| 00:00.01 | igcewieling | ChannelZ: if it isn't a typoe it would explain a few things. |
| 00:00.22 | igcewieling | carbinemonoxide: come back when you are not brain dead. |
| 00:00.29 | volga629 | Hello Everyone, having trouble with RTP stream from remote extension and registration working ok. |
| 00:00.50 | igcewieling | ~ask |
| 00:00.50 | 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. |
| 00:00.59 | carbinemonoxide | You've been very helpful but very impolite. You get my thanks but not much of it. |
| 00:01.27 | ChannelZ | volga629: still firewall/nat issues. |
| 00:01.30 | igcewieling | carbinemonoxide: the best advice you will get from anyone is to come back when you are not tired, frustrated, etc. |
| 00:02.17 | carbinemonoxide | Yes, but name calling and teasing is not very professional. It's fine that you're that way, just don't expect anyone to really think positively of you. |
| 00:02.23 | volga629 | possible, let me put debug. I want confirm that from asterisk box is ok. But in front I have fortigate and on debug I see traffic flow |
| 00:02.36 | carbinemonoxide | Let's not turn this channel into what it shouldn't be though. |
| 00:04.10 | ChannelZ | carbinemonoxide: we don't know anything about your network setup, haven't seen your configuration, any console output, or sip debug. It's hard to help. |
| 00:04.24 | ChannelZ | Start by getting a SIP device configured and working. |
| 00:04.31 | ChannelZ | Forget Google Voice. |
| 00:04.59 | carbinemonoxide | That's what I'm working on now. I'm not noticing anything in my logs about my SIP client even trying to connect to my asterisk server at this point. |
| 00:05.44 | ChannelZ | First, it's asterisk -rvvv (one r, more v's) to turn up the verbosity. |
| 00:06.26 | ChannelZ | What sip device are you using? A softphone on another machine on the same network, an actual sip phone, etc? |
| 00:06.57 | j4jackj | Does anyone know of any services other than google voice that allow free calling? |
| 00:07.15 | carbinemonoxide | I've tried with a softphone (Ekiga) and I also have an Aastra 9133i, no luck on either. |
| 00:07.50 | ChannelZ | Have you configured a peer for either in sip.conf? |
| 00:08.35 | carbinemonoxide | Yes, I have. One sec. |
| 00:09.21 | carbinemonoxide | http://pastie.org/8292924 |
| 00:10.38 | volga629 | here call debug https://fpaste.networklab.ca/8hYS/ getting retransmission after 7 sec in call |
| 00:10.59 | ChannelZ | So assuming you configure your device to talk to the IP of the machine asterisk is running on, with the name 'carbine' and whatever password, get it to register. You should see *something* on the console. |
| 00:11.02 | ChannelZ | Particular if you "sip set debug on" |
| 00:11.37 | carbinemonoxide | No such command 'sip set debug on' (type 'core show help sip set' for other possible commands) |
| 00:13.45 | carbinemonoxide | Hmm. |
| 00:13.52 | navaismo | uh, module load chan_sip.so |
| 00:14.20 | carbinemonoxide | Just tried that. No such file or directory. |
| 00:14.42 | j4jackj | OMG |
| 00:14.46 | ChannelZ | Then your asterisk build has failed |
| 00:14.47 | navaismo | ha your asterisk didnt compile chan sip |
| 00:15.10 | navaismo | so go back, and do ./configure then make menuselect |
| 00:15.26 | navaismo | and see if chan_sip is not XXX'ed |
| 00:16.34 | ChannelZ | volga629: I don't know why your device either isn't getting Asterisk's packets or if it is the replies aren't making it back |
| 00:16.55 | igcewieling | ChannelZ: I vote for externip= or SIP ALG |
| 00:18.43 | ChannelZ | could be, the device's IP is 100.x but they're coming from 216.x |
| 00:18.57 | *** join/#asterisk russum (~russum@ool-4574dc67.dyn.optonline.net) |
| 00:19.05 | ChannelZ | It sends the invite, gets denied, says 'OK' and re-tries witrh the correct auth.. but then stops replying |
| 00:20.06 | *** part/#asterisk russum (~russum@ool-4574dc67.dyn.optonline.net) |
| 00:21.22 | carbinemonoxide | Hah yeah it wasn't selected when I compiled it. |
| 00:22.15 | carbinemonoxide | I was working with 11.5.1 which had it selected by default. When it wasn't compiling I dropped down to 11.2-cert2 and out of habit I didn't even check. |
| 00:23.19 | ChannelZ | well it should be selected by default unless you're missing a dependency. |
| 00:27.50 | ChannelZ | volga629: Your device is contacting 99.x.x.222 (the external IP of your asterisk?) but it's replying as 10.x.x.50 |
| 00:28.47 | ChannelZ | Your device thinks it's 100.x.x.1 but is coming from 216.x.x.129 |
| 00:29.58 | ChannelZ | port numbers are all over the place |
| 00:32.07 | volga629 | because I am testing from cell which telus 3G |
| 00:32.16 | volga629 | on asterisk set externip= 99 |
| 00:33.00 | volga629 | I disabled on fortigate any possible stuff which related to sip |
| 00:33.19 | ChannelZ | But there's all sorts of munging going on by the looks of it and the packets are getting screwed up somewhere |
| 00:33.26 | volga629 | let me double check again all settings |
| 00:34.40 | volga629 | that might be possible because mobile provider android phone is on internal network. example 10.10.10.10 _----. public ip 216.... |
| 00:35.15 | ChannelZ | I have doubts your externaddr is set correctly because it's giving out a 10.x.x.x IP |
| 00:35.17 | volga629 | 99.x.x.222 mapped public ip |
| 00:35.44 | volga629 | yes, might softphone settings need to play |
| 00:36.03 | volga629 | I am testing zoiper and bria |
| 00:36.18 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 00:36.53 | volga629 | I have running stun server, but didn't tried use it ? |
| 00:37.22 | ChannelZ | stun will only help the remote device figure out what its own external IP is, if it even supports STUN |
| 00:37.52 | ChannelZ | but your own asterisk is giving out its LAN IP to remote clients, which isn't right. |
| 00:39.50 | volga629 | ok going check everything this need to be fixed |
| 00:40.58 | volga629 | ok look like I forgot sip reload |
| 00:41.04 | volga629 | I see right now o=root 1472667817 1472667817 IN IP4 99.224.107.222 |
| 00:41.06 | volga629 | s=Asterisk PBX 11.4.0 |
| 00:41.08 | volga629 | c=IN IP4 99.224.107.222 |
| 00:41.12 | ChannelZ | This also assumes your mobile provider doesn't block SIP in the first place |
| 00:41.41 | volga629 | it not I see traffic on fortigate |
| 00:41.46 | ChannelZ | that it isn't letting some stuff through but dicking with the INVITE acknowledgement or something |
| 00:42.19 | ChannelZ | A lot of providers (in the US anyway) frown upon doing voip over their data |
| 00:42.41 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:43.22 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 00:43.24 | volga629 | Is put scb in from asterisk might will be more helpful ? |
| 00:43.29 | volga629 | sbc |
| 00:43.38 | ChannelZ | huh? |
| 00:43.59 | volga629 | that will handle transcoding |
| 00:44.28 | ChannelZ | I don't know what you're talking about |
| 00:45.00 | *** join/#asterisk classix (~salven@silenceisdefeat.com) |
| 00:46.27 | volga629 | on rtp debug I see sent/received then Got RTP packet from 216.218.29.129:30706 (type 00, seq 007862, ts 2822462428, len 000160) |
| 00:46.30 | volga629 | [2013-09-02 20:45:34] WARNING[2354]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission |
| 00:46.53 | volga629 | How I can do better control on rtp range |
| 00:48.21 | ChannelZ | You can't control what port the remote end wants asterisk to send RTP to. |
| 00:48.58 | ChannelZ | You also don't have an RTP problem (yet) - Your device sends an INVITE. Asterisk replies 'Ok, let me try' and then follows up by saying 'OK, that worked, here's where you send RTP to me at.' But the device fails to respond 'OK' to Asterisk. |
| 00:49.50 | ChannelZ | Why, I don't know... either because the device isn't getting the packets, or it is and is trying to reply but its replies are not making it back to asterisk. |
| 00:53.16 | ChannelZ | It's curious that the initial INVITE conversation seems to happen fine (device sends INVITE, asterisk asks for auth, device says OK and tries again with a good auth response) but it breaks down afterwards. |
| 00:59.33 | volga629 | I am trying check on fortigate if any traffic get dropped |
| 01:00.06 | volga629 | if any thing was dropped that explain a lot |
| 01:03.47 | *** join/#asterisk classix (salven@silenceisdefeat.com) |
| 01:04.33 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 01:13.18 | volga629 | I see proper mapping right on fortigate |
| 01:13.20 | volga629 | id=13 trace_id=4 func=__ip_session_run_tuple line=2519 msg="DNAT 99.224.107.222:11449->10.18.130.50:11449" |
| 01:14.06 | volga629 | and call going through as should be |
| 01:18.10 | volga629 | I think the trouble is that I am using from outside not port 5060 and I think asterisk should listen non standard port ? |
| 01:19.18 | volga629 | here working debug https://fpaste.networklab.ca/ku1U/ |
| 01:24.16 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:29.06 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:34.38 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 01:35.41 | igcewieling | volga629: DNAT? You should use Asterisk's NAT support and disable any other NAT traversal |
| 01:35.49 | igcewieling | Including STUN, TURH, ETC. |
| 01:36.04 | volga629 | this on fortigate border firewall |
| 01:36.38 | igcewieling | ChannelZ: The Certified Asterisk versions have a LOT of stuff disabled by default which are enabled in the regular Asterisk versions. |
| 01:38.12 | volga629 | Currently it working in this test case: remote ext ---INTERNET --- FORTIGATE --- ASTERISK |
| 01:38.44 | igcewieling | volga629: then turn off SIP ALG and/or SIP SPI on your firewall |
| 01:39.03 | volga629 | yes I disabled |
| 01:39.19 | volga629 | call is going through no problem |
| 01:39.27 | igcewieling | cool Glad you got it working. |
| 01:40.06 | carbinemonoxide | Okay. I managed to get everything working. Thanks for everyone's help. |
| 01:41.54 | carbinemonoxide | Anyone ever work with getting notices for voicemail in Google Voice on SIP phones? |
| 01:42.13 | volga629 | Huge thank you for EVERYONE for they support and time |
| 01:44.49 | volga629 | last small question is inkeys= and outkey= is related only to iax trunk settings ? |
| 02:05.15 | WIMPy | They are used when you enable encryption. |
| 02:11.56 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 02:13.01 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:13.01 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:15.19 | volga629 | ok so I can use on sip trunk too ? |
