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| 01:12.10 | *** join/#asterisk TehRabbitt (ad467d21@gateway/web/freenode/ip.173.70.125.33) |
| 01:12.13 | TehRabbitt | hey |
| 01:12.49 | TehRabbitt | so i've been struggling to get this cisco 7970 to register. I was able to get the firmware updated, and it's grabbing the cnf.xml correctly, but it just sits at "Registering" and i'm not sure if it's the phone or the server that's the issue :/ |
| 01:13.36 | TehRabbitt | I'm using FreePBX as the front-end but obviously asterisk at the backend. I'm wondering if there's a log or such that would show said failed registration attempts |
| 01:15.09 | lvlinux | TehRabbitt: asterisk -rvvvvv and watch for the failed registrations. |
| 01:15.58 | lvlinux | TehRabbitt: IIRC the 7970 may need to use TCP, rather than UDP to work correctly, so you might want to check into that. I know the 8961 I think it was would only work over TCP. |
| 01:15.59 | [TK]D-Fender | Which you shouldn't see unless you actualy have SIP debug enabled |
| 01:16.11 | lvlinux | oh yes that's right. |
| 01:16.13 | TehRabbitt | Ah maybe that's why |
| 01:16.25 | lvlinux | TehRabbitt: and they run "sip set debug on" |
| 01:16.27 | TehRabbitt | I know it just spins on "Registering" and then "Resetting" and keeps looping |
| 01:16.27 | [TK]D-Fender | And I've never seen a phone that didn't work with UDP |
| 01:17.22 | lvlinux | [TK]D-Fender: yep some of the Cisco's need TCP with the SIP firmware. I thought it was the stupidest thing ever when I found that out. Had a client that just had to use his Cisco phones... |
| 01:17.27 | TehRabbitt | firmware version SIP70.8-5-4S |
| 01:17.52 | lvlinux | I'm not positive the 7970 is one of them though. Could be another issue. |
| 01:21.45 | TehRabbitt | SIP/2.0 401 Unauthorized |
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| 01:27.23 | [TK]D-Fender | straight-up auth so far |
| 01:32.14 | TehRabbitt | unless I have the username wrong or something |
| 01:33.10 | [TK]D-Fender | Which is clearly at least half of what "auth" is |
| 01:33.11 | TehRabbitt | http://pastebin.com/wxEEzegT |
| 01:33.49 | [TK]D-Fender | <--- Transmitting (NAT) to 192.168.1.41:49280 ---> |
| 01:33.54 | [TK]D-Fender | phone is NOT behind NAT |
| 01:33.59 | [TK]D-Fender | fix your extension |
| 01:34.36 | TehRabbitt | okay, NAT disabled |
| 01:34.52 | TehRabbitt | o_o |
| 01:34.55 | TehRabbitt | that... fixed it |
| 01:35.10 | TehRabbitt | so NAT == Auth issues? |
| 01:35.56 | [TK]D-Fender | you are restricting PORTS when you enable that |
| 01:36.13 | TehRabbitt | Ah I see :) that makes sense |
| 01:36.28 | TehRabbitt | now I just have to mess with the dialplan.xml file so that I can actually dial numbers lol |
| 01:39.32 | *** join/#asterisk jmordica (uid18332@gateway/web/irccloud.com/x-uwwfhxcpjpzidaib) |
| 01:39.59 | jmordica | does anyone have experience with realtime sip peers and pruning "Unreachable" ones? |
| 01:40.54 | jmordica | I wrote a small script to prune unreachable realtime peers and it works (removes them) but they come back shortly after as unreachable. Even a "sip reload" doesn't do the trick. Only after an asterisk restart do the unreachable peers go away. |
| 01:48.28 | TehRabbitt | w00t... I have 2-way audio on my first inbound test call |
| 01:48.46 | TehRabbitt | now I have to figure out why the phone keeps saying "reorder" every time I dial ANY digit :/ |
| 01:48.48 | lvlinux | TehRabbitt: yay! |
| 01:49.08 | TehRabbitt | lol :P |
| 01:49.13 | lvlinux | yep has something to do w ur dialplan.xml file |
| 01:49.36 | TehRabbitt | http://pastebin.com/CcW7SEDY |
| 01:49.43 | TehRabbitt | it looks good to me :/ so not sure why it's not working |
| 01:51.37 | lvlinux | http://pastebin.com/tM3KLfVy |
| 01:51.45 | lvlinux | ^^^^^^^^Here's mine^^^^^^^^^^ |
| 01:51.50 | lvlinux | take a look and compare. |
