| 00:00.53 | j4jackj | ChannelZ: do you remember G.FSH? |
| 00:00.58 | j4jackj | it was a fun joke |
| 00:02.38 | j4jackj | It's actually implemented in G729b |
| 00:03.08 | Penguin | Does it require an additional license? |
| 00:08.32 | j4jackj | Penguin: a-yup. I don't use 729. |
| 00:10.13 | Penguin | I didn't know -- I always disable the G.FSH extension before I compile G.729. |
| 00:16.17 | j4jackj | Penguin: G729 requires the license, G.Fsh just uses that license. |
| 00:16.21 | j4jackj | Sorry for any confusion. |
| 00:16.30 | Penguin | I see. |
| 00:18.49 | j4jackj | G.fsh itself MAY also require a license depending on whether the codec itself has VAD. |
| 00:22.05 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 00:24.51 | j4jackj | YOU GOT PRANKED until I reverse classed that... |
| 00:29.25 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
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| 01:22.24 | raidghost | <PROTECTED> |
| 01:22.24 | raidghost | [2013-09-16 03:20:44] WARNING[668][C-00000003]: channel.c:5956 ast_request: No channel type registered for 'A2B' |
| 01:22.24 | raidghost | 013-09-16 03:20:44] WARNING[668][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'A2B' (cause 66 - Channel not implemented) |
| 01:22.47 | raidghost | How to fix this ANNOYING errormessage. Its the only thing that holding me from make a outbound call |
| 01:23.47 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:23.50 | [TK]D-Fender | <PROTECTED> |
| 01:23.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.196) |
| 01:24.02 | [TK]D-Fender | What would give you the impression that it exists? |
| 01:24.09 | [TK]D-Fender | that is not a channel technology |
| 01:24.40 | [TK]D-Fender | Would not appear you are paying attention to what you are dialing |
| 01:24.50 | raidghost | Been using soon 18 hours on this annoying crap. Would be nice if we could solve it before im thinking of going to bed. |
| 01:25.34 | [TK]D-Fender | Look at what you are dialaing and realize that that is not a valid tech. |
| 01:25.46 | WIMPy | Maybe you should find out whay you try to do? |
| 01:26.20 | raidghost | So i should use IAX2 instead of sip? |
| 01:26.27 | *** join/#asterisk apb1963 (~apb1963@174.134.98.138) |
| 01:26.37 | [TK]D-Fender | raidghost: you aren't SPECIFYING SIP |
| 01:26.48 | [TK]D-Fender | raidghost: lok at the format of your dial command. |
| 01:26.54 | WIMPy | You should use something that exists instead of "a2b". |
| 01:27.05 | [TK]D-Fender | raidghost: You put the word "A2B" where a TECHNOLOGY belongs |
| 01:27.11 | raidghost | NO i didnt |
| 01:27.22 | raidghost | It says SIP in my a2billing trunk settings |
| 01:27.23 | [TK]D-Fender | [21:22]raidghost[2013-09-16 03:20:44] WARNING[668][C-00000003]: channel.c:5956 ast_request: No channel type registered for 'A2B' <- oh hell yes. |
| 01:27.44 | raidghost | I dont know where this A2B F* thing comes from |
| 01:27.45 | [TK]D-Fender | that message is not lying. Asterisk was told to dial(A2B/SOMETHING) |
| 01:27.50 | [TK]D-Fender | Its your server |
| 01:28.54 | raidghost | Ok. So the A2B thing i dont know where is. Is somewhere in my configs |
| 01:33.16 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:39.40 | raidghost | Looking for s in from-sip-external (domain sip.myprovider.com) |
| 01:39.59 | nam3l3zz | hello people |
| 01:40.14 | raidghost | a normal message? or is this "s" thingy supposed to not be there |
| 01:40.49 | [TK]D-Fender | raidghost: taht looks like a call from somewhere you registered to |
| 01:48.43 | raidghost | Neeh. Its time for some hours rest. [TK]D-Fender: Thanks for help out troublershooting |
| 01:48.51 | [TK]D-Fender | raidghost: and it would go to "s" because you didn't tell them to send to something else by specifying the return exten in your REGISTER statement |
| 01:51.59 | raidghost | Bedtime. BBL |
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| 02:31.49 | *** join/#asterisk mohadib (~mohadib@unaffiliated/mohadib) |
| 02:31.51 | mohadib | hello |
| 02:33.37 | mohadib | I have a sip trunk form didlogic with two DIDs. I am seeing reject messages from asterisk from that trunk. The rejects are to non existant extensions, coming from an ip in Palestinian. |
| 02:34.19 | mohadib | I dont really understand why or how this is happening, I'm pretty new to telephony. Can anyone offer any insite? I'm concerned that security somewhere has been breached. |
| 02:35.03 | mohadib | er.. insight :s |
| 02:40.54 | Penguin | If the IP address is from Palestine, what gives you the impression it is a call from didlogic? |
| 02:41.07 | mohadib | yeah, i think im wrong about that |
| 02:41.18 | mohadib | i had to allow anon sip sessions to use didlogic |
| 02:41.25 | mohadib | and didlogic is the default context for incoming calls |
| 02:41.26 | Penguin | That was a mistake. |
| 02:41.30 | mohadib | yeah? |
| 02:41.33 | Penguin | That's another mistake. |
| 02:41.34 | mohadib | i didnt like it |
| 02:41.43 | Penguin | Properly create a peer for didlogic. |
| 02:41.44 | mohadib | flowroute didnt need anon sip |
| 02:42.19 | mohadib | i think i did? I made a entry in sip.conf for didlogic. one to register, then a sip entry |
| 02:42.27 | Penguin | Any good ITSP is going to give you the IP address(es) for their server(s). |
| 02:42.51 | Penguin | Pastebin your entries that you created in sip.conf. |
| 02:42.57 | mohadib | ok, thanks! |
| 02:43.22 | Penguin | Mask your passwords only. |
| 02:44.28 | mohadib | http://pastebin.com/RxjC6tN4 |
| 02:44.42 | mohadib | ah, i masked my user and did # too, sorry |
| 02:45.22 | Penguin | http://pastebin.com/fKArHCNp |
| 02:45.44 | Penguin | Crap, username should be changed to defaultuser. I missed that. |
| 02:47.08 | mohadib | so, change the type and use remotesecret instead of secret? |
| 02:48.11 | Penguin | type=friend is for when devices need to authenticate to asterisk using a username. secret is for when a device needs to authenticate to asterisk. |
| 02:48.24 | Penguin | Neither of those criteria match your use case. |
| 02:48.25 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 02:48.26 | mohadib | oh, i see you removed insecure too |
| 02:48.29 | mohadib | ok |
| 02:49.26 | mohadib | ok, updated and reloaded, thanks |
| 02:49.42 | Penguin | Don't forget to set allowguest back to no. |
| 02:49.45 | mohadib | ah |
| 02:50.26 | mohadib | do i still need a default context? |
| 02:50.38 | Penguin | In extensions.conf? |
| 02:50.43 | mohadib | in sip.conf |
| 02:50.53 | Penguin | Does not compute. |
| 02:51.00 | mohadib | context=didlogic ; Default context for incoming calls |
| 02:51.14 | mohadib | that is under general in sip.conf |
| 02:51.35 | Penguin | That would ideally be something like context=unauthenticated |
| 02:51.43 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 02:51.44 | mohadib | ok |
| 02:52.17 | Penguin | And then be sure to create that context in extensions.conf and put in it only those extension that you want unauthenticated devices to use. Probably none. |
| 03:06.22 | mohadib | Penguin: now that I have turned quest access off I am rejecting calls from didlogic |
| 03:06.29 | mohadib | handle_request_invite: Sending fake auth rejection for device |
| 03:06.39 | Penguin | Let's look at that for a minute. |
| 03:06.51 | Penguin | Look at the IP address that the calls are coming from. |
| 03:06.51 | mohadib | do you want to see the full message? |
| 03:06.55 | Penguin | Not yet. |
| 03:07.37 | mohadib | ok, i got the ip |
| 03:07.44 | Penguin | Now share it with me. |
| 03:07.57 | mohadib | 27.50.90.165 |
| 03:09.47 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 03:10.11 | Penguin | If you are sure the calls using that address are originating with didlogic, we can add that to the peer. |
| 03:10.44 | mohadib | yes, that is coming from didlogic |
