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00:14.38 | Prelude2004c | hey guys.. good day... some question.. i got opus going on my phone with asterisk but something is not right.. i setup codecs.conf to only have opus8 and its sample rate of 8khz and maxbitrate of 6000 , yet when on my phone i use full or wideband it still works... does it not force the system to use 8khz and respect the bw ? i mean the idea is to save bw and it doesnt' matter what i chooce its always the same bit rate |
00:14.40 | Prelude2004c | any ideas? |
00:15.44 | shido6 | limit the codecs the phone accepts |
00:15.54 | Prelude2004c | i did.. i only allow=opus |
00:16.13 | Prelude2004c | oh yes.. i did.. but still my bw stats still how 60kbs |
00:16.17 | shido6 | show the call - the list of codecs are listed in the sip trace |
00:16.26 | Prelude2004c | down is 60kbs and up is 60kbs |
00:18.13 | Prelude2004c | i select only 8khz and still in logs it says " codec_opus.c:203 lintoopus_frameout: [Encoder #98 (48000)] 960 samples, 1920 byte " |
00:18.32 | shido6 | http://listening-tests.hydrogenaud.io/igorc/results.html |
00:18.39 | shido6 | https://www.opus-codec.org/comparison/ |
00:20.44 | Prelude2004c | yes i have seen all that |
00:21.36 | shido6 | show the call , please :) |
00:21.58 | Prelude2004c | i need to be on the low size of opus where it auto adjusts to g729 or better quality when needed but if bw is good and can pickup then it uses better quality |
00:22.23 | Prelude2004c | trying to get to all the logs :( .. its a lot , i set verbose at 255 and debug at 255 |
00:22.32 | shido6 | and your codecs.conf |
00:22.51 | shido6 | no not ALL the logs - just the sip call would be a nice start [pastebin.com] |
00:23.26 | shido6 | heh |
00:24.43 | Prelude2004c | http://pastebin.com/raw/8YsbW3S8 |
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00:52.36 | Prelude2004c | so any ideas? |
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01:43.58 | [[thufir]] | inbound calls show up in the Asterisk CLI: http://pastebin.com/QitSuwjZ but there's a problem with SIP retransmits. Basically, start with turning on sip debug? |
01:45.23 | [[thufir]] | I can dial out, for what it's worth. does that exclude NAT and firwall issues? |
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02:46.11 | snadge | "Is it possible to add H.264 for video on that trunk?" .. heh |
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03:02.58 | boris_rrh | hello, I am trying to enable ARI at my fresh asterisk installation |
03:03.24 | boris_rrh | trying curl -v -u asterisk:asterisk -X POST "http://localhost:8088/ari/channels" |
03:04.11 | boris_rrh | and get 404 with "message": "Resource not found" |
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03:04.42 | boris_rrh | when I try endpoints instead of channels everything works fine |
03:04.59 | boris_rrh | I also get events over websocket |
03:05.05 | boris_rrh | what might be wrong? |
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03:26.15 | [TK]D-Fender | [[thufir]], yes you should be looking with sip debug enabled |
03:27.24 | [[thufir]] | [TK]D-Fender: thanks. can I eliminate NAT and firewall issues because I'm able to dial out? (using telnyx) |
03:27.33 | [TK]D-Fender | no |
03:31.07 | [[thufir]] | ok |
03:31.26 | [[thufir]] | sipscanner.voicefraud.com can detect 5060 on my IP address. |
03:35.18 | [[thufir]] | [TK]D-Fender: what positive tests can I run to check NAT and firewall? |
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03:35.46 | [TK]D-Fender | Look at the call |
03:35.50 | mnathani | whats the best codec for use with voip calls on a PC |
03:36.05 | [TK]D-Fender | mnathani, the biggest one the other side supports |
03:36.09 | mnathani | and which provider / voip software supports such codec |
03:36.32 | mnathani | are you familiar with voip.ms offerings |
03:36.33 | [TK]D-Fender | Basically the PST runs on G.711. So that's where you hsould probably b |
03:36.35 | [TK]D-Fender | be |
03:36.45 | [TK]D-Fender | PSTN* |
03:37.03 | [TK]D-Fender | so that'd be ulaw |
03:37.49 | mnathani | thanks |
03:38.22 | mnathani | your nick makes me curious - are you affiliated with any software release group? |
03:38.51 | [TK]D-Fender | noe |
03:38.52 | [TK]D-Fender | nope |
03:45.10 | drmessano | Lies |
03:45.34 | drmessano | [TK]D-Fender wrote the keygen for PKZIP for the Kaypro |
