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| 00:31.56 | freetown2 | good morning all. I had a working asterisk 1.2 that I was forced to upgrade to 1.8. Dialplans was updated to new syntax and works but I seem to have problems now with the sip configuration. Under 1.2, I had insecure=very, canreinvite=yes for the internal Cisco callmanager peer and nat=no for the cisco and provider peers. Incoming calls get through okay but outgoing calls would connect but there would be no sound and the call will drop s |
| 00:31.57 | freetown2 | hortly after the call was answered. What has changed from asterisk 1.2 to 1.8 in the sip department? |
| 00:32.44 | cusco | insecure and nat changed |
| 00:32.47 | cusco | see the wiki |
| 00:33.00 | cusco | something like force_rport,commedia |
| 00:33.03 | cusco | or something like that |
| 00:33.07 | WIMPy | And 1.8 is quite out of date still. |
| 00:33.25 | cusco | is it no longer supported (lts) ? |
| 00:33.28 | WIMPy | Yes, and port,invite for the other one. |
| 00:33.43 | WIMPy | But Asterisk should tell you when it loads the settings. |
| 00:37.12 | WIMPy | Yes, but you still miss the advantages of two or three years. |
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| 00:40.32 | freetown2 | I suppose I can try bugging the Ubuntu Dev team to upgrade their packages to asterisk 11. |
| 00:41.51 | WIMPy | I wouldn't install something as configurate at build time as Asterisk as some pre-built package. |
| 00:43.21 | freetown2 | heh, the guide as a section for OpenSolaris...I wonder if I should give illumos another shot. |
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| 04:06.00 | stevePearPear | hi |
| 04:06.00 | stevePearPear | im using asterisk with ws, routed through nginx layer for ssl. once in a while, Iâll get 502 gateway error which is resolved when i reset nginx or asterisk. I checked around and it might be due to opened file, any idea? |
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| 04:29.11 | elcontrastador | Where can I ask a question about DPMA? I just upgraded from 11.0_1.7.0 to 11.0_2.0.0 and to asterisk-11.11 from 11.6. Phones are failing to authenticate. Based on the version numbers, it should be a minor upgrade and syntax should still work. |
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| 08:38.31 | dan_j | Hi. Does anyone know what would cause chan_sip to stop receiving packets for a few seconds. I've just implemented asterisk with drbd and pacemaker. But now, about every 10 minutes, most of the peers get cut off and active calls receive no audio. |
| 08:39.26 | dan_j | tcpdump shows replies to OPTIONs, but asterisk doesnt seem to listen to them, and continues to send out OPTIONs and eventually marks the peer as UNREACHABLE |
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| 09:16.09 | pa | hi |
| 09:16.23 | pa | i installed asterisk on a ubuntu 14.04 |
| 09:16.43 | pa | i had to install dahdi from sources because the package dahdi-dkms is apparently broken |
| 09:16.51 | pa | but when i run asterisk i get this: http://paste.ubuntu.com/7950328/ |
| 09:17.50 | WIMPy | Do you need dahdi? |
| 09:18.11 | pa | for an ISDN TA |
| 09:18.17 | pa | i use it as landline |
| 09:18.21 | pa | it was zapata before |
| 09:18.55 | WIMPy | ok |
| 09:20.38 | WIMPy | Is your system.conf ok? What does dahdi_cfg -v say? |
| 09:21.26 | pa | i check |
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| 09:22.18 | pa | WIMPy, it says there are 3 chnels to configure, and ends with DAHDI_SPANCONFIG failed on span 1: No such device or address (6) |
| 09:22.40 | pa | maybe i have to load the dahdi module with some parameters? |
| 09:22.41 | WIMPy | Is the driver loaded? |
| 09:22.49 | pa | i modprobed dahdi |
| 09:22.52 | pa | it worked |
| 09:22.54 | pa | it's there |
| 09:23.10 | WIMPy | That's only the framework. |
| 09:23.17 | WIMPy | You need a driver for your card. |
| 09:23.39 | pa | this is what lsmod finds: http://paste.ubuntu.com/7950380/ |
| 09:23.48 | pa | i guess maybe hisax is the driver? |
| 09:23.58 | pa | it used to be zaphfc |
| 09:24.02 | WIMPy | There's no driver. |
| 09:24.19 | WIMPy | If it's a hfc card, it's dahdi_hfcs. |
| 09:24.38 | WIMPy | Hisax only worked with Asterisk <1.2. |
| 09:25.00 | WIMPy | But if you're using DSS1, you can use mISDN instead. |