| 03:11.24 | mohadib | at didlogic site they instruct me to turn on anon guest, but you think I can work around that? |
| 03:11.50 | Penguin | http://pastebin.com/7821U7zg |
| 03:12.25 | Penguin | It depends on how many hosts they are going to send calls from. |
| 03:12.26 | mohadib | the new part you added goes in sip.conf too? |
| 03:12.31 | mohadib | ah i see |
| 03:12.47 | mohadib | can i use cidr addressing in host? |
| 03:12.57 | Penguin | No. |
| 03:13.18 | Penguin | Not unless that is a new feature in a newer asterisk version than what I am familiar with. |
| 03:13.22 | mohadib | i see |
| 03:13.34 | mohadib | so the new config you added goes in sip.conf too? |
| 03:13.47 | Penguin | That should have been apparent. |
| 03:14.25 | mohadib | ah yes |
| 03:16.31 | mohadib | ok that did it |
| 03:16.34 | mohadib | Thanks Penguin |
| 03:16.45 | mohadib | Ill watch for rejections if they use another ip |
| 03:16.58 | Penguin | Let's just hope they don't suck so badly that they start sending calls from a new address each time. |
| 03:17.21 | mohadib | yeah |
| 03:17.29 | mohadib | if so ill just ditch them |
| 03:17.46 | Penguin | If they just have a few IPs, it won't be that much of a problem to expand the peer with a few more host entries. |
| 03:18.08 | mohadib | can you suggest anyone good for sip trunking and DIDs? |
| 03:18.24 | Penguin | First of all, there is no such thing as SIP trunking. |
| 03:18.41 | Penguin | Second of all, what continent are you located on? |
| 03:18.56 | mohadib | ah, ok on the first point |
| 03:19.20 | mohadib | what is the right terminology to use for my connection to didlogic? |
| 03:19.34 | Penguin | SIP? |
| 03:19.38 | mohadib | hah ok |
| 03:19.39 | Penguin | They are an ITSP. |
| 03:19.42 | mohadib | ah |
| 03:19.47 | Penguin | They are a peer to your asterisk. |
| 03:19.59 | mohadib | so i have read about sip trunking i thought? |
| 03:19.59 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 03:20.04 | mohadib | people are abusing that term? |
| 03:20.18 | Penguin | SIP doesn't trunk. People who use the phrase don't understand what trunking is. |
| 03:20.23 | mohadib | ah |
| 03:20.28 | mohadib | trunking is for pstn? |
| 03:20.35 | Penguin | ~trunk |
| 03:20.35 | infobot | [trunk] a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 03:20.57 | mohadib | ah thank you |
| 03:21.12 | mohadib | ok, second Im in the US, but i also want a did with good rates in AU |
| 03:21.40 | Penguin | ~itsplist-us |
| 03:21.41 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 03:21.44 | mohadib | currently im using flowroute for my US dids and didlogic for AU |
| 03:21.55 | mohadib | ok, thanks again |
| 03:22.16 | mohadib | ~itsplist-au |
| 03:22.18 | Penguin | I don't know if we have any AU ones on the bot... |
| 03:22.22 | mohadib | ok |
| 03:22.56 | Penguin | I would hope there is more than just one in AU, though. |
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| 04:10.50 | wafflejock | Hello all is this an appropriate place to ask for help or can someone point me in the right direction? |
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| 04:57.28 | ChannelZ | ~ask |
| 04:57.28 | 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. |
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| 06:35.13 | wafflejock | Okay, so here's my setup. I have a raspberry Pi that I loaded up with one of the RasPBX images and I want to use it for Google Voice (Motif) based conference calls. That is I have a Google account with voice and I want to set it up in RasPBX. |
| 06:36.28 | wafflejock | I've done this before but this time around I'm running into errors when I try to add my Google Voice account. So I'm wondering if I can get some help debugging the errors I'm seeing. Specifically this is coming up: [2013-09-15 23:13:34] WARNING[7102]: res_xmpp.c:3587 xmpp_client_thread: JABBER: socket read error |
| 06:36.29 | wafflejock | [2013-09-15 23:13:34] WARNING[7102]: res_xmpp.c:3528 xmpp_client_receive: Parsing failure: Hook returned an error. |
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| 06:38.12 | wafflejock | I've tried with a couple of different Google accounts to verify nothing in the username/password was causing a problem but I'm just not sure how to move forward. Optionally I can install Aterisk and FreePBX on my Kubuntu laptop setup if that will work better, but really just looking for a solution to connection Google Voice with Asterisk. |
| 06:44.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.248) |
| 06:46.22 | deegen | maybe |
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| 07:20.45 | ChannelZ | Those errors seem a bit odd with the socket read error |
| 07:21.49 | ChannelZ | seems like perhaps you've got firewall issues or something else preventing access. Beyond that without seeing any configs or debug it's just guessing |
| 07:23.32 | ChannelZ | I will get socket read errors from time to time, probably when Google's servers barf |
| 07:24.05 | ChannelZ | but it otherwise works. |
| 07:28.21 | ChannelZ | That said Google Talk is pretty much dead, and I don't know if Google Voice is long for this world either |
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| 08:10.42 | wafflejock | ChannelZ what are my alternatives in the way of getting a DID to point to a local Asterisk/FreePBX setup? |
| 08:11.14 | wafflejock | ChannelZ: sorry I'm a complete noob here but any advice is welcome... may be knocking out soon but will leave the IRC open for suggestions |
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| 10:02.46 | asghar144 | hi |
| 10:03.49 | apurvtwr | hi I am using asterisk 1.8.21. There is a background noise that I occassionally get on some channels. |
| 10:04.45 | apurvtwr | it's intermittent. However, here are few observations which I ahve made. |
| 10:05.20 | *** join/#asterisk aruntomar (~Thunderbi@117.195.39.222) |
| 10:05.42 | apurvtwr | the noise is almost always on only 1 one of the 2 channels of a call. |
| 10:07.09 | apurvtwr | I have checked the codec used on the two channels, in first test (when it was reproduced) it was ulaw on one and alaw on the noisy channel. However, second time it was ulaw on both. |
| 10:08.09 | apurvtwr | on the other hand, most of the other calls work well on ulaw codec. |
| 10:08.40 | apurvtwr | Can anyone give me an idea, what I should look at next to debug the issue? |
| 10:09.36 | asghar144 | i am upgrading from asterisk 1.6.2 to asterisk 1.8.23.1 and using cdr_mysql backend, i always used usegmtime=yes and cdr was inserted in gmt time but with 1.8 no matter what i configure in cdr_mysql.conf records alway are inserted in localtime. |
| 10:10.36 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:10.44 | apurvtwr | this is my configuration in sip.conf |
| 10:10.45 | apurvtwr | disallow = all allow = ulaw allow = alaw allow = g729 qualify=yes |
| 10:11.05 | asghar144 | asked on mailling list http://lists.digium.com/pipermail/asterisk-users/2013-September/280541.html |
| 10:11.26 | asghar144 | withot any responce |
| 10:13.38 | apurvtwr | anyone? |
| 10:20.20 | asghar144 | so i have created a patch that resolve this issue if here can test the patch? |
| 10:26.32 | asghar144 | if somebody here can test the patch? |
| 10:27.13 | Greenlight | You asked on the mailing list on Saturday. It's not even Monday morning yet in US. I'd say wait a little longer |
| 10:29.43 | asghar144 | sorry i was expecting some answer not only from US but also other parts of the world |
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| 10:32.34 | Greenlight | It's a rather specific question, likely best answered by whoever actually altered the code in the first place. My guess is that that must have been a reason for the change, but I can't think what that might be. |