| 01:52.44 | TehRabbitt | thanks lvlinux |
| 01:52.51 | lvlinux | np |
| 01:53.00 | TehRabbitt | I'm gonna try going line by line |
| 01:53.08 | TehRabbitt | I'm thinking syntax error somewhere in it |
| 01:53.13 | lvlinux | possibly |
| 01:53.55 | drmessano | What are you dialing? |
| 01:53.58 | drmessano | TehRabbitt, |
| 01:54.10 | TehRabbitt | drmessano: trying to dial 1-xxx-xxxx |
| 01:54.11 | drmessano | Oh ANY digit |
| 01:54.17 | TehRabbitt | as soon as I hit ANY digit |
| 01:54.18 | TehRabbitt | "Reorder" |
| 01:54.58 | TehRabbitt | lvlinux: yeah, i tried using your xml... same issue, goes right to "reorder" |
| 01:55.17 | lvlinux | hmmm |
| 01:55.22 | lvlinux | idk |
| 01:55.43 | TehRabbitt | it shows TFTP Error: dialplan.xml |
| 01:55.53 | TehRabbitt | also TFTP Access Error: XMLDefault.cnf.xml |
| 01:55.59 | TehRabbitt | even though they are both there. |
| 01:56.49 | lvlinux | well that means it never even got to my file, or yours either for that matter. |
| 01:57.04 | lvlinux | make sure you have it case sensitive exactly how the phone is looking for it. |
| 01:57.20 | lvlinux | dialplan.xml vs DIALPLAN.xml vs DIALPLAN.XML |
| 02:01.27 | drmessano | Permissions too |
| 02:03.24 | TehRabbitt | it's so weird :/ |
| 02:03.29 | TehRabbitt | perhaps it's perms. |
| 02:03.38 | TehRabbitt | though i'm' pretty sure everything should have read access |
| 02:03.53 | TehRabbitt | it worked for 1 call, "Your call cannot be completed as dialed"... but subsequent calls go "reorder" |
| 02:07.40 | drmessano | TFTP Access error means just that |
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| 02:07.59 | TehRabbitt | yeah, I changed the TFTP server to my local desktop instead of my server, put the xml there and it works. |
| 02:08.06 | TehRabbitt | so it has to be a server config. |
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| 02:21.18 | TehRabbitt | w00t it works :D it all works |
| 02:21.19 | TehRabbitt | lol |
| 02:21.23 | TehRabbitt | thanks again guys! |
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| 06:53.10 | slav3_sergal | stupid question of the day. |
| 06:53.47 | slav3_sergal | how complicated is it to setup a dialplan to ring an extension based on incoming caller's phone number? |
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| 06:58.26 | WIMPy | slav3_sergal: Not at all. Take a look a the "/callerid" option of extensions. |
| 06:59.01 | slav3_sergal | thanks, googling now |
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| 07:06.21 | igcewieling | slav3_sergal: I perfer using the more modern GotoIf($["${CALLERID(num)}" == "1234567890"]?match-label) ExecIf is also handy |
| 07:06.51 | igcewieling | you can also google for Asterisk ed-girlfriend or mother-in-law options |
| 07:06.57 | slav3_sergal | lol |
| 07:08.01 | igcewieling | WIMPy: earlier this evening I managed to do a remote executing command using DTMF transport to encode the ASCII 8-) |
| 07:08.59 | igcewieling | roughly 1/10th the speed of the earliest modem, but it worked. |
| 07:17.23 | slav3_sergal | god i feel dumb, it's been so long since i played with this phone network lol |
| 07:17.43 | slav3_sergal | and i forgot basically everything |
| 07:19.09 | slav3_sergal | but i got it fixed so mom's not so frustrated dealing with my ivr when she calls me |
| 07:20.39 | slav3_sergal | out of curiosity has anyone messed with those real fancy looking asterisk voip phones? I'm rocking cisco setup for sip |
| 07:20.57 | slav3_sergal | they do the job but it'd be nicer to maybe upgrade to a fancy new phone |
| 07:30.21 | sgelardi | igcewieling: what command did you end up executing? |
| 07:33.40 | WIMPy | igcewieling: So next level is IP then :-) |
| 07:39.54 | igcewieling | sgelardi: 'ls' |
| 07:41.07 | igcewieling | WIMPy: it is 8-bit clean so, in theory, you should be able to run almost protocol over it. |
