| 00:00.39 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
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| 00:18.34 | moe` | oli-work: sorry, I cannot help you. You'll have to wait for one of the asterisk yoda masters to return. |
| 00:18.49 | moe` | [TK]D-Fender is one of them, he just bailed |
| 00:18.53 | *** join/#asterisk cyford33 (~support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 00:19.11 | oli-work | no problems :) thanks for the ack though |
| 00:19.17 | moe` | pokes file and malcolmd |
| 00:19.25 | oli-work | i'm not really in a rush |
| 00:19.43 | oli-work | i found some discrepancies between working and not working calls though |
| 00:20.02 | oli-work | i can't apply a fix for another 7 hours though, because production :( |
| 00:20.19 | moe` | yeah working on hot equipment sucks |
| 00:20.20 | oli-work | mostly working is better than not working :p |
| 00:21.39 | moe` | I will say that when the yoda masters are here, and you give them the info they need to help you, they are awesomely good at troubleshooting. |
| 00:23.57 | moe` | for example, they got my asterisk instance working once I gave them what they needed. |
| 00:24.21 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 00:24.29 | [TK]D-Fender | oli-work |
| 00:24.29 | [TK]D-Fender | internal > internal > pickup inbound call > return to internal call > no audio or moh <- sounds like re-invite failure |
| 00:24.45 | oli-work | we haven't enabled directmedia/reinvites |
| 00:25.02 | moe` | ah, there's a yoda master. |
| 00:25.31 | [TK]D-Fender | show the call |
| 00:25.38 | oli-work | i was going to try binding asterisk to one interface |
| 00:26.16 | oli-work | are you familiar with voipmonitor [TK]D-Fender? also what would you like to see of the call? |
| 00:34.00 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-cymsjinrkutmogyk) |
| 00:36.25 | [TK]D-Fender | the full call with SIP debug from * CLI |
| 00:36.29 | [TK]D-Fender | nothing 3rd party |
| 00:37.20 | oli-work | <oli-work> http://pastebin.com/hafYvPHB here is a sip debug |
| 00:37.41 | oli-work | i think the first part is cut off though |
| 00:38.03 | oli-work | ... waiting for someone onsite to replicate so i can get better logs :( |
| 00:52.44 | moe` | [TK]D-Fender : you must have immense patience for this |
| 00:52.49 | moe` | hats off to you dude |
| 00:52.54 | [TK]D-Fender | I have youtube. |
| 00:52.58 | moe` | ahha |
| 00:53.11 | [TK]D-Fender | when my screen flashes I start caring(ish) again |
| 00:53.45 | moe` | just for shit and giggles, what's your desktop? I'm FreeBSD fvwm2 with xchat |
| 00:54.14 | moe` | mind you, I've tweaked fvwm2 six ways from Sunday |
| 00:54.20 | moe` | but still, fvwm2 |
| 00:55.48 | [TK]D-Fender | Win 8.1 |
| 00:55.58 | *** join/#asterisk vehk (~vehk@unaffiliated/vehk) |
| 00:56.29 | [TK]D-Fender | I used Ubuntu for a couple of years but got tired of flash issues, etc |
| 00:56.59 | [TK]D-Fender | I miss the warm fuzzy OSS feeling, but appreciate that I'm more "functional" on Windows. |
| 00:57.47 | oli-work | [TK]D-Fender, did you manage to take a look at the SIP debug? |
| 00:58.28 | *** join/#asterisk vehk (~vehk@unaffiliated/vehk) |
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| 00:59.18 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:00.10 | [TK]D-Fender | I was waiting for that "better" output |
| 01:00.31 | oli-work | no problem |
| 01:01.28 | [TK]D-Fender | And nothing in there that says much to me |
| 01:01.29 | [TK]D-Fender | [2015-07-27 15:00:03] NOTICE[1646]: chan_sip.c:16778 check_auth: Correct auth, but based on stale nonce received from '"6414 Ingrid/Carol" <sip:6414@192.168.200.50>;tag=A2391835-FF21070A' |
| 01:01.29 | [TK]D-Fender | [2015-07-27 15:00:05] NOTICE[1646]: chan_sip.c:16778 check_auth: Correct auth, but based on stale nonce received from '"MARTEC USED 1" <sip:6494@192.168.200.50>;tag=dd6e8d08c4fce740o1' |
| 01:01.35 | [TK]D-Fender | This can't be good clearly... |
| 01:06.37 | oli-work | mmm... |
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| 01:13.02 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:17.25 | oli-work | is that really a problem? it appears that it's caused by asterisk reloading without the handsets being rebooted. |
