValencia | well I'll get it |
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| 09:30.28 | ruied | Asterisk doesn't bundle pjsip does it? I seemed that once I have downloades asterisk that bundled the pjsip... is it the certified asterisk? |
| 09:31.56 | dadrc | Asterisk 12+ ships with chan_pjsip, if that's what you're asking |
| 09:32.47 | ruied | ah |
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| 10:12.19 | RudyValencia | ruied: how do I install the latest version of Asterisk on Debian? |
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| 10:15.03 | ruied | wget <asterisk-url> |
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| 10:20.58 | RudyValencia | so that's a source build? |
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| 10:30.59 | Samot | And most likely not current. |
| 10:31.11 | Samot | No, that's a package build. |
| 10:33.52 | Samot | RudyValencia: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz |
| 10:34.01 | RudyValencia | OK |
| 10:35.00 | Samot | https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source |
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| 11:26.18 | slima | I have a problem, I have in dialplan: exten => h,n,System(/etc/asterisk/SCRIPTS/fax2mail.sh "${CDR(SRC)}" "${CDR(DST):4}" "${FAXEMAIL}" "${FAXFILE}") I see in logs: Executing [h@incoming:17] System("IAX2/e1-2362", "/etc/asterisk/SCRIPTS/fax2mail.sh [...] But script not exec, When i do su -s /bin/bash asterisk and copy/paste line from logs, script exec, so it is not permissions problem. Any idea? |
| 11:29.35 | ruied | I have been using extensions.conf up to now. Now I'm looking for a possible change to ael but then I saw also lua. What are the benefits of using lua vs ael? I would like to go to bet on one of them, but do not know where to bet... |
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| 12:14.17 | RudyValencia | So robert_ and I are working on our new PBX and while we get great audio on our calls, the music on hold sounds like it only wants to use 8KHz, is there a setting I can change to increase the quality? |
| 12:15.20 | Samot | What sound files are you using? |
| 12:15.52 | RudyValencia | MP3s at 320Kbps, 44.1 or 48KHz |
| 12:22.17 | RudyValencia | Samot: ^ |
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| 12:23.23 | Samot | Sounds like they are being transcoded to a lower compression. |
| 12:23.43 | robert_ | how do we stop that? |
| 12:28.27 | Samot | Try wav files. |
| 12:28.32 | Samot | Or slin. |
| 12:42.59 | RudyValencia | We had it before to where it sounded great, but we can't remember how |
| 12:43.04 | RudyValencia | Samot: ^ |
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| 12:50.16 | Samot | I'm not sure. I don't use Mp3s for MoH |
| 12:50.35 | robert_ | slin sounds awful |
| 12:50.40 | robert_ | it's loud and distorted |
| 12:50.48 | RudyValencia | I think I may have it |
| 12:50.53 | RudyValencia | I just need to lower the volume |
| 12:50.53 | Samot | I just use wav files. |
| 12:51.07 | robert_ | also |
| 12:51.30 | robert_ | on our test hold-line, it wants to use PCMu 8khz |
| 12:51.52 | RudyValencia | Hold on robert_ |
| 12:52.12 | robert_ | that was just so I could be sure |
| 12:52.17 | RudyValencia | I'm fixing the distortion |
| 12:53.19 | RudyValencia | ugh |
| 12:53.24 | RudyValencia | something went wrong there. |
| 12:53.35 | robert_ | what the hell |
| 12:58.07 | RudyValencia | robert_: I think it might be the transcoding |
| 12:59.29 | robert_ | hmm |
| 12:59.57 | RudyValencia | Also, Speex is designed to encode speech. |
| 13:00.50 | RudyValencia | G722 16KHz is going to be the best format. |
| 13:01.02 | ruied | can I set several endpoints to use the same auth in pjsip ?? |
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| 13:35.34 | ruied | I'm starting to migrate to pjsip. I'm configuring endpoints, but have realized that I need a lot more work (more lines per extention(endpoint) in using pjsip.conf than sip.con file. For now is there advantages in using pjsip? for now, I see a pjsip.conf file that will become a lot heavier than sip.conf. |
| 13:38.19 | ruied | do not know if it worth to migrate now to pjsip or if I shall wait. Any opinions? |
| 13:42.12 | file | as of Asterisk 13 we (Digium) focus on chan_pjsip and chan_sip has become community supported, and yes - using pjsip.conf directly results in larger stuff (unless you use templates which reduces it) but gives more control, there's also a wizard option which simplifies it |
| 13:42.13 | file | http://blogs.asterisk.org/2016/05/04/pjsip-configuration-wizard/ |
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| 13:52.07 | ruben23 | hi there guys |