| 07:42.30 | igcewieling | s/almost protocol/almost any protocol/ |
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| 07:57.53 | skrusty | #j asternet |
| 07:57.56 | skrusty | oops |
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| 08:09.56 | sgelardi | igcewieling: cool, any interesting reads on this? To be honest I never heard of this before |
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| 08:36.36 | igcewieling | sgelardi: something I "invented" |
| 08:37.12 | igcewieling | I hope to document the protocol and release it. more of a toy, but there are a few edge cases I can think of where it could be handy. |
| 08:42.00 | igcewieling | "Hello World" on the wire is "7b000048656c6c6#20576#726c6413*d" |
| 08:44.11 | igcewieling | start (8-bits), sequence number (16-bits), data (variable), ccitt crc (16-bits), end (8-bits) |
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| 09:06.55 | sgelardi | igcwieling: pretty interesting nevertheless, what do you use for authentication? |
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| 09:12.24 | thothcastel | Would an asterisk device seat behind the firewall or directly to the line (Fiber line) |
| 09:12.25 | thothcastel | ? |
| 09:21.46 | sgelardi | thothcastel: what kind of network infrastructure do you have? |
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| 09:23.11 | sgelardi | thothcastel: the simple answer is though, it should not be public ally exposed. |
| 09:32.05 | sgelardi | igcewieling: I could see a pretty pratical use for it. It maybe slow, but RAS from a cell phone |
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| 12:04.10 | Jesterboxboy | is there a table somewhere where i can see which sip code maps to which dialstatus in asterisk? |
| 12:08.16 | WIMPy | It's in the source. |
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| 12:09.18 | Jesterboxboy | that i assumed :D, i just thought maybe theres a mapping table somewhere floating around |
| 12:09.37 | Jesterboxboy | any hint as to where i should look? |
| 12:10.14 | WIMPy | Actually, I think I have seen it on the wiki as well. |
| 12:10.38 | WIMPy | But it may be easier to find in chan_sip.?. |
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| 12:18.42 | Jesterboxboy | mhm there i can find it mapped to AST_CAUSES, but wher are thes then set to the dialstatus ? |
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| 12:19.13 | thothcastel | mouse settings won't be saved after a reboot - why not? |
| 12:19.18 | thothcastel | el capitan |
| 12:19.24 | WIMPy | They are the DIALSTATUSes. |
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| 12:27.43 | Jesterboxboy | then iam not getting something, for example i cant find CHANUNAVAIL or ANSWER |
| 12:29.37 | Jesterboxboy | while for example sip response code 404 is returning AST_CAUSE_UNALLOCATED in channel_sip.c, but |
| 12:30.05 | Jesterboxboy | i cant find this dialcode listed in the info about the variable $DIALSTATUS for example in wiki.asterisk.org |
| 12:30.11 | nixhr | I've setup a queue and MOH and everything works fine. However, when a queue member makes a blind transfer to another extension, the caller hears MOH while on hold, which is not preferred. I want him to hear a ringing tone while on hold waiting for the transfer to finish. |
| 12:31.17 | nixhr | How do I set different MOH for the queue and for the transfers between members? |
| 12:32.59 | nixhr | if I set different moh class before the Queue command in the dialplan, then the moh class is applied to the queue, which is not what I want. And I don't see another place where I could put the moh class selection. Any idea would be very apreciated. thanks. |
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| 13:10.44 | Jesterboxboy | <PROTECTED> |
| 13:10.44 | Jesterboxboy | in chan_sip.c it switches to AST_CAUSE_UNALLOCATED , but i cant find what dialstatus this is |