| 01:17.53 | oli-work | i can't find any information indicating the affect of the above log entries |
| 01:20.11 | [TK]D-Fender | Well no hold in there... |
| 01:20.22 | [TK]D-Fender | We need something real and usable to work with |
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| 01:38.46 | oli-work | [TK]D-Fender, http://pastebin.com/KSuTHw28 |
| 01:39.16 | oli-work | this debug is of this scenario: 6404 calls 6410 ; mobile calls 6404 ; 6404 answers (which puts 6410 on hold; 6410 has no MoH nor audio when calls is picked up again by 6404 |
| 01:39.42 | oli-work | the debug is on peer 6404 |
| 01:40.03 | oli-work | appreciate your help and time :) |
| 01:40.31 | [TK]D-Fender | I'm seeing multiple subnets already |
| 01:41.04 | oli-work | they can all route to eachother |
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| 01:41.06 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:41.13 | [TK]D-Fender | automatic first suspision is improper localnet steetings' |
| 01:42.12 | oli-work | the networking for the PABX is that is has an internal IP routable from the other subnets, and a public IP which is SNAT from the firewall |
| 01:42.18 | oli-work | no public IP assigned to the pabx directly |
| 01:42.49 | [TK]D-Fender | verify your localnet settings |
| 01:43.03 | oli-work | ok |
| 01:44.30 | oli-work | curiously, if there is no externally routable IP for the PABX |
| 01:44.37 | oli-work | should externip be set to the internal address? |
| 01:46.14 | [TK]D-Fender | no |
| 01:46.20 | [TK]D-Fender | It should be set to your WAN IP |
| 01:46.29 | [TK]D-Fender | and you should be specifying your local networks |
| 01:46.42 | oli-work | thanks, i will test when i can |
| 01:46.59 | [TK]D-Fender | What have you changed? |
| 01:47.08 | oli-work | i cannot change networking settings right now |
| 01:47.23 | [TK]D-Fender | As in? |
| 01:47.27 | [TK]D-Fender | I'm talking ASTERISK here.... |
| 01:47.33 | oli-work | i know |
| 01:47.49 | oli-work | i can't make any changes that may affect its functionality (ableit already kind of broken) |
| 01:47.49 | [TK]D-Fender | so the reason you can't verify this all first is....? |
| 01:47.55 | oli-work | oh to verify only |
| 01:47.59 | oli-work | sure, localnets are empty |
| 01:48.04 | [TK]D-Fender | BAD |
| 01:48.40 | [TK]D-Fender | Don't slck on the basics. They will, over time, eventualy bite you in the ass |
| 01:48.46 | [TK]D-Fender | slack* |
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| 02:18.35 | tompaw | Hello. |
| 02:23.49 | tompaw | Can I mix ARI channels.originate with Dial(${PJSIP_DIAL_CONTACTS(blah)}) in the dialplan? OR should I handle multiple contacts in ARI? |
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| 05:07.57 | Coolguy3289 | Anyone Online? |
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| 05:18.55 | ChannelZ | nooooooope |
| 05:27.15 | moe` | hey guys, a question |
| 05:27.58 | moe` | ah crap, [TK]D-Fender is not around |
| 05:28.00 | moe` | oh well |
| 05:28.34 | moe` | ok, say a user out there calls 69@voip.mydomain |
| 05:28.37 | moe` | and it connects |
| 05:28.47 | moe` | not a SIP peer on my asterisk instance |
| 05:28.51 | moe` | but it works |
| 05:29.03 | moe` | what UDP ports would that involve? |
| 05:29.10 | moe` | or is it dynamic? |
| 05:29.42 | robmal | 5060 for signalling and 10000-20000 for data |
| 05:29.52 | moe` | ah, thank you |
| 05:37.41 | ChannelZ | * results may vary |
| 05:38.36 | moe` | a friend installed linphone and was able to call my asterisk box ###@vop.mydomain, not being connected to my asterisk server at all |
| 05:38.43 | moe` | kinda stranged me out at first |
| 05:38.49 | ChannelZ | 5060 is SIP (default). RTP can be anything. rtp.conf specifies the range of ports Asterisk will request the remote end send their RTP to; The remote end can request any port it wants for Asterisk to send its RTP to. |
| 05:39.41 | moe` | over the control channel it must be call-back to the client, cuz I have inbound ports locked |
| 05:39.46 | moe` | outbound are keep-state |
| 05:39.47 | ChannelZ | If he was able to call you, then you have unauthenticated calls turned on. Which is possibly a bad idea if you don't design your dialplan around that. |
| 05:40.10 | moe` | yeah still learning here, that's why I'm asking these questions |