| 02:16.37 | WIMPy | no |
| 02:16.46 | WIMPy | You can use tls and srtp with sip. |
| 02:17.59 | volga629 | ok thanks |
| 02:27.15 | igcewieling | volga629: all valid sip.conf options should be listed in....... sip.conf.sample |
| 02:29.34 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 02:31.22 | j4jackj | hi pabelanger |
| 02:34.26 | volga629 | reading right now |
| 02:41.51 | *** join/#asterisk lunaphyte (~lunaphyte@unaffiliated/lunaphyte) |
| 02:42.00 | *** part/#asterisk lunaphyte (~lunaphyte@unaffiliated/lunaphyte) |
| 02:44.57 | Penguin | So the problem was that you forgot to sip reload after setting externaddr and localnet? |
| 02:47.25 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 03:04.10 | volga629 | yes, and was need add 5060 vip on fortigate |
| 03:46.59 | *** part/#asterisk P3r3gr1n3 (~chwall@63.225.228.54) |
| 04:00.26 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 04:00.26 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 05:13.14 | *** part/#asterisk volga629 (~bendersky@CPE085b0e07d3f2-CM7cb21b15b251.cpe.net.cable.rogers.com) |
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| 05:35.18 | apb1963_ | life? |
| 05:36.07 | apb1963_ | birds? hamburger lovers? pants removers? etc? |
| 05:37.29 | *** join/#asterisk Blashyrkh (~chatzilla@043-054-094-081.as39912.net) |
| 05:37.34 | ChannelZ | I'm not wearing any pants. |
| 05:37.41 | apb1963_ | perfect |
| 05:37.51 | apb1963_ | just stay seated |
| 05:37.53 | apb1963_ | :) |
| 05:38.13 | apb1963_ | So I've decided to join the retransmissions club |
| 05:38.27 | apb1963_ | I checked everything I could think of |
| 05:38.31 | apb1963_ | dropped all my firewalls |
| 05:38.44 | apb1963_ | checked my NAT settings as I understand them... |
| 05:38.58 | apb1963_ | still getting critical retransmission timeouts |
| 05:39.08 | apb1963_ | Here's the call: http://ix.io/7NT |
| 05:40.13 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.151) |
| 05:40.25 | ChannelZ | well it's sending out its LAN IP which is not going to get you very far afterwards |
| 05:40.27 | apb1963_ | Router is at 192.168.0.1 and has the public IP of course... forwarding to asterisk at .101; softphone is at .100 |
| 05:41.53 | ChannelZ | So this call is incoming from a softphone behind some other firewall? |
| 05:42.01 | apb1963_ | One router |
| 05:42.22 | apb1963_ | both asterisk server and softphone are on the same subnet. asterisk as .101, softphone at .100 |
| 05:42.40 | ChannelZ | then why is the call coming from an external IP? |
| 05:42.59 | apb1963_ | because the calls is being made from a landline, to ipkall DID |
| 05:43.15 | apb1963_ | softphone shouldn't even really be involved just yet |
| 05:43.25 | apb1963_ | It's hitting the IVR |
| 05:43.35 | apb1963_ | and then it times out before the message finishes playing. |
| 05:44.12 | ChannelZ | well again your asterisk is sending out its LAN IP for RTP so unless IPkall ignores you that won't work |
| 05:44.13 | apb1963_ | If I were to hit an option on the IVR, THEN the softphone would come into play... but again.. it doesn't make it that far. |
| 05:44.17 | [TK]D-Fender | And your NAT settings are not right |
| 05:44.18 | ChannelZ | But doesn't explain the retransmissions |
| 05:44.25 | [TK]D-Fender | So it's handing out the IP on the box itse.f |
| 05:44.45 | apb1963_ | sip settings are NAT=yes |
| 05:44.53 | j4jackj | Hello? |
| 05:44.54 | apb1963_ | but I'll doublecheck |
| 05:45.03 | ChannelZ | no you need to look at localnet and externaddr |
| 05:45.04 | [TK]D-Fender | Contact: <sip:2532431532@192.168.0.101:5060> <--- the other side would NOT be responding due to being told a useless return address |
| 05:45.19 | [TK]D-Fender | [01:44]apb1963_sip settings are NAT=yes <- this is vague and meaningless |
| 05:45.33 | [TK]D-Fender | apb1963_: there are SEVERAL settings to be made and WHERE you put things counts |
| 05:45.50 | [TK]D-Fender | apb1963_: Also, ipkall themselves are NOT behind NAT so even their peer entry is wrong. |
| 05:46.55 | ChannelZ | doh I always forget about Contact |
| 05:47.21 | apb1963_ | localnet is correct at 192.168.0.0 and externHOST is correct with my fqdn |
| 05:48.10 | [TK]D-Fender | apb1963_: Why are you wasting time telling us "I did everything right" and not showing us ACTUAL configs? |
| 05:48.14 | apb1963_ | ok in sip.conf, nat=yes |
| 05:48.16 | [TK]D-Fender | apb1963_: You have done this wrong somewhere |
| 05:48.33 | [TK]D-Fender | apb1963_: your little snippits are not going to show us where you made your mistake |
| 05:49.03 | [TK]D-Fender | [01:48]apb1963_ok in sip.conf, nat=yes <- perhaps I wasn't clear with "this is vauge and meaningles" and "location counts" |
| 05:49.41 | apb1963_ | http://ix.io/7NV |
| 05:50.12 | apb1963_ | that's actually sip_general_additional.conf |
| 05:50.30 | [TK]D-Fender | apb1963_: ping asterisk.saveabunny.com |
| 05:50.34 | [TK]D-Fender | from your OS CLI |
| 05:50.35 | apb1963_ | which I'm sure I don't need to tell you is included by sip.conf |
| 05:51.01 | apb1963_ | well it gets the private ip... because I put it in /etc/hosts |
| 05:51.10 | [TK]D-Fender | then you have shot yourself in the face |
| 05:51.12 | [TK]D-Fender | ^ |
| 05:51.15 | apb1963_ | ouch |
| 05:51.17 | [TK]D-Fender | DUMBN move |
| 05:51.20 | ChannelZ | is your DNS split horizon, such that it returns LAN addresses when queried locally? |
| 05:51.28 | ChannelZ | nm |
| 05:51.30 | [TK]D-Fender | You told * how to get your public IP... and then LIED TO IT |
| 05:52.09 | apb1963_ | ok. something else wasn't doing what I wanted so I did it that way... |
| 05:52.25 | apb1963_ | apache maybe... I don't remember what it was |
| 05:52.27 | ChannelZ | bang random things with hammers till it works |
| 05:52.30 | [TK]D-Fender | ..... |
| 05:52.49 | [TK]D-Fender | You don't even know why you are doing these things... wonderful.. |
| 05:52.50 | apb1963_ | hey I got the other thing working |
| 05:52.56 | [TK]D-Fender | Things. |
| 05:52.58 | [TK]D-Fender | And stuff |
| 05:53.04 | apb1963_ | I did at the time... I just don't remember |
| 05:53.06 | [TK]D-Fender | Yeah, the vague just keeps on coming |
| 05:53.14 | ChannelZ | Does your IP change a lot? |
| 05:53.37 | apb1963_ | I moved off a VM onto a PM... so yeah things got changed around |
| 05:54.01 | apb1963_ | added a router to the network that I didn't have |
| 05:54.03 | apb1963_ | before |
| 05:54.15 | apb1963_ | so... things have been in flux for the last couple of weeks |
| 05:54.34 | *** join/#asterisk Kalidarn (~Kalidarn@unaffiliated/kalidarn) |
| 05:54.46 | apb1963_ | so let me put my hosts file back |
| 05:55.02 | apb1963_ | and wait for something to break :) |
| 05:55.04 | Kalidarn | so i seem to have an interesting problem, my SIP line works fine, except after a while i see that the remote server is not reachable, rebooting my cisco router seems to fix it. |
| 05:55.31 | ChannelZ | Kalidarn: NAT timeout perhaps? |
| 05:55.32 | Kalidarn | should mention that the cisco router is local (ie just outside of the asterisk box) |
| 05:55.39 | Kalidarn | ChannelZ: i thought you might say that |
| 05:55.57 | ChannelZ | Something is closing down. Or your router is broken. |
| 05:56.12 | Kalidarn | ChannelZ: yeah im getting that feeling |
| 05:56.40 | apb1963_ | Yup... that fixes asterisk |
| 05:56.43 | Kalidarn | ChannelZ: ill give that further investigation, i did read somewhere about cisco and SIP and NAT being an issue |
| 05:56.44 | ChannelZ | Do you use qualify? |
| 05:56.52 | Kalidarn | not using any QoS atm |
| 05:57.11 | Kalidarn | as it's a fairly low traffic line |
| 05:57.13 | ChannelZ | no, in sip.conf - basically an occasional 'ping' to a peer |
| 05:57.19 | Kalidarn | ah |
| 05:57.51 | Kalidarn | ChannelZ: yes i do |
| 05:58.06 | Kalidarn | ive noticed in the logs after a while that pinging changes to "unreachable" |
| 05:58.11 | apb1963_ | see I wasn't lying to asterisk... I was simply trying to tell it there was more than one way to get there :/ |
| 05:58.20 | ChannelZ | at what interval? Perhaps it's too long. |
| 06:06.13 | apb1963_ | <[TK]D-Fender> apb1963_: Also, ipkall themselves are NOT behind NAT so even their peer entry is wrong. <<< Can you elaborate on this? |
| 06:07.14 | apb1963_ | puts on his flame retardant suit |
| 06:07.50 | ChannelZ | You have told asterisk that IPkall is behind nat, but they are not. |
| 06:08.00 | apb1963_ | yah, that much I understood |
| 06:08.11 | apb1963_ | but I don't know what makes him say that |
| 06:08.24 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.38) |
| 06:08.34 | apb1963_ | how is the peer entry wrong is what I'm asking |
| 06:08.45 | apb1963_ | there's no mention of nat at all |
| 06:08.50 | ChannelZ | <--- Transmitting (NAT) to 66.54.140.46:5060 ---> |
| 06:09.19 | apb1963_ | yeah I saw that... I presumed it meant that asterisk was behind NAT |
| 06:09.42 | ChannelZ | no, it means it thinks the peer it's talking to is |
| 06:10.31 | apb1963_ | does it automagically assume that? Because there is nothing about NAT in any ipkall context |
| 06:11.49 | ChannelZ | It's probably inherited from your [general] section |
| 06:12.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.38) |
| 06:12.56 | apb1963_ | ok.... but I thought that section was what specified that asterisk was behind nat |
| 06:13.56 | j4jackj | apb1963_: hi |
| 06:14.16 | ChannelZ | nat= has nothing to do with what asterisk is.. it controls peers |
| 06:14.25 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 06:14.27 | ChannelZ | externhost/externaddr and localnet are what control asterisk |
| 06:15.26 | *** join/#asterisk wolfmitchell (~wolfmitch@botters/wolfmitchell) |
| 06:16.41 | apb1963_ | ok this is making sense now |
| 06:27.17 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 06:29.08 | apb1963_ | ok that's all fixed now |
| 06:29.11 | apb1963_ | yay |
| 06:29.21 | apb1963_ | hi j4jackj |
| 06:41.27 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 06:48.59 | j4jackj | hi apb1963_, so what hawve you been doing in your microcosm of the world? |
| 06:49.42 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
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| 07:35.21 | bulkorok | hellooooooo :-) |
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| 07:40.46 | onats | hello, is there a ready made appliance running on asterisk? Switchvox? |