| 13:10.56 | Jesterboxboy | in respect to what dial i.e sets |
| 13:17.04 | catphish | is DAHDI still required for a software-only asterisk install? |
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| 13:35.06 | Jesterboxboy | how would i catch any number with 2-5 numbers between 0-9 ? |
| 13:35.30 | Jesterboxboy | extension i mean |
| 13:35.37 | [TK]D-Fender | catphish: DAHDI is never required for just "Asterisk". At this point only MeetMe requires it AFAIK. |
| 13:36.05 | [TK]D-Fender | Jesterboxboy: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 13:36.18 | catphish | [TK]D-Fender: thanks, meetme was likely the reason i've always needed it, is meetme still the best way to do conferences? |
| 13:36.30 | [TK]D-Fender | No, Confbridge |
| 13:38.33 | Jesterboxboy | thanks TK, but i have already looked at exactly this document, but couldnt find it |
| 13:39.08 | Jesterboxboy | i want something that matches any number with 2 to 4 digits |
| 13:39.27 | [TK]D-Fender | Jesterboxboy: "asterisk dialplan patterns" <- easiest google search ever |
| 13:40.18 | [TK]D-Fender | Jesterboxboy: Have you finished reading that whole doc on dialplan patterns? |
| 13:41.35 | Jesterboxboy | yes |
| 13:42.41 | [TK]D-Fender | Jesterboxboy: Then you'd see there is nothing to specify a variable length except for "." which doesn't do it. That tells you immediately that you can't do it in 1 shot. |
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| 13:43.19 | Jesterboxboy | okay thanks for the info, i asked because i thought mabye it isnt documented or i just overlooked it |
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| 13:44.26 | [TK]D-Fender | Jesterboxboy: Nope. |
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| 15:15.45 | hirogen1 | hi |
| 15:16.15 | hirogen1 | how do i setup a freephone number in avaya suite administrator |
| 15:16.29 | hirogen1 | avaya amdinistrator as the telecoms guy is on annual leave lol i do have access to it |
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| 15:24.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:25.01 | bounceman | In my extensions.conf I have the following exten => _X.,1,Verbose("do something"), just before that line I include another context with exten => 6001.,1,Verbose("Do something different"). |
| 15:25.13 | bounceman | I read that asterisk will look for a match before even looking at the include. |
| 15:25.20 | bounceman | Since _X. will match everything, am I screwed? |
| 15:25.52 | igcewieling | hirogen1: I believe you are on the wrong channel. |
| 15:26.05 | igcewieling | bounceman: no. X. will match 2 or more characters. |
| 15:26.32 | igcewieling | . will match all characters, that's why Asterisk throws up warnings if you use it. |
| 15:27.01 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 15:27.01 | bounceman | http://pastebin.com/nGzfRC5Q |
| 15:27.16 | bounceman | when I dial 6000, I do not get the Hello this time |
| 15:28.20 | hirogen1 | ah ok |
| 15:28.48 | igcewieling | bounceman: show us a cli output of a failed call. |
| 15:28.50 | igcewieling | pastebin it. |
| 15:29.09 | igcewieling | I suspect a context issue. |
| 15:30.04 | bounceman | The call does not fail, it dials 6000 |
| 15:30.14 | bounceman | but it does not match on 6000 but on _X. |
| 15:30.21 | igcewieling | I still need the pastebin or I cannot help further. |
| 15:30.43 | bounceman | Hang on, I will get a new paste |
| 15:31.51 | catphish | [TK]D-Fender: thanks, i'll look at confbridge, i'm completely reworking my config to remove all deprecated/unsupported modules |
| 15:33.20 | bounceman | http://pastebin.com/QzxDsRKd |
| 15:33.23 | bounceman | start with this |
| 15:33.56 | bounceman | 080000000 is just a pseudo num |
| 15:33.58 | igcewieling | bounceman:that does not look like Asterisk CLI output. |
