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| 00:35.08 | drab | I'm trying to use callfiles to pull a bunch of extensions into a confbridge and trying to do so with just one extension, but the other phones hung up as soon as I pick up. Here's the config and script: http://pastebin.ca/3260533 |
| 00:35.40 | drab | if I don't filter with the /5000 (the only phone I want to allow to initiate that process), the confbridge actually works, but then all the other phones are also triggering the script |
| 00:36.28 | drab | if I put the /5000 filter on the callfile call, then the other phones ring, but when I pick up the call ends and I see a "statys UNKNOWN" message |
| 00:36.58 | drab | even tho the other phones show up in sip show peers and work otherwise just fine |
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| 00:57.24 | drab | ok, I got it to work with a gotoif, so if the callerid is not the extension I want to enable to trigger the callfile it just skips to same,n(join), otherwise it executes system |
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| 01:27.08 | elcontrastador | Anyone having issues or know what's happening with wiki.asterisk.org? |
| 01:27.28 | elcontrastador | just came back |
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| 08:49.02 | elcontrastador | Hi guys. |
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| 08:50.53 | elcontrastador | I'm messing with the ARI on 13.6.0 and noticing that none of the list all paths are working. For instance, get /applications |
| 08:51.11 | elcontrastador | get /channels, etc |
| 08:51.23 | elcontrastador | more specific queries work |
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| 10:36.11 | dan_j | Hi. Does asterisk provide any debug reports on jitter or packet loss? |
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| 11:14.12 | aberrios | dan_j: I believe you can grab them via rtcp debug... i think |
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| 13:43.46 | WIMPy | dan_j: It sets some variables at the end of Dial. |
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| 14:09.55 | lmkone | hello all |
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| 14:23.53 | lmkone | I got passed a simple internal asterisk configuration and moved on to configure external sip trunk. The strange thing is that I am unable to make any calls via that trunk (asterisk responds with 503 Service Unavailable and does not even talk to sip trunk provider) and inbound calls sometimes work (for every 10 tries I get 2 that make it and ring internal phones). I have followed mutliple configuration samples but unable to get it t |
| 14:26.50 | [TK]D-Fender | Show us the calls |
| 14:26.52 | [TK]D-Fender | ~pb |
| 14:26.52 | 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. |
| 14:26.53 | [TK]D-Fender | ^^^ |
| 14:27.03 | [TK]D-Fender | "sip set debug on" |
| 14:27.06 | [TK]D-Fender | "core set verbose 10" |
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| 14:31.40 | lmkone | http://pastebin.com/viMKSJqM |
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| 14:40.49 | doobeh | I've got a odd little problem-- I've setup an asterisk box-- for now it's all fed by analog lines |
| 14:41.15 | doobeh | when I call in, I get connected and the audio is fine, I can hear them, they can hear me |
| 14:41.32 | doobeh | when I dial out though, it's all quiet on both sides |
| 14:41.57 | doobeh | the call does connect, just no audio |
| 14:41.58 | [TK]D-Fender | lmkone: It's looking like you didn't enable SIP debug globally, only for your calling peer |
| 14:42.31 | wdoekes | (or the other peer is unreachable) |
| 14:42.40 | [TK]D-Fender | doobeh: What are those analog lines connected to? And the bigger suspect is the OTHER end of the call. |
| 14:43.04 | [TK]D-Fender | wdoekes: No, we see a SIP COS 5 which indicates a new packet is being formed |
| 14:43.07 | doobeh | the analog lines connect to FXO ports on the xorcom box |
| 14:43.08 | [TK]D-Fender | wdoekes: Which we never see |
| 14:43.22 | [TK]D-Fender | doobeh: Show us the full call |
| 14:43.24 | [TK]D-Fender | ~pb |
| 14:43.24 | 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. |
| 14:43.25 | [TK]D-Fender | ^^^ |
| 14:46.01 | doobeh | https://gist.github.com/doobeh/04f5c102f4578bce9f19 |