| 05:40.28 | moe` | thanks for your input guys, this is exactly why I ask |
| 05:40.44 | moe` | hmmm |
| 05:40.47 | ChannelZ | I dunno what you mean by 'must be call-back' |
| 05:40.59 | ChannelZ | If he called your system, that's an incoming connection. |
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| 08:08.28 | nunne | Trying to get webrtc to work properly on asterisk 11.18 on ubuntu 14.04. I'm getting "Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd". Which seems to be a problem with uuid / uuid-dev. I have both installed from the repository. But anyone have any insight if I need to build libuuid for source? (and where can I find the source |
| 08:08.28 | nunne | for it). Or any other idea what might be my problem? |
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| 08:23.43 | Milos | What's the default key to cancel an attended transfer (say the person who you're holding the attended transfer with says that they have to go) - on hangup I expect it would tranfer, so how would one cancel the transfer? |
| 08:25.09 | nunne | Milos: on a phone or using dtmf? |
| 08:25.22 | Milos | both |
| 08:25.28 | Milos | preferably dtmf |
| 08:25.56 | Milos | I assume I can just google how to do it on a SPA504G |
| 08:26.00 | Milos | so yes dtmf |
| 08:26.15 | nunne | you are using the axfer asterisk function or are you doing attended transfer with a hardware phone? |
| 08:26.36 | *** join/#asterisk v0lZy (~Thunderbi@cpe-92-37-93-207.dynamic.amis.net) |
| 08:26.41 | v0lZy | hi |
| 08:26.41 | Milos | asterisk |
| 08:27.26 | v0lZy | I'm looking at the output of sip show channelstats ... I see a column that says 'jitter' and has a number like N.NNNN |
| 08:28.04 | v0lZy | I'm guessing it represends seconds? |
| 08:28.12 | v0lZy | 1.0003 sec for example? |
| 08:28.40 | Milos | no |
| 08:28.43 | Milos | it shows jitter |
| 08:28.45 | Milos | jitter is a thing |
| 08:28.57 | Milos | "slight irregular movement, variation, or unsteadiness, especially in an electrical signal or electronic device." |
| 08:29.06 | Milos | in other words, the latency jitter |
| 08:30.03 | nunne | Milos: try hitting *0 again (default for axfer). but I only think that would work if you started the attended transfer via asterisk axfer-function (not by pressing a button on the phone). If that doesn't work I'm sure it's not to hard to do some dialplan magic to get that functionality :) |
| 08:30.18 | Milos | v0lZy, but I think you're right, it'll be in seconds |
| 08:30.33 | Milos | but 1.0 is quite high |
| 08:31.57 | Milos | nunne, *0 worked :) thank you, is that defined in one of the config files? |
| 08:32.26 | Milos | "disconnect => *0" ? |
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| 08:35.35 | v0lZy | Milos: yeah, i was just giving a generic example... |
| 08:35.38 | v0lZy | supposing it is seconds |
| 08:35.52 | v0lZy | 0.0018 would still be small, right? |
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| 08:36.08 | nunne | Milos: from the top of my head i think *0 is default for axfer.. but might be wrong (haven't used the defaults for those things in ages) take a look at /etc/asterisk/features.conf there are comments there also about the defaults :) |
| 08:36.08 | v0lZy | though i see the number vary s bit between 0.0018 and 0.0016 on an active call |
| 08:36.26 | Milos | nunne, yeah I saw it but didn't know what it meant (comments were poor - disconnect from what?) |
| 08:36.29 | nunne | Milos: and you can change them as well to something that makes more sense (for you, that is :)) |
| 08:37.37 | v0lZy | I have this weird problem I can't get a grip on ... I call in on a line, and it has a lot of distortion in the call.. I thought It might be jitter |
| 08:37.45 | nunne | disconnect the call in that sense. so that you get axfer call back. but disconnect is just * i believe. |
| 08:37.52 | v0lZy | but my jitter values for the call are 0.0018 .. |
| 08:37.58 | v0lZy | so thats probably not it |
| 08:39.54 | v0lZy | the odd thing is, it happens only for incoming calls, and after I am put on hold and picked up again, the distortion goes away (our hold plays MOH) |
| 08:40.05 | v0lZy | and MOH is clear and crisp |
| 08:40.22 | v0lZy | and the jitter values are still there |
| 08:40.22 | Milos | maybe it's a codec thing? |