| 15:34.31 | bounceman | The CLI output gives you no errors or anything, it just tells us it's hitting on this |
| 15:34.31 | bounceman | exten => _X.,1,Verbose(2,==> INCOMING CALL FROM PSTN - ${CALLERID(num)} TO ${EXTEN}) |
| 15:34.45 | bounceman | Why would it be beneficial to see it? |
| 15:34.50 | igcewieling | bounceman: I think the call is coming into the sip-custom context, not the sip-in context. However that is entuirely speculation until I see the CLU output. |
| 15:35.21 | igcewieling | bounceman: Perhaps [TK]D-Fender has the required patience. |
| 15:36.16 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 15:36.24 | bounceman | Again, the asterisk CLI give you nothing you're asking for. |
| 15:36.36 | igcewieling | btw, next time tell people you are using FreePBX. |
| 15:37.06 | bounceman | SECCOBELIX01*CLI> core show version |
| 15:37.06 | bounceman | Asterisk 11.7.0~dfsg-1ubuntu1 built by bui |
| 15:37.19 | bounceman | We do not use Freepbx |
| 15:38.17 | igcewieling | For example this tells me the call came into the switch_manager context. [Nov 24 10:37:40] -- Executing [+17183455940@switch_manager:1] NoOp("SIP/vzb-regular-7-00100ef0", "*****************************************************************************************") in new stac |
| 15:38.41 | igcewieling | now, if I was expecting the call to come into a differnet context, then I just found my problme. |
| 15:39.06 | igcewieling | anyway, I've wasted enough time here. back to work for me. |
| 15:39.10 | bounceman | "Asterisk will always try to find a matching extension in the current context first, and only follow the include statement to a new context if there isn't anything that matches in the current context." <- This is what I was talking about |
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| 16:09.48 | [TK]D-Fender | bouncemanWhy would it be beneficial to see it? <- because you might want to see it to make it easier to search your CLI output & logs |
| 16:09.58 | [TK]D-Fender | bounceman: The same reason you put comments in code |
| 16:11.28 | bounceman | But I already said 4 times that the CLI gives nothing to this issue except for what I already said. |
| 16:11.57 | [TK]D-Fender | Show us the actual call output |
| 16:12.23 | bounceman | it is solved now, I had to do some reorgainisation. Asterisk wiki says what it says |
| 16:12.32 | Kobaz | anyone have any issues with receiving a call and having delayed audio setup |
| 16:13.17 | Kobaz | someone calls a DID on an itsp, call goes over to asterisk, asterisk does a Dial() (audio's not going yet). And then the polycom endpoint picks up, and says hello, but the outside caller did not hear the hello |
| 16:13.21 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 16:13.41 | Kobaz | me personally I've gotten into the habit of knowing i'm on an asterisk system and waiting like 2-3 seconds after picking up to say hello |
| 16:13.55 | igcewieling | Kobaz: is direct media enabled? |
| 16:13.59 | Kobaz | no |
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| 16:14.12 | [TK]D-Fender | bounceman: Ok, that's good. Any other questions currently unanswered? |
| 16:14.36 | bounceman | No thank you |
| 16:14.52 | dorphalsig | Hey all. Can anybody give me a hand with an (stupid) ICE error? |
| 16:15.02 | Kobaz | igcewieling: nope, doing b2bua |
| 16:17.21 | Kobaz | if i Answer() before Dial()'ing. I wonder if the first few ms wont get cut off |
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| 16:41.57 | bounceman | General question, modifying FROM and TO headers are not a good idea? |
| 16:45.35 | [TK]D-Fender | In general you simply can't |
| 16:45.48 | bounceman | So r uri is the way to do it? |
| 16:45.54 | [TK]D-Fender | not really a thing |
| 16:46.05 | igcewieling | Doing it outside of Asterisk, such as in Kamailio |
| 16:46.07 | [TK]D-Fender | What specifically are you looking to do? |
