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| 09:02.13 | freemanls | Hello guys, I have an audio loss in the middle of the call, could it be from NTP not configured on Cisco SIP phones? |
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| 10:31.25 | yun1989 | hello all |
| 10:31.34 | yun1989 | i have one problem in asterisk 13 |
| 10:31.35 | yun1989 | http://pastebin.com/Xe07pLzF |
| 10:31.44 | yun1989 | do you know why it's happen ? |
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| 10:37.54 | WIMPy | Might help if you write what you've got a problem with, so people know if that pastbin might contain something they know about. |
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| 10:52.15 | kchehab | hi |
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| 10:53.04 | Brokennoze | Hi all. Is anyone aware of an issue with Asterisk 11.5 and IAX2? every 50th call or so Asterisk deplays send of the VoiceFullFrame |
| 11:00.52 | kchehab | i am using asterisk with chan_mobile ,i follow all instructions (simple-agent.py) and 1st mobile conected and works perfecly but when i attach another bluetooth device with same mobile brand to pair the mobile is paired but not connected as i can see it as headset in mobile search results and second mobile always recieve invites to connect to hci0 while its paied to hci1 |
| 11:00.55 | kchehab | kindly advise |
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| 12:06.37 | RPerre | Anyone know any TTS software that is free and multi-lang? (I'm aiming Portuguese) |
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| 12:15.11 | Chainsaw | RPerre: It looks like the "mbrola" project has portuguese voices. |
| 12:15.18 | Chainsaw | RPerre: You can use that through the festival engine. |
| 12:16.08 | Chainsaw | RPerre: However, potentially awkward licensing: "This synthesizer is provided for free, for non commercial, non military applications only." |
| 12:16.26 | RPerre | non commercial |
| 12:16.27 | RPerre | crap |
| 12:16.34 | RPerre | thanks for your input Chainsaw |
| 12:16.54 | Chainsaw | RPerre: Any time. |
| 12:20.11 | RPerre | Chainsaw, I've found this: http://espeak.sourceforge.net/ |
| 12:20.52 | Chainsaw | RPerre: That just gives me a 503 Varnish error. |
| 12:21.23 | RPerre | not me :o |
| 12:21.42 | WIMPy | gets a 104 Connection reset by peer |
| 12:21.50 | RPerre | o.O |
| 12:21.51 | RPerre | dafuq |
| 12:22.19 | RPerre | oh, the download page gives me a 404 |
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| 12:48.50 | zamba | on a LAN, which audio codec should be used for the best quality? |
| 12:49.03 | zamba | currently using alaw, but maybe there's an ever better codec? |
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| 12:49.31 | Chainsaw | zamba: G722a is probably the most wideband codec that's still well supported. |
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| 12:49.45 | Chainsaw | zamba: Polycom brands that as "HD Audio". |
| 12:49.53 | Chainsaw | zamba: Or "HD voice", even. Sorry. |
| 12:49.58 | Chainsaw | (Laptops on the mind) |
| 12:51.30 | zamba | Chainsaw: if i don't specify which codec to use, will the client automatically negotiate the best possible codec? |
| 12:51.50 | Chainsaw | zamba: In a perfect world, yes. |
| 12:52.01 | zamba | Chainsaw: our clients are polycom phones |
| 12:52.29 | zamba | we interface with lync for the outside communication.. apart from that all communication is internally on the asterisk server |
| 12:52.40 | Chainsaw | zamba: I've specified allow=!all,ulaw,alaw,g722 for ours. |
| 12:52.49 | Chainsaw | zamba: They are IP670 & IP7000, if that helps. On the UCS firmware train. |
| 12:53.10 | zamba | and then the phone will automatically select the best? |
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| 12:54.11 | zamba | is g711a a better protocol to use than ulaw/alaw, for instance? |