| 09:25.33 | pa | ah i see.. i think i dont have this module.. how possible? i compiled dahdi from latest sources, dahdi-linux-complete-2.9.2+2.9.2/ |
| 09:26.02 | WIMPy | It's not part of dahdi. |
| 09:26.07 | WIMPy | It's a 3rd party add-on. |
| 09:26.40 | pa | aha.. and where can i get it? it used to come with ubuntu, but since the package is broken.. |
| 09:29.02 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/6 |
| 09:29.41 | pa | thanks! |
| 09:46.41 | pa | shit.. it does not compile against kernel 3.13 :( |
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| 09:46.54 | pa | gosh, nightmare again |
| 09:47.22 | pa | no wonder, the source is from 2011 |
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| 09:49.24 | pa | but maybe i might be doing it wrong... |
| 09:49.36 | pa | WIMPy, does this look correct to you? http://paste.ubuntu.com/7950524/ |
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| 09:56.13 | WIMPy | That's not the git I had. |
| 09:56.30 | pa | well that git does not clone anymore |
| 09:56.50 | WIMPy | I just did it. |
| 09:57.21 | pa | right, now it works -_- |
| 09:57.22 | pa | sorry |
| 09:57.52 | WIMPy | But I haven't tried compiling it for a long time. |
| 09:58.42 | pa | WIMPy, yes, in fact i get the same error |
| 09:59.26 | pa | would that mean that hfcs is abandoned? |
| 09:59.47 | WIMPy | Well, having to downgrade the kernel after installing a new system is one of the reasons I prefer not to use dahdi. |
| 10:00.16 | pa | :( |
| 10:00.18 | WIMPy | Possibly. I guess ISDN support is kind of dead. |
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| 10:15.05 | sekil | openvox dahdi drivers have zaphfc already |
| 10:15.19 | sekil | with oslec integrated |
| 10:19.50 | WIMPy | The openvox cards will also work with the Linux drivers. |
| 10:20.35 | sekil | hfc modules? |
| 10:21.22 | sekil | I think not when it comes to one-port cards |
| 10:21.35 | pa | mine has only one port :( |
| 10:21.44 | WIMPy | Do they have single port ones? |
| 10:21.46 | sekil | oh yeah..hisax modules |
| 10:22.54 | WIMPy | I'm not even sure hisax/hfc-s ever worked correctly with voice. |
| 10:23.59 | WIMPy | I used it with Siemens Chipset cards for voice. But when I no longer had ISA slots, I had to find something else. |
| 10:24.42 | WIMPy | Although the hisax stuff can still be usefull. AFAIK it's still the only way to manage call forwarding. |
| 10:27.42 | pa | so basically there's no other way than downgrading the kernel to access an ISDN channel via asterisk? |
| 10:28.04 | WIMPy | There are many. |
| 10:28.33 | WIMPy | You have an el-cheapo hfc-s pci card? |
| 10:28.43 | pa | exactly |
| 10:28.45 | WIMPy | What protocoll do you use? |
| 10:28.59 | pa | protocol? like SIP? |
| 10:29.18 | WIMPy | The ISDN protocol. |
| 10:29.21 | pa | ah |
| 10:29.23 | pa | hm |
| 10:29.33 | pa | good question, i'm not 100% sure.. is there some way to check? |
| 10:29.40 | pa | maybe i can google, let me see |
| 10:29.45 | WIMPy | Country? |
| 10:30.10 | pa | EURO i think |
| 10:30.17 | pa | Italy |
| 10:30.28 | WIMPy | Great. |
| 10:30.40 | WIMPy | Then you can use all existing options. |
| 10:30.53 | pa | and i use this TA for calling landlines, and for logging (within asterisk) incoming calls |
| 10:32.24 | WIMPy | I prefer to use the kernel supplied drivers and LCR. |
| 10:32.47 | WIMPy | But you could also use the "old and deprecated mISDN" with chan_misdn. |
| 10:32.57 | WIMPy | It's all explained on the link I gave you. |
| 10:33.38 | pa | thanks! i'll read that. I also would prefer kernel drivers, but i don't know what LCR is, so now i will read up:) |
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| 10:56.53 | goddva | does asterisk support the "a=direction" in the SDP ? |
| 11:03.38 | WIMPy | If I get it right, only outgoing. |
| 11:03.54 | WIMPy | For chan_sip, that is. No idea about chan_pjsip. |
| 11:05.26 | pa | WIMPy, so if i understand you, the solution of choice would be mISDN2 + this sort of additional PBX to connect to asterisk? |
| 11:06.30 | WIMPy | It is *my* choice. You have to make your own. |
| 11:07.18 | WIMPy | But having the drivers in the kernel is a big plus for me. |
| 11:07.47 | WIMPy | And it has also been the most stable for me, but ymmv. |