| 16:46.49 | [TK]D-Fender | Rememebr * is a B2BUA, not a proxy. So nothing is getting "apseed on" so much |
| 16:46.52 | [TK]D-Fender | passed* |
| 16:47.22 | [TK]D-Fender | new invite may take the "from" based on callerID unless overriden elsewhere |
| 16:47.30 | bounceman | Kamailio is a proxy for us, we would love to keep these sip transormations outside of it if possible |
| 16:48.02 | [TK]D-Fender | Give us a specific sample we can confirm |
| 16:48.20 | igcewieling | bounceman: Asterisk is not a proxy. If you want proxy functions such as rewriting headers, do it outside of Asterisk. |
| 16:51.06 | *** join/#asterisk MadHatter42 (~MadHatter@unaffiliated/madhatter42) |
| 16:51.31 | bounceman | We just have some new requirements on FROM and TO Header display, knowing that modifying FROM and TO is "forbidden" in SIP 2.0 I am just looking for someone to bounce with |
| 16:52.06 | [TK]D-Fender | Well * is a B2BUA ... there is no such thing as "modify" really. |
| 16:52.14 | *** join/#asterisk genpaku (~genpaku@107.191.100.185) |
| 16:52.16 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 16:52.30 | [TK]D-Fender | The words "entirely new call" come to mind.... |
| 16:53.37 | igcewieling | ~book |
| 16:53.41 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:54.55 | mjordan | I think a better question is why are you even bothering to modify the From and To header, and what parts are you trying to modify |
| 16:55.12 | mjordan | Doing that - particularly modifying the From header - is really a very, very bad idea. |
| 16:55.16 | mjordan | VERY. |
| 16:55.28 | mjordan | Unless you like your system not working, in which case, modify away |
| 16:55.31 | bounceman | Ofc, but request-uri? |
| 16:55.32 | bounceman | :-) |
| 16:55.39 | [TK]D-Fender | Well again there isn't even such thing as "moduify" really since you aren't forwarding the SIP traffic |
| 16:55.48 | mjordan | Er |
| 16:55.54 | mjordan | You determine the outbound request-uri yourself |
| 16:56.01 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 16:56.07 | mjordan | and since Asterisk supports dialling by SIP URI, I'm not sure why you would need to modify it |
| 16:56.25 | bounceman | Because we interconnect with other operators |
| 16:56.31 | bounceman | and they have certain demands. |
| 16:57.16 | mjordan | That doesn't actually answer my question. If different operators require different things in the request URI, then somewhere you will have logic that says if it is an operator of one particular type, send this; if not, send this other thing |
| 16:57.27 | mjordan | That might as well be your dialplan. |
| 16:57.41 | mjordan | That is Asterisk's preferred way of modifying things |
| 16:57.54 | mjordan | If you need something that is operating at a lower "SIP" layer, then Kamailio is more appropriate. |
| 16:58.28 | bounceman | And exactly that it what we are doing. My question has been answered, we will do it in the kamailio |
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| 17:41.18 | *** join/#asterisk thinkgnu (~matt@216.67.183.210) |
| 17:43.01 | thinkgnu | I'm sure my problem is unique and I hope someone more experienced than myself has a simple fix: I lose RTP data to/from my SIP provider using asterisk behind a NAT. Packet sniffing shows it's dropping between my WAN Interface and my LAN Interface. Any ideas? |
| 17:46.33 | igcewieling | sounds like a typical firewall / nat issue. What is your externip= and localnet= set to and what, if any, ports are you port forwarding to your Asterisk |
| 17:47.57 | thinkgnu | I have my localnet= set to my lan subnet and externip= set to our public IP, which is a static IP. |
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| 17:48.32 | igcewieling | make sure you have directmedia=no too. |
| 17:48.37 | thinkgnu | I get most packets but usually I hover right under 1% loss and sometimes pile up to 5% loss(more of a rarity). |