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| 07:45.41 | j4jackj | Hello all! |
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| 08:45.08 | Blashyrkh | what is sip error 603 |
| 08:45.18 | Blashyrkh | i get it when sending an invite to an asterisk |
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| 08:50.41 | Chainsaw | Blashyrkh: Declined. It does not approve. |
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| 09:16.32 | wdoekes | Blashyrkh: that's odd.. asterisk 11 doesn't list 603 in the chan_sip cause2sip. |
| 09:17.23 | wdoekes | oh, never mind. it's sprinkled all over the source as various responses |
| 09:18.10 | wdoekes | technically the 603 is the global 403. 403 says the request won't work here, 603 says it won't work anywhere |
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| 13:20.12 | vbrinza | hi all. connecting an external client to asterisk behind nat. when calling external numbers via trunk, getting 500 server internal failure if nobody pickup and can't hear the voice if the other end answers. any idea how to solve it? |
| 13:20.26 | vbrinza | the sip port forwarding on router is done |
| 13:20.54 | [TK]D-Fender | need more than SIP forwarded.. |
| 13:20.58 | [TK]D-Fender | SIP doesn't carry the voice |
| 13:20.59 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 13:21.06 | pensmit | Anyone use flowroute? |
| 13:21.25 | pensmit | They have been dropping a lot lately. What are some other options? |
| 13:21.40 | igcewieling | ~itsp |
| 13:21.41 | 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. |
| 13:29.18 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
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| 13:33.35 | pensmit | ~itsplist-us |
| 13:33.35 | 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 |
| 13:34.25 | pensmit | Can you set up least cost routing with multiple options in case one itsp goes down |
| 13:35.21 | igcewieling | pensmit: yes. |
| 13:35.33 | pensmit | I need to look into that. |
| 13:35.42 | pensmit | Ever hear of these guys? http://www.siproutes.com/ |
| 13:35.43 | igcewieling | assuming "write my own dialplan" when you say "least cost routing" |
| 13:36.47 | pensmit | Basically flowroute has gone down a lot for us lately. We have a few customers(just getting started) and some are wanting to leave because of it. |
| 13:37.05 | [TK]D-Fender | Feel free to choose your outbond peer based on weather and the winner of last night's sports game.... |
| 13:38.04 | igcewieling | [TK]D-Fender: I've getting a FreePBX vibe from him. |
| 13:38.05 | Nugget | What am I gonna do with 40 subscriptions to Vibe?! |
| 13:42.13 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 13:52.44 | pensmit | We're using AsteriskNow at this point with FreePBX...yes. |
| 13:52.48 | pensmit | Is this a no no here? |
| 13:53.03 | Qwell | ~freepbx |
| 13:53.04 | 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 |
| 13:53.22 | pensmit | Ahh...maybe we'll get off it then. |
| 13:53.34 | pensmit | My questions really aren't about freepbx. |
| 13:53.39 | [TK]D-Fender | pensmit: Only as far as asking how to effect changes to your dialplan flow from within the GUI |
| 13:53.56 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 13:53.58 | [TK]D-Fender | pensmit: Because that's other peoples code (modules) |
| 13:54.07 | pensmit | I see. |
| 13:54.24 | [TK]D-Fender | pensmit: Which we didn't write, and aren't goign to guess how to support |
| 13:54.39 | pensmit | I can't blame you. |
| 13:54.45 | pensmit | My questions are more general. |
| 13:55.06 | [TK]D-Fender | pensmit: Yes, and FreePBX is a very specific hindrance. |
| 13:55.43 | pensmit | Is there a way to get around flowroute going down issues? It seems like outbound is doable but DIDs have to go through someone correct? |
| 13:55.53 | igcewieling | I can smell the stench of a GUI from a mile away. |
| 13:56.21 | igcewieling | pensmit: Go into sip.conf and....oh, that's right you can't go into sip.conf because it will be overwritten by the GUI... |
| 13:56.40 | pensmit | Ok..Ok. I'll switch. |
| 13:56.47 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:56.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:56.55 | bacobart | there's a sip_custom.conf where you can put options which cannot be set by the gui |
| 13:56.56 | Greenlight | He's asking a more general question; and no, with inbound DID's your screwed if the ITSP goes down |
| 13:56.57 | pensmit | Assuming I move to bare bones asterisk. What can be done? |
| 13:56.58 | igcewieling | pensmit: we don't care if you switch or not, just go to a channel which supports your GUI |
| 13:57.34 | igcewieling | like, I dunno #FreePBX or something like that. |
| 13:57.41 | pensmit | It sounds to me like you simply don't have the capability to answer a question igcewieling |
| 13:57.45 | pensmit | You just want to start shit |
| 13:57.47 | pensmit | go fuck yourself |
| 13:57.48 | pensmit | anyone else |
| 13:57.56 | Greenlight | Woah chill |
| 13:57.57 | *** mode/#asterisk [+b *!*pensmit@*unaffiliated/pensmit] by Qwell |
| 13:57.57 | *** kick/#asterisk [pensmit!~north@pdpc/sponsor/digium/Qwell] by Qwell (pensmit) |
| 13:59.04 | *** mode/#asterisk [-b *!*pensmit@*unaffiliated/pensmit] by Qwell |
| 13:59.17 | igcewieling | *shrug* I was going to put him on /ignore, but banning works too. |
| 13:59.24 | Greenlight | lol |
| 13:59.33 | Formater | :) |
| 13:59.40 | Qwell | igcewieling: I'm not convinced that I removed the right person. |
| 14:00.13 | *** join/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 14:00.20 | igcewieling | Qwell: time will tell. |
| 14:01.12 | pensmit | So does anyone here have any real advice on my issue |
| 14:01.21 | Greenlight | [02:56pm] <Greenlight> He's asking a more general question; and no, with inbound DID's your screwed if the ITSP goes down |
| 14:01.23 | pensmit | Or is the room full of igcewieling clones? |
| 14:01.29 | pensmit | thanks greenlight |
| 14:02.06 | pensmit | So with outbound my problem is solvable by least cost routing? |
| 14:02.09 | Greenlight | pensmit: Please don't swear or be rude. igcewieling helps a lot of people in here. And his dislike of FreePBX came accross rather strong today. |
| 14:02.12 | pensmit | or simply multiple outbounds. |
| 14:02.32 | Greenlight | The best advice I can give is to get a more reliable ITSP |
| 14:02.40 | Greenlight | If they keep going down |
| 14:03.06 | [TK]D-Fender | [09:55]pensmitIs there a way to get around flowroute going down issues? It seems like outbound is doable but DIDs have to go through someone correct? <- If your provider of the DID goes down it's "game-over" |
| 14:03.22 | [TK]D-Fender | pensmit: outbound is outbound.... you can failover on that all you want |
| 14:03.42 | Greenlight | YOu just need to decide what is a "failure" and how to failover. |
| 14:03.53 | [TK]D-Fender | [10:01]pensmitSo with outbound my problem is solvable by least cost routing? <- do not mix the terms "least cost routing" and "failover". They are NOT the same |
| 14:04.14 | Qwell | s/least cost/most available/ |
| 14:04.43 | [TK]D-Fender | pensmit: And the failover option for your outbound is sitting right there in your Outbound Route. |
| 14:05.40 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:05.42 | pensmit | So for newbies like me it's best to possibly have two itsps for outbound but I'm screwed on the inbound. |
| 14:05.53 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 14:06.08 | Greenlight | Or get a reliable ITSP. |
| 14:06.17 | vbrinza | <[TK]D-Fender> anything about SIP/2.0 500 Server internal failure ? |
| 14:06.37 | vbrinza | i know RTP carry the voice |
| 14:06.40 | pensmit | greenlight out of curiousiyty which one do you like the best? |
| 14:06.44 | Greenlight | On small scale failvoer with multiple ITSP's can be done. But try asking an ITSP to set aside 1000 channels just incase you need them when your main provider goes down. |
| 14:06.52 | [TK]D-Fender | vbrinza: I would not comment on a single message you say you see in a string of comms. I'd have to see the ENTIRE thing from beginning to end. |
| 14:06.53 | [TK]D-Fender | ~pb |
| 14:06.54 | 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. |
| 14:06.55 | [TK]D-Fender | ^^^ |
| 14:07.11 | Greenlight | pensmit: I'm in the UK, I think you're accross the pond so different ITSPs |
| 14:07.26 | pensmit | Ahh ok, thank you sir. |
| 14:07.46 | Greenlight | I'm also at the lower end of whole sale with 1000+ channels so different providers to choose from |
| 14:09.43 | pensmit | On the cursing and igcewieling situation. People like that are caustic to a channel. I see no reason to hate on FreePbx users. More flies are caught with honey than vinegar. He's not really hating on FreePbx, but rather using this channels general dislike of freepbx as an excuse to take his life frustrations out on newcomers. I've seen this pattern of behavior many times before. It doesn't help to not call |
| 14:09.43 | pensmit | <PROTECTED> |
| 14:09.52 | pensmit | Anyhow thanks for all the help. Later guys. |
| 14:11.02 | Greenlight | His analogy comparing freepbx to a ladyboy would have been perhaps more apt. |
| 14:11.33 | [TK]D-Fender | pensmit: As caustic goes, openly spewing obscenities is possibly worse. |
| 14:12.09 | pensmit | Agree to disagree. Don't start nothing won't be nothing. |
| 14:12.30 | vbrinza | <[TK]D-Fender> please take a look at https://gist.github.com/anonymous/6424477 |
| 14:12.39 | vbrinza | the error appears in 351 line |
| 14:12.43 | [TK]D-Fender | pensmit: Sounds more like "if you're bad I'm entitled to match your inappropriateness" |
| 14:12.58 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 14:13.23 | [TK]D-Fender | vbrinza: repaste masking ONLY passwords |
| 14:13.38 | pensmit | Sounds to me more like this channel doesn't like to deal with entrenched trolls. Anyhow that's all I have to say. Thanks for the help. |
| 14:14.36 | [TK]D-Fender | pensmit: We also didn't say he wasn't harsh. Just don't go around using to justify your own reaction there. |
| 14:14.57 | pensmit | If he was my buddy. I'd defend him too. |
| 14:15.29 | [TK]D-Fender | pensmit: I wasn't defending him. I'm calling YOU out. |
| 14:15.37 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-gsypucskdfuxjlbi) |
| 14:15.37 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:15.42 | pensmit | I cursed. I was kicked. Nothing to call out on that. |