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| 14:46.54 | [TK]D-Fender | SIP/2008-00000058 <- we aren't seeing the debug for this end |
| 14:47.02 | [TK]D-Fender | [09:26][TK]D-Fender"sip set debug on" |
| 14:47.04 | [TK]D-Fender | [09:27][TK]D-Fender"core set verbose 10" |
| 14:50.52 | lmkone | this is so weird - now I am able to make outbound calls - so it must be something with my sip.conf and external sip trunk configuration right ? |
| 14:51.26 | [TK]D-Fender | lmkone: You aren't showing debug for the call you are placing out |
| 14:51.27 | lmkone | have not changed anything |
| 14:51.42 | [TK]D-Fender | lmkone: that is ONLY the debug for the inbound leg |
| 14:52.02 | [TK]D-Fender | [09:26][TK]D-Fender"sip set debug on" <- Do NOT attempt to restrict to a specific peer |
| 14:52.20 | lmkone | it is what I did |
| 14:52.39 | doobeh | https://gist.github.com/doobeh/a67f903cea27e01ebab1 is this more helpful [TK]D-Fender ? |
| 14:52.47 | lmkone | after inputing "sip set debug on" and "core set verbose 10" i tried to make a call |
| 14:53.02 | [TK]D-Fender | Show us a call that actually goes out, a peer dump, etc "sip show peer freeconet-out", etc |
| 14:54.43 | lmkone | sip show peer freeconet-out --> http://pastebin.com/PxP017Y3 |
| 14:55.49 | [TK]D-Fender | I don't see it answering |
| 14:55.54 | [TK]D-Fender | doobeh: ^ |
| 14:56.15 | [TK]D-Fender | lmkone: New call.... |
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| 15:01.01 | doobeh | peer-freeconet not found |
| 15:02.28 | [TK]D-Fender | ? |
| 15:02.45 | [TK]D-Fender | Show us, don't just tell us. |
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| 15:03.33 | [TK]D-Fender | So far I see you did previously PB it |
| 15:03.39 | lmkone | i was able to make a call again, so not sure if this logs going to help but here it is anyway http://pastebin.com/XHi2TZtG |
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| 15:03.46 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:03.53 | [TK]D-Fender | Where is the new call attempt where I can also see you issuing the debug types requested? |
| 15:03.55 | *** join/#asterisk kharwell (kharwell@nat/digium/x-ynohairdkogdlfgd) |
| 15:04.34 | lmkone | but the sound quality was really bad, is it because of the codec settings ? |
| 15:05.02 | [TK]D-Fender | no, it's your internet connection, or your provider |
| 15:05.17 | [TK]D-Fender | Reliably Transmitting (NAT) to 213.218.116.66:5060: <- and your provider is NOT behind NAT. Fix your peer settings for them |
| 15:05.36 | [TK]D-Fender | No real ITSP would even require this |
| 15:06.03 | [TK]D-Fender | The call you just showed also shows your SPA303 hanging up before answering |
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| 15:06.25 | [TK]D-Fender | So we don't get to see the rest of what's negotiated |
| 15:08.18 | doobeh | Sorry, must be very trying on your patience-- when I run `sip show peer freeconet-out` during, or when a call is not active in rasterisk I just get: "Peer freeconet-out not found." |
| 15:08.24 | doobeh | https://gist.github.com/doobeh/bb870edd2cb3fd873feb |
| 15:08.31 | doobeh | is with the verbose mode set to 10 for core |
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| 15:10.30 | doobeh | oh |
| 15:10.43 | doobeh | I see, sorry-- crossed wires, I should be asking for peer 2008 right? |
| 15:11.25 | doobeh | https://gist.github.com/doobeh/f5d6691fa5610169d284 for the peer info |
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| 15:24.12 | [TK]D-Fender | doobeh: Conversations are getting mixed |
| 15:24.29 | [TK]D-Fender | so far your call looks fine except I don't see DAHDHI answering that call. |
| 15:26.19 | lmkone | bad sound quality was because of the codec setting on the spa303. I am still unable to figure out why it started working all of the sudden - have tried different settings for a few hours prior but was unsucessful |
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| 15:37.22 | doobeh | hmm, so when I call from an extension to my cellphone-- it's all quiet, but when I hang up I do get a little noise just before the call terminates |
| 15:37.57 | doobeh | could that speak to a codec issue in that when the extension dials via the analog line-- it's just not talking in an analog way? |
| 15:38.08 | [TK]D-Fender | no |
| 15:38.12 | doobeh | damn :P |
| 15:38.29 | [TK]D-Fender | DAHDI = TDM. There is no "codec" to that per-se |