| 08:40.24 | v0lZy | so this isnt jitter |
| 08:40.37 | v0lZy | Milos: I suspect it might be a codec thing |
| 08:40.38 | v0lZy | but |
| 08:40.44 | v0lZy | I'm not sure how to check |
| 08:42.46 | v0lZy | hm |
| 08:43.16 | v0lZy | In sip.conf i changed the preferred codec from ulaw to alaw to match what I have set up on my incoming sip trunk |
| 08:43.36 | v0lZy | appears to help |
| 08:47.06 | v0lZy | the issue seems to be between alaw and ulaw transcoding |
| 08:47.14 | v0lZy | would asterisk be terribly bad at that? |
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| 08:58.09 | nunne | v0lZy: not really, but avoiding transcoding is generally a good idea. depends on the hardware you're running. but transcoding ulaw to alaw shouldn't be to painfull for the cpu. |
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| 10:30.30 | whizzi | good day all. Anyone here who knows how I can enable the SIP-header Subscription-State: in Asterisk? My Aastra phone is complaining about missing the header while things like allowsubscribe=yes on the particular peer (type=friend) ? |
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| 11:15.40 | mccar | Hi! |
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| 11:16.42 | mccar | I cant compile asterisk 11 on Debian Jessie stable. Dont know how to fix this error: Unknown value '@PBX_PJPROJECT@' found in build_tools/menuselect-deps for PJPROJECT |
| 11:17.27 | *** join/#asterisk ketas (~ketas@229-211-191-90.dyn.estpak.ee) |
| 11:18.05 | mccar | When I run "make" I got this error: "gcc: error: @PJPROJECT_INCLUDE@: no such file or folder" |
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| 11:18.40 | v0lZy | hi |
| 11:18.42 | mccar | Can you, please, help me? |
| 11:18.56 | v0lZy | mccar: what's the issue? |
| 11:19.15 | mccar | I cant compile asterisk |
| 11:19.27 | v0lZy | everyone: quick question... if i have a soft phone and a desk phone and both use the same SIP account... shouldnt they be ringing at the same time when a call comes in? |
| 11:19.36 | mccar | make[1]: *** [res_rtp_asterisk.o] Error 1 |
| 11:20.27 | v0lZy | a gcc error |
| 11:21.43 | mccar | Yes, because @PJPROJECT_INCLUDE@ is not defined |
| 11:23.04 | mccar | I have no idea where I can define it. Googling didn't help me |
| 11:25.57 | v0lZy | mccar: its not mentioned anywhere or its value is empty? |
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| 11:33.49 | v0lZy | not sure if this helps: http://lists.digium.com/pipermail/asterisk-commits/2013-March/059812.html |
| 11:35.47 | v0lZy | https://github.com/asterisk/pjproject |
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| 11:53.29 | v0lZy | any ideas how to register the same SIp account on 2 phones and have them ring at the same time? User has desktop phone and softphone when on the road with their laptop... |
| 11:53.42 | v0lZy | ideally they'd always use the same number and everyone would see the BLF for it |
| 11:57.17 | mccar | They will not work at the same time |
| 11:57.35 | v0lZy | yeah, i read up on it thanks to your earlier reply, mccar |
| 11:57.54 | v0lZy | just still wondering how to get that functionality |
| 11:58.00 | mccar | You're welcome |
| 11:58.58 | v0lZy | mccar: seems asterisk 12 can do it with pjsip ;D |
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| 11:59.27 | mccar | I'm using deban and I dont want to use backports |
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| 11:59.45 | mccar | Just try to compile stable version |
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| 12:55.14 | Ice_Strike | Hello |
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| 15:12.44 | vassilux | I use astersk 11 into 2 boxes connected by IAX2. I want to disable CDR in the 1 box if the call answered by 2nd box. Tried php script from 2nd box to update cdr table and it does not work cause the cdr is not completed. Any idea ? |
| 15:13.42 | vassilux | So my dial command Dial(SIP/phone1&SIP/phone2&IAX/mytrunk/someone) |
| 15:14.31 | vassilux | cdruniqueid passed by IAXVAR |
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| 15:19.14 | WIMPy | Hmm. What about using a gosubfrom Dial and use NoCDR from there if the IAX channel answered? |
| 15:20.32 | *** part/#asterisk whizzi (~whizzi@mailserver.mondago.com) |