| 14:15.59 | pensmit | I'm done cursing. What's his face got the attention he want and you defended your buddy. |
| 14:16.14 | [TK]D-Fender | pensmit: Again, I never defended him. |
| 14:18.07 | *** part/#asterisk pensmit (~pensmit@unaffiliated/pensmit) |
| 14:18.45 | bacobart | so... |
| 14:18.49 | bacobart | good thing it's almost friday |
| 14:19.11 | Greenlight | That's a tenius defenition of "almost" |
| 14:19.53 | bacobart | think positive;) |
| 14:20.20 | Greenlight | .. I'll be retiting "soon" :) |
| 14:20.24 | Greenlight | *retiring |
| 14:20.43 | Greenlight | Only a mere 30 odd years to go |
| 14:21.36 | [TK]D-Fender | vbrinza: Reliably Transmitting (NAT) to 217.10.68.150:5060: |
| 14:21.44 | [TK]D-Fender | vbrinza: o=root 860572820 860572820 IN IP4 217.10.77.102 |
| 14:22.08 | [TK]D-Fender | vbrinza: Their media is NOT carried on the same server as their SIP signalling and you are dialing a DIRECT URI, instead of a properly configured peer entry |
| 14:22.33 | [TK]D-Fender | vbrinza: And it is treating them as though signalling and media have to be forced through the dialed IP. |
| 14:22.45 | [TK]D-Fender | vbrinza: Go make a proper peer with "nat=no" |
| 14:23.06 | *** join/#asterisk deegen (~deegen@S01060023bee90320.gv.shawcable.net) |
| 14:28.00 | *** join/#asterisk aruntomar (~Thunderbi@49.248.158.90) |
| 14:29.58 | [TK]D-Fender | vbrinza: also note private part in PM |
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| 14:32.35 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:34.23 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:37.50 | hjf | damn |
| 14:38.00 | hjf | i need to recover an account password from a SPA2102 |
| 14:38.32 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 14:38.42 | hjf | i have admin access but i dont know the account pasword |
| 14:39.31 | Greenlight | Maybe Google will know. Or the user manaul. |
| 14:40.00 | Greenlight | Can't you just reset it ? |
| 14:40.09 | hjf | but i will lose the SIP account |
| 14:40.17 | Greenlight | And ? |
| 14:40.26 | hjf | i dont know the SIP account password |
| 14:40.27 | Greenlight | Call your proivider and get the details. |
| 14:40.40 | Greenlight | *provider |
| 14:40.58 | hjf | it's from some reseller guy, i can't call the provider directly. i called the reseller and he said sure i'll reset the password but he never did :/ |
| 14:41.50 | hjf | i was thinking about sniffing the traffic and catch the password on the wire, but there's no option to make it Basic authentication |
| 14:41.50 | Greenlight | What an odd setup. |
| 14:42.22 | Greenlight | Sounds like you're pissing into the wind. |
| 14:42.27 | hjf | lol |
| 14:43.09 | Greenlight | Get a proper ITSP |
| 14:43.28 | hjf | yah but this is free :P |
| 14:43.47 | [TK]D-Fender | Then go ask for your money back |
| 14:43.52 | hjf | hahaha |
| 14:43.57 | Greenlight | Why do people thing time is free |
| 14:44.10 | Greenlight | You were in here yesterday with the same problem. You've spent days trying to fix this. |
| 14:44.16 | Greenlight | Your time is NOT free. |
| 14:44.40 | hjf | Greenlight: i work in a comic book store. there's no one in the shop right now so i hack things |
| 14:45.40 | Greenlight | Well, ok, my time isn't free, so... good luck with that! |
| 14:48.15 | Chainsaw | hjf: Does a config backup from the SPA2102 not contain the password in plaintext? |
| 14:48.23 | Chainsaw | hjf: Sometimes you get lucky. |
| 14:49.00 | hjf | Chainsaw: i haven't found an option to get a backup config file o.o |
| 14:49.12 | Chainsaw | hjf: Often on the same tab/page as the firmware upgrade. |
| 14:49.41 | Chainsaw | hjf: I'm not guaranteeing you that the option is there, having never seen the device myself... but it's rather common. |
| 14:50.49 | hjf | Chainsaw: this is so lame. Cisco's documentation says to SAVE THE CONFIG PAGE to a file |
| 14:54.06 | deegen | If two phones in a ring group are ringing, and someone dials *8, what line gets picked up? |
| 14:54.26 | *** join/#asterisk vlad_starkov (~vlad_star@87.117.139.75) |
| 14:54.33 | Greenlight | What's a ring group ? |
| 14:54.47 | deegen | Call group, sorry. |
| 14:55.12 | Greenlight | And, what is *8 setup to do? |
| 14:56.30 | deegen | If one phone in the call group is ringing, *8 will pick it up. |
| 14:57.19 | igcewieling | deegen: does it matter, the person dialing *8 will still get the call. |
| 14:58.14 | hjf | igcewieling: i think he means a third person in a phone thats not ringing |
| 14:58.15 | igcewieling | there may be more than one devices ringing, but there is still only one call coming in. |
| 14:58.38 | deegen | If more than one line is ringing, what call though? I'm guessing it's the first one, but that's based on nothing. |
| 14:59.24 | igcewieling | deegen: I must not be understanding your question. |
| 14:59.44 | Greenlight | I'm still not sure what you mean by call group, or what exactly *8 is setup to do. |
| 14:59.55 | igcewieling | call comes in, gets sent to two devices. someone else dials *8 and starts talking to me. |
| 15:00.03 | igcewieling | Greenlight: see features.conf |
| 15:00.22 | igcewieling | and callgroup= and pickupgroup= |
| 15:00.55 | *** join/#asterisk dnram (~Adium@67.63.143.254) |
| 15:01.18 | Greenlight | Ahh of course; I've not played with that stuff too much |
| 15:02.54 | deegen | igcewieling: Right. So, if two devices in a callgroup ring, and a third person dials *8 which is configured to pick up a call in the callgroup, what call gets picked up? |
| 15:03.14 | igcewieling | deegen: the person calling gets picked up. |
| 15:03.50 | deegen | igcewieling: If two devices in the callgroup are both being called, what one will get picked up? |
| 15:04.06 | igcewieling | now if you have multuple people calling the same callgroup at the same time --- well I don't know the answer to THAT. |
| 15:04.19 | igcewieling | deegen: I don't know. |
| 15:04.52 | igcewieling | deegen: does anything useful show on the CLI when doing the *8? |
| 15:05.57 | deegen | igcewieling: Ok, thanks. I'll check.. |
| 15:08.20 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:09.11 | hjf | hehe i found why the analog phone wasn't ringing |
| 15:09.30 | eirirs_ | it wasn't plugged in? :P |
| 15:09.31 | hjf | i thought it was the SPA2102, shitty supply not being able to generate 90V ring voltage |
| 15:09.41 | hjf | no... roaches got the phone o.o |
| 15:09.48 | eirirs_ | haha, so basically it wasn't "plugged in" :P |
| 15:10.05 | eirirs_ | haha even better |
| 15:10.06 | hjf | http://files.sharenator.com/nasty_Spending_my_time_differently_for_a_while-s480x306-332166-580.jpg |
| 15:10.22 | eirirs_ | those unplugged it for you :P |
| 15:10.30 | eirirs_ | ... on their own way |
| 15:10.33 | hjf | the bastards shit on the pcb |
| 15:10.47 | eirirs_ | and fried their own ass? |
| 15:10.48 | hjf | and their crap is so strong it eats through the solder mask and the copper |
| 15:10.53 | eirirs_ | LOL |
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| 15:15.35 | *** join/#asterisk navaismo (~navaismo@189.241.120.161) |
| 15:15.37 | hjf | http://i.imgur.com/C8FDwdx.jpg |
| 15:15.41 | hjf | that's roach damage |
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| 15:25.55 | hjf | overlapping dial plans...i want to be able to call extensions at home by a single digit, but also call outside numbers transparently. will this cause trouble? |
| 15:26.42 | hjf | that is, dial 5 to call a phone inside, or dial 555-1212 for an external call |
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| 15:29.12 | [TK]D-Fender | hjf: your phone should wait at the end of that signe digit and then proceed as expected |
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| 15:30.04 | hjf | good |
| 15:30.30 | hjf | i also have configured internal calls with canreinvite=yes and directrtp and all that |
| 15:30.44 | hjf | so all my call audio is phone to phone |
| 15:30.57 | hjf | what if i want to configure an external SIP provider? |
| 15:31.10 | [TK]D-Fender | Then go do it. |
| 15:31.11 | hjf | i will need all phones to go through asterisk in that case |
| 15:31.18 | [TK]D-Fender | And do NOT allow reinvites to it |
| 15:31.28 | hjf | just canreinvite=no ? |
| 15:31.39 | [TK]D-Fender | canreinvite=yes <- and this is the wrong parameter for 1.6+ |
| 15:31.43 | [TK]D-Fender | it's "directmedia" |
| 15:31.47 | hjf | i see |
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| 15:32.39 | hjf | i have directmedia=yes directrtpsetup=yes |
| 15:32.47 | hjf | in the [general] section |
| 15:33.02 | [TK]D-Fender | don't |
| 15:33.25 | [TK]D-Fender | disable it in [general] and enable only in the specific peers that should be allowed to within your LAN |
| 15:33.28 | *** join/#asterisk lorsungcu (~anonymous@50-78-230-69-static.hfc.comcastbusiness.net) |
| 15:33.52 | hjf | well, all the peers in my LAN use it |
| 15:34.29 | hjf | ok really dumb question i guess: whatever is set in [general] i don't need to copy in each client? i see many examples where they copy the same config for every peer |
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| 15:41.04 | [TK]D-Fender | hjf: [general] should be the most secure and safest option. EVERYTHING should be explicit. [general] is what is used when things slip through the cracks |
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| 17:33.17 | treborsux | anyone need these ill let em fly cheap! http://cgi.ebay.com/190895859784 |
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| 18:04.29 | igcewieling | quiet day today |
| 18:04.56 | WIMPy | Where? |
| 18:05.51 | igcewieling | here |
| 18:06.44 | WIMPy | 50 lines/hour |
| 18:06.55 | igcewieling | must be all from the two people I have on /ignore. |
| 18:07.37 | igcewieling | last thing I saw is the spam from treborsux and before that [TK]D-Fender saying something about 2.5 hrs ago. |
| 18:08.14 | [TK]D-Fender | nope, it IS that quiet |
| 18:08.15 | WIMPy | Yes. The last 2.5h were quiet. |
| 18:08.39 | WIMPy | But I've been away for over 400 lines. |
| 18:08.59 | igcewieling | I've not seen anything "on-topic" today. |
| 18:09.59 | igcewieling | ah, no there was an Asterisk question about 4.5 hours ago. |
| 18:10.01 | WIMPy | The usual SIP/NAT shit. |
| 18:10.29 | igcewieling | WIMPy: I have hjf on /ignore |
| 18:11.58 | WIMPy | I guess I could grep|cut|sort|uniq my history, but I don't care. |