03:46.17 | [TK]D-Fender | is afraid the Osbourne he used to have will reaniamte and hunt him down... |
03:47.24 | drmessano | Also, he cracked the app that turns your iPad into a bathroom scale |
03:47.30 | drmessano | Would you like a copy? |
03:55.05 | [[thufir]] | I can't seem to unregister telnyx: http://pastebin.com/LxvPFd3J because it doesn't show in "sip show peers". How do remove it from the results of "sip show registry"? |
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04:01.22 | [TK]D-Fender | Well since it's been a half hour wirthout seeing the actual call as requested .. |
04:01.24 | [TK]D-Fender | I'm out. |
04:01.27 | [TK]D-Fender | heads off |
04:18.15 | [[thufir]] | [TK]D-Fender: wait one sec. |
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04:19.26 | [[thufir]] | [TK]D-Fender: failed call http://pastebin.com/upwU3G8X |
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07:16.14 | [[thufir]] | inbound calls show up in the Asterisk CLI: http://pastebin.com/QitSuwjZ but there's a problem with SIP retransmits. Sip debug on shows http://pastebin.com/upwU3G8X for a failed call to the DID from outside. Why? |
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09:30.52 | confused_recruit | Hello - is there someone I could have a quick chat with (community member is fine) re: Asterisk and the coding behind it? |
09:32.45 | confused_recruit | I'm a recruiter - I work for a company which uses the ASterisk system to underpin their phone systems - they have asked me to recruit a Release Manager to managage their development process. Which is fine - although they can't answer me what language(s) Asterisk is written in. It's Linux based for them, so I'm guessing PHP / Python or something similar? Can someone confirm this for me? Many thanks! :) |
09:33.37 | WIMPy | You can't do telephony in a scripting language. It's all C. |
09:34.33 | confused_recruit | C - sweet, that works for me :) |
09:35.55 | confused_recruit | They were going on about projects with AngularJS and CSS3 and I was trying to explain to them that's the front end, not the back end - but the HR lady didn't understand and just kept giving me words like JavaScript etc - so I thought perhaps it has some PHP/Python type code behind it. But IIRC both are based on C anyway and C is a fine answer, thank you for your time :) |
09:37.53 | WIMPy | Asterisk dies not come with a frontend. So if it's about something like that, you have to find out what it it they're using. |
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10:02.25 | [[thufir]] | how do I know or see, from the dialplan, whether a number is allocated for incoming calls? I have: http://pastebin.com/ZP1E6Nd4 apparently I'm returning a 404 not found to the sip provider for my DID. |
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10:59.03 | Samot | [[thurfir]] Change your inbound to match the DID as it's sent to you. |
10:59.26 | Samot | And do you have your trunk setup properly? |
11:01.38 | [[thufir]] | Samot: one sec. |
11:04.22 | [[thufir]] | Samot: right now I'm using '_x.' => Â Â Â Â Â 1. NoOp() but have tried many patterns. Outbound dialing works. AFAIK the trunk is correct. I'm reading about NAT and am leaning towards thinking that it's NAT at the root of this. How do I run a SIP trace? |
11:04.44 | Samot | What's the context set to? |
11:04.52 | Samot | In the trunk settings? |
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11:05.53 | [[thufir]] | Samot: inbound, line 22 |
11:05.55 | [[thufir]] | http://pastebin.com/M3Nn41Vw |
11:06.38 | Samot | asterisk -rvvvvvvvvvvvv |
11:06.46 | Samot | sip set debug on |
11:06.52 | Samot | Make the call, watch for the output |
11:06.58 | Samot | ~pb the results. |
11:07.04 | Samot | ~pb |
11:07.05 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
11:08.00 | [[thufir]] | that's alota alota v's. ok. |
11:08.58 | Samot | No pms, please. |
11:09.02 | Samot | Just paste it here. |
11:09.13 | [[thufir]] | it shows my public ip, though. |
11:09.18 | Samot | So? |
11:09.35 | Samot | It's not like it's a secret to the world what your public IP is. |
11:09.41 | Samot | You send it to everyone you talk to. |
11:09.49 | [[thufir]] | ok |
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11:12.06 | [[thufir]] | Samot: calling the DID: http://pastebin.com/UfEyVUB7 |
11:13.07 | Samot | Nope. |
11:13.14 | Samot | There is zero verbosity in there |
11:13.19 | Samot | core set verbose 10 |