| 15:40.19 | lmkone | now unable to take incoming calls -> http://pastebin.com/LP6CZkBf |
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| 15:42.10 | [TK]D-Fender | No matching peer for 'xxx' from '213.218.116.65:5060' |
| 15:42.22 | [TK]D-Fender | you are failing to match the inbound call |
| 15:43.10 | [TK]D-Fender | Which means the IP isn't matching what they are sending, and you are not allowing unidentified calls to fall through anywhere |
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| 15:44.38 | lmkone | does this line not match every call ?? exten => freeconet_in,1,Dial(SIP/10,Ttr) |
| 15:48.48 | [TK]D-Fender | nothing to do with it |
| 15:49.04 | [TK]D-Fender | diaplan only gets executed when the call is ACCEPTED |
| 15:49.07 | [TK]D-Fender | they are not accepted |
| 15:49.16 | [TK]D-Fender | * cannot identify them as a peer it knows about |
| 15:49.23 | [TK]D-Fender | [10:42][TK]D-FenderNo matching peer for 'xxx' from '213.218.116.65:5060' <- |
| 15:49.39 | [TK]D-Fender | Dialplan is what they ask for. They are not accepted as somebody * knows |
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| 15:56.12 | ThomasKeller | res_odbc.c:1412 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away |
| 15:56.20 | lmkone | [TK]D-Fender : can you advise what should i do then as I am buffled right now |
| 15:56.24 | ThomasKeller | I am getting above the error |
| 15:56.51 | ThomasKeller | looks like connection to my MySQL database times out |
| 15:57.07 | ThomasKeller | are there some "keep alive" settings ? |
| 15:57.37 | lmkone | shall i add all the ip addresses / hosts of the provider to sip.conf ?? |
| 15:58.00 | ThomasKeller | or how else can i prevent this error ? |
| 15:59.32 | igcewieling1 | ThomasKeller: that is not always an error. Asterisk will attempt to re-connect automatically. |
| 16:00.17 | igcewieling1 | ThomasKeller: there is an idlecheck option for res_odbc.conf |
| 16:02.13 | [TK]D-Fender | lmkone: As I said. Your peer doesn't match. It is not coming from an IP that matches what you have in your peer |
| 16:02.32 | [TK]D-Fender | lmkone: And if they give you the list... then yes, that's what you're going to have to do. |
| 16:04.50 | lmkone | [TK]D-Fender : thanks |
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| 16:07.30 | ThomasKeller | igcewieling1: so this idlecheck option should be some number smaller than "wait_timeout" settings in my MySQL database ? |
| 16:10.10 | igcewieling1 | ThomasKeller: Ignore the message unless you are experiencing problems. |
| 16:12.23 | igcewieling1 | I get these when the server has been idle, it reconnects and there is not a problem. |
| 16:12.23 | igcewieling1 | [Nov 19 04:15:01] WARNING[18425] res_odbc.c: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away |
| 16:12.23 | igcewieling1 | [Nov 19 04:15:01] NOTICE[18425] res_odbc.c: Re-connecting asterisk-rw |
| 16:14.13 | ThomasKeller | igcewieling1: yes, it reconnects again and everything works |
| 16:14.26 | ThomasKeller | I am just bothered by the warnings in my logs |
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| 16:15.31 | igcewieling1 | ThomasKeller: it will stop when you have enough calls to keep the connection active. I only get the message when there has been no calls for a while. |
| 16:15.44 | ThomasKeller | yes, me too |
| 16:16.05 | ThomasKeller | I don't have many calls |
| 16:16.19 | ThomasKeller | I have set "idlecheck => 600" in cdr_odbc.conf |
| 16:16.22 | rrittgarn | when using webrtc with dtls, can the certs be self signed, or do they need a valid CA? |
| 16:16.26 | ThomasKeller | lets see if that makes any difference |
| 16:19.36 | doobeh | [TK]D-Fender: got it working-- thanks for the help-- it was to do with the country settings in dahdi config. Again, thanks a million for helping narrow it down |
| 16:20.06 | [TK]D-Fender | You're welcome |
| 16:25.17 | rrittgarn | chan_sip.c:5717 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance '0x7f5dd8cc01b8', was it compiled with support for it? <--- this is just looking for res_srtp right? |
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| 16:42.20 | file | no, it requires DTLS-SRTP support in OpenSSL |
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| 16:54.20 | rrittgarn | file is there a particular package that that would be in (debian)? I have the dev packages for libsrtp but i'm on squeeze so wondering if they aren't up to date enough? |