| 18:12.15 | hjf | ok i want to configure my asterisk server for outbound calls |
| 18:12.36 | hjf | i configured a sip provider following t his guide: http://www.rjsystems.nl/en/2100-asterisk.php |
| 18:13.07 | hjf | but when i try to make a call i get a 403 error |
| 18:14.51 | hjf | my asterisk server is inside the LAN |
| 18:14.59 | hjf | behind a NAT |
| 18:15.54 | *** join/#asterisk gmalsack (~gmalsack@23.30.198.161) |
| 18:16.52 | gmalsack | hey guys and gals.... any opinions on what the most preferred way to reload the dialplan is? better service asterisk reload, or module reload pbx_conf? |
| 18:17.51 | WIMPy | What about 'dialplan reload'? |
| 18:18.02 | j4jackj | WIMPy: right on |
| 18:18.05 | gmalsack | well yea, there's that to |
| 18:18.16 | j4jackj | It's likely THE way to. |
| 18:18.54 | gmalsack | does just reloading an individual module risk any kind of risk of running into a sync issue with other modules? |
| 18:19.15 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-104-246-231.ph.ph.cox.net) |
| 18:20.32 | j4jackj | Unlikely, but it's not advisable. |
| 18:21.41 | gmalsack | ok thanks! :-) |
| 18:22.15 | hjf | http://pastebin.com/RPjqeBu8 |
| 18:22.21 | hjf | thats what i get when i try to place a call |
| 18:23.39 | gmalsack | hjf: you already working with someone on this? |
| 18:23.59 | hjf | no? |
| 18:24.28 | gmalsack | line 36 unauthorized. can you post sip.conf? |
| 18:25.01 | gmalsack | for the hjf sip peer/friend ?\ |
| 18:25.53 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 18:26.08 | gmalsack | I'm assuming your asterisk server is 10.42.42.35... |
| 18:26.46 | *** join/#asterisk citrusfizz (~chatzilla@70.184.40.66) |
| 18:27.38 | citrusfizz | is there a simple phone to sip hardware box? i have a normal phone that needs to register to our hosted sip provider. |
| 18:27.40 | hjf | http://pastebin.com/bLpzamdC |
| 18:27.52 | hjf | gmalsack: yes, inside the LAN |
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| 18:29.42 | gmalsack | hjf: looks like your sip device is sending the wrong credentials. at least that's what I'm seeing thus far. |
| 18:30.27 | citrusfizz | better question i should ask, is what would you guys recommend for an analog to sip device? |
| 18:30.48 | hjf | gmalsack: hmm... the device should be configured with the same user/pass as the remote server? |
| 18:31.08 | hjf | gmalsack: what i want is a home PBX. calls between home devices, and outbound calls through an IP provider |
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| 18:32.03 | gmalsack | the device 10.42.42.104 seems to be sending the wrong username/password combination to 10.42.42.35 |
| 18:33.09 | gmalsack | look at lines 33, 35 and 36 of your http://pastebin.com/RPjqeBu8 |
| 18:33.11 | hjf | gmalsack: internal calls work. i can call the voicemail for example |
| 18:33.58 | gmalsack | hjf: oh, ok. then what about your dialplan. users context to be precise... can you post that? |
| 18:34.38 | hjf | 1 sec |
| 18:35.29 | _Corey_ | corey@voneto.comh6bxca16 |
| 18:35.51 | _Corey_ | well then |
| 18:35.54 | _Corey_ | whoops |
| 18:36.10 | gmalsack | well, that clears up my question on _Corey_ |
| 18:36.21 | hjf | gmalsack: http://pastebin.com/YKAFVD9t |
| 18:36.25 | _Corey_ | :) |
| 18:36.32 | hjf | _Corey_: RUN |
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| 18:43.01 | gmalsack | in sip.conf I'm not seeing anything that tells asterisk what to nat. i.e. localnet externalip directives. I'm also not seeing anything in the cli output log where your asterisk system is trying to send traffic out to your provider. can you post dialplan show? |
| 18:44.18 | gmalsack | sorry. spoke too soon. hold off on that one. |
| 18:45.35 | gmalsack | though I'm still thinking you need to add your localnet and extern directives. |
| 18:46.23 | gmalsack | your provider is seeing the call come from your internal 10. ip address. |
| 18:46.45 | hjf | yes but the provider is magical |
| 18:46.53 | gmalsack | lol |
| 18:46.55 | hjf | they gave me a spa2102 that had nothing configured |
| 18:47.02 | hjf | no stun or anything like that |
| 18:47.10 | hjf | and it Just Works |
| 18:47.17 | hjf | i have no idea how |
| 18:47.44 | gmalsack | remote provisioning??? idk??? |
| 18:48.22 | hjf | no lol |
| 18:48.28 | hjf | it just works |
| 18:48.34 | *** join/#asterisk aruntomar (~Thunderbi@49.248.156.37) |
| 18:48.44 | hjf | i also configured a Panasonic IP phone with that provider's setup and it works too |
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| 18:52.12 | gmalsack | you're trying to call 080808 on the providers network? |
| 18:52.24 | gmalsack | doesn't seem like a valid phone number.... |
| 18:52.37 | hjf | gmalsack: it doesn't work with valid numbers either :P |
| 18:53.33 | gmalsack | do you get a 403 forbidden from the provider with a number the provider requires. i.e. 10 digits, 11 digits , +11digits.... |
| 18:54.18 | hjf | yes |
| 18:54.23 | hjf | 7 digits, that's what we use here |
| 18:54.44 | hjf | i can call 7 digit numbers with the SPA2102 |
| 18:55.52 | gmalsack | try adding the localnet and externaddr directives and dialing a standard number. I would try 10 digits. all of the providers I know require at least 10 digits. flowroute requires 11. |
| 18:56.00 | hjf | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 18:56.08 | hjf | this is the dialplan from the provider |
| 18:57.53 | hjf | weird |
| 18:58.01 | gmalsack | yea |
| 18:58.01 | hjf | i called a free number from them |
| 18:58.08 | hjf | and it rings |
| 18:58.12 | gmalsack | whos the provider? |
| 18:58.25 | hjf | they have 6-digit numbers starting with 1, which you can call for free if you're a user |
| 18:58.30 | hjf | and it was ringing |
| 18:58.38 | hjf | it's some wholesale provider in Argentina |
| 18:58.45 | gmalsack | hmmmm |
| 18:59.17 | drmessano | Argentina? AlpacaVOX? |
| 19:00.35 | hjf | lol |
| 19:00.42 | hjf | well i get this from a friend |
| 19:00.52 | hjf | who buys from a provider called sidecoms |
| 19:00.59 | hjf | which seems to buy from someone called iplatense |
| 19:01.03 | hjf | or vice versa |
| 19:02.24 | gmalsack | sorry hjf. I'm out of ideas. you can dial voicemail, so your sip peer is good. but your provider is sending a forbidden, so I wouldn't expect it to be a nat issue since you're getting a response back from them. |
| 19:02.43 | hjf | yeah |
| 19:02.53 | hjf | maybe that dial plan mutates the numbers some how? |
| 19:02.59 | gmalsack | I would still try the localnet and externaddr just because I've seen that cause a forbidden in the past. |
| 19:03.26 | gmalsack | but short of that the only other thing I've seen cause a forbidden is an invalid number match on the server side. |
| 19:06.01 | hjf | gmalsack: what i dont understand is, what does asterisk do with the credentials? does it pass the credentials from my phone to the remote peer, or does it dial the number and connect with his name? |
| 19:07.22 | *** join/#asterisk aruntomar (~Thunderbi@49.248.157.213) |
| 19:07.40 | gmalsack | asterisk does not pass the phones credentials. the phones credentials are passed to asterisk to validate the call on your asterisk system. your asterisk system then needs it's own credentials to send to the provider. i.e. your registration line |
| 19:07.56 | gmalsack | or the sip peer configuration in sip.conf |
| 19:09.28 | gmalsack | as far as your provider is concerned, the asterisk system is the phone, it doesn't have any idea what's really behind the asterisk system. |
| 19:10.00 | gmalsack | though I did run into a provider that sent forbiddens based on the user-agent being asterisk. so that could also be the case. |
| 19:24.21 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.54) |
| 19:25.40 | hjf | lol |
| 19:25.48 | hjf | they even have a guide how to set up asterisk |
| 19:25.51 | hjf | http://www.sidecoms.com/guias.htm |
| 19:25.59 | hjf | insecure=very |
| 19:26.03 | hjf | haha |
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| 19:41.16 | [TK]D-Fender | hjf: Outdated, but accurate at that time. Also nothing irregular there |
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| 19:45.38 | j4jackj | Oh, the scourge of an international call going to voicemail... |
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| 20:13.05 | roderickm | Does res_pjsip require that the endpoint ID == auth username? Registration is successful when they match, but unauth'd when pjsip is configured for a separate username. |
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| 20:20.01 | putnopvut | roderickm: that's not a requirement of res_pjsip. |
| 20:20.44 | putnopvut | roderickm: do you have a username= in your type=auth section? |
| 20:20.56 | roderickm | putnopvut: thanks, I must be doing something wrong, then. If I set a username other than the endpoint name, I get unauth. |
| 20:21.24 | putnopvut | Is the device that is authenticating configured to use that alternate name as well? |
| 20:21.34 | roderickm | I saw the config help that said it's usually equivalent, but thought that might have become assumed as always equivalent. |
| 20:21.40 | roderickm | Yes |
| 20:22.14 | roderickm | And only on inbound auth. Outbound appears to honor the username in the auth section. |
| 20:23.36 | putnopvut | roderickm: I'll run some tests on my end and see if I get the same issue. |
| 20:23.43 | roderickm | ok, thanks. |
| 20:24.21 | roderickm | Wouldn't bother you if I hand't triple-checked. I've read the Configuring res_pjsip wikipage about six times now. |
| 20:26.22 | putnopvut | roderickm: no problem. Would rather it get fixed sooner rather than having a bunch of people complain about it. |
| 20:26.58 | putnopvut | I'm changing a phone on my desk to use a different auth username from the endpoint name to see if it'll work properly. I'll let you know how it turns out. Having to compile code and reboot phones. |
| 20:27.38 | roderickm | No hurry on my behalf — once I figured out that it worked when they're identical (like the example), I moved along… but wanted to let someone know. |
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| 20:31.05 | gmalsack | refresh my memory.... what's the new alternative to agi? |
| 20:33.35 | zigg | ARI? |
| 20:35.35 | *** join/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 20:35.55 | rrittgarn | anybody use the builtin snmp features of asterisk with Cacti? |