| 10:36.35 | Greenlight | apurvtwr: From what I can see you just listed a bunch of sip configuration options. You've not actually asked a question, did you mean to ? |
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| 10:38.16 | asghar144 | yes there should be some reason but in doc,s and in changelog nothing is mantioned even in cdr_mysql.conf there is still old options that should be removed if they removed GMT support. |
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| 10:40.19 | apurvtwr | Greenlight: I meant to ask, what else could be a reason for this issue? Since everywhere I ahve read, it seems like a codec issue |
| 10:40.49 | apurvtwr | even here https://issues.asterisk.org/jira/browse/ASTERISK-1323 |
| 10:41.18 | Greenlight | You are taking SIP, that issue is IAX. |
| 10:41.56 | Greenlight | Or perhaps I missed the start of your question. I only join #asterisk at 10 past |
| 10:42.33 | apurvtwr | oh yes, I see you joined after i have already asked the question |
| 10:42.41 | apurvtwr | hi I am using asterisk 1.8.21. There is a background noise that I occassionally get on some channels. |
| 10:43.03 | apurvtwr | it's intermittent. the noise is almost always on only 1 one of the 2 channels of a call. |
| 10:43.16 | apurvtwr | I have checked the codec used on the two channels, in first test (when it was reproduced) it was ulaw on one and alaw on the noisy channel. However, second time it was ulaw on both. |
| 10:43.24 | Greenlight | What codec are you using ? |
| 10:43.32 | apurvtwr | on the other hand, most of the other calls work well on ulaw codec. |
| 10:44.00 | Greenlight | ANd you've ruled out a faulty microphone etc ? |
| 10:44.02 | apurvtwr | I have tried using ulaw only as well as the other time alaw only as well |
| 10:44.45 | apurvtwr | yes, because it happens only once in 7-8 calls |
| 10:45.07 | Greenlight | And always remains for the duration of the call ? |
| 10:45.22 | apurvtwr | yes |
| 10:45.33 | Greenlight | Sounds like the endpoint could be faulty |
| 10:45.55 | apurvtwr | by endpoint you mean the carrier? |
| 10:46.18 | Greenlight | I was presuming you had a sip phone as an endpoint ? |
| 10:46.20 | asghar144 | patch is available here http://www.world-call-trade.com/asterisk/cdr_mysql_cdrzone.patch |
| 10:47.28 | apurvtwr | actually, I have a gateway registered as a SIP trunk to asterisk... however, gateway hosts PRI lines |
| 10:47.36 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:47.44 | apurvtwr | so for asterisk it's SIP |
| 10:47.45 | Greenlight | And on the other side of the call |
| 10:48.42 | apurvtwr | both sides call are routed through the gateway, however I can try taking the call on SIP phone if you give me 5 mins |
| 10:49.01 | Greenlight | I would suggest the issue is on the gateway, or the PRI side |
| 10:49.54 | Greenlight | I would recommend getting a capture of the traffic when a call is in progress, and then playing it back using something like wireshark |
| 10:51.08 | apurvtwr | ok. Let me do that. I will reproduce the issue first with a SIP phone on one side and gateway on the other. |
| 10:51.25 | apurvtwr | then capture the packets with wireshark. |
| 10:51.49 | Greenlight | You might even try connecting the SIP phone directly to the PRI<->SIP gateway |
| 10:52.35 | Greenlight | Is your PRI alaw or ulaw ? |
| 10:53.34 | *** join/#asterisk evilman_work (~evilman@87.244.43.210) |
| 10:54.35 | apurvtwr | the PRI is alaw |
| 10:55.26 | apurvtwr | I am trying a couple of calls right now to reproduce the issue on SIP phone. |
| 10:57.50 | Greenlight | If the PRI is alaw then i'd suggest trying to use that exclusively |
| 11:02.05 | *** join/#asterisk aruntomar (~Thunderbi@117.195.49.190) |
| 11:05.28 | apurvtwr | ok.. I tried 20 calls, one side on SIP phone the other through gateway, none of them had this background noise issue |
| 11:06.13 | Greenlight | That's odd |
| 11:06.24 | Greenlight | No noise even on PSTN side |
| 11:06.36 | apurvtwr | the codec I was using was ulaw on both channels (asterisk). Gateway however was doing codec negotiation to alaw |
| 11:06.51 | apurvtwr | but there was no noise at all on either channels. |
| 11:10.17 | Greenlight | Why are you using ulaw ? |
| 11:13.18 | apurvtwr | actually there are mutliple PRI carriers on the same gateway. Some use Ulaw, others alaw. Gateway is configured to use the appropriate translation. |
| 11:13.57 | WIMPy | o.O |
| 11:14.03 | apurvtwr | I ahve tried using only alaw as well.. but it didn't work |
| 11:14.12 | *** join/#asterisk modesto916 (~modesto@189-90-192-72.isimples.com.br) |
| 11:15.44 | Greenlight | My gut feeling is that the gateway is doing something odd |
| 11:15.50 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 11:16.10 | Greenlight | ANd perhaps not detecting what codec should be getting used. |
| 11:16.28 | Greenlight | I seem to recall if you interpert ulaw as alaw you get noise |
| 11:16.32 | Greenlight | Maybe someone can confirm ? |
| 11:16.59 | Chainsaw | Greenlight: There is an agreed way to transcode it without too much bother. But failing to do so wouldn't sound very good. |
| 11:17.48 | Chainsaw | Greenlight: I tend to allow both alaw & ulaw on all sides of the link in the vain hope that it finds a way to not transcode the entire way. |
| 11:18.51 | Greenlight | I'm lucky; everything's alaw over here |
| 11:19.34 | Chainsaw | I do find the ulaw vs alaw naming is the other way round then I'd expect. (The "a" for the American way) |
| 11:21.12 | Greenlight | Indeed, it's counter intuative |
| 11:22.02 | Chainsaw | Greenlight: I should have waved by the way, on my way back from Inverness. |
| 11:22.19 | Greenlight | Ahh you stayed up there for the weekend ? |
| 11:22.26 | Chainsaw | Greenlight: Whole week :) |
| 11:22.35 | Greenlight | Ahh cool - hope it was nice |
| 11:22.38 | Chainsaw | Getting some rest at the end of the world. |
| 11:22.46 | Chainsaw | Definitely. Even had good weather. |
| 11:22.50 | Greenlight | Wow |
| 11:23.28 | Chainsaw | Well, we had to walk back a few miles totally soaked one day. We forgot our enchanted umbrella. |
| 11:23.41 | Chainsaw | (You don't even have to open it. But don't ever leave it behind.) |
| 11:23.57 | Greenlight | Enchanted by Sod's Law ? :) |
| 11:24.26 | Chainsaw | I thought by faeries? |
| 11:24.28 | WIMPy | Only tourists carry umbrellas with them. |
| 11:24.48 | WIMPy | The locals know they don't work in heavy wind. |
| 11:24.56 | *** join/#asterisk davlefouAMD (~david@41.227.50.138) |
| 11:25.32 | Chainsaw | WIMPy: There wasn't much wind though. It was raining straight down, not sideways. |
| 11:25.51 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 11:26.09 | WIMPy | Downward rain only happens if Xmas and easter fall on the same day here. |
| 11:26.27 | Chainsaw | wonders where WIMPy is |
| 11:27.18 | Chainsaw | A little cottage deep in the black forest perhaps. |
| 11:28.14 | WIMPy | The "Land between the seas". At a fjord of the baltic sea, but the north sea is less than 50km away as well. |
| 11:28.21 | WIMPy | That always makes for good wind. |
| 11:28.24 | Chainsaw | Ah yes, that wouldn't help. |
| 11:28.50 | Greenlight | "The Land Between The Seas" -- sounds very Game of Thrones esque |
| 11:29.06 | Chainsaw | Totally does. |
| 11:30.05 | WIMPy | Schleswig-Holstein |
| 11:31.47 | Chainsaw | Must have good beer. |
| 11:32.21 | Chainsaw | Greenlight on the other end is more in a whisky area. |
| 11:32.49 | WIMPy | The one with the famous swing top. |
| 11:34.14 | *** join/#asterisk TobSnyder (~schneider@146-52-43-241-dynip.superkabel.de) |
| 11:34.17 | bulkorok | Flens?! |
| 11:34.35 | WIMPy | exactely |
| 11:34.37 | Chainsaw | other end? hand! |
| 11:34.43 | Chainsaw | really must learn to type properly one day |
| 11:35.14 | WIMPy | certainly perfers whisky. |