11:13.27 | Samot | That was the point of asterisk -rvvvvvvvvvvvv |
11:13.37 | Samot | To put you in the proper verbosity for a debug. |
11:14.07 | Samot | Do it again, as told. Then pastebin the results. |
11:14.47 | [[thufir]] | I did all those v's. asterisk -rvvvvvvvvvvvv I checked just now. I'll do core set verbose 10 also. |
11:15.13 | Samot | And call the DID from the outside |
11:15.22 | Samot | Not from your PBX. |
11:15.28 | Samot | <PROTECTED> |
11:17.15 | [[thufir]] | Samot: http://pastebin.com/LwN4aBWg |
11:18.10 | Samot | How are you making the test call? |
11:18.28 | [[thufir]] | skype |
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11:18.48 | Samot | To the DID? |
11:19.08 | [[thufir]] | yes. skype dial out to a phone number. it rings. |
11:19.23 | Samot | The call isnt hitting the system. |
11:19.37 | [[thufir]] | telnyx say that I'm returning back a 404. |
11:20.05 | Samot | Then fix your firewall |
11:20.20 | Samot | Because the call is not hitting the pbx |
11:20.37 | [[thufir]] | what's weird is that I can dial out. how do you know that it's a firewall? you are inferring this because it doesn't hit the pbx? |
11:21.45 | Samot | Yes. |
11:22.05 | [[thufir]] | is there some positive test I can do? sip trace? |
11:22.12 | Samot | Also, i doubt telnyx is behind NAT |
11:22.23 | Samot | You just did one |
11:23.32 | [[thufir]] | ahh, ok. thanks. |
11:23.45 | Samot | And dialing out is outbound traffic. I doubt you have outbound rules in your firewall. |
11:24.24 | [[thufir]] | dmz? |
11:24.29 | Samot | Make sure you have it set up with the proper nat rules |
11:24.31 | Samot | No. |
11:26.53 | [[thufir]] | which NAT rules? listen on port 5060 (?) and forward to ip address x.x.x.x.8 for Asterisk? |
11:27.22 | Samot | 5060, 10000-20000 on UDP for them. |
11:29.39 | [[thufir]] | 5060 is tcp/udp? 10000-20000 is udp only? ok. |
11:30.34 | Samot | It's all UDP. |
11:30.50 | [[thufir]] | ok. no tcp then. I'll fix that. thanks. |
11:35.50 | [sID] | Samot: Hi, I wrote and posted about a2 me that is From the UI |
11:36.18 | Samot | OK. |
11:37.01 | [sID] | Only probably lie because I do not see anything like that :) |
11:43.13 | Samot | Are you saying they replied? |
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12:11.07 | [[thufir]] | I'm looking at http://the-asterisk-book.com/1.6/einleitung-regex.html but don't see how match a number starting with a + and then NANP? |
12:12.02 | WIMPy | Just add a +1 to the pattern. |
12:12.24 | WIMPy | Or are you really using REGEX? |
12:13.24 | [[thufir]] | change this: '_1NXXNXXXXXX' => 1. NoOp() to '_+1NXXNXXXXXX' => 1. NoOp() ? easy enough. Not trying to use regex. |
12:13.48 | WIMPy | Yes, just that. |
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12:46.10 | [[thufir]] | telnyx tells me that I'm sending many, many 200 OK's but that otherwise it looks fine for incoming calls to the DID. It rings, I answer, then hangup. I forwarded 5060 UDP and 10000-20000UDP to the internal IP address for the Asterisk box. |
12:46.40 | [[thufir]] | The messages are that a critical packet wasn't received, reference https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. |
12:47.56 | TandyUK | I get that a LOT when using TCP/TLS for handsets |
12:48.09 | TandyUK | if youre using udp exclusively, not sure whats wrong |
12:48.24 | [TK]D-Fender | that makes no sense for TCP |
12:48.39 | [TK]D-Fender | TCP takes care of retransmission |
12:48.43 | TandyUK | it does, when the tcp session has dropped |
12:48.59 | [TK]D-Fender | Normally shouldn't happen |
12:49.13 | TandyUK | it still tells asterisk it fails, so once asterisk tries again, but now theres no tcp session to send the packet down, and a tthat point asterisk realises the phone is unavailable |
12:49.22 | [TK]D-Fender | but I suppose if your router gets flooded it may rank things as idle and allowed to flush |
12:49.31 | [TK]D-Fender | Better routers let you set rules for that though |
12:49.40 | TandyUK | I had my voip platform provider modify the code to make it a notice not a warning as it was just flooding our logs |
12:50.17 | TandyUK | its not the router lol, its when the connection (from the other end) has been interrupted, eg customer broadband went down, or mobile lost 3g data, etc |
12:50.42 | TandyUK | possible customer end routers not being conservative enoug hwith states, but thats all out of our control |