| 20:36.19 | newtonr | gmalsack, zigg, this will help some http://lists.digium.com/pipermail/asterisk-users/2013-September/280379.html |
| 20:36.43 | zigg | newtonr: that reply was actually to me ;) |
| 20:37.45 | putnopvut | roderickm: Just ran a test with an endpoint called "bob" with auth=bob-auth. In section bob-auth, I had username=england and password=hunter2. Authenticated fine. |
| 20:40.05 | gmalsack | rrittgarn: i looked into the snmp features a while back. since I've never done anything with them and I'm huge on snmp for monitoring my systems, I lean towards thinking I didn't really find anything I was looking for.... |
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| 20:40.31 | roderickm | putnopvut: thanks for checking. I have endpoint 3020 with auth=auth3020. In section auth3020, username=3020 will authenticate, any other username value will fail (with corresponding agent username set appropriately, of course). Tested with polycom 331 and osx tlphn app. |
| 20:41.01 | putnopvut | roderickm: Interesting. I was using a polycom 430 for my test. |
| 20:41.35 | roderickm | Chalk it up to configuration error until I can prove differently. |
| 20:41.36 | putnopvut | In the xml for the polycom, I had reg.1.auth.userId="england" and reg.1.auth.password="hunter2". |
| 20:41.50 | rrittgarn | @gmalsack: well... I will say when it works its pretty darn nifty... i have Cacti grabbing 4 of 10 of my boxes correctly. One of my higher volume boxes snmpwalks just fine, but doesn't show up in cacti... so I'm not sure if its a cacti problem, or an asterisk problem. |
| 20:42.04 | putnopvut | Not sure if the markup for a 331 is the same or not. |
| 20:42.26 | rrittgarn | @gmalsack: or even an SNMP problem, as a few of my scripts sometimes time out when walking some of the OIDs |
| 20:42.34 | gmalsack | what type of metrics are you getting from it? |
| 20:42.48 | putnopvut | roderickm: quick test would be to look at the REGISTER or INVITE from the phone that has an Authorization header in it. What does the username= part of the line have? If it's still set to 3020, then probably the phone has been misconfigured. |
| 20:42.56 | roderickm | Depends on the sip.ld version more than model number. FWIW, I had reg.2.auth.userId="3020-1" |
| 20:43.02 | roderickm | Never changed the password. |
| 20:43.30 | rrittgarn | gmalsack: The thing i care most about is concurrent channels in use, and that walks fine |
| 20:43.32 | roderickm | The displayname and label were 3020 the whole time; I only changed the userid. |
| 20:43.51 | putnopvut | yeah, displayname and label don't play into authentication, so that should be fine. |
| 20:43.59 | gmalsack | oh, yea, I just wrote a script for that... |
| 20:44.20 | rrittgarn | gmalsack: how often to you query for it? And what are you doing to store that info? |
| 20:44.44 | putnopvut | roderickm: Like I was saying though, if the Authorization header of a REGISTER or INVITE still shows username="3020" then the phone is not using the userid for authentication. |
| 20:45.09 | gmalsack | stored in sql |
| 20:45.59 | roderickm | putnopvut: It's showing the modified username (3020-1) https://gist.github.com/roderickm/e9b69e5b55ce23d762df |
| 20:46.11 | rrittgarn | how often though? (honestly not even sure how often rrdtools/cacti queries) |
| 20:46.25 | gmalsack | rrittgarn: query every 20 seconds, and stores channels, concurrent calls, active calls, transcoders, encoders, and decoders.... |
| 20:47.05 | gmalsack | rrittgarn: I think cacti out of the box only queries every 5 minutes... |
| 20:47.24 | putnopvut | roderickm: Man, this would be a good time to ask you to issue a CLI command to show what Asterisk thinks the value is for that auth section. Unfortunately, such a command doesn't exist :-/ |
| 20:47.43 | roderickm | I spent a few minutes looking for debug pjsip... |
| 20:47.48 | gmalsack | rrittgarn: yea, every 5 minutes is the cron job cacti setup on my box. |
| 20:49.19 | rrittgarn | gmalsack: good to know. I don't manage the cacti stuff here at the office, but i figure if give enough info to the people that do, they should be able to print me a pretty picture |
| 20:52.24 | *** join/#asterisk saxa (~saxa@201-50-67-77.user.veloxzone.com.br) |
| 20:52.43 | saxa | hi, it seems that I can get through with sip info |
| 20:52.48 | saxa | with dtmf tones |
| 20:54.37 | saxa | i have dtmfmode=rfc2833 , but selecting rfc2833 on the phone it does not work |
| 20:54.53 | saxa | selecting via sip info in the phone it works. |
| 20:55.15 | saxa | i have dtmfmode=rfc2833 in sip.conf |
| 20:55.26 | saxa | strange |
| 20:57.39 | saxa | is there a way how can i figure out why is this working in that wrong way and in the correct way (haveing both phone and sip.conf) with dtmfmode=rfc2833 does not ? |
| 20:58.03 | saxa | sip set debug on ? |
| 20:58.13 | navaismo | and dtmf log |
| 20:58.24 | navaismo | and rtp set debug |
| 21:00.11 | saxa | thx navaismo |
| 21:00.21 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:01.14 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:02.22 | saxa | navaismo: but rtp does only show a bunch of lines with ips |
| 21:02.24 | saxa | Sent RTP packet to 187.115.169.227:19974 (type 00, seq 005093, ts 058848, len 000160) |
| 21:02.43 | saxa | but does not show anything on digist typed or so |
| 21:02.53 | saxa | s/digist/digits |
| 21:03.52 | navaismo | it will show the type of packet including dtmf(¿?) |
| 21:04.06 | navaismo | anyway bbl going to eat |
| 21:04.53 | saxa | enjoy and thx |
| 21:07.35 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:12.21 | *** join/#asterisk nightrid3r (~kvirc@62.205.72.166) |
| 21:12.22 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 21:14.24 | hjf | how can i check which codecs are installed? |
| 21:14.45 | kaldemar | module show like codec |
| 21:15.55 | hjf | cool |
| 21:16.24 | hjf | which codec would you recommend for ip over the internet calling? |
| 21:16.43 | hjf | i tried g.729 this morning, directly from a linksys device, and it was quite good |
| 21:17.09 | hjf | for LAN calls i use g722 |
| 21:17.12 | kaldemar | any one that suits your conditions. |
| 21:18.15 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 21:19.23 | danfromuk | Hi. In 1.8, do you still add __ before the variable name to pass it to newly created channels? The reason I ask is that when transfer a call, the variables don't seem to be carried over. |
| 21:19.45 | [TK]D-Fender | hjfwhich codec would you recommend for ip over the internet calling? <- doesn't matter what it's over (if you can afford it), it's a question about what the OTHER side supports |
| 21:20.01 | [TK]D-Fender | danfromuk: Standard inheritance rules stand |
| 21:20.21 | danfromuk | Any idea why a transfer would break the inheritance? |
| 21:20.46 | [TK]D-Fender | danfromuk: Look for the key differntiator there... |
| 21:21.53 | danfromuk | [TK]D-Fender: I dont understand what you mean, however this code was working fine when I was running 1.4. |
| 21:22.21 | [TK]D-Fender | danfromuk: You aren't being precise on the action you're doing.... |
| 21:22.40 | danfromuk | [TK]D-Fender: let me see if I can fish out the dial plan for pasting. |
| 21:22.43 | hjf | [TK]D-Fender: i know but there's also a codec preference list. how would you arrange it for internet calling? |
| 21:22.51 | [TK]D-Fender | danfromuk: NOt what i'm getting at... |
| 21:23.12 | [TK]D-Fender | hjf: Stop talking about it as "internet calling". That is not a "thing" |
| 21:23.18 | [TK]D-Fender | hjf: Look at what you are calling TO |
| 21:24.53 | hjf | [TK]D-Fender: i mean calling over the internet, as opposed to calling over a lan |
| 21:25.03 | hjf | you know, when it's not a <1ms gigabit ethernet connection |
| 21:25.20 | [TK]D-Fender | hjf: Again... no. Let try this again to see if it wil register. This depends on what you are CALLING |
| 21:25.26 | hjf | ok |
| 21:25.27 | hjf | forget it |
| 21:26.11 | [TK]D-Fender | hjf: You keep asking as if "internet" was the iimportant factor. That by itself says nothing |
| 21:26.24 | [TK]D-Fender | hjf: It depends what the device you are CALLING supports |
| 21:26.41 | [TK]D-Fender | hjf: And what you need to accomodate for as far as bandwidth is concerned |
| 21:27.00 | [TK]D-Fender | The answer is pretty much "the highest quality codec both ends support that your bandwidth can handle" |
| 21:27.41 | [TK]D-Fender | but just saying "over the internet" means nothing. I could have a gigibit link to the internet. Would I use a LOWER quality one in that case? |
| 21:27.45 | [TK]D-Fender | Of course not. |
| 21:28.11 | [TK]D-Fender | So pick the best quality codec you can afford to use |
| 21:28.23 | hjf | yeah whatever |
| 21:28.24 | [TK]D-Fender | And has practical load implications |
| 21:29.33 | [TK]D-Fender | G.722 is a waste to an ITSP as that gets transcoded back to G.711 anyways and would notmally incur a loss. |
| 21:29.42 | [TK]D-Fender | So you need to look at what is on the other end |
| 21:29.50 | [TK]D-Fender | not the fact that you get to it via "the internet" |
| 21:30.21 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 21:31.51 | *** join/#asterisk przerull (~philip@philzerull.com) |
| 21:32.18 | [TK]D-Fender | You pick codecs based on CPU, bandwidth, cost, support, etc. |
| 21:32.25 | przerull | hello all. so has anyone had any luck connecting a webrtc session to the PSTN? |
| 21:33.07 | danfromuk | [TK]D-Fender: here is what i'm trying to do. http://pastebin.com/UM4dnDZp |
| 21:33.52 | [TK]D-Fender | danfromuk: So lets rewind again... what very specific action are you doing here? |
| 21:34.07 | hjf | i tested g726-32 and it sounds like shit |
| 21:34.22 | [TK]D-Fender | hjf: Depends what you're coming from. |
| 21:34.30 | danfromuk | [TK]D-Fender: that pb is not just a dial plan. Its explains the steps i'm taking. |
| 21:34.44 | danfromuk | I'm trying to set a variable which is present, even if the call is transferred. |
| 21:34.44 | hjf | g.711 |
| 21:34.50 | hjf | sounds really bad |
| 21:34.56 | hjf | as if i had a sore throat |
| 21:35.25 | [TK]D-Fender | danfromuk: I see you mention recording... but that's not what we're looking into... |
| 21:35.36 | [TK]D-Fender | it isn't G.711's fault |
| 21:35.42 | [TK]D-Fender | that is what the entire PSTN uses |
| 21:35.58 | [TK]D-Fender | is asterisk TRANSCODING there? |
| 21:36.07 | [TK]D-Fender | What are you using for endpoints? |