| 11:44.23 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:49.33 | apurvtwr | ok.. I have checked gateway codecs as well |
| 11:49.34 | *** part/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 11:50.32 | apurvtwr | gateway uses alaw for this PRI everytime (with or without noise ) |
| 11:50.54 | apurvtwr | Gateway and asterisk are communicating over ulaw in both cases. |
| 11:51.16 | Greenlight | Capture the packets |
| 11:51.29 | Greenlight | Throw it into wirehshark, listen to the streams |
| 11:51.42 | Chainsaw | With the utmost respect apurvtwr, Greenlight needs raw data, not your interpretation of it. |
| 11:51.58 | Greenlight | Also, are you allowing direct media ? |
| 11:52.22 | apurvtwr | thanks Chainsaw . :) |
| 11:52.53 | Greenlight | Yup - an actual call trace would help :) |
| 11:54.02 | apurvtwr | ok. I am sharing that with you. |
| 11:54.41 | Greenlight | ~pb |
| 11:54.41 | 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. |
| 12:00.27 | *** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz) |
| 12:08.30 | asghar144 | anyone wish to test patch for asterisk 1.8 for mysql cdr backend, it enable insert cdr in any timezone. |
| 12:12.07 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 12:20.50 | *** join/#asterisk paule32 (~paule32@dslb-188-106-253-140.pools.arcor-ip.net) |
| 12:22.34 | apurvtwr | Greenlight: pastebin wasn't allowing more than 500 KB per paste.. so I am sharing the full logs of two calls in this link |
| 12:22.35 | apurvtwr | https://www.dropbox.com/s/fpouh0j9d5ya9ue/full |
| 12:22.37 | *** join/#asterisk vlad_starkov (~vlad_star@195.68.180.118) |
| 12:23.39 | apurvtwr | 192.168.1.7 is asterisk server and 192.168.1.55 is my gateway |
| 12:23.45 | *** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net) |
| 12:24.44 | apurvtwr | in the middle of the logs there is line "Call with noise starts here" , after which are the logs of the call that is noisy |
| 12:25.47 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:25.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:28.51 | apurvtwr | Greenlight: are you able to download the file from the link? |
| 12:33.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:37.28 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 12:49.36 | apurvtwr | Greenlight: are you there? :) |
| 12:51.37 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:54.30 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 12:55.54 | *** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net) |
| 12:57.56 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:00.48 | *** part/#asterisk ManxPower (~manxpower@ip98-183-25-31.pn.at.cox.net) |
| 13:04.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:07.47 | *** join/#asterisk jsjc (~Adium@123.157.5.58) |
| 13:09.15 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 13:26.27 | vlad_starkov | Question: Say I run Originate(SIP/1234,exten,my_context,100,1) in dialplan. Is it possible to pass some channel variables to "SIP/1234" peer to make it possible to receive it at "my_context,100,1" after "SIP/1234" is being answered? |
| 13:26.43 | *** join/#asterisk serafie (~erin@nat/digium/x-cpuehfmhvvaecccd) |
| 13:28.06 | [TK]D-Fender | vlad_starkov: No. use a call-file or AMI originate instead as those support it |
| 13:28.34 | vlad_starkov | [TK]D-Fender: thanks! |
| 13:30.53 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 13:31.38 | hjf | [Sep 16 10:30:41] NOTICE[102760]: chan_sip.c:15071 sip_reg_timeout: -- Registration for 'xxxx@xxxx.226.247' timed out, trying again (Attempt #2608).226.247' timed out, trying again (Attempt #2608) |
| 13:31.52 | hjf | i get that error all the time (well, 2608 times as you can see |
| 13:31.56 | *** join/#asterisk danjenkins_ (~danjenkin@213.106.234.250) |
| 13:31.59 | hjf | but i can place calls through that ITSP |
| 13:33.56 | [TK]D-Fender | hjf: Because registration has nothing to do with sending authed calls out a peer |
| 13:34.20 | [TK]D-Fender | hjf: And you should really, really be looking at the SIP debug of your registration attempt |
| 13:34.48 | hjf | hmmm... it was the router. very weird. there was a conntrack entry with type SIP there. i deleted it, and asterisk registered |
| 13:35.58 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 13:36.29 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zgukwrsbantpxfiv) |
| 13:36.29 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:36.33 | Chainsaw | hjf: Ask jkroon about SIP and conntrack. You basically do not want it. |
| 13:36.52 | Chainsaw | hjf: Not when you have a SIP server behind the NAT router. It is very client-orientated and will get it wrong. |
| 13:37.23 | PLMg | hello, can anyone help me with a bit of scripting? (1-2 lines) |
| 13:38.03 | WIMPy | I think that's likely been due to it running out of memory. |
| 13:38.29 | hjf | Chainsaw: well, i need conntrack if i'm using NAT anyway, no? |
| 13:38.32 | [TK]D-Fender | PLMg: describe what you need |
| 13:38.39 | WIMPy | The conntrack has wrked fine for me. |
| 13:38.40 | [TK]D-Fender | hjf: No/. |
| 13:38.41 | PLMg | need to edit fax-process.pl to change the way emails are sent. (subject...) |
| 13:39.15 | [TK]D-Fender | PLMg: Show us the code (keeping in mind this isn't an Asterisk issue) |
| 13:39.43 | hjf | [TK]D-Fender: from my router's manufacturer "By disabling the conntrack system you will lose functionality of the NAT and most of the filter and mangle ..." |
| 13:40.08 | [TK]D-Fender | hjf: And how much of that do you need? |
| 13:40.42 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:40.42 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.12 | *** join/#asterisk davlefouAMD (~david@41.227.50.138) |
| 13:42.10 | PLMg | need a few minutes, seems like I have no editor installed. And also thx for the help even if this is not an asterisk issue |
| 13:49.09 | PLMg | http://pastebin.com/VBLdUmJe |
| 13:50.12 | PLMg | what I need is for the number the fax was sent to be included in the message |
| 13:50.16 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 13:50.22 | *** join/#asterisk davlefouAMD (~david@41.227.50.138) |
| 13:51.07 | hjf | I found the problem. the internet went out during the night and the router failed over to the secondary connection. conntrack was still tied to the secondary IP |
| 13:51.39 | hjf | so i added a little script to purge SIP conntrack entries when there's a failover |
| 13:52.46 | WIMPy | That's the difference between SNAT and MASQUERADE. |
| 13:52.52 | [TK]D-Fender | PLMg: The destination number for this fax is ".$dest." |
| 13:52.59 | [TK]D-Fender | PLMg: looks like it's there already |
| 13:53.50 | PLMg | uhm yeah, I do recive emails that are send to me via fax, but they do not show from who |
| 13:53.52 | PLMg | what number |
| 13:54.03 | PLMg | they all say my fax number |
| 13:55.08 | Penguin | Edit your command to include the callerid number of the sender. |
| 13:55.25 | PLMg | ok... and how do I do that? :) |
| 13:55.34 | PLMg | I mean, what do I type and where |
| 13:55.34 | [TK]D-Fender | plnYou said "sent TO", not "sent FROM" |
| 13:56.39 | Penguin | the number the fax was sent ... to be included |
| 13:56.42 | Penguin | Unclear. |
| 13:57.53 | PLMg | can you give me an example? I didn't find sent from |
| 13:58.01 | PLMg | I really suck at this, sry |
| 13:58.40 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-68-145-81.home.otenet.gr) |
| 13:58.58 | PLMg | under #default parameters the $to = "noreply.... this line? |
| 14:00.01 | *** join/#asterisk OS_Florent2 (~chatzilla@62.244.88.2) |
| 14:01.58 | OS_Florent2 | hi, how can i redirect the 2 leg of call bridged via bridge application after a defined time (in ms or second) ? |
| 14:02.25 | PLMg | I think I am not expresing myself right. So I have a fax, people send me faxes I recive them in tiff format. Those files are sent to a email adress but in the email it says they come from my fax number. Not the senders fax number. |