| 11:07.49 | pa | WIMPy, but with chan_capi i wouldnt need the LCR, or would i still? |
| 11:08.02 | pa | i mean mISDN2 and chan_capi |
| 11:08.06 | WIMPy | No, that's only needed with mISDN2. |
| 11:08.28 | WIMPy | Although you can use CAPI with mISDN2 now. |
| 11:08.38 | pa | yes, that's what i meant |
| 11:08.50 | pa | skipping LCR, and using mISDN2 through chan_capi |
| 11:09.28 | WIMPy | You can do that, yes. |
| 11:09.48 | pa | and that would work also with voice and everything else? |
| 11:09.52 | WIMPy | Although the Asterisk capi channels haven't been that active, wither. |
| 11:10.21 | WIMPy | I think the mISDN2 CAPI has round about no features. |
| 11:11.10 | pa | hm.. meaning no voice or? |
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| 11:11.26 | WIMPy | No, voice does work. |
| 11:11.47 | pa | ah, so maybe it would do it.. |
| 11:11.52 | pa | i try and let you know :) |
| 11:13.51 | pa | i wonder if that's also ok for el-cheapo cards, though |
| 11:14.09 | WIMPy | It doesn't matter. |
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| 11:28.58 | gavimobile | is it normal to have these messages in the cli several times a day? I don't expirence any problems but it looks like these are small disconections in my internet which drive me mad! http://pastebin.com/SzmxSbEf |
| 11:29.27 | pa | WIMPy, do you know whether ubuntu asterisk supports chan_capi? i followed this: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+with+CAPI, installed capiutils, enabled capi in modules.conf, but it seems i don't have the "capi" command within asterisk CLI, and when i call nothing happens.. |
| 11:29.46 | pa | oh, i also "capiinit"ed |
| 11:30.30 | pa | i think there's no chan_capi |
| 11:31.05 | pa | no.. |
| 11:31.09 | pa | bloody ubuntu |
| 11:32.33 | WIMPy | gavimobile: You seem to have a bad or saturated connection |
| 11:32.44 | pa | 14.04 is the sloppiest LTS ever.. |
| 11:33.09 | WIMPy | chan_capi is a 3rd party add-on again. |
| 11:36.36 | pa | WIMPy, do you remember the repo? :) |
| 11:37.04 | pa | ah found |
| 11:37.06 | WIMPy | All I know is on the link I gave you. |
| 11:37.08 | pa | in your page |
| 11:37.16 | pa | but you are right |
| 11:37.20 | pa | it's abandoned |
| 11:37.25 | pa | it's the same as it was for 10.04 |
| 11:38.05 | pa | well not exactly |
| 11:38.11 | pa | but almost |
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| 11:46.21 | pa | ok seems that the latest chan capi does not compile against recent asterisks.. |
| 11:51.24 | pa | what i find a little odd is that the build system identifies my asterisk version |
| 11:51.25 | pa | http://paste.ubuntu.com/7951233/ |
| 11:51.35 | pa | but then it fails compiling |
| 11:55.16 | pa | i guess i have to use your way :) |
| 11:55.21 | pa | the only one left |
| 11:57.40 | pa | WIMPy, what distribution do you use? |
| 11:57.49 | gavimobile | WIMPy: how would I diagnose something like this? I don't know if its my isp or my router |
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| 12:05.11 | WIMPy | pa: Usually Slackware |
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| 12:05.40 | WIMPy | gavimobile: Check your throughput or set up somethign like smokeping. |
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| 12:29.43 | XATRIX | Hi guys, i have a trouble. I have 3G modem Huawei E1550 for the voice calls. And when i do a call, VOIP->GSM->mobile phone, I can hear mobile partner well in my headphones. But my speech is like a robot |
| 12:30.01 | XATRIX | What can it be ? Outcoming codec ? |
| 12:30.18 | XATRIX | My Phoner says i'm using in:A-Law; out:A-Law |
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| 12:56.37 | tparcina | XATRIX: You have a good signal on 3G modem? |
| 12:56.50 | XATRIX | I suppose yes |
| 12:57.04 | XATRIX | Because of i can hear remote side clearly |
| 12:57.17 | XATRIX | Only my speech is robotized... |
| 12:58.02 | XATRIX | Also, my voice sounds, for example "you cannont dial this number" - are robotized |
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| 13:05.00 | XATRIX | How can i find out my signal quality from Linux ? |
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| 13:14.23 | stevePearPear | hi |
| 13:14.57 | stevePearPear | anyone gotten this error before? |