| 21:36.10 | danfromuk | hjf: have you run a voip tester on your connection to see if its an issue with your internet connection? http://myspeed.visualware.com/indexvoip.php |
| 21:36.11 | [TK]D-Fender | What's your bandwidth? |
| 21:36.35 | danfromuk | hjf: that link will run a latency and packet loss test. |
| 21:37.08 | hjf | g711 sounsd good. 726 sounds awful |
| 21:37.26 | hjf | i'm on a spa2102 to asterisk to a remote voip provider |
| 21:37.26 | danfromuk | [TK]D-Fender: I mention recording because thats the job of that code. However the cause of the issue is the variable inheritance is broken. |
| 21:39.12 | hjf | :\ |
| 21:39.13 | hjf | wat |
| 21:39.23 | hjf | i tested with g729 and it sounds great |
| 21:39.33 | hjf | but... i didnt install g729 in asterisk? |
| 21:39.44 | danfromuk | You don't need to |
| 21:40.02 | danfromuk | You only need to install g729 in asterisk if you are changing from g729 to something else. |
| 21:40.10 | hjf | but |
| 21:40.13 | hjf | the other day |
| 21:40.17 | hjf | i tried g.722.1 |
| 21:40.31 | [TK]D-Fender | Or * has to record in any other format. Or generate tones. Or mix. |
| 21:40.39 | hjf | and asterisk was complaining that it didint recognize g7221 @ 32k... only at 48k |
| 21:41.15 | [TK]D-Fender | danfromuk: So on your case.... what precisely is the user doing? |
| 21:41.47 | hjf | also what's it with the licensing thing? i have a linksys device that has g729. does that mean i'm entitled to use, because linksys already paid for the license? |
| 21:41.52 | danfromuk | User 1 receives a call, transfers it to user 2. User 2 speaks to the caller and presses *1 to trigger a macro. |
| 21:41.57 | [TK]D-Fender | hjf: On the SPA. |
| 21:42.02 | [TK]D-Fender | hjf: Not on Asterisk |
| 21:42.12 | hjf | [TK]D-Fender: but asterisk isn't doing anything there? |
| 21:42.15 | hjf | just connecting the calls |
| 21:42.19 | [TK]D-Fender | hjf: Not YET |
| 21:42.37 | [TK]D-Fender | hjf: It is running in PASSTHROUGH until it has to transcode. At which point it will dump your call |
| 21:42.56 | hjf | yes |
| 21:43.09 | hjf | if it runs in passthrough i dont need a license for it? |
| 21:43.19 | nightrid3r | hjf: you can use G729 without a license in paastrough mode, that is if both endponts have a G729 license and nothing changes inbetween |
| 21:43.20 | [TK]D-Fender | correct |
| 21:44.01 | [TK]D-Fender | But that means no indications tones, no recording in any other format that G.729, no meetme/confbridge, no paging, etc |
| 21:44.12 | [TK]D-Fender | And all of your prompts have to be in g.729 format |
| 21:44.19 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:44.52 | nightrid3r | and you need a license to convert prompts to G729 |
| 21:45.58 | [TK]D-Fender | no, you can do that right on Digium's site, amongst other places |
| 21:46.18 | Chainsaw | I found the licensing idea tedious. I just got one of those encoder cards. |
| 21:46.21 | nightrid3r | oh nice |
| 21:46.44 | nightrid3r | hjf: see http://wiki.villagetelco.org/G.729 |
| 21:47.36 | danfromuk | [TK]D-Fender: Is my last message what you were after? |
| 21:47.38 | [TK]D-Fender | nightrid3r: Do not link that again in here |
| 21:47.49 | nightrid3r | ok |
| 21:47.52 | [TK]D-Fender | [17:41]danfromukUser 1 receives a call, transfers it to user <- vague |
| 21:48.10 | [TK]D-Fender | danfromuk: I asked "specific action" several times... |
| 21:48.32 | danfromuk | [TK]D-Fender: I dont understand. They hit the Transfer key on the phone and dial the extension number of the other SIP Peer. |
| 21:48.49 | danfromuk | [TK]D-Fender: An attended transfer. |
| 21:48.54 | [TK]D-Fender | danfromuk: and THAT is your failure |
| 21:49.05 | [TK]D-Fender | attended transfer has NO inheritance |
| 21:49.15 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 21:49.35 | [TK]D-Fender | they are 2 completely separate channels that just happen to get glued at some later point |
| 21:49.46 | [TK]D-Fender | blind will inherit |
| 21:50.01 | danfromuk | So there is no way to pass variables across? |
| 21:50.17 | [TK]D-Fender | no |
| 21:50.32 | hjf | hm... now i can't get the SPA to talk to the softphoens |
| 21:50.46 | hjf | it's configured for g729 first, alaw second |
| 21:50.57 | hjf | the softphones are g722 first, then alaw |
| 21:51.04 | Chainsaw | nightrid3r: But yes. If you want the "easy way out" on G729, just do Digium a favour and buy one of those cards. It'll always work, for 120 channels. This is enough for most deployments. |
| 21:51.15 | przerull | 4 |
| 21:51.27 | danfromuk | [TK]D-Fender: I'm not sure youre correct as I've noticed that variables set after the transfer is complete, can't be accessed either. |
| 21:51.43 | hjf | also that's the configuration i'm using for the peers |
| 21:51.54 | hjf | disallow=all, allow=g729, allow alaw |
| 21:51.59 | hjf | disallow=all, allow=g722, allow alaw |
| 21:53.07 | hjf | shouldn't they agree on alaw? |
| 21:54.05 | navaismo | przerull with or without asterisk? |
| 21:54.27 | danfromuk | [TK]D-Fender: This bug report and subsequent fix seems to imply that variables should be inherited during an attended transfer. https://issues.asterisk.org/jira/browse/ASTERISK-5351 |
| 21:56.23 | hjf | Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) |
| 21:57.08 | hjf | [Sep 3 21:55:43] WARNING[41821]: chan_sip.c:6031 sip_call: No audio format found to offer. Cancelling call to hjf |
| 22:02.02 | danfromuk | [TK]D-Fender: Are you still around? |
| 22:02.21 | [TK]D-Fender | danfromuk: exten => 1,1,Dial(SIP/1||tT) <- also keep in mind differnces between a SIP-based transfer and an * DTMF one |
| 22:03.25 | danfromuk | I dont understand why this cause ever worked then. |
| 22:04.36 | danfromuk | [TK]D-Fender: So back to my original question, is there no way to access or pass on the variables? |
| 22:04.53 | [TK]D-Fender | Should not be |
| 22:05.09 | [TK]D-Fender | those are 2 channels with awell estrablshed variable that just happen to get bridged |
| 22:05.27 | danfromuk | I can't believe i'm the first person to have this issue. |
| 22:05.30 | [TK]D-Fender | a blind transfer would keep them |
| 22:05.53 | [TK]D-Fender | And it isn't an "issue", it's a "fact of life" |
| 22:06.40 | WIMPy | danfromuk: I know that feeling. |
| 22:08.40 | danfromuk | Hmm. |
| 22:09.16 | danfromuk | Not very impressed. It means that after an attended transfer, the callee can't do anything smart with the call apart from hangup. |
| 22:09.48 | WIMPy | wonders what that means |
| 22:10.46 | danfromuk | WIMPy: if you've just joined us... I've just been told that variables can't be inherited through an attended transfer. |
| 22:12.12 | WIMPy | Is it only about attended transfers? I'd expect the result to be the same, once the transfer is completed. |
| 22:12.31 | danfromuk | Blind transfers work fine and variables are inherited. |
| 22:13.48 | danfromuk | I thought that once the attended transfer is complete, the variables should copy over, but thats not the case. But i'm sure it worked fine in 1.4 as the code was working without any problems. Now in 1.8, i'm getting complaints from users. |
| 22:16.56 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 22:19.42 | danfromuk | So now i'm trying to work out how to use global variables to do the job i need, but thinking about it, there will be no way to match up the global variable with the original channel, since the new channel can't access any data about it. |
| 22:27.51 | danfromuk | Ok. I give up. Looks like I might have to say bye to asterisk after 4 years. |
| 22:28.43 | rrittgarn | you could store the channel info into a DB and pull it back out based on bridged channel |
| 22:28.52 | WIMPy | Just because you try to use it as a PBX. |
| 22:28.55 | rrittgarn | not the most efficient but possible |
| 22:29.23 | WIMPy | Where's the difference to globals? |
| 22:29.30 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 22:29.48 | rrittgarn | DB can store unique info per call (and add reporting later), vs Globals you could have collisions no? |
| 22:30.08 | rrittgarn | and im not saying astdb, thinking more mysqld... but thats just because I'm familiar with it |
| 22:30.57 | WIMPy | You could likewise use unique names. |
| 22:31.11 | WIMPy | The question is how you know that name? |
| 22:31.49 | danfromuk | WIMPy: [23:28] <WIMPy> Just because you try to use it as a PBX. |
| 22:32.06 | danfromuk | WIMPy: what else am I supposed to use asterisk for if not a PBX? |
| 22:32.26 | WIMPy | For what it's good at. |
| 22:32.40 | WIMPy | IVR, Conferencing, maybe as a gateway. |
| 22:33.14 | [TK]D-Fender | it works fine as a PBX |
| 22:33.21 | [TK]D-Fender | there are limitations to one kind of transfer. |
| 22:33.30 | [TK]D-Fender | beat them over the head on "this is how you have to do it. |
| 22:33.35 | [TK]D-Fender | If they can't handle it... oh well |
| 22:33.59 | WIMPy | There are lots of limitations. |
| 22:34.03 | danfromuk | [TK]D-Fender: Works fine? I'm sorry, but plenty of offices use attended transfer daily. |
| 22:36.19 | navaismo | and what is the issue with attended xfers? |
| 22:36.46 | danfromuk | It 'breaks' variable inheritance. But [TK]D-Fender will say its 'by design' |
| 22:37.04 | navaismo | adn why you want to inherit variables? |
| 22:37.13 | navaismo | so whats the real issue? |
| 22:37.41 | danfromuk | I need to be able to set a variable at the start of a call which needs to be accessed later on, even after an attended transfer. |
| 22:38.18 | danfromuk | And I'm sure it was working in 1.8, but since i moved to 11, it stopped working. |
| 22:38.37 | navaismo | you can wait to 12 hehe |
| 22:39.19 | WIMPy | still has one-way-audio there. |
| 22:39.25 | navaismo | danfromuk, use astdb to retreive the value then destroy the key when the call id hangup |
| 22:39.35 | WIMPy | Not sure waht the trigger is, yet. |
| 22:39.38 | saxa | too |
| 22:39.55 | danfromuk | navaismo: what key would I use to save and retrieve the data? |
| 22:40.12 | ChannelZ | the channel name or something? |
| 22:40.17 | navaismo | what you want |
| 22:40.47 | danfromuk | Both the orignal channel and the subsequent channels need to be able to know the same key. |
| 22:41.24 | danfromuk | I'll have to see what channel variables are available after an attended transfer to see what options there are for a key |
| 22:41.53 | *** join/#asterisk bananapie (~david@76-10-155-177.dsl.teksavvy.com) |
| 22:41.54 | navaismo | uh? astdb is visible to asterisk at any point of your dialplan |