| 14:02.51 | davlefouAMD | hi, some one have use SipAddHeader with sflphone. I create an in my dialplan but i don't how to use it with commande line of anrochage. |
| 14:04.49 | [TK]D-Fender | davlefouAMD: please rephrase... |
| 14:06.11 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-zumhjauzqrlnwuzi) |
| 14:06.11 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:07.12 | *** join/#asterisk Draecos (~Draecos@58-7-130-209.dyn.iinet.net.au) |
| 14:10.36 | *** join/#asterisk roderickm (~roderickm@67.63.143.254) |
| 14:16.11 | *** join/#asterisk Geek-Linux (~mubbashir@static-host210-2-165-210.link.net.pk) |
| 14:17.27 | Geek-Linux | Hello every one, how can be the concurrent calls limit /sec set in asterisk can any one guide me ? |
| 14:17.59 | Greenlight | There is no hard "limit". It depends on a vast number of factors. |
| 14:18.12 | Greenlight | Or, you mean to set a limit? |
| 14:18.18 | Penguin | Are you trying to set a limit |
| 14:18.19 | Penguin | ? |
| 14:18.19 | Geek-Linux | means in a same second at a time , how many calls can be accepted. |
| 14:18.23 | Geek-Linux | yes |
| 14:18.34 | Penguin | You'll have to write it in dial plan. |
| 14:18.48 | Greenlight | Most likely with a DB backend to a dialplan script |
| 14:18.55 | Geek-Linux | because when a bulk of calls land on my servers at the same time they crash and restarts |
| 14:19.11 | Penguin | Why would you need a db or a script? |
| 14:19.36 | Geek-Linux | yes i have writed it by groupcount and hangup the calls if limit exceeds but it is for simultaneous calls |
| 14:19.42 | Penguin | I would probably use channel groups and something to check the time. |
| 14:19.53 | Penguin | Check the time, too. |
| 14:20.06 | Penguin | Not just the count, but also the time. |
| 14:20.10 | Geek-Linux | time in which sense ? |
| 14:20.18 | Greenlight | The "per sec" bit |
| 14:20.41 | Greenlight | But... how many calls are being dumped on your server ? |
| 14:20.42 | Geek-Linux | means in a same condition call + time |
| 14:21.01 | Geek-Linux | it has limit of 240 calls |
| 14:21.11 | Geek-Linux | means 8 E1s |
| 14:21.12 | Greenlight | Ok, so it should not be restarting. |
| 14:21.16 | Penguin | I would probably use STRFTIME unless I thought of something better. |
| 14:21.37 | Greenlight | Which version of Asterisk are you running ? |
| 14:21.47 | Geek-Linux | 1.6.2.6 |
| 14:22.06 | Greenlight | That's unsupported now, and quite old. |
| 14:22.14 | Greenlight | ~version |
| 14:22.14 | infobot | rumour has it, version is for the kernel "uname -r", for your distro "cat /etc/*-release" or "lsb_release -d". For other applications, try running it with a --version command. |
| 14:22.22 | Greenlight | ~upgrade |
| 14:22.22 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 14:22.23 | [TK]D-Fender | And far from even the latest in that branch |
| 14:22.25 | Geek-Linux | all the servers were stable a month ago but now they and restarting after every 10 minutes |
| 14:22.39 | Greenlight | Bleh infobot |
| 14:22.46 | Penguin | Unless we're asked about a PROBLEM with asterisk, why is the version important? |
| 14:23.05 | Greenlight | Sorry, random restarts sounded like a problem to me. My bad. |
| 14:23.38 | Greenlight | It's not like the whole SIP stack has been re-written since 1.6 |
| 14:23.49 | Penguin | The only thing I got was a request to write some dial plan. |
| 14:24.01 | Penguin | I don't see anything about restarting anything in that question. |
| 14:24.11 | file | Greenlight, I see what you did there. |
| 14:24.24 | Greenlight | :) |
| 14:24.46 | Penguin | There it is. I see it now. "crash and restarts" |
| 14:25.26 | Greenlight | I'm not one of the "OMG YOU MUST UPGRADE" fanbois, but if he's getting restarts randomly, then it's surely a good start.. |
| 14:25.32 | Geek-Linux | We are using TDM and then IAX. and i observe a WARNING. exceptionally long voice queue and they suddenly restart. |
| 14:25.42 | Greenlight | Ahh that error |
| 14:25.44 | Penguin | It somehow got lost in the scrolling. |
| 14:25.52 | *** join/#asterisk danjenkins (~danjenkin@213.106.234.250) |
| 14:26.21 | Geek-Linux | yup but still no answer on the forums. even no reason. |
| 14:26.38 | Geek-Linux | i was in doubt on network but networks seems good |
| 14:26.51 | Greenlight | I beleive it's a deadlock that causes it |
| 14:26.54 | file | it means that the thread which is supposed to read in the media is somehow locked or otherwise doing something else |
| 14:27.01 | file | why that occurs in your situation, who knows |
| 14:27.36 | Geek-Linux | thats why tried to ask you all about, any ideas or reasons |
| 14:27.54 | Greenlight | Yea, you're running and old version with lots of since-fixed bugs. |
| 14:28.18 | davlefouAMD | [TK]D-Fender, i have that i can create an headers message for my softphone: http://the-asterisk-book.com/1.6/applikationen-sipaddheader.html |
| 14:28.43 | Qwell | ~upgrade asterisk |
| 14:28.43 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 14:28.46 | davlefouAMD | and i want get the message in softphone who is sflphone. |
| 14:28.46 | asghar144 | anyone wish to test patch for cdr mysql backend that enable insert cdr in mysql in any timezone, patch is for asterisk-1.8.23.1 |
| 14:28.48 | Greenlight | You're getting a deadlock somewhere, but I doubt anyone is going to give any time to help fixing a deadlock on a non-current version |
| 14:28.54 | Penguin | If you've got bugs, you definitely need to upgrade to see if the problem goes away. |
| 14:28.59 | Geek-Linux | can it be the issue of communication of old version with the new ones. because i am using 1.6.2.6 + 10.12.1 + 10.5.1 |
| 14:29.08 | [TK]D-Fender | davlefouAMD: What "message"? |
| 14:29.26 | Greenlight | Geek-Linux: It's hard to say. |
| 14:29.35 | Penguin | With IAX2? Could be. |
| 14:29.35 | [TK]D-Fender | davlefouAMD: Sounds like you need to learn what your softphone supports... this does not appear to be an * issue |
| 14:29.47 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 14:29.56 | davlefouAMD | [TK]D-Fender, ok, i ll look about that. |
| 14:30.24 | WIMPy | Geek-Linux: I can't remember when it was, but there was a version of IAX that didn;t want to play nicely with other versions. |
| 14:30.46 | WIMPy | I'd guess in the 1.6.2 area. |
| 14:31.33 | Geek-Linux | WIMPY: hmmm i am realy frustrated by this issue. |
| 14:31.36 | Penguin | It used to be recommended that both sides of an IAX2 trunk were the same asterisk version. I assume that recommendation remains today. |
| 14:32.02 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 14:32.38 | WIMPy | I don;t see why they should be the same. I've been using it between different versions all the time. But at one time there has been an issue in doing so. |
| 14:32.42 | Greenlight | Geek-Linux: I never much liked the 1.6 versions; current 11 is really nice. It would thourghly recommend it |
| 14:32.55 | Geek-Linux | OK: you know this will make me to upgrade my OS version too. and which will be a headache :( |
| 14:33.24 | WIMPy | Yes. 1.6.* was bad. Use at least the latest 1.8, better the latest 11. |
| 14:33.53 | Geek-Linux | Greenlight: what would you recommend the OS verison. debian wheezy or squeez. |
| 14:33.54 | [TK]D-Fender | Geek-Linux: How would this have anything to do with your OS? |
| 14:34.16 | Greenlight | Personally we always use CentOS. Currently 6. |
| 14:34.36 | Greenlight | I'm not sure about Debian, or why you'd *need* to even upgrade it. |
| 14:34.53 | Greenlight | If anything, I guess, ensure it supports timerfd |
| 14:37.34 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-zytsjnwtokclzcwx) |
| 14:41.00 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 14:41.55 | Geek-Linux | GreenLight: i am checking for the support timerfd but what it is used for |