| 13:14.58 | stevePearPear | 8018#0: *24183 upstream prematurely closed connection while reading response header from upstream, client: 116.15.31.xxx, server: xxx.yyy.io, request: "GET / HTTP/1.1", upstream: "http://127.0.0.1:8088/ws", host: "xxx.yyy.io" |
| 13:15.15 | stevePearPear | when using nginx as websocket redirect to asterisk |
| 13:16.02 | XATRIX | tparcina: any idea on my problem ? |
| 13:17.15 | tparcina | XATRIX: How is Asterisk connected to 3G modem? |
| 13:17.32 | XATRIX | chan_dongle |
| 13:17.37 | XATRIX | Asterisk 11.11.0 |
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| 13:18.12 | pa | WIMPy, so after all plenty of options reduces to one, with the late kernels :) |
| 13:20.18 | XATRIX | tparcina: as a dongle show state <device> shows me out: |
| 13:20.20 | XATRIX | <PROTECTED> |
| 13:20.20 | XATRIX | <PROTECTED> |
| 13:20.28 | *** join/#asterisk hindi (~hindi@030-138-088-212.ip-addr.vsenet.de) |
| 13:20.54 | XATRIX | The signal is not so bad i think |
| 13:21.03 | tparcina | XATRIX: I don't know what 30 means, but -53 dBm is a good signal. |
| 13:21.22 | XATRIX | Me neither. So , what's the problem with |
| 13:21.38 | XATRIX | May i ask you to call my number to take a look at the sound |
| 13:21.39 | XATRIX | ? |
| 13:22.20 | tparcina | XATRIX: Sorry, I'm not able to do that. :( |
| 13:26.23 | *** join/#asterisk Kattyroo (61577062@gateway/web/cgi-irc/kiwiirc.com/ip.97.87.112.98) |
| 13:26.33 | XATRIX | tparcina: So, what's the problem with outgoing audio can be ? |
| 13:26.37 | XATRIX | Any debug ? |
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| 13:38.49 | nunne | Anyone else has issus with asterisk 11 and sip lockups? After the system has been online for maybe a month or so people will start complaining about delays on answers/pickups of about 5 seconds. And also we start seeing a lot of failed calls. You don't see anything "wrong" in the console and all phones seem okay as well. |
| 13:41.35 | [TK]D-Fender | A failure would have to show something..... |
| 13:41.49 | [TK]D-Fender | And you should be specific on what version you're running. |
| 13:48.34 | XATRIX | Guys, what's wrong with my GSM-trunk ? |
| 13:48.41 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 13:49.00 | XATRIX | When i call to my asterisk from the mobile, i have the following sound ftp://anime.ihome.ua/InComiNg/1.3gpp |
| 13:49.07 | XATRIX | Is it a codec issues ? |
| 13:49.43 | XATRIX | P.S this sound was recorded on my mobile |
| 13:51.25 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 13:54.16 | stevePearPear | hi anyone has gotten this error when using nginx as a reverse web socket proxy for asterisk? |
| 13:54.17 | stevePearPear | 8018#0: *24183 upstream prematurely closed connection while reading response header from upstream, client: 116.15.31.xxx, server: xxx.yyy.io, request: "GET / HTTP/1.1", upstream: "http://127.0.0.1:8088/ws", host: "xxx.yyy.io" |
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| 14:06.16 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
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| 14:11.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:17.18 | nunne | [TK]D-Fender: It's 11.10, but I had this problem in 11.7 as well. I don't see anything in any log when the call fails. So I guess asterisk is just rejecting the calls.. And for the 5 second delay.. I have no idea. Even here the call will "look" normal in the logfile |
| 14:17.50 | [TK]D-Fender | Logs = largely useless. |
| 14:17.53 | [TK]D-Fender | live CLI..... |
| 14:18.00 | [TK]D-Fender | Guessing is bad... |
| 14:19.40 | *** part/#asterisk igcewieling (~ewieling@ip98-183-26-100.pn.at.cox.net) |
| 14:20.28 | nunne | [TK]D-Fender: I know. I should do a tcpdump and look into it. But was wondering if anyone else has stumbled upon this. Since it seems to be an overload issue. Since a simple restart of asterisk solves it. |
| 14:21.20 | [TK]D-Fender | no failed call, no status dumps, nothing to look at... |
| 14:21.40 | [TK]D-Fender | And ther is also an update to take |
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| 14:36.47 | roxlu | hi! |
| 14:37.16 | WIMPy | nunne: Did you try 'sip show channels'? |
| 14:38.22 | *** join/#asterisk nysander (~nysander@afrc73.neoplus.adsl.tpnet.pl) |