| 12:54.23 | Chainsaw | zamba: G711a -> aLaw. G711u -> uLaw. |
| 12:54.35 | Chainsaw | zamba: It's actually the same thing, just written out differently. |
| 12:55.12 | Chainsaw | zamba: It's worth leaving ulaw & alaw specified for when you want to talk to the plain old telephone system over ISDN, in which case that's the best audio quality you're going to get. |
| 12:55.13 | zamba | and the client will select codec on a call-by-call basis, right? |
| 12:55.19 | Chainsaw | zamba: Correct. |
| 12:55.36 | Chainsaw | zamba: As I said, in an ideal world both ends will select the best codec they have in common. |
| 12:55.48 | Chainsaw | zamba: In the real world... sometimes you have to help things along a little. |
| 12:56.06 | zamba | hehe, ok |
| 12:56.07 | Chainsaw | (So I'd supervise that first dance between your Polycoms) |
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| 13:07.37 | WIMPy | Chainsaw: In practice, probably. But the ISDN has supported G.722 from the very beginning. |
| 13:08.44 | Chainsaw | WIMPy: Not seen it used. |
| 13:10.01 | WIMPy | From what I've read, some CAT-iq bases can make use of it. |
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| 13:28.23 | ThomasKeller | I am using mysql for CDR logging. It seems, the connection to mysql times out after some time |
| 13:28.37 | ThomasKeller | I often see following errors in my log: |
| 13:28.39 | ThomasKeller | res_odbc.c: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away |
| 13:28.39 | ThomasKeller | res_odbc.c: SQL Execute error! Verifying connection to MySQL-asterisk [MySQL-asterisk]... |
| 13:28.39 | ThomasKeller | res_odbc.c: Connection is down attempting to reconnect... |
| 13:28.39 | ThomasKeller | res_odbc.c: Connecting MySQL-asterisk |
| 13:29.22 | ThomasKeller | can somebody advice how to fix this ? |
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| 13:51.56 | yun1989 | hello all |
| 13:52.14 | yun1989 | last day I install the asterisk version 13 |
| 13:53.03 | yun1989 | and I tried to make a call between sipml5 and "microSIP" sofphone |
| 13:53.38 | yun1989 | I followed this link https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 |
| 13:53.49 | yun1989 | but I don't have sucess in this adventure |
| 13:55.18 | yun1989 | the sofphone respond with " Everyone is busy/congested at this time" |
| 13:55.33 | yun1989 | anyone know what is happen ? |
| 13:59.08 | [TK]D-Fender | No. Yo need to actually llook at thee complete call deebug |
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| 14:09.04 | yun1989 | @ [TK]D-Fender http://pastebin.com/D58VCDAv |
| 14:09.21 | yun1989 | It is the complet call debug |
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| 14:10.06 | [TK]D-Fender | "sip set debug on" <- |
| 14:11.26 | yun1989 | http://pastebin.com/ZC42Fzda |
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| 14:12.00 | yun1989 | I don't receive any event in softphone |
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| 14:14.15 | [TK]D-Fender | SIP/2.0 488 Not Acceptable Here |
| 14:14.22 | [TK]D-Fender | because it doeesn't like what * is offeereing |
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| 14:17.15 | yun1989 | @TK]D-Fender it's possible softphone don't support this ? |
| 14:17.52 | [TK]D-Fender | they don't agree. |
| 14:17.56 | [TK]D-Fender | Make them agreee |
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| 14:28.28 | yun1989 | I see this link and currently I have call WebRTC to sofphone but i think don't have audio |
| 14:28.31 | yun1989 | https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/ |
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| 14:34.21 | file | you don't appear to have pjproject installed or ICE enabled |