| 17:49.24 | thinkgnu | directmedia=no in the main portion of sip config? |
| 17:49.55 | thinkgnu | I do have it set for my endpoints that are behind NAT with Asterisk. |
| 17:53.17 | igcewieling | set it everywhere 8-) |
| 17:53.52 | thinkgnu | lemme give that a go |
| 17:54.17 | thinkgnu | I have an ATA the SIP provider gave me and I can't seem to find anything in their config that is magic and yet it works behind NAT wonderfully |
| 17:54.51 | igcewieling | thinkgnu: packet loss is usually a transport / router issue. |
| 17:54.55 | thinkgnu | everything behind my NAT is 0% packet loss. I tend to agree it is the NAT but I've tried two different firewalls with all protocol helpers turned off |
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| 17:55.48 | igcewieling | thinkgnu: are you forwarding any ports? |
| 17:57.12 | thinkgnu | I have it setup to forward all udp sip 5060 and all rtp udp 10000-20000 |
| 17:57.47 | thinkgnu | If it matters it is a Sonicwall TZ200 |
| 17:58.47 | *** join/#asterisk Coralisk (~coral@jonasbengtson.se) |
| 17:58.48 | Coralisk | hey |
| 17:58.52 | Coralisk | anyone that knows sip.js well here? |
| 17:58.58 | Coralisk | and asterisk |
| 17:59.11 | igcewieling | Try forwarding only 5060. The outbound audio from the server to the client should automatically create the NAT translations for audio from client to server. |
| 17:59.53 | thinkgnu | I'll give that a try real quick |
| 18:01.12 | igcewieling | thinkgnu: I manage a large number of Asterisk boxes. A few of them are behind NAT with no clients on the internet, then no port forwarding is done. Some are behind NAT and have clients on the internet, for those we have to portforward 5060 |
| 18:01.29 | igcewieling | 5060/UDP not TCP. |
| 18:01.50 | *** part/#asterisk infinity1 (~brendon@web2.artsopolis.com) |
| 18:02.11 | thinkgnu | So right now I have port forwarding from my WAN Interface IP to my asterisk box which includes 5060 and RTP ranges. I'll remove the RTP ranges and see what I get real quick |
| 18:04.22 | thinkgnu | that was worse actually |
| 18:04.28 | thinkgnu | that was worse actually |
| 18:04.35 | thinkgnu | whoops, didn't mean to repeat that |
| 18:05.33 | igcewieling | define "worse" |
| 18:05.38 | igcewieling | more packet loss? |
| 18:05.48 | *** join/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 18:05.57 | thinkgnu | higher packet loss and it is evident in the audio |
| 18:06.02 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 18:06.29 | igcewieling | weird. are you sure all SIP alg is disabled in the sonic wall? |
| 18:07.01 | thinkgnu | it is disabled, yes |
| 18:08.14 | thinkgnu | I ran a packet monitor and the RTP loss to WAN is 0% but the loss that my asterisk box shows and that the LAN Interface is showing (this is incoming) is identical so it must be NAT. |
| 18:08.37 | thinkgnu | Is there anything my provider could be doing that could provoke such behavior with NAT? |
| 18:09.20 | igcewieling | A few providers like Verizon Wireless who activly try and block sip., doesn't seem all that common. |
| 18:09.35 | thinkgnu | my sip provider is also my ISP so I know they are not doing that |
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| 18:11.04 | thinkgnu | what is the preferred NAT setting in sip.conf? I've tried everything out of desperation |
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| 18:15.28 | igcewieling | nat=force_rport,comedia |
| 18:15.57 | thinkgnu | I'll give that a go |
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| 18:29.58 | thinkgnu | this could well be a fluke but that seems to have helped igcewieling |
| 18:30.40 | thinkgnu | there are times when loss gets really really low. 0.05% for example. but it goes back up. This is with other network traffic and without. |