| 14:38.28 | nysander | hello |
| 14:39.59 | WIMPy | hi nysander |
| 14:40.17 | nysander | how can I make conditional dialplan based on number called (for example _915. as entry point and then _9156. redirect to some other exten and rest of _915. parse in this exten) |
| 14:41.28 | WIMPy | You don't have to. You can just write your extensions the way you want to handle it. |
| 14:41.45 | WIMPy | Or you can GotoIf or the like. |
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| 14:42.45 | roxlu | I'm trying to add ICE/WebRTC support in my application and I want to start by letting Chrome connect to my app. I'm receiving a Binding-Request and I reply with a Binding-Success-Response, though the browser keeps sending Binding-Requests and in the log I see "Received non-STUN packet from an unreadable connection". I was hoping someone to find someone with WebRTC/ICE experience here who maybe knows what I'm doing wrong? |
| 14:43.27 | Kattyroo | roxlu: good morning. |
| 14:43.44 | roxlu | Kattyroo: you a good morning too :) |
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| 14:50.58 | navaismo | hi malcolmd here is another thread, can you merge it with the previuos one or locke that too. Thanks in advance--->http://forums.digium.com/viewtopic.php?f=1&t=91007&sid=ac2799c9c61f6d871e975e16741b0921 |
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| 14:55.59 | XATRIX | Anyone can advice on 3G modem sound corruption ? |
| 14:58.52 | *** join/#asterisk nysander (~nysander@afrc73.neoplus.adsl.tpnet.pl) |
| 14:58.57 | nysander | back |
| 14:59.59 | nysander | WIMPy I know but I need to pass all _915. connections to 2 carriers (trunks) and _9156. only to one of them |
| 15:00.07 | XATRIX | I have 3G modem Huawei E1550 for the voice calls. And when i do a call, VOIP->GSM->mobile phone, I can hear mobile partner well in my headphones. But my speech is like a robot What can it be ? Outcoming codec ? My Phoner says i'm using in:A-Law; out:A-Law |
| 15:00.20 | XATRIX | Any idea how to debug this things ? |
| 15:01.31 | *** join/#asterisk [1]nysander (~nysander@adnm144.neoplus.adsl.tpnet.pl) |
| 15:01.56 | [1]nysander | XATRIX, do you have newest firmware? what distro and asterisk version? |
| 15:02.42 | WIMPy | nysander: That's ok. You can have overlapping extensions. Asterisk will use the most specific match. |
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| 15:04.10 | XATRIX | [1]nysander: Yeap, i have a recent firmware, Using asterisk 11.11.0 |
| 15:04.45 | XATRIX | My current kernel is 2.6.18-371.11.1.el5 , and dongle has RSSI: -53 dBm |
| 15:04.59 | XATRIX | As for the signal - should be excellent level |
| 15:05.01 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 15:05.48 | XATRIX | Also, when i talk to my mobile phone - i can hear my voice perfectly on the headset of my PC, but when i talk to the mic of the PC - i hear something like ftp://anime.ihome.ua/InComiNg/1.3gpp |
| 15:06.18 | XATRIX | It's a record of PC->VoIP->GSM-mobile phone voice |
| 15:06.31 | XATRIX | I'm totally stucked in this |
| 15:07.10 | XATRIX | It it suppose to be a RF interference, i should hear distortion in both directions |
| 15:07.32 | XATRIX | And my SIP works correctly. I can talk to my another trunk successfully |
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| 15:07.45 | WIMPy | Not neccessarily, but I still agree that it's probably not. |
| 15:09.44 | XATRIX | Any ideas on my trouble ? |
| 15:11.54 | WIMPy | I'd suspect some internal timing/packetization thing. I came across such a thing, but I can't remember the circumstances. |
| 15:12.51 | XATRIX | I checked my CONFIG_HZ_1000 is set in kernel |
| 15:14.05 | XATRIX | But wait a bit. If it possible can be a timings, it should be affected in both directions |
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| 15:14.29 | XATRIX | Why do i hear a clear sound from mobile->GSM->VoIP->PC ? |
| 15:14.38 | XATRIX | And distorted vice versa |
| 15:14.43 | Qwell | that sounds like a codec mismatch |
| 15:15.05 | Qwell | trying to play a ulaw as slin or something |
| 15:15.23 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 15:15.30 | Qwell | or probably more like alaw<>ulaw, since there's some semblance of voice left |
| 15:16.42 | pa | has anybody here tested chan_capi_hps? |