| 14:42.52 | Greenlight | Anything that needs a timing source |
| 14:43.00 | Geek-Linux | GreenLight: because all the softwares used for setting needs latest kernal to work fine. |
| 14:43.16 | Greenlight | used for setting ? |
| 14:43.49 | Geek-Linux | setting for Asterisk machine including TDM, wanpipe, dahdi, ss7 |
| 14:43.59 | Qwell | ...says who? |
| 14:44.33 | Greenlight | I've installs running fine on 2.5 kernels |
| 14:44.45 | Geek-Linux | me :) because i have experiece when moved my some server latest release on old OS |
| 14:45.24 | Qwell | Your experience is wrong. |
| 14:45.38 | *** join/#asterisk Pullphinger (~Pullphing@12.40.23.68) |
| 14:46.15 | Geek-Linux | OK. |
| 14:46.21 | Geek-Linux | :( |
| 14:47.39 | Kobaz | the OS is more than just the kernel |
| 14:47.51 | [TK]D-Fender | 2.5 ...? |
| 14:48.05 | Penguin | Upgrading the kernel is sometimes useful. |
| 14:48.15 | Penguin | No one uses 2.5 kernel. |
| 14:48.49 | Greenlight | I'm pretty sure CentOS5 is still 2.5 |
| 14:49.03 | Qwell | No distros used 2.5. |
| 14:49.04 | Greenlight | ALthough I've not access handy to any boxes running it |
| 14:49.19 | Penguin | 2.5 has never been released publicly. |
| 14:49.20 | *** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:91:a4bb:9a23:9bc8) |
| 14:49.36 | Greenlight | I know that CentOS5 kernels didn't support timerfd, and CentOS6's kernel did |
| 14:49.47 | Greenlight | ANd I know CEntOS6 is 2.6 |
| 14:49.51 | Penguin | 2.4, 2.6, 3.0 |
| 14:50.23 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 14:50.43 | Penguin | I still remember using 2.2. |
| 14:51.27 | Kobaz | 2.2 ? |
| 14:51.36 | Kobaz | i remember building 1.2.13 |
| 14:51.49 | Penguin | I haven't used Linux for THAT long. |
| 14:51.52 | Kobaz | that kernel had the famous ld_preload remote root exploit vulnerability |
| 14:52.03 | Penguin | I didn't start with Linux until around 2000. |
| 14:52.14 | Kobaz | 95/96 for me |
| 14:52.44 | Kobaz | i remember installing slackware 96, after having downloaded the whole cd image on my 14.4 |
| 14:52.54 | Kobaz | it was much newer than the slackware that came with my 'linux unleashed book' |
| 14:53.02 | Penguin | My first Slackware was 8.0. |
| 14:53.09 | Greenlight | Downloading CD images over 14.4 :) |
| 14:53.23 | Greenlight | Tieing up the phone line for days |
| 14:54.03 | Kobaz | Slackware 3.1, released in July 1996, shipped with Linux kernel 2.0.0 and was called "Slackware 96" in allusion to Windows 95 |
| 14:54.09 | Kobaz | we had a second phone line just for data |
| 14:54.25 | Kobaz | so it was on 24/7 downloading for 3 weeks to get that image, hah |
| 14:54.36 | Greenlight | We eventually got ISDN2e (?) for data |
| 14:54.41 | Greenlight | Wow was it QUICK |
| 14:56.21 | Kobaz | i should have just ordered the cd for $2 |
| 14:56.22 | fullstop | Kobaz: I downloaded every floppy image for that, wrote them all to disks.. just to find out that it didn't support my scsi controller. |
| 14:56.41 | Kobaz | probably took $50 in electricity to download |
| 14:56.41 | Greenlight | hehe |
| 14:58.07 | fullstop | leonard zubkoff eventually wrote the scsi driver the next year, and I was able to install. |
| 14:58.19 | fullstop | but the poor guy died in a helicopter crash a few years later. :( |
| 14:59.39 | Kobaz | aw |
| 15:04.30 | Qwell | New policy: People in the Open Source community aren't allowed to die anymore. |
| 15:04.35 | Qwell | glares at lilo and skvidal |
| 15:04.55 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
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| 15:14.51 | hjf | stupid freenode |
| 15:14.59 | hjf | anyway |
| 15:15.38 | hjf | i can download the config file and it looks like this: |
| 15:15.38 | hjf | interface GigabitEthernet1/0/2 port link-type hybrid port hybrid vlan 1 untagged |
| 15:17.16 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.7.96) |
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| 15:37.34 | pabelanger | Anybody using cisco SG300-20 on their voip networks? |
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| 15:52.03 | *** join/#asterisk devdvd (~Melissa@nc-184-3-20-206.dhcp.embarqhsd.net) |
| 15:55.54 | devdvd | hey, im looking for a good wisip phone that can easily go between different access points. it doesn't have to maintain it's connection between access points, just needs to be able to pick one up and use it. I currently have a spectralink 8002 and it's horrible so that one is out. Any recommendations. I would also be fine with a regular cordless phone with multiple sip base stations but it would need the same extension |
| 15:56.00 | devdvd | any recommendations? |
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| 16:49.51 | vlad_starkov | Question: Is it possible to use pound '#' key in confbridge.conf? |
| 16:50.20 | vlad_starkov | now it returns WARNING[894]: config.c:1362 process_text_line: Unknown directive '#=no_op' at line 76 of /etc/asterisk/confbridge.conf |
| 16:50.55 | Greenlight | '#' isnt a config option though? |
| 16:51.16 | Greenlight | From what I remember of confbridge.conf |
| 16:51.36 | vlad_starkov | I'd like to use '#' in [menu] context |
| 16:51.41 | Greenlight | Ahh in the menu |
| 16:51.49 | newtonr | vlad_starkov, https://issues.asterisk.org/jira/browse/ASTERISK-22478 |
| 16:52.32 | newtonr | Appears that you cannot, currently. The # gets processed as a directive, like for includes |
| 17:08.38 | vlad_starkov | newtonr: Thanks |
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| 17:44.28 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 17:46.04 | asghar144 | anybody wish to test patch against cdr mysql backend fot configurable cdr timezone? |
| 17:51.53 | *** join/#asterisk felipealmeida (~user@177.40.162.84) |
| 17:51.57 | boom^time | Hey guys, is there a good PHP to AMI set of tools or are you better off just writing them yourself? |
| 17:53.09 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 17:55.27 | [TK]D-Fender | boom^time: php-agi library is commonly used and jsut about everyone who uses it is just fine with it |
| 18:00.48 | drmessano | boom |
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| 18:02.15 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 18:06.44 | Cubber | can anyone recommend a reliable SIP trunk service to replace my now useless google voice trunks? I require PSTN access in the US. |
| 18:06.47 | boom^time | [TK]D-Fender, thanks but that is for AGI right? I'm looking for AMI communication. |
| 18:08.06 | boom^time | Cubber, I haven't tried them yet but someone here recommended them to me and I'm going to give them a shot here soon: http://voip.ms/ |
| 18:08.19 | boom^time | actually I think the bot will give you a list |
| 18:08.21 | boom^time | ~itsp |
| 18:08.21 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 18:08.29 | boom^time | ~itsplist-us |
| 18:08.30 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 18:08.32 | ChannelZ | I think there's some AMI handling things in php-agi |
| 18:10.59 | Cubber | boom^time: thanks |
| 18:11.00 | *** join/#asterisk felipealmeida (~user@177.40.162.84) |
| 18:12.34 | boom^time | ChannelZ, I'm confused on how to use it. With AGI the PHP is executed from within the dial-plan. While with what I'm doing the AMI will need to be called from within the PHP. |
| 18:13.29 | [TK]D-Fender | boom^time: php-agi has AMI methods in there |
| 18:14.48 | ChannelZ | It's just a toolkit. Presumably you would be launching your AMI script from the shell with 'php' |
| 18:15.07 | ChannelZ | You just use their functions/callbacks/however it's implemented instead of writing your own IO |
| 18:21.09 | *** join/#asterisk krunal (ca83777a@gateway/web/freenode/ip.202.131.119.122) |
| 18:21.17 | krunal | Hello guys |
| 18:21.50 | krunal | I am getting a weird issue with asterisk Asterisk 10.12.2 |