| 16:54.42 | file | libssl-dev but whether that OpenSSL has DTLS-SRTP support or not in squeeze I wouldn't know |
| 16:54.50 | file | and without it you don't get DTLS-SRTP support |
| 16:55.12 | igcewieling1 | rrittgarn: that info is often in "make menuconfig" |
| 16:55.33 | rrittgarn | menuconfig didn't complain about anything - i didn't even see a dtls option in make menuconfig though |
| 16:56.14 | rrittgarn | just res_srtp was the only thing that required the crypto libraries - and thats with adding the --with - crypto,ssl, and srtp flags on the ./configure |
| 16:56.32 | rrittgarn | i appreciate the direction either way. thank you for the help as always |
| 16:57.24 | rrittgarn | i do have one more question - working with sipml5.org trying to test my configuration and i keep showing the reg contact as peername@[random].invalid - with errors trying to resolve that. Thought it was stun servers, but that doesn't seem to be helping either? |
| 16:57.41 | igcewieling1 | rrittgarn: resource modules / res_srtp ? |
| 16:57.49 | igcewieling1 | (I/m on Asterisk 11) |
| 16:58.05 | rrittgarn | yeah i have that - it allowed me to install without issue (and 13 here) |
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| 17:43.39 | roler | Hey guys... Trying to figure out why audio is jittery (not laggy, but actual cut outs) on inbound audio from cell phones. Inbound audio from land lines is perfect. We have a Sangoma PRI card which worked perfectly on a Time Warner Circuit. After moving and using AT&T BVoIP, the issue started. Do you guys think it could be an audio codec? They say that we need to support G.711 and G.729 codecs (protocols) and honestly i'm not sure where that is |
| 17:43.39 | roler | <PROTECTED> |
| 17:45.49 | [TK]D-Fender | You are PRI to their gatway which is basically just VoIP |
| 17:46.11 | [TK]D-Fender | Which means like anything else that is packetized that way, subject to loss, jitter, etc |
| 17:46.12 | igcewieling1 | roler: I doubt cell phones have anything to do with it. |
| 17:46.22 | [TK]D-Fender | And you should cutt out the middleman and just go direct to them |
| 17:46.45 | igcewieling1 | roler: if it was a codec issue you will never have audio |
| 17:47.13 | igcewieling1 | roler: PRIs do not support codecs. |
| 17:47.23 | roler | i see... |
| 17:47.26 | igcewieling1 | (for all reasonable values of "not" |
| 17:48.08 | igcewieling1 | if your carrier is talking about codecs, then you don't have a PRI. |
| 17:48.24 | roler | it was a trouble shooting guide, maybe it was generic |
| 17:49.23 | igcewieling1 | roler: you should throw out that guide. |
| 17:49.31 | roler | [TK]D-Fender ; when you say cut out the middle man, what do you mean? They provide us with a Cisco 3900. It has one ethernet in from the MPOE, and two out - one ethernet for internet and one PRI for our PBX. Are you suggesting there is a way to eliminate the Cisco 3900 and use VoIP directly? |
| 17:50.08 | [TK]D-Fender | you go PRI to it, it goes VoIP out. Cut out the PRI interface and that Cisco and just go direct |
| 17:50.35 | roler | [TK]D-Fender ; which would make my ears happy, that SOB has 5 hovercraft fans on it |
| 17:50.39 | igcewieling1 | [TK]D-Fender: I don't think that's what he has. |
| 17:51.36 | igcewieling1 | roler: contact your carrier and complain. |
| 17:52.15 | roler | igcewieling1 ; I will complain again.. Those guys suck honestly... can't talk directly to techs. you can only submit tickets |
| 17:52.59 | igcewieling1 | roler: assuming what you are saying is true then the issue is the carrier. |
| 17:53.28 | roler | i will definitely inquire about VoIP though... I love cutting out unnecessary equipment |
| 17:53.36 | igcewieling1 | roler: are you providing them with calla33ac0543dd5e74b5ba9480844f5ca19794b20f08cfb8532cffbefc92f9ee82553d84558ff5a45c7d15a4d5cd534b1ac72dc1715f1cd83aaab3c |
| 17:53.41 | igcewieling1 | oopd. |
| 17:54.02 | igcewieling1 | roler: are you providing them with call samples? |
| 17:55.02 | roler | igcewieling1 ; when I first set up the service, someone from India took call samples from their cisco router... They had me try an 800 number, landline, cellphone, and I think a 900 number. But no, I haven't provided call samples from my own equipment. Having a test for a landline and cell phone makes me think they have something screwed up on their end |