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12:57.41 | [[thufir]] | here is the sip trace for an inbound call. outbound calls are fine. again, I setup port forwarding, but "guess" that it's NAT and/or firewall problem. http://pastebin.com/P76z3zzZ |
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13:00.10 | [TK]D-Fender | <--- Transmitting (NAT) to 192.76.120.10:5060 ---> |
13:00.24 | [TK]D-Fender | Virtuall no provider is behind NAT. Fix your peer |
13:00.42 | [TK]D-Fender | Contact: <sip:+16044494243@192.168.1.8:5060> |
13:01.09 | [TK]D-Fender | And you seem to have been sending the wrong IP for contact |
13:01.33 | [TK]D-Fender | So it's looking like you messed up your otehr basics in [general] |
13:01.58 | [[thufir]] | pardon: http://pastebin.com/HjeG086P |
13:03.26 | [[thufir]] | the provider isn't behind NAT, but I am. the wrong IP for contact for ____? |
13:03.54 | [TK]D-Fender | FIX YOUR TRUNK |
13:04.03 | [TK]D-Fender | <[TK]D-Fender> Virtuall no provider is behind NAT. Fix your peer |
13:04.55 | [TK]D-Fender | And you have still screwed up your general NAT settings. You are transmitting your PRIVATE IP as the Contact: and not your PUBLIC ONE. |
13:05.08 | [TK]D-Fender | 225-237 |
13:05.09 | [[thufir]] | ah. |
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13:09.44 | [[thufir]] | nat=never ? |
13:10.04 | [TK]D-Fender | NO |
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13:30.17 | [[thufir]] | I set externip in the peer to transmit my public IP as the contact. correct? nat=force_rport,comedia because I'm using NAT. correct? |
13:30.40 | [TK]D-Fender | externip is NOT a peer-level parameter |
13:30.45 | [TK]D-Fender | You're doing it wrong |
13:31.04 | [TK]D-Fender | <[TK]D-Fender> <[TK]D-Fender> Virtuall no provider is behind NAT. Fix your peer |
13:34.13 | sharad | I have two asterisk boxes one is using as voip gateway another is using as media server. when i make a call through my SIP phone which is registered on media server, using the voip gateway server to the PSTN network and PSTN guy hold the call then i get default MOH from my VOIP gateway How is it possible to get this MOH from the media it self |
13:34.16 | [[thufir]] | externip goes under [general] then. |
13:49.34 | [[thufir]] | [TK]D-Fender: thanks, it works. I don't understand, why, though. what does [general] do? by putting externip in general, that...sends my external IP? but why can't I put it in [telnyx]? what's the distinction? |
13:50.05 | Samot | Because it's not a peer level setting. |
13:50.05 | [TK]D-Fender | because you don't get so specify the advertieed address PER PERR |
13:50.55 | [[thufir]] | ok |
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13:51.10 | [TK]D-Fender | * will send either the interface's IP, or what you set as external |
13:51.36 | [TK]D-Fender | based on your localnet |
13:51.44 | [TK]D-Fender | which is ALSO not a peer setting |
13:51.49 | [TK]D-Fender | read the SAMPLE CONFIG |
13:51.53 | [TK]D-Fender | The local matters |
13:51.58 | [TK]D-Fender | location* |
13:53.49 | file | wobbles |
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14:40.40 | ronybeck | Hi all. I have a problem with a SIP trunk on Asterisk. When Asterisk sends an invite, the SIP Trunk provider sends back a 401 unauthorized (authentication challenge). Asterisk should now answer the authentication challenge. But it doesnt. Who do I fix that? |
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15:00.47 | mirela666 | ronybeck, did you try setting trunk with insecure=invite ? |
15:07.46 | mirela666 | maybe even you should remove the insecure section |
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15:10.28 | ronybeck | Milenco: I did |
15:10.31 | [TK]D-Fender | Not relevant |
15:10.32 | ronybeck | I just tried removing it |
15:10.38 | ronybeck | Makes no difference |
15:12.07 | ronybeck | It is like Asterisk doesn't see the 401 unauthorized message |
15:12.26 | ronybeck | Even though it prints it in the asterisk console when I enable sip debugging |
15:13.16 | [TK]D-Fender | You should probably be showing us the debug & config masking only the secret at this point.... |
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15:15.46 | ronybeck | sure |
15:16.42 | ronybeck | [mdc_trunk_conf-1] |
15:16.42 | ronybeck | ; phonestar - friend |
15:16.42 | ronybeck | type=friend |
15:16.42 | ronybeck | context=from-trunk |
15:16.42 | ronybeck | defaultuser=YYYYYYYYYYY |