| 18:37.51 | thinkgnu | Yes, loss came back up. I'd correlate it with traffic if I hadn't done testing with the voice traffic entirely isolated |
| 18:38.40 | igcewieling | can you check the firewall cpu util? |
| 18:39.44 | thinkgnu | yes. 18%-35% |
| 18:47.46 | *** join/#asterisk iulhk (~iulhk@host-208-net-68-160-119.mobilinkinfinity.net.pk) |
| 18:48.32 | thinkgnu | Here is another really strange thing...When I call phone numbers that are withing my provider's direct network, their NOC for example my incoming traffic is perfect but outgoing takes a major dive |
| 18:48.52 | thinkgnu | It only happens when calling the provider though. I can't put my finger on this really bizarre issue |
| 18:49.31 | iulhk | can i get connected call info after every minute and update database by using asterisk perl agi ? |
| 18:51.12 | WIMPy | iulhk: What information are you looking for? But No, not via AGI. |
| 18:52.22 | iulhk | <WIMPy>: if call is connected then i want to update call duration after every minute ? |
| 18:53.56 | WIMPy | You have to do it in another task. But the concept smalls funny anyway. |
| 18:54.05 | igcewieling | iulhk: Asterisk will not allow you to update the call duration -- that is all handled internally to Asterisk. |
| 18:54.52 | igcewieling | You should be able to use the Asterisk Manager Interface if you want to get the call duration info. Can't imagine why though. |
| 18:55.56 | iulhk | <igcewieling>: ok , thank you |
| 18:59.35 | thinkgnu | At this point, I think the best solution is to throw a second NIC into the machine and just put it on a public IP with no NAT. I am not gaining any ground on this issue at all. |
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| 19:27.11 | pjensen00 | sure |
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| 20:17.40 | deweydb | hey guys! |
| 20:17.56 | deweydb | can someone tell me what setting i need to change to make asterisk save voice mail recordings to disk |
| 20:18.05 | deweydb | right now it seems like they just get emailed and the original disapears |
| 20:19.49 | MrTAP | delete=no |
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| 20:23.40 | lgaetz | Att: @mjordan, working a Sangoma ticket and the question of the status of SILK codec for *13 came up. Is there anything beyond "it's on the list" that I can relay to them? |
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| 21:19.50 | tacky | lvlinux: Hey. I just a got a test phone call from John Todd to ISN UNIX*1917 (aka 8649*1917) and audio worked no problem. He agreed that since we got audio when you called via SIP URI directly but not when via ISN/ITAD that it made no sense. In any case, my incoming is working for sure. I still don't know what audio problem we had between us, but it is definitely strange. Anyway, thanks again for helping me test. |
| 21:25.56 | mjordan | lgaetz: unfortunately, not much else right now. Sorry :-\ |
| 21:26.12 | mjordan | lgaetz: higher priority stuff just keeps bouncing up. I'll try to see if we can't push it up a bit higher. |
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| 21:52.55 | lgaetz | @mjordan, this is a decent sized client. If they were interested in sponsoring the migration, what would the starting point be as far as a contact at Digium |
| 21:58.08 | mjordan | Probably Hardy |
| 21:58.14 | mjordan | dhardy@digium.com |
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| 22:02.52 | lvlinux | tacky: hmm, well the last call was the sip call. Maybe it would have worked if I had tried it again. :-) |
| 22:03.42 | tacky | lvlinux: yeah, good point. well, you still have fingers and i still have ears. lol |
| 22:05.08 | lvlinux | yeah maybe I'll have a chance to try later this eve or sometime. (mobile now) |
| 22:05.59 | tacky | lvlinux: right on |
| 22:08.12 | lgaetz | thanks @mjordan |
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