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| 18:13.42 | rrittgarn | anybody play with debian8 much? I'm trying to be better than an average user and not say i don't like change... tried to systemctl enable asterisk and then start it - but after a re-compile it will no longer start. I'm probably missing something but as stated - change |
| 18:16.03 | rrittgarn | and it was a problem caused by my recompile - not debian8... nevermind |
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| 18:26.29 | cloud9 | Hello everyone, I have a question I could use help with if anyone has a moment. I've had an asterisk server recording CDR's to a mysql db for a few years. As of recently I've been running into an issue where asterisk generates a uniqueid that's already been used, can't be inserted into the table, and asterisk crashes as result. |
| 18:26.52 | cloud9 | is there a way I can configure asterisk to prepend something I specify to the uniqueid? |
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| 18:44.08 | mjordan | cloud9: well, if you're repeating unique IDs, you probably have some very interesting time issues going on. |
| 18:44.42 | mjordan | cloud9: that aside, Asterisk will always prepend the hostname to the unique ID if you specify that in asterisk.conf |
| 18:45.26 | mjordan | cloud9: So if you have reset the time on your server somehow, or have otherwise made it so that the epoch timestamp is repeating with entries in the DB, you can specify the system name in asterisk.conf to avoid duplicates |
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| 19:09.41 | rrittgarn | with web rtc, is a reg contact containing 'invalid' normal? or do I have an issue to troubleshoot - googling for that term leads me to stuff from last year and i'm not sure how relevant it is |
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| 20:10.20 | Apocryphal | Evening! When doing a Blind Transfer (SIP blind transfer), is it possible to get the callerid of the one did the transfer. Right now all I'm getting is the callerid of the one being transferred |
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| 20:12.15 | igcewieling1 | Apocryphal: attended transfer shows the callerid of the one doing the transfer. |
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| 20:12.25 | Apocryphal | YEah, but this is for a blind transfer :/ |
| 20:12.48 | robmal | CEL can has. |
| 20:12.51 | igcewieling1 | Apocryphal: Then you'll have to hack something into the dialplan. |
| 20:13.04 | igcewieling1 | robmal: um, CEL is a logging system. |
| 20:13.45 | Apocryphal | igcewieling1: yeah, I've done that for one scenario, simply setting a __transferer=blabla variable. But in this scenario the Dial happens in a Local Channel, and __variables doesn't seem to inherit out of a local channel into a SIP channel |
| 20:15.53 | [TK]D-Fender | ${BLINDTRANSFER} <------------- |
| 20:16.16 | Apocryphal | That contains the channelname of the one who did the transfer - can I exchange that for a callerid somehow? |
| 20:16.47 | igcewieling1 | Apocryphal: have you looked at the f() and o() options to Dial? |
| 20:16.51 | [TK]D-Fender | Lookup the info from the peer |
| 20:17.33 | Apocryphal | igcewieling1: the o-option changes the callerid for the one receiving the call - we don't want that. Let me check f real quick |
| 20:17.49 | igcewieling1 | if not, then [TK]D-Fender is a good way. |
| 20:18.02 | igcewieling1 | if not, then [TK]D-Fender's way is a good way. |
| 20:18.28 | Apocryphal | Hmm, yeah something like SIPPEER(${BLINDTRANSFER},callerid_num) |
| 20:18.57 | Apocryphal | But will that sip peer still be present when the code runs though. I'm doing this entirely with Fast AGI |
| 20:19.18 | igcewieling1 | Apocryphal: Oh! Then I can't help you. |
| 20:19.31 | igcewieling1 | Best of luck though. |
| 20:20.55 | Apocryphal | Thanks anyway igcewieling1 ;) |
| 20:21.51 | igcewieling1 | Apocryphal: next time mention that at the start. |
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| 20:23.33 | Apocryphal | Yeah - sorry was so focused on maybe just missing some channel variable |
| 20:25.04 | WIMPy | Well, you have to extract the peer name from the chnnel name. |
| 20:25.21 | WIMPy | Off course, that will only work for chan_sip. |
| 20:25.35 | [TK]D-Fender | WELL THE LOOKUP FUNCTION AFTERWARDS THAT IT |