| 15:22.25 | vassilux | Yes , but how I can know if the IAX2 channel answered and not SIPs phones , ${DIALSTATUS} is for my dial command , I'm missing something ? |
| 15:23.57 | [TK]D-Fender | BRIDGECHAN |
| 15:24.15 | vassilux | yesss thank :-) |
| 15:24.23 | WIMPy | Or what was it DIALEDPEERNAME? |
| 15:24.31 | [TK]D-Fender | yeah, that |
| 15:24.53 | [TK]D-Fender | Or set a var in that gosub/macro |
| 15:25.10 | vassilux | thank boys I will learn the documentation |
| 15:26.14 | WIMPy | I actually wonder, where DIALEDPEERNAME is documented. |
| 15:28.14 | WIMPy | DIALEDPEERNAME, DIALEDPEERNUMBER and BRIDGEPEER. I ghuess they shoul appear somewhere in app_dial. |
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| 15:53.32 | lvlinux | Could someone listen to this helicopter-voice voicemail and tell me if you have any ideas? |
| 15:53.40 | lvlinux | http://ruel.io/msg0003.WAV |
| 15:54.18 | lvlinux | I've gotten several of these lately and can't figure it out. When I call that DID and leave a message it's fine. |
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| 15:58.16 | lvlinux | it thought at first it might be a codec mismatch or something but the only codecs I allow from that peer are ulaw and alaw. |
| 15:59.26 | [TK]D-Fender | can't be a codec mismatch |
| 15:59.38 | lvlinux | well i didn't think so either. |
| 15:59.56 | lvlinux | * will reject the call if a compatible one is not offered right? |
| 16:00.13 | [TK]D-Fender | correct |
| 16:00.48 | lvlinux | did you listen to it? |
| 16:01.00 | lvlinux | sounds as if there is a voice there but I can't make out anything. |
| 16:01.28 | lvlinux | the other two that I received like this were from different caller id's (not that that really means anything) |
| 16:09.29 | Chainsaw | I've had helicopter-like sound effects due to a G722 codec mismatch. |
| 16:09.31 | Chainsaw | But never on alaw/ulaw. |
| 16:10.44 | lvlinux | what exactly do you mean by "G722 codec mismatch"?? I use G722 in the office with that Asterisk box, but for that call it should have been strictly 711. |
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| 16:15.25 | Chainsaw | lvlinux: It's a clock rate issue based on a historical error https://en.wikipedia.org/wiki/G.722#RTP_encapsulation |
| 16:17.21 | lvlinux | Chainsaw: ah, ok interesting. |
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| 16:42.58 | *** join/#asterisk Coolguy3289 (~Coolguy32@2602:30a:2c54:ae90:7c8d:bcf5:b339:a5e1) |
| 16:43.05 | Coolguy3289 | Anyone online? |
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| 16:43.55 | Coolguy3289 | I need done help with asterisk |
| 16:44.04 | Coolguy3289 | some* |
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| 16:47.27 | *** join/#asterisk cyford33 (support@c-73-137-1-6.hsd1.ga.comcast.net) |
| 16:47.55 | cyford33 | does anyone know if google tts still working? |
| 16:48.01 | Coolguy3289 | Idk |
| 16:48.21 | Coolguy3289 | Do you have any exp setting up asterisk correctly? |
| 16:49.27 | cyford33 | depends on what it is being setup for |
| 16:49.50 | cyford33 | why whats up |
| 16:50.21 | Coolguy3289 | Well, I have gotten it to refister with flowroute (VoIp provider) and I can call ppl with the phone, but no callback tone |
| 16:50.37 | Coolguy3289 | And no sound whatsoever when I call that phone |
| 16:50.47 | Coolguy3289 | There is absolutely no sound at all |
| 16:51.05 | cyford33 | are u natting |
| 16:51.23 | Coolguy3289 | Natting? |
| 16:51.35 | cyford33 | dos your server have a public ip address |
| 16:51.39 | cyford33 | does |
| 16:51.44 | ttl- | I'm having Asterisk PBX and serveral SIP phones on my LAN. I have a SIP trunk with 2 telephone numbers from a local SIP-provider here in Belgium. When i dial an external number the receiving party gets robotic/broken audio for the first 5 to sometimes 40 seconds of the call, after that the audio is good. |
| 16:51.58 | Coolguy3289 | Yeah |
| 16:52.13 | Coolguy3289 | It is a Hosted Dedicated server |
| 16:52.22 | ttl- | I know this hqs to be some kind of RTP problem but i can't figure it out... |
| 16:52.45 | Coolguy3289 | I'm new to this, I wish I could help |
| 16:53.09 | ttl- | Coolguy3289: NP ;-) |
| 16:53.28 | Coolguy3289 | Thatbks |
| 16:53.30 | Coolguy3289 | Thanks |