| 18:22.07 | krunal | whenever any sip extension tries to registere there |
| 18:22.07 | drmessano | Is it screaming that it's unsupported? |
| 18:22.09 | drmessano | Stab it |
| 18:22.43 | krunal | getting Correct auth, but based on stale nonce received from |
| 18:22.53 | krunal | and no extensions are getting registered |
| 18:23.00 | krunal | what should be wrong with it? |
| 18:23.33 | krunal | please help me out |
| 18:30.56 | krunal | is anybody there to help me out? |
| 18:31.32 | [TK]D-Fender | krunal: 10.12.3 is out, and that entire branch is only getting security fixes at this point, and not for long. We highly recommend you upgrade |
| 18:31.55 | newtonr | krunal, try Googling "asterisk stale nonce" there is quite a lot of reading on it from discussions that have already been had |
| 18:32.39 | krunal | thanks for the update |
| 18:32.40 | *** join/#asterisk ledoktre (c766d0a9@gateway/web/freenode/ip.199.102.208.169) |
| 18:36.15 | ledoktre | Greetings. I've got a quick question regarding the use of the : and =~ operators. I am trying to do a gotoif based on if the number returned contains a number. For example : GotoIf($["${CALLERID(NUM)}" : "*.[5551212]+)"]?true) . I am sure I've got a typo, but can't seem to nail it. I did not include the area code on that number because some calls if they are local seem to come in without showing the area code in the CID. He |
| 18:37.29 | ledoktre | Oops - typo there. On the second part after : "*.[5551212]+" (I read the () is required if you want to see the return value. I just want true or false. Tried it with and without () marks. |
| 18:39.33 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 18:42.25 | boom^time | I see nothing in any documentation or even on during a google search that claims you can use phpagi to originate via AMI |
| 18:42.41 | boom^time | I must be missing something... |
| 18:49.11 | ledoktre | anyone here use : and =~ for regex ? |
| 18:57.52 | *** join/#asterisk navaismo (~navaismo@189.241.77.253) |
| 18:58.58 | Penguin | ledoktre: I've never tried to use : in a GotoIf(). Try the equal sign like everyone else uses. Also, take a look at the REGEX() function. |
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| 19:13.41 | *** join/#asterisk ageis (kevin@ageispolis.net) |
| 19:14.20 | ageis | can anyone point out what's causing this "syntax error, unexpected '=', expecting $end" |
| 19:14.26 | ageis | here's the line: exten => h,2,GotoIf($[${REC} = 1]?recorded,s,1) |
| 19:14.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.219) |
| 19:19.54 | Penguin | ageis: Is ${REC} a null value? |
| 19:21.26 | ageis | do I just need quotes? |
| 19:21.48 | Penguin | If it can be null, make it not a null value. Use double quotes on both sides of the comparison or add a static bit of data on both sides, such as $[x${REC} = x1]. |
| 19:22.12 | ageis | yeah, I think I just need quotes on both sides. thanks |
| 19:22.14 | Penguin | The quotes will also be compared, so you have to use them on both sides. |
| 19:23.40 | ageis | so it would be like so: GotoIf($["${REC}" = "1"]?recorded,s,1) |
| 19:24.01 | Penguin | Looks okay to me... if a null value was the problem. |
| 19:24.51 | Penguin | Now it can compare "" to "1" in the event the variable has no data. |
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| 19:38.38 | rrittgarn | afternoon gentlemen, anybody ever work with Cetis 2800/2802 series phones before |
| 19:41.05 | ChannelZ-Wk | never even heard of 'em |
| 19:41.41 | ncrollo | Call 1 enters the queue and is answered by Agent A. Call 2 then enters the queue and rings all phones except Agent A. Agent A then ends call 1 and should be able to immediately answer call 2 |
| 19:41.50 | ncrollo | is this possible or am I crazy? |
| 19:41.52 | ChannelZ-Wk | (their website is completely busted) |
| 19:44.01 | navaismo | ncrollo, is possble, retry & wrapuptime are the hints |
| 19:44.39 | ncrollo | thats what I thought... |
| 19:46.01 | ncrollo | strategy=ringall; ringinuse = no; wrapuptime=1; timeoutrestart = yes ;timeout=45 ;retry = 1 |
| 19:46.09 | rrittgarn | @ChannelZ yeah... used google cache to find their number just now... they are hotel style phones. |
| 19:47.24 | ChannelZ-Wk | Do you have some that aren't working or were you asking in general to see if they did? |
| 19:48.02 | rrittgarn | bidding on a job that wants to re-use them... needed to know if they were analog or digital with some form of a controller |
| 19:48.12 | rrittgarn | ended up finding a cached version of their site that had their number and called |
| 19:48.25 | rrittgarn | answer was purely analog, also branded Telematrix |
| 19:49.13 | ChannelZ-Wk | ah. |
| 19:49.15 | navaismo | ncrollo, retry must be setted to 1 aswell to start ringing again all members |
| 19:49.34 | [TK]D-Fender | [15:41]ncrollois this possible or am I crazy? <- not, it is not possible |
| 19:49.56 | ncrollo | navaismo, it is set to 1 |
| 19:50.28 | navaismo | oh i think the ; was actullay a ; in the file |
| 19:51.24 | ncrollo | no, just a bad new line seperator decision ^.^ |
| 19:56.32 | Penguin | Do you have two separate queues for this? |
| 20:00.35 | ncrollo | no its one queue |
| 20:01.05 | [TK]D-Fender | ncrollo: Three is no ability to immediately answer that waiting call. |
| 20:01.22 | Penguin | How do you ring only one agent with a ringall strategy? |
| 20:01.39 | [TK]D-Fender | Penguin: Uniquely :) |
| 20:02.00 | Penguin | Very strange. |
| 20:02.15 | ncrollo | there are 4 agents, 1 main operator and 3 other "part time" operators, right now the operator gets off a call and either has to wait for the timeout (45) seconds or run over to another phone |
| 20:02.41 | [TK]D-Fender | ncoCorrect. That is "how it is" |
| 20:02.49 | [TK]D-Fender | ncrollo: Correct. That is "how it is" |
| 20:04.19 | rrittgarn | F1 is acting up again |
| 20:04.24 | rrittgarn | getting no-dial tones on a few lines |
| 20:04.46 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 20:05.24 | ncrollo | [TK]D-Fender, then what is timeoutrestart, retry and wrap-up time for? |
| 20:05.57 | [TK]D-Fender | ncrollo: Wrap up cuts you out from distribution when distribution happens |
| 20:06.22 | ncrollo | which i understand. |
| 20:06.30 | [TK]D-Fender | ncrollo: if there is a distribution in progress and then 1 phone newly becomes availa it will NOT get "added" to the ringing bunch. |
| 20:06.45 | ncrollo | but and that makes sense |
| 20:06.53 | [TK]D-Fender | ncrollo: the ones chosen are at the start and if another becomes available 1 second in, too bad |
| 20:07.07 | Penguin | Isn't wrapuptime for how long the phone remains unavailable after it has ended its call? |
| 20:07.12 | Penguin | s/phone/agent/ |
| 20:07.34 | [TK]D-Fender | [16:07]infobotPenguin meant: Isn't wrapuptime for how long the agent remains unavailable after it has ended its call? <- yes |
| 20:07.46 | [TK]D-Fender | which is counted agains the next distribution start |
| 20:07.55 | ncrollo | yeah and wrapup time actually makes sense |
| 20:08.26 | ncrollo | but from what I rad on asteriskinfo "agent to answer is reset if a BUSY or CONGESTION is received" |
| 20:08.58 | ncrollo | so if Agent A is on a call it sends a BUSY |
| 20:09.05 | Penguin | Are you using the agent channel or some other channel tech for members? |
| 20:09.09 | [TK]D-Fender | that sentence does not make sense.. and could you clarify that source? |
| 20:09.19 | [TK]D-Fender | What is "asteriskinfo"? |
| 20:10.25 | ncrollo | http://www.voip-info.org/ |
| 20:10.48 | Penguin | I wouldn't have guessed that. |
| 20:10.53 | ncrollo | specifically http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 20:15.35 | ncrollo | what I was trying to say is from what I read on voip-info.org timeoutrestart is the critical part. When Agent A sends a busy the time out for an agent to answer is reset |