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| 15:42.41 | SamDaMan | so im running an onsite pbxtra server from fonality (not my choice) I found out that it is running asterisk 1.6.0 .. would like to get it up to at least 1.8, has anybody had any experience with this or have any advice? |
| 15:43.22 | file | just FYI - 1.8 is in security fix only status |
| 15:44.48 | sruffell | jamespass: Did anything ever shake out about the RTP jitter? |
| 15:44.57 | SamDaMan | 1.6.0 has had no support for almost 5 years, I'll take what I can get |
| 15:45.59 | SamDaMan | figured the shortest version jump would better keep compatability |
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| 16:10.28 | y_ateya | How much packet loss can codec g729 handle in asterisk? |
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| 16:24.37 | ChannelZ | How much Cylon do you speak? |
| 16:25.06 | pjensen00 | 11000101 00010101 1101000 1000110101 |
| 16:25.07 | [TK]D-Fender | codec_lpc10.so ;) |
| 16:25.15 | [TK]D-Fender | Domo arigato! |
| 16:25.38 | pjensen00 | Como estas? |
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| 17:04.09 | notze | hey guys i successfully installed a asterisk server, now i need the easiest way to setup 20 users that can call each other. |
| 17:04.32 | notze | is there any tutorial on this, i was trying just to add them to sip.conf, but now they need to be in a context and and |
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| 17:08.10 | Penguin | notze: You won't set up users. You'll set up phones. |
| 17:08.15 | Penguin | ~primer |
| 17:08.15 | infobot | New to asterisk configuration? Check out this primer to get started. http://burner.com/asterisk-primer |
| 17:08.23 | Penguin | ~book |
| 17:08.23 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:08.45 | Penguin | notze: Yes, they will need to be assigned a context. |
| 17:10.48 | notze | Penguin, i assigned the sip phones a context, but now i get to extension '1002' rejected because extension not found in context 'interna |
| 17:11.08 | Penguin | notze: Then you didn't create extension 1002 in that context. |
| 17:11.49 | Penguin | notze: I guess you wrote the context, but did you forget the extension? |
| 17:13.46 | notze | Penguin, i added to extension.conf the [internal] section from the link you gave me. i created 2 sip entries 1001 and 1002. now i actually search howto assing them to the context |
| 17:14.11 | Penguin | 1001 and 1002 are not very good names for phones. |
| 17:14.28 | Penguin | But you still have to create the extensions within the internal context. |
| 17:15.05 | notze | Penguin, i try exactly this:) i need 20 phones so they need to be numbers |
| 17:15.11 | Penguin | no |
| 17:15.23 | Penguin | They should be something unique to each piece of hardware. |
| 17:15.23 | Penguin | ~devices |
| 17:15.23 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 17:16.13 | Penguin | We always use the MAC address. I recommend you do the same. |
| 17:17.37 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:21.14 | Penguin | notze: It simplifies administration to have a good plan going in. |
| 17:23.28 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 17:24.31 | notze | Penguin, basically its working now. |
| 17:24.34 | notze | still i have no voice |
| 17:27.06 | notze | now its getting bad |
| 17:27.11 | notze | [Mar 17 17:25:42] WARNING[27350]: chan_sip.c:4204 retrans_pkt: Hanging up call QmQk2WOeJdA4fdQ.jjcZ2YHQXHljA1rn - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).[Mar 17 17:26:24] WARNING[27350]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission |
| 17:27.40 | notze | i think it can be the reasong that this server has a virtual ip 10.0.0.x which is natted to the outgoing.. its an openstack setup |
| 17:28.18 | notze | Penguin, sadly your link ends here :D |
| 17:31.55 | *** join/#asterisk danjenkins_ (anonymous@nat/digium/x-jmixieygswkdtjlo) |