15:16.42 | ronybeck | secret=XXXXXXXXXX |
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15:16.45 | ronybeck | host=ps1.voipgateway.org |
15:16.46 | ronybeck | nat=force_rport |
15:16.50 | ronybeck | progressinband=yes |
15:16.52 | ronybeck | qualify=no |
15:16.54 | ronybeck | fromuser=YYYYYYYYYYY |
15:16.56 | ronybeck | That is the trunk configuration |
15:17.06 | mirela666 | use the pastebin ~pastebin |
15:17.11 | ronybeck | Sorry |
15:17.12 | mirela666 | ~pb |
15:17.12 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
15:17.16 | ronybeck | ~pastbin |
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15:22.50 | ronybeck | http://paste.lisp.org/display/319814 <----------------------- here is the TCP dump from the invite |
15:23.01 | [TK]D-Fender | no providers should be behind NAT |
15:23.42 | ronybeck | [TK]D-Fender: are you referring to this: nat=force_rport |
15:23.48 | [TK]D-Fender | yes |
15:23.57 | ronybeck | What does it do? |
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15:24.14 | [TK]D-Fender | set's who to trust for ports & IP's |
15:24.23 | [TK]D-Fender | you should be taking what they tell you at face value |
15:25.09 | ronybeck | Just remove that line then? |
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15:26.09 | [TK]D-Fender | nat=NO |
15:26.27 | [TK]D-Fender | and type=peer |
15:26.49 | ronybeck | But with type=peer I can't receive calls right? |
15:26.57 | [TK]D-Fender | incorrect |
15:27.40 | ronybeck | ok |
15:28.06 | [TK]D-Fender | make the changes and show a new call with * SIP debug (not external) |
15:28.45 | ronybeck | ok |
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15:52.51 | mirela666 | ronybeck, type=friend never use unless you know what you are doing, it will create both user and peer and it can be very insecure if you don't set a password |
15:53.32 | ronybeck | ok |
15:53.35 | ronybeck | Understood |
15:53.44 | ronybeck | How do I call "*"? |
15:53.59 | ronybeck | I know how to do this stuff on Freeswitch. Asterisk is a new world to me. |
15:54.32 | [TK]D-Fender | <ronybeck> How do I call "*"? <- as in? |
15:54.59 | ronybeck | <[TK]D-Fender> make the changes and show a new call with * SIP debug (not external) |
15:55.15 | [TK]D-Fender | Asterisk CLI |
15:55.16 | [TK]D-Fender | sip debug |
15:55.23 | ronybeck | ah |
15:55.26 | ronybeck | understood |
15:55.45 | ronybeck | I thought you meant don-t call external. But you mean don't use an external tool |
15:55.47 | ronybeck | right |
16:01.45 | ronybeck | http://paste.lisp.org/display/319817 <------------ here is the logs |
16:02.50 | ronybeck | That is with the changes suggested so far |
16:03.19 | [TK]D-Fender | That is a pretty old release, you should upgrade |
16:03.32 | [TK]D-Fender | <PROTECTED> |
16:04.04 | ronybeck | It is part of a software suite called MobyDick. This is the latest supported version |
16:05.10 | ronybeck | Otherwise I would have taken the latest version :-( |
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17:23.00 | drmessano | Wow |
17:23.02 | drmessano | https://www.pascom.net/en/pricing/ |
17:28.03 | file | hmm? |
17:28.12 | file | oic |
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20:21.48 | _abc_ | Hello. Can anyone suggest a iax2 phone client for linux, non gnome, non kde? Preferrably a statically compiled one? |
20:21.59 | _abc_ | sflphone seems to be one but no idea what libs it requires. |
20:23.05 | _abc_ | Strangely sflphone is advertised as SIP and something else (its own protocol) but the debian docs claim it is IAX2 compatible too. |
20:27.08 | _abc_ | http://www.voip-info.org/wiki/view/IAXComm is gone gone? |
20:36.47 | _abc_ | Basically I can't find a iax2 client for linux which is included in any distribution besides sflphone. |
20:38.47 | _abc_ | https://sourceforge.net/projects/kiax/ is still around but not included in any distribution as far as I checked |
20:41.21 | _abc_ | Ahh kiax is static compiled. Nice. |
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20:45.32 | _abc_ | Package verified and installed: lies, it is not static. Dll compiled and depends on a lot of distribution specific libs |
20:46.10 | _abc_ | Package name is kiax-2.1-beta1-linux-static.tar.gz |
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22:37.07 | drmessano | IAX Softphone = Uncommon |
22:37.18 | drmessano | IAX Softphone for Linux = Less Common |
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