| 20:16.12 | Penguin | (1509.05) <Penguin> Are you using the agent channel or some other channel tech for members? |
| 20:22.56 | ncrollo | Penguin, I'm not sure I understnad that |
| 20:23.10 | ncrollo | I have static member extensions |
| 20:23.31 | Penguin | member => Agent/123? member => Local/123@agents? |
| 20:23.57 | Penguin | or the worst, member => SIP/123? |
| 20:24.55 | navaismo | use the worst, A L W A Y S |
| 20:25.27 | ncrollo | lol |
| 20:25.28 | ncrollo | member => SIP/201122 |
| 20:25.40 | ncrollo | yeah I'm using the worst :/ |
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| 22:15.44 | riantsoa | hi all |
| 22:15.51 | ChannelZ-Wk | waves his boobs |
| 22:16.03 | riantsoa | lol |
| 22:16.58 | riantsoa | i have trouble with DTMF, i use chan dongle and when i press touch on my gsm phone asterisk cant recognise the DTMF |
| 22:17.49 | riantsoa | i connect succesfully to asterisk with my mobile phone via an huawei 1750 usb 3g dongle |
| 22:18.10 | riantsoa | but i cannot navigate on my IVR via the dtmf tone |
| 22:18.39 | *** join/#asterisk luke_ftw (~luke_ftw@unaffiliated/luke-ftw/x-9245329) |
| 22:19.10 | riantsoa | please, can somebody help me, i have spent 5 days trying to solve this |
| 22:21.37 | ChannelZ-Wk | well the gsm codec destroys DTMF as far as I know and has to be handled as data by the network, kinda like rfc2833 does |
| 22:22.17 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 22:23.24 | riantsoa | how can i do that on asterisk? |
| 22:24.29 | ChannelZ-Wk | I assume it's something either handled by chan_dongle by default or a config option.. but I've no idea |
| 22:26.59 | *** join/#asterisk Wiretap (~wiretap@unaffiliated/wiretap) |
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| 23:16.49 | *** join/#asterisk wafflejock_ (~quassel@c-24-14-54-125.hsd1.il.comcast.net) |
| 23:17.26 | wafflejock_ | ~itsplist-us |
| 23:17.26 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:21.50 | wafflejock_ | So I have a situation, I posted last night but didn't see a response so trying again. I was previously using Google Voice to connect to Asterisk (keep in mind I'm a complete noob here, but a developer). I'm wondering what I need now to get a DID for inbound calls so I can setup a conference call line (I also want to just be able to explore asterisk/freepbx for the sake of learning and being able to handle more complicated |
| 23:21.50 | wafflejock_ | deployments). So I see the ITSPs list but I don't know what the easiest/cheapest way to go about this is, or what it is I'm looking for really. Do I need SIP trunk service? what should I be searching for? |
| 23:22.15 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.12) |
| 23:22.23 | tm1000 | ~freepbx |
| 23:22.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 23:22.28 | Penguin | ~trunk |
| 23:22.28 | infobot | trunk is, like, a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 23:22.44 | *** join/#asterisk felipealmeida (~user@177.40.162.84) |
| 23:22.55 | wafflejock_ | Penguin: thx |
| 23:23.04 | wafflejock_ | tm1000: thx |
| 23:25.20 | wafflejock_ | So if I don't have a physical line coming into the place can I still configure Asterisk via just the internet to connect to one of these providers, would that be considered an IAX trunk? |
| 23:27.07 | tm1000 | wafflejock_: no SIP |
| 23:27.18 | tm1000 | I mean. sure you can get an IAX trunk…if you want... |
| 23:27.24 | tm1000 | mainly SIP though |
| 23:31.08 | wafflejock_ | So, general question, do most small/medium businesses who are using some sort of SIP service or Asterisk usually go to cloud services now or is it still beneficial to have an on site server for any particular reason? |
| 23:32.56 | ChannelZ-Wk | I suppose you save a leg of latency having it local. |
| 23:35.45 | *** join/#asterisk aypea[3] (~aypea@CPE-58-172-192-180.bqzk1.ken.bigpond.net.au) |
| 23:36.22 | aypea[3] | hi. trying to script usage of menuselect but I cannot figure out how to select app_meetme to be compiled. anyone able to help (point at a doc)? |
| 23:37.02 | ChannelZ-Wk | well you can configure things with makeopts |
| 23:38.13 | aypea[3] | hrm. unless I'm mistaken that has a dodgy feel to it :) |
| 23:41.08 | [TK]D-Fender | wafflejock_: A few reasons to have the server hosted externally, and probably a few more to have it locally. |
| 23:42.12 | aypea[3] | found an example on the internet. it seems it's what I thought it'd be. lets see how she flies. |
| 23:42.46 | [TK]D-Fender | aypea[3]: Menuselect will tell you what you're missing... |
| 23:42.57 | [TK]D-Fender | aypea[3]: What is pretty much ... DAHDI' |
| 23:43.25 | *** join/#asterisk jsjc (~Adium@123.157.5.58) |
| 23:43.37 | aypea[3] | yeah. it's more of a case of getting 11.5.1 to try and compile it since it's been undefaulted (and semi-deprecated). |
| 23:45.09 | aypea[3] | w00t! :) -rw-r--r-- root/root 170312 2013-09-17 09:43 ./usr/lib/asterisk/modules/app_meetme.so |
| 23:45.17 | aypea[3] | built and packaged. |
| 23:45.33 | wafflejock_ | [TK]D-Fender: Yeah I had previously played with the PIAF (PBX in a Flash) raspberry Pi image for running asterisk/freepbx just to check it out and like it all a lot in terms of all the configurability and could potentially code some stuff around the server. I'm interested in setting up a server for a small business as well who would like to have 5-15 offices setup with phones, but am looking to do some more experimenting on my |
| 23:45.33 | wafflejock_ | own before trying to deploy something for another business. Mostly I stick to web development work, but have a BS in CS from DePaul so I have some knowledge in the arena just nothing specific to telecom. It seems like all of the ITSPs just offer some sort of hosted deal, I'd really like to have the server local here just to learn on though. |
| 23:45.34 | aypea[3] | for the curious: menuselect/menuselect --enable app_meetme menuselect.makeopts |
| 23:47.59 | [TK]D-Fender | wafflejock_: Having your own server means not having to rent the service |
| 23:48.39 | [TK]D-Fender | wafflejock_: and having direct control over it. Also if your internet goes down your users can still call amongst each other and you could have alternate means of dialing out. |
| 23:48.56 | wafflejock_ | [TK]D-Fender: yeah so out of the list of ITSPs from this IRC channel this is the first one I've seen that seems to offer just the DID and line service separate from the cloud hosted virtual PBX http://vitelity.net/services_voip/ |
| 23:49.08 | wafflejock_ | [TK]D-Fender: thx for the pointers on the advantages too |
| 23:49.49 | [TK]D-Fender | wafflejock_: ALL of those providers offer staright services, not just "hosted: |
| 23:50.43 | [TK]D-Fender | infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:50.45 | [TK]D-Fender | ^ |
| 23:50.58 | [TK]D-Fender | All of them offer straight-up service |
| 23:51.08 | [TK]D-Fender | not tied to running hosted PBX |
| 23:51.36 | wafflejock_ | [TK]D-Fender: okay they are apparently just pushing the hosted service on their sites and I wasn't seeing the deal for just hooking up service, any one you think is particularly better personally? |
| 23:52.34 | [TK]D-Fender | wafflejock_: Flowroute has one of the better reps these days, Vitelity is pretty strong as well, and voip.ms is a vitelity reseller who is often a little cheaper, but don't support T.38 faxing. |
| 23:52.50 | [TK]D-Fender | So I'd probably look at Flowroute and vitelity first |
| 23:53.09 | wafflejock_ | [TK]D-Fender: awesome information thank you very much I'll look into those |
| 23:53.27 | [TK]D-Fender | Compare them all, but what I am giving you is based on a alrge volume of community feedback substantiated by people I respect here |
| 23:59.52 | *** join/#asterisk jasonwert (~w3rt@96-42-150-164.dhcp.trcy.mi.charter.com) |