| 16:53.36 | cyford33 | Coolguy3289 so u cant hear them an they cant hear you? |
| 16:53.43 | Coolguy3289 | correct |
| 16:54.15 | Coolguy3289 | Flowroute said it was something about a private ip being displayed in the header instead of a public |
| 16:55.01 | Coolguy3289 | They also said the system is giving a 200 OK right away instead of an in progress |
| 16:55.14 | cyford33 | Coolguy3289 send ya debug info... but make sure your sending the correct codec to the provider as well |
| 16:55.42 | Coolguy3289 | how can I get that? |
| 16:57.48 | cyford33 | ttl- i think it may be your network, and mabe asterisk is correcting the jitter |
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| 17:03.09 | ttl- | cyford33: No it's not the network, i'm on cable 10Mbit up and 100Mbit down, latency to the SIP provider is less than 15ms and dev is only a few ms. When i register a SIP phone directly to my trunk there are no audio problems. |
| 17:04.52 | cyford33 | sip phone from outside your network? |
| 17:06.02 | ttl- | cyford33: no, both Asterisk and the sip phones are in the same LAN 192.168.12.0/22 |
| 17:07.35 | cyford33 | this just proves asterisk is ok internally... try register a sip client outside ya network an make a call |
| 17:08.39 | cyford33 | outside client too inside client |
| 17:11.31 | ttl- | cyford33: ok thank you, i will try it later this evening and report here. |
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| 17:33.32 | Coolguy3289 | Anyone else have any input on my problem? |
| 17:35.02 | [TK]D-Fender | Coolguy3289: PASTEBIN your failed call for us to look at |
| 17:35.03 | [TK]D-Fender | ~pb |
| 17:35.04 | 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. |
| 17:35.05 | [TK]D-Fender | ^^^ |
| 17:35.46 | Coolguy3289 | Okay, How do I get the logs? |
| 17:35.56 | [TK]D-Fender | * CLI |
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| 17:35.59 | [TK]D-Fender | "sip set debug on" |
| 17:36.03 | Coolguy3289 | Okay |
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| 17:45.40 | Coolguy3289 | Here is the log |
| 17:45.59 | Coolguy3289 | Pastebin.com/qw7WVBA1 |
| 17:46.18 | cyford33 | wow, google tts enabled a captia on it |
| 17:47.09 | cyford33 | whats the best tts to use other than google |
| 17:48.43 | [TK]D-Fender | Coolguy3289: Contact: <sip:12345424534@127.0.0.1:5060> |
| 17:48.44 | [TK]D-Fender | 883 |
| 17:48.57 | [TK]D-Fender | you have not properly set up * to know your WAN IP |
| 17:49.36 | Coolguy3289 | I know the IP, I just don't know where to change that |
| 17:49.45 | [TK]D-Fender | and that is a VERY big telltale sign that you likely put a HOST entry for some FQDN directly on your box which is screwing you over from resolving one externally. |
| 17:50.47 | Coolguy3289 | The only entries that were changed were the Sip.conf and extensions.conf files |
| 17:50.56 | [TK]D-Fender | ^ |
| 17:50.58 | [TK]D-Fender | <PROTECTED> |
| 17:50.59 | [TK]D-Fender | ^ |
| 17:51.01 | [TK]D-Fender | check it |
| 17:51.09 | [TK]D-Fender | and pastebin your sip.conf [general] |
| 17:52.14 | Coolguy3289 | I would have to exclude some of thatbdata |
| 17:54.05 | [TK]D-Fender | no more than 3 line should need masking |
| 17:54.10 | [TK]D-Fender | I don't need register's |
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| 18:15.45 | [TK]D-Fender | Coolguy3289: Issue resolved |
| 18:15.53 | Coolguy3289 | Again, thanks |
| 18:16.02 | [TK]D-Fender | np |
| 18:16.16 | [TK]D-Fender | [13:49][TK]D-Fenderand that is a VERY big telltale sign that you likely put a HOST entry for some FQDN directly on your box which is screwing you over from resolving one externally. <- this was it |
| 18:17.01 | Coolguy3289 | I like irc better than the forums...faster help most of the time |
| 18:17.57 | [TK]D-Fender | The more you're able to show the faster the better... |
| 18:18.56 | Coolguy3289 | Yeah, well I will definitely be hanging around! |
| 18:21.26 | lvlinux | Hey, would having srvlookup enabled keep Asterisk from accepting registrations if Asterisk can't reach a DNS server?? |
| 18:23.01 | lvlinux | I lost internet on my home Asterisk server the other day and to my horror, no calls would go through until it was restored. Even *97 for voicemail. |
| 18:23.52 | lvlinux | I'm trying to figure out how to prevent that next time. |