| 20:25.37 | [TK]D-Fender | IS |
| 20:25.40 | [TK]D-Fender | </caps> |
| 20:26.44 | Apocryphal | The channel is hung up at the point that I need it though |
| 20:28.10 | Apocryphal | Ah, and SIPPEER won't do - all the users are registered with a kamailio server. |
| 20:28.13 | Apocryphal | So there are two peers |
| 20:28.17 | Apocryphal | The two kamailio servers |
| 20:30.58 | WIMPy | Do the blonde transfer then and hope the caller ID update won't work. |
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| 21:19.36 | Kobaz | soooo |
| 21:19.45 | Kobaz | i have a customer who does logistics for a fortune 500 |
| 21:19.52 | Kobaz | is this bad... |
| 21:20.09 | Kobaz | 2 agents logged in. 25 calls waiting in queue, longest waiting is 35 minutes |
| 21:20.32 | Kobaz | going into thanksgiving and black friday sales |
| 21:20.34 | Kobaz | TWO ??? |
| 21:21.44 | paraxor | as a consumer, yes |
| 21:21.51 | paraxor | 35 minutes is ridiculous |
| 21:21.58 | Kobaz | hehe |
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| 21:28.50 | hetii | Hi :) |
| 21:29.30 | hetii | Q:I have set videosupport=yes in my sip.conf and I use zoiper app for andorid |
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| 21:29.45 | hetii | as I see I`m not able to send/get any video |
| 21:29.51 | hetii | do I need to do something elese ? |
| 21:43.38 | newtonr | Probably have the video codecs enabled for the device and in sip.conf |
| 21:44.06 | newtonr | I haven't configured a video call in a long time so I'm not too much help |
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| 23:17.19 | thinkgnu | Looks like I have a Sonicwall that likes to eat RTP traffic. Anyone else here have that experience? |
| 23:21.54 | newtonr | Maybe look for SIP ALG options. Sometimes firewalls/routers try to be helpful and end up mangling things. When in doubt simply see if you can turn the SIP ALG off. |
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| 23:23.27 | thinkgnu | I turned that off |
| 23:24.07 | thinkgnu | Strange thing is that it will do it frequently and other times not at all. I've attempted to configure traffic shaping and QoS and still can't seem to affect it in any positive way |
| 23:25.56 | newtonr | Hopefully someone here is familiar with Sonicwall. It has been years since I've messed with Sonicwall specifically. It may come down to examining the SIP packets on both sides and verifying where each side is being instructed to send RTP. That may help you know if it is only the RTP being affected itself or if it is information in the SIP SDP that is being affected. |
| 23:26.38 | thinkgnu | I examined the packets from my wan interface to my lan interface and that seems to be where I'm losing them |
| 23:26.56 | thinkgnu | They are all making it to the WAN interface and addressing is taking place there |
| 23:27.30 | robmal | I've dealt with a few sonicwalls and none ate anything. |
| 23:27.38 | thinkgnu | Most of the RTP packets are making it. I'm usually losing anywhere from .50-1.5% |
| 23:28.12 | robmal | Maybe the WAN connection? |
| 23:28.26 | newtonr | Do the lost packets account for entire calls? Or do you mean out of a single call you are losing that percentage of packets? |
| 23:28.35 | thinkgnu | out of a single call |
| 23:28.51 | thinkgnu | there are times(few) where I will lose 0.05% packets on a call |
| 23:29.09 | thinkgnu | this is when nothing is changing on my asterisk config or my sonicwall config. Just losing less packets |
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| 23:29.40 | newtonr | Odd, yeah that sounds like a more fundamental problem than configuration, but I'm not sure |
| 23:30.00 | thinkgnu | I am particularly suspect of the sonicwall anyway. Last week it started killing my openvpn traffic and I had to default it and fully reconfigure it to make it do anything sensible at all |
| 23:32.45 | thinkgnu | I was just curious if anyone else had experienced /strange/ sonicwall voip behavior before |
| 23:35.31 | thinkgnu | I'm replacing it with something else. If I were really on the ball I supposed I'd add a NIC to my Asterisk box and stick one NIC on my WAN and on in the LAN. Maybe another time. |
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| 23:44.30 | lvlinux | thinkgnu: go pfsense or astlinux |
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