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| 15:18.38 | XATRIX | Qwell: any idea how to fix ? |
| 15:19.03 | XATRIX | Do i have to set 'disallow=all' and 'allow=gsm' in my extension ? |
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| 15:26.31 | XATRIX | I've fixed it! |
| 15:26.47 | XATRIX | I don't know how, but i've installled elrepo kernel for centos 5.10 |
| 15:26.54 | XATRIX | 3.2.61-1.el5.elrepo |
| 15:27.04 | XATRIX | And now - everything sounds ok! |
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| 15:50.25 | Kattyroo | oh where is my hair brush |
| 15:50.37 | Kattyroo | infobot: ohwhereismyhairbrush |
| 15:50.37 | infobot | extra, extra, read all about it, ohwhereismyhairbrush is NOT FAIR, MY POOR HAIRBRUSH! |
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| 17:16.27 | wrxed | having audio issues trying to get Altigen (H323) to talk to Asterisk 11. I've established a VPN between the sites where the asterisk server is and the Altigen box is. I've turned off NAT and turned on direct media and still am not getting audio. Anybody have expierience in getting h323 to work with asterisk? Its new to me and I would appreiate the help. The debugging message dont seem too |
| 17:16.28 | wrxed | helpful either |
| 17:18.07 | newtonr | I haven't touched H.323 in years! I'll be of no help. |
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| 17:20.21 | wrxed | yeah... that seems to be about how long ago any thing i google for is... YEARS old |
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| 18:15.20 | malcolmd | navaismo: got it |
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| 18:24.17 | apb1963 | anyone else using ipkall? Having problems with it today? |
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| 18:39.03 | ro23 | hello |
| 18:40.15 | ro23 | i've been using pbxinaflash for some time and now i'm installing asterisk manually. i'm on the menu select screen but i'm not sure what i need - does anyone know which ones are the default for piaf? |
| 18:41.22 | [TK]D-Fender | just about everything |
| 18:45.33 | ro23 | ok thanks |
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| 19:55.42 | youjelly | Hey guys |
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| 20:25.15 | TazzNZ | Morning all |
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| 21:31.59 | iulhk | using asterisk 10.7.0 i have 2 sippeers type=friend i have set rtptimeout=60 seconds, call should be disconnected if there is no rtp, but my call doesn't hangup, even i have set rtpholdtimeout=300 seconds and at one client i have hold the call more than 300 seconds call doesn't hangup, any idea please ? |
| 21:35.43 | [TK]D-Fender | your version is old even within your branch, and the branch itself is longer supported |
| 21:38.58 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:41.50 | iulhk | <[TK]D-Fender>: should i replace with asterisk-11 LTS? can i use existing sippeers table with new asterisk 11 LTS version ? does it effect on performance ? |
| 21:41.53 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:42.13 | [TK]D-Fender | yes, you should upgade./ |
| 21:42.19 | [TK]D-Fender | Your version isold and not support |
| 21:42.21 | [TK]D-Fender | ed |
| 21:43.02 | iulhk | what about sippeers table in mysql ? should i change with new sippeers schema or existing will work ? |
| 21:47.40 | [TK]D-Fender | Should be the same |
| 21:48.39 | wrxed | only difference is the nat value |
| 21:48.49 | wrxed | from 10 -> 11 syntax went from 'yes' to force_rport,comedia |
| 21:49.00 | wrxed | or i should say 'yes' became depricated |
| 21:49.30 | iulhk | thanks :) |
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| 23:36.52 | iulhk | <[TK]D-Fender>, i have upgraded asterisk 11 , but still rtpholdtimeout doesn't work, any idea please ? |
| 23:39.09 | Gugge | what do you have rtpholdtimeout set to on the peer, and what do you have rtptimeout set to? |
| 23:39.26 | Gugge | and are you sure there is no rtp when you are testing hold? |
| 23:41.26 | iulhk | <Gugge>: i have set rtpholdtimeout=300 , i called peer A to peer B, call connected and then at peer A i click on hold button. at receiver end i was getting music on hold, i wait about 10 minutes but call still connected , ? |