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| 13:59.37 | ruied | What I'm liking in pjsip is the max_contacts |
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| 14:30.50 | Syrex_ | hi everyone |
| 14:35.13 | [TK]D-Fender | ruied, pjsip allows multiple contacts to register simultaneously, to have multiple fixed IP's for auth (good for providers witha range they send from), etc |
| 14:37.27 | ruied | yes |
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| 14:47.57 | Syrex_ | My SIP provider has a distributed environment and requires me to send outbound calls via IP x whilst incoming calls originate from range y. How can I modify my SIP trunk to associate any incoming calls from range y with my trunk? |
| 14:49.41 | Syrex_ | [ECN] host=41.221.2.94 canreinvite=yes username=********* secret=********* sendrpid=yes type=peer insecure=port,invite context=custom-fix-wrong-prefix |
| 14:53.25 | Syrex_ | Can I define a subnet (eg 41.221.2.0/24) so that any inbound call from those IPs is associated with my trunk definition? |
| 15:03.38 | Samot | For each IP they send from, you need to have a PEER for it. |
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| 15:12.04 | voipmonk | do they have a domain Syrex_ ? |
| 15:12.37 | voipmonk | when you look into the sip headers are they using a domain? |
| 15:12.48 | Syrex_ | voipmonk: Perhaps, I assume you are referring to me enabling SIP debug and reviewing an invite? Let me have a look... |
| 15:13.11 | voipmonk | yes but in the meantime if time is of the essence , please continue with Samot's suggestion |
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| 15:20.43 | Syrex_ | voipmonk: Wouldn't appear so but they do present a consistent User-Agent: ECN Megaswitch |
| 15:21.47 | Syrex_ | The 'via', 'contact' and 'call-id' values include IPs out of a consistent subnet (would match 41.221.0.*) |
| 15:26.55 | [TK]D-Fender | Syrex_, you can't do this with chan_sip. Some of these you can do with chan_pjsip |
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| 15:30.06 | Syrex_ | D-Fender: Thanks, any way you could think of to associate calls with a certain peer? I currently have them coming in anonymously, perhaps I can change the associated peer via a context? |
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| 15:31.59 | [TK]D-Fender | Syrex_, "change the associated peer via a context" doesn't eman anything |
| 15:32.07 | [TK]D-Fender | mean* |
| 15:32.18 | [TK]D-Fender | You either MATCH, or you don't |
| 15:32.37 | [TK]D-Fender | Ther is not "change" and no "via a context" |
| 15:32.37 | Syrex_ | ie: /etc/asterisk/sip_general_custom.conf has context=custom-from-sip-external /etc/asterisk/extensions_custom.conf has [custom-from-sip-external] exten => _.,n,Set(SIP_CODEC=g729) exten => _.,n,Goto(custom-fix-wrong-prefix,${EXTEN},1) is there a way I can associate calls later via a dial plan? |
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| 15:33.13 | [TK]D-Fender | Ditch this word "associate" |
| 15:33.20 | [TK]D-Fender | <[TK]D-Fender> You either MATCH, or you don't <- |
| 15:33.32 | [TK]D-Fender | if you do NOT match a peer then it is an UN-AUTHED CALL |
| 15:33.34 | [TK]D-Fender | The end. |
| 15:33.45 | [TK]D-Fender | What you DO with those afterwards is up to you. |
| 15:34.07 | [TK]D-Fender | There is no "change association". The channel now exists. If you go on to process it then that's on you |
| 15:34.47 | [TK]D-Fender | But your CDR, peer=use count, etc will not be changing |
| 15:35.14 | Syrex_ | [TK]D-Fender: Thanks, I'll work around it for now... Is anyone here using chan_pjsip in production? ie: Is it considered stable? |
| 15:37.41 | [TK]D-Fender | It's in an LTS brach for more than a year now... |
| 15:37.48 | [TK]D-Fender | And passed the devel it was in. |
| 15:37.55 | [TK]D-Fender | This shouldn't be a question |
| 15:38.09 | [TK]D-Fender | And yes obviously people are using it in production. |
| 15:38.19 | [TK]D-Fender | chan_sip is in community-maint-only status now |
| 15:39.02 | Syrex_ | Thank you, I'll build the code... |
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| 19:32.14 | jeffspeff | is there anything special about the cert generated by ast_tls_cert.sh or will buying any ssl cert be fine? looking at this cheap cert https://www.ssls.com/ssl-certificates/comodo-positivessl |
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| 20:52.55 | drmessano | Buying a cert is fine |
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| 21:36.44 | freebs | oh that's nice freepbx removed ur ban |
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