| 17:34.05 | Penguin | notze: You're saying The Book doesn't address NAT settings? |
| 17:34.13 | Penguin | I find that extremely hard to believe. |
| 17:34.29 | notze | http://burner.com/asterisk-primer/firewalls-nat/ |
| 17:34.34 | notze | The part of the guide is still under construction but will be available soon. |
| 17:35.06 | notze | Penguin, this is what i meant :D |
| 17:35.46 | *** join/#asterisk areski (~areski@80.174.128.5.dyn.user.ono.com) |
| 17:36.01 | Penguin | notze: You should probably consult the book, which I also showed you. |
| 17:36.13 | notze | Penguin, doing it right now :D |
| 17:38.44 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 17:40.44 | Penguin | ~sipnat |
| 17:40.44 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 17:40.59 | Penguin | This could use some updating, but it is generally accurate. |
| 17:45.45 | *** join/#asterisk nny (~Scott@cpe-174-107-209-219.sc.res.rr.com) |
| 17:46.11 | nny | can someone poke my addled brain and remind me which variable shows the SIP peer who initiated the channel? |
| 17:46.33 | nny | guess I can use channel and strip the id off |
| 17:46.58 | notze | Penguin, got it working :DDD JUHUHUHUHUHUHU |
| 17:47.59 | notze | Penguin, thx |
| 17:48.04 | nny | oh duh, SIPPEER |
| 17:48.05 | nny | kill me |
| 17:48.51 | pjensen00 | Is there any way to grab all of a channel's variables? |
| 17:49.00 | nny | core show channel |
| 17:49.12 | [TK]D-Fender | dumpchan |
| 17:49.13 | Penguin | DumpChan() |
| 17:49.17 | nny | ooh or that |
| 17:49.25 | pjensen00 | oh duh! Thanks! |
| 17:49.26 | nny | does core show channel show all as well? |
| 17:49.31 | Penguin | no |
| 17:49.31 | pjensen00 | I don't think so |
| 17:49.35 | nny | ahh |
| 17:49.39 | Penguin | I don't think it shows any. |
| 17:50.05 | pjensen00 | Also, I get annoyed when core show channel truncates the channel name. If you do the "concise" flag you get a csv and it's not truncated |
| 17:50.08 | nny | it does but I think only contrext based ones |
| 17:50.17 | Penguin | oh |
| 17:51.07 | nny | and CDR variables |
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| 20:24.49 | jeev | any suggestions how to best debug iax2 calls? i have iax2(C) to iax2(B), to iax2(A) to dahdi having break up issues but iax2(B) to iax2(A) to dahdi is fine.. i don't see anything up at C though |
| 20:25.49 | *** join/#asterisk lorsungcu_ (uid65806@gateway/web/irccloud.com/x-ewitjusbltebhfnq) |
| 20:26.05 | lorsungcu_ | anyone know if the DPMA requires dnsmasq? |
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| 20:32.01 | newtonr | sgriepentrog ^ |
| 20:33.20 | newtonr | jeev, I'd start with packet captures looking at jitter, latency, loss, etc. Then you might go on to looking at the logs for each system. |
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| 20:39.56 | sgriepentrog | lorsungcu_: do you mean to ask if DPMA requires active dns resolution to be functioning, or if it has a dependency specifically on that dns caching solution? |
| 20:42.45 | lorsungcu_ | sgriepentrog: I should preface this with this being a freepbx system, but can't find any documentation regarding this. |
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| 20:43.25 | lorsungcu_ | sgriepentrog: in regards to your question, the former, i think; I've got a single system with DPMA, and it has a ton of reverse lookup requests. Only thing different from anything else is that it's using DPMA and for whatever reason, dnsmasq is running. |
| 20:45.03 | sgriepentrog | That's interesting. Can you detail the reverse lookups? Is it looking up the system's ip, phone's ip, or something on internet? |
| 20:47.50 | sgriepentrog | BTW, dnsmasq is installed by default on (at least recent) freepbx distro. |
| 20:49.04 | sgriepentrog | I'm testing it now and not seeing any traffic to/from dnsmasq. |