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| 19:09.41 | dork | any sip trunk providers anyone can recommend for US/can/int that allow sales/call center type ops and supports pjsip? we're having problems getting our pbx up on vitelity's SBC because they're acting like they don't know what pjsip is |
| 19:10.11 | [TK]D-Fender | pjsip works fine, ans does vitelity as a provider |
| 19:10.15 | TazzNZ | pjsip is a lib, it should need to be supported |
| 19:10.31 | [TK]D-Fender | I'd focus on the likely minor task of finishing some minor debugging |
| 19:10.34 | TazzNZ | shouldn't* |
| 19:11.02 | dork | so anyone want to answer my actual question? |
| 19:11.16 | dork | we're going to have a second provider for redundancy anyways |
| 19:11.38 | [TK]D-Fender | I don't think you'll find any who make any specific mention of pjsip for support. |
| 19:11.40 | [TK]D-Fender | they are all just SIP |
| 19:11.54 | [TK]D-Fender | as for otehr vendors just for redundency: |
| 19:11.57 | [TK]D-Fender | ~itsplist-us |
| 19:11.57 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 19:12.10 | [TK]D-Fender | flowroute would be a good one |
| 19:12.20 | dork | voip.ms doesn't allow call center type ops |
| 19:12.22 | [TK]D-Fender | voip.ms uses vitelity as their back-end IIRC |
| 19:12.24 | dork | ok thanks i'll check on them |
| 19:12.51 | [TK]D-Fender | dorkvoip.ms doesn't allow call center type ops <- this I've nevber seen before. Is it in print somewhere? |
| 19:13.10 | dork | yes on their SLA |
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| 19:13.25 | TazzNZ | geepers - what are you doing then dork ? |
| 19:13.42 | dork | TazzNZ: what do you mean? |
| 19:15.14 | TazzNZ | that is a messed up SLA |
| 19:15.28 | TazzNZ | 25. The customer will not use VoIP.ms termination (outbound) services for telemarketing purposes (Including, but not limited to Automated Dialers, Call Centers and collection agencies). VoIP.ms will suspend such activities without warning. |
| 19:16.17 | [TK]D-Fender | Guess they want to avoid getting complaints due to spammers |
| 19:16.40 | TazzNZ | I can understand Automated Dialers, but the rest ? |
| 19:16.58 | TazzNZ | unless by call centres they also mean outbound |
| 19:17.20 | [TK]D-Fender | they said outbound explicitly |
| 19:17.34 | [TK]D-Fender | "VoIP.ms termination (outbound) services" |
| 19:17.36 | TazzNZ | true |
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| 19:19.40 | pjensen00 | I'm crashing Asterisk somehow. I don't see any error messages before it goes down in either the console or the log file "messages". Is there a way to catch the actual error that is crashing Asterisk? |
| 19:19.57 | TazzNZ | but surely outbound means money, where as inbound doesn't ? |
| 19:20.15 | TazzNZ | pjensen00, edit asterisk.conf and up the verbose and debug settings |
| 19:20.25 | pjensen00 | thanks |
| 19:21.19 | pjensen00 | wait, will that increase the verbosity of the console? If so, I already have both of those at 10 while it's running |
| 19:21.50 | pjensen00 | would turning on dumpcore capture it? |
| 19:21.55 | [TK]D-Fender | TazzNZbut surely outbound means money, where as inbound doesn't ? <- outbound means fraudsters forging callerID and violating DNC lists, etc |
| 19:22.04 | [TK]D-Fender | And all sorts of scammers |
| 19:22.43 | TazzNZ | it should also increase what is logged at startup pjensen00 |
| 19:22.50 | pjensen00 | ah ok, thanks |
| 19:23.12 | TazzNZ | [TK]D-Fender, I must admit, I have never understood why people don't implement callerid restrictions.... |
| 19:23.43 | TazzNZ | the DNC thing I don't understand fully, since I am not in the US or any where where it is enforced |
| 19:23.50 | [TK]D-Fender | because they can technically off it and just don't want to give it to assholes who will cause them trouble |
| 19:23.58 | dork | TazzNZ: i was ready to sign up for voip.ms and then i came across that, it's weird that they will not allow 'business' outbound traffic |
| 19:24.05 | dork | doesn't sound like a sound business model but wtf do i know |
| 19:24.38 | [TK]D-Fender | They allow "business" |
| 19:24.48 | [TK]D-Fender | they just don't want outbound spammers |
| 19:24.57 | dork | not if the business relies on outbound cold calls to generate revenue |