| 22:41.56 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 22:42.01 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-gsypucskdfuxjlbi) |
| 22:42.17 | bananapie | hello, I did make menuselect and I want to copy the config only to a new source tree. Which file has my config saved? |
| 22:42.31 | bananapie | ( linux kernel used to use .config to save it, but there is not .config in asterisk ) |
| 22:42.46 | navaismo | makeopts.something |
| 22:42.49 | navaismo | let me see |
| 22:43.02 | bananapie | menuselect.makeopts |
| 22:43.03 | danfromuk | navaismo: i know, but if a call comes in and it adds a variable to astdb, then after an attended transfer, how would the data be retrieved? I can't use a static key because then other channels would pick up the wrong data. |
| 22:43.04 | WIMPy | menuselect.makeopts |
| 22:43.17 | bananapie | cool |
| 22:43.17 | bananapie | thanks |
| 22:43.40 | navaismo | yeah that |
| 22:44.36 | navaismo | danfromuk, the other channels write in another key, lets say you will use an UNIQUE ID as family and the ky will be your var |
| 22:45.31 | ChannelZ | use the caller channel name |
| 22:46.13 | danfromuk | ChannelZ: THe caller channel name won't help because after the attended transfer, that information isn't available to the peer receiving the transfer. |
| 22:47.38 | navaismo | well you have many options to create the new entry in the astdb, we cant tell you more since we dont know what are you doing, but in my experience astdb is a life saver when you need to retreivi dynamic values during a call to do another stuff in other contexts and then safely you can destroy the entry in astdb |
| 22:48.11 | danfromuk | Ok. I'll have a play with it. Thanks |
| 22:48.27 | danfromuk | Ok I'm off. Thanks for your help. Have a good night. |
| 22:48.38 | WIMPy | Yes, but how do you transport the identifier? |
| 22:48.59 | danfromuk | WIMPy: thats what i was wondering. |
| 22:49.51 | navaismo | identifier of what? |
| 22:50.28 | WIMPy | You need a unique identifier per call. |
| 22:51.16 | danfromuk | Thats available anywhere in the dial plan, regardless of what happens to the call. |
| 22:54.12 | navaismo | uniqueid? I dont know what are you doing exactly. but let me put an example: Peer1 call another number, Peer2 try to spy it, but peer 2 doesnt know from where(device) is calling peer1, so when peer 1 made the call asterisk ask a pin, then create an entry on astdb ith that pin(unique pin for every peer) with the channel name. So peer 2 has the pin list, so peer 2 use the pin to retrieve the correct channel to spy. |
| 22:54.28 | navaismo | obviously its just an example |
| 22:54.35 | navaismo | <PROTECTED> |
| 22:55.29 | navaismo | now in terms of dialplan names and vars you can create a 'name_uniqueID' family and assign the current key |
| 22:55.36 | *** join/#asterisk timahvo1 (~rogue@197.237.174.93) |
| 22:56.11 | WIMPy | And how do you get it to the destination channel if variables aren't inherited? |
| 22:57.06 | danfromuk | WIMPy: I need to check, but i'm hoping there is a variable that closely matches the original channel which i can use as a key |
| 22:59.35 | navaismo | WIMPy, you pull the vlaue from astdb |
| 22:59.39 | navaismo | value* |
| 22:59.47 | WIMPy | How? |
| 22:59.53 | [TK]D-Fender | Bridged channel <- |
| 22:59.57 | WIMPy | You need to know the key. |
| 23:00.23 | [TK]D-Fender | look at the bridged channel FOR the key |
| 23:00.25 | navaismo | ok can you guys tell me what are you trying to do |
| 23:00.46 | navaismo | its hard to get somewhere just shooting in the dark |
| 23:01.30 | ChannelZ | He needs to track some data for a complete call, even if the original call gets transferred somewhere |
| 23:01.59 | danfromuk | navaismo: A call comes in. I set a variable called 'recordingname' which is used if the call recording is triggered. Then the call gets transferred through an attended transfer. That means that if the 2nd operator tries to start the call recording, the 'recordingname' variable is unavailable. |
| 23:02.09 | ChannelZ | So at the end of the call, in the 'h' extension I presume, he needs to retrieve the user data for the call. But apparently attended transfers lose channel variables. |
| 23:02.27 | hjf | so... what could be causing the problem with the codecs? why doesn't asterisk negotiate a common code between two peers? |
| 23:02.33 | danfromuk | ChannelZ: not a completed call |
| 23:02.49 | hjf | if i have disable=all, allow=g729,alaw in one and disable=all, allow=g722,alaw |
| 23:03.01 | hjf | why doesn't asterisk make them use alaw in both? |
| 23:03.07 | danfromuk | hjf: remove g729 and try again |
| 23:03.19 | hjf | danfromuk: yes, if i remove g729 it works |
| 23:03.22 | hjf | but i want g729 |
| 23:03.32 | ChannelZ | ok, sort of same difference as far as the exercise is concerned. The trick is "sometime after a transfer" really |
| 23:03.34 | hjf | for my SIP provider and the SPA that supports it |
| 23:03.43 | hjf | and alaw for the softphones |
| 23:04.03 | danfromuk | hjf: ok so the spa is probably trying to use g729 and then asterisk can't complete the call with the other side because you don't have g729 on the other side. |
| 23:04.23 | danfromuk | hjf: if you want it to work like this, you'll probably need a g729 license. |
| 23:04.28 | hjf | danfromuk: the spa is configured for g729 first, alaw second, only prefered codecs=no |
| 23:04.41 | hjf | i dont want asterisk to transcode |
| 23:04.49 | navaismo | goes and take a look on the freepbx recording macro |
| 23:04.54 | ChannelZ | how can it not if both sides don't do g729? |
| 23:05.10 | hjf | ChannelZ: but they both do alaw |
| 23:05.15 | hjf | Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) |
| 23:05.19 | danfromuk | hjf: it will try to transcode if you don't have g729 on both sides. |
| 23:06.09 | danfromuk | hjf: from experience, its much easier to get a g729 license. Asterisk often gets mixed up. I spent days trying to get passthrough to work. In the end I gave up a bought some licenses. |
| 23:06.23 | hjf | yes but i'm on freebsd 9.1 |
| 23:06.28 | hjf | no g729 binaries |
| 23:06.39 | hjf | and i dont feel like fetching IPP to build them |
| 23:06.41 | danfromuk | hjf: no idea bout freebsd |
| 23:07.03 | hjf | there are binaries for 8.1,which may work, but only optimized for intel processors |
| 23:07.12 | hjf | and this is an amd box |
| 23:08.17 | hjf | danfromuk: passthrough (ata-asterisk-voip provider) works |
| 23:08.38 | danfromuk | hjf: so whats the issue? |
| 23:08.41 | navaismo | danfromuk, so you are using a special macro or script to trigger the recording? |
| 23:09.36 | hjf | danfromuk: i want to use softphones, without g729, to the provider - which also supports u/alaw |
| 23:10.02 | danfromuk | navaismo: yes. Its line 58 in this PB http://pastebin.com/UM4dnDZp |
| 23:10.49 | hjf | sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x40 (slin)/0x40 (slin) |
| 23:11.10 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.245) |
| 23:11.26 | danfromuk | hjf: i believe that means its trying to transcode. |
| 23:11.49 | [TK]D-Fender | hjf: You are only looking at bits and pieces of TWO different call legs and not the big picture |
| 23:13.27 | hjf | [TK]D-Fender: yes? |
| 23:14.00 | [TK]D-Fender | pastebin... the ENTIRE CALL |
| 23:15.40 | hjf | http://pastebin.com/jJVjrU5a |
| 23:16.41 | [TK]D-Fender | hjf: "core set verbose 10" |
| 23:16.45 | [TK]D-Fender | And do it again |
| 23:16.54 | [TK]D-Fender | You always want basic dialplan included |
| 23:17.43 | danfromuk | navaismo: from what i can tell, freepbx also relies on inheritance. |
| 23:17.44 | *** join/#asterisk felipealmeida (~user@177.98.104.248) |
| 23:18.11 | danfromuk | navaismo: maybe it works in 1.8 |
| 23:18.46 | hjf | http://pastebin.com/FVwgXnwX |
| 23:19.26 | *** join/#asterisk danfromuk (~androirc@unaffiliated/danfromuk) |
| 23:20.14 | hjf | Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x502 (gsm|g729|ilbc) |
| 23:20.15 | hjf | what |
| 23:20.33 | hjf | i use them with alaw all the time |
| 23:23.16 | [TK]D-Fender | surprise |
| 23:31.24 | jeev | well surprise surprise |
| 23:32.03 | ChannelZ | wierd collection |
| 23:32.14 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 23:36.11 | *** join/#asterisk Draecos (~Draecos@124.148.202.165) |
| 23:36.14 | Katty | hi kids. |
| 23:36.45 | WIMPy | Hi mum! |
| 23:36.59 | Katty | hugs WIMPy |
| 23:37.03 | Katty | how was your day dear? |
| 23:37.40 | WIMPy | Was quite ok. |
| 23:37.57 | Katty | horay! |
| 23:38.55 | navaismo | freepbx also use asterisk 11 |
| 23:39.01 | navaismo | and works |
| 23:39.05 | navaismo | danfromuk^ |
| 23:39.47 | WIMPy | Yes, I guess bridgedpeer is the key, but he's gone. |
| 23:41.12 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:43.47 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 23:43.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 23:43.58 | ChannelZ | the variable? |
| 23:44.09 | WIMPy | yes |
| 23:45.38 | ChannelZ | but it sounded like he needs to do something before the Dial in his dialplan in which case they aren't bridged yet? |
| 23:46.19 | ChannelZ | (I missed if he ever pasted an example of what he was trying to do) |
| 23:46.37 | WIMPy | As far as I got it it was just about getting variables from the first call leg to the third (the one the 2nd was transferred to). |
| 23:46.59 | WIMPy | I Didn't get the beginning, either. |
| 23:47.06 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 23:48.18 | WIMPy | wonders why bridgedpeer is always set twice per channel. |
| 23:50.30 | *** join/#asterisk aruntomar (~Thunderbi@49.248.157.73) |
| 23:51.22 | ChannelZ | Ah, found where he did paste. He has a *1 feature that launches a macro to record a call. The filename it should use is set in his dialplan when the call comes in, __RecordingFilename. He's saying if the original callee transfers the call elsewhere, the new callee's channel doesn't get the RecordingFilename for when they decide to hit *1 |
| 23:52.08 | WIMPy | Yes, I read that from what he said. |
| 23:53.40 | ChannelZ | so in that context I guess his macro would be occuring after the bridge so yeah |
| 23:54.09 | ChannelZ | anyhoo |
| 23:54.31 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 23:57.54 | navaismo | ohhh sh** |
| 23:58.00 | navaismo | i was doing a mini lab |
| 23:58.06 | navaismo | and hes gone haha |
| 23:58.58 | *** join/#asterisk serafie (~erin@24.96.64.240) |