| 20:49.08 | jeev | newtonr, is seeing LAGRQ a possible issue ? |
| 20:49.35 | sgriepentrog | How & on what port are you seeing lookup requests? |
| 20:51.56 | lorsungcu_ | sgriepentrog: sorry, one second. |
| 20:52.48 | lorsungcu_ | sgriepentrog: dnsmasq is installed, but not running on latest distro |
| 20:53.03 | lorsungcu_ | on my DPMA install, it's running and listening |
| 20:53.26 | lorsungcu_ | I'm not entirely sure what it's looking up |
| 20:53.36 | lorsungcu_ | I can send you a capture, though |
| 20:53.52 | lorsungcu_ | mind if I PM? |
| 20:53.53 | sgriepentrog | yes please. |
| 20:53.56 | sgriepentrog | sure |
| 20:54.16 | lorsungcu_ | thanks |
| 20:58.02 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:00.03 | jeev | newtonr, i looked at the other end of the call and jitter is at 20000ms .. |
| 21:00.27 | jeev | i dont even know how |
| 21:28.48 | *** join/#asterisk alexw (~textual@unaffiliated/alexw) |
| 21:29.32 | *** join/#asterisk notze (~georg@ip1f126505.dynamic.kabel-deutschland.de) |
| 21:29.54 | notze | whats the best sip phones for asterisk |
| 21:30.47 | robmal | Zoiper! |
| 21:32.12 | [TK]D-Fender | lol. |
| 21:32.15 | [TK]D-Fender | Polycom > All |
| 21:33.09 | robmal | [TK]D-Fender: About that, is there a way to change the sort order of the contact directory? |
| 21:35.08 | [TK]D-Fender | Yes, there is even a magical index field RIGHT THERE |
| 21:35.10 | Penguin | It won't arrange by penis. |
| 21:35.46 | robmal | You mean speed_index, but that's not the same. |
| 21:36.01 | [TK]D-Fender | functionally it is |
| 21:36.08 | pjensen00 | First in, first out. Heyooo |
| 21:36.09 | robmal | I want speed index to be arranged by the user, but the directory to be arranged by extension, not the last name. |
| 21:36.59 | robmal | (because people with last names beginning with 'z' are never important, sorry) |
| 21:37.50 | robmal | s/'z'/'a'/ |
| 21:37.57 | robmal | Typo ;-) |
| 21:41.41 | robmal | [TK]D-Fender: Also, what's the difference between BLF and buddy watch? I'm new to this contact directory world. |
| 21:42.14 | robmal | Except 'BLF can blink the light green, and buddy watch can't' |
| 21:42.44 | [TK]D-Fender | buddy watch = presence = blf |
| 21:43.06 | robmal | But buddy watch can't blink the light green afair. |
| 21:43.51 | [TK]D-Fender | it does |
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| 21:44.51 | robmal | Good to know. |
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| 22:46.28 | pjensen00 | is it possible to execute dialplan applications through the ARI? |
| 22:46.40 | pjensen00 | I see the option to execute functions, but I'm missing applications. |
| 22:46.47 | file | it is not |
| 22:46.56 | pjensen00 | ah ok |
| 22:49.04 | pjensen00 | what's the difference between functions and applications anyways? I never thought to ask the question until recently. |
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| 22:50.23 | newtonr | pjensen00, https://wiki.asterisk.org/wiki/display/AST/Types+of+Asterisk+Modules |
| 22:50.40 | pjensen00 | newtonr: thanks |
| 22:50.55 | newtonr | pjensen00, np! |
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| 23:22.55 | pjensen00 | I'm assuming the reason why applications aren't allowed is because ARI is supposed to give you the tools to do applications on your own? That's 100% cool, though is there a way to replicate DumpChan then? |
| 23:23.12 | pjensen00 | it's not mission critical, but it's something I'd like in my toolbox for troubleshooting |
| 23:23.16 | *** part/#asterisk kharwell (kharwell@nat/digium/x-igrrhkztzmdlmaoe) |
| 23:39.15 | newtonr | pjensen00, I'm not sure, you might want to E-mail the app-dev list. |
| 23:39.18 | newtonr | I've got to run! |
| 23:39.23 | pjensen00 | latah |
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