| 19:25.07 | dork | then tht is not business conducive no matter how pedantic one tries to be |
| 19:25.23 | [TK]D-Fender | There my be a grey line in terms of how that might get classified. |
| 19:25.29 | [TK]D-Fender | Have you called to confirm? |
| 19:25.35 | [TK]D-Fender | may* |
| 19:25.38 | dork | im sure there is, but i would never onboard with a company with verbiage like that |
| 19:25.42 | dork | their loss not mine |
| 19:25.51 | [TK]D-Fender | ok/fine/sure. |
| 19:25.58 | [TK]D-Fender | NEXT!@@!@11!oneoneone |
| 19:26.34 | TazzNZ | lol |
| 19:26.53 | dork | haha, how can you go on to the next when you contributed nothing but geek nostril flaring in the first place |
| 19:27.13 | TazzNZ | that is [TK]D-Fender and we love him for it |
| 19:27.17 | dork | s/NEXT/FAILED |
| 19:27.41 | [TK]D-Fender | dorkhaha, how can you go on to the next when you contributed nothing but geek nostril flaring in the first place <huh? |
| 19:28.32 | dork | (kidding, thanks for the info) |
| 19:29.19 | TazzNZ | VoIP.ms does not encrypt its transmissions, or sensitive parts of customer information in its database. <-- That would be a worry |
| 19:30.10 | pjensen00 | TazzNZ: Sadly, that didn't spit out the error. Just a crash from asterisk and silence. |
| 19:31.12 | TazzNZ | try running this |
| 19:31.26 | TazzNZ | asterisk -f -d -C /etc/asterisk/asterisk.conf |
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| 20:28.05 | Coolguy3289 | Anyone know how to get asterisk to automatically add the 1 for US numbers? So that 10 digit numbers can be dialed? |
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| 20:38.37 | lvlinux | I'll try this again: i have srvlookup set to yes in sip.conf. Callcentric requires it. When I lose internet, my PBX pretty much dies as it won't do any sip connections. Any way to prevent this? (I have a PSTN line as well, so it still needs to work with no internet access.) |
| 20:47.53 | Cuzner | Coolguy3289: how do you determine if something is a US number? |
| 20:48.38 | Coolguy3289 | The calling code for US and Canada is +1 |
| 20:49.16 | Coolguy3289 | The only way asterisk knows if I am dialing am outgoing number is to make sure a 1 is before the number |
| 20:50.07 | lvlinux | Coolguy3289: you can do this: |
| 20:50.57 | lvlinux | exten => _XXXNXXXXXX,1,Dial(SIP/voippeer/1${EXTEN},,) |
| 20:51.18 | lvlinux | that prefixes a 1 before the number dialed if it is a 10 digit format |
| 20:51.43 | Coolguy3289 | If I remember correctly, I have something similar to that inly with a 1 before the XXX |
| 20:52.08 | lvlinux | exten => _NXXXXXX,1,Dial(SIP/flowroute/1985${EXTEN},,) |
| 20:52.09 | Coolguy3289 | So would I just Add that to the outgoing dialplan? |
| 20:52.28 | lvlinux | ^^^ that one also adds the area code if you want to dial a 7 digit number |
| 20:52.46 | lvlinux | yes put that in your extensions.conf where you catch outgoing dial strings. |
| 20:52.55 | Coolguy3289 | Okay, thanks |
| 20:53.07 | Coolguy3289 | I will try it in a minute |
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| 20:56.07 | litn | hey guys, is there a way for me to find out what my highest number of concurrent calls has been? |
| 20:56.29 | litn | I want to switch from a minutes based billing to a concurrent call based billing, but I'm not sure how to go about finding out how many I need |
| 20:56.38 | litn | I can randomly check call volume but I don't think that gives me a good idea |
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| 21:01.51 | malcolmd | not that asterisk can tell you about interactively, no, not that i can think of. |
| 21:03.25 | litn | any suggestion then outside of asterisk? |
| 21:03.42 | litn | should I write a scrip to call sip show calls every few seconds and let it count for a week or something? |
| 21:06.26 | malcolmd | you could...it's not a good operation to run on a busy system prior to 12/13 as it'll lock the channels to give you the list. 12/13 it'll pull that information out of stasis instead. if you're logging calls or channels in cel you could look there to see how many you had going at once |
| 21:10.31 | litn | oh that's true, I do log in CDR in MySQL, I could figure out a query for that |
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