| 00:04.15 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:09.05 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 00:14.32 | Katty | hi kids. |
| 00:14.59 | Penguin | Would you like me to call you mommy? |
| 00:15.09 | Penguin | It seems a little weird, but I'll do it. |
| 00:16.00 | WIMPy | does |
| 00:16.14 | WIMPy | Katty: Hi mom! |
| 00:16.29 | Katty | hi dear. |
| 00:16.32 | Katty | hugs WIMPy |
| 00:16.32 | Penguin | Oh. I thought YOU wanted me to call you mommy. |
| 00:16.53 | Katty | it doesn't matter. i'm already pratically channel mom anyway |
| 00:17.38 | WIMPy | Just that we only get talk about food. |
| 00:17.56 | Katty | but food's good. |
| 00:18.04 | Katty | it's a nice common topic that everyone seems to like |
| 00:18.07 | Katty | and few get pissed off about |
| 00:18.09 | WIMPy | yes |
| 00:24.39 | WIMPy | Oh, Cisco has a new protocoll? SPCP instead of SCCP? Or is it just a new name? |
| 00:26.00 | Katty | maybe it was a typo. |
| 00:26.39 | WIMPy | No |
| 00:26.53 | WIMPy | "Smart Phone Control Protocol" |
| 00:27.44 | Katty | fancy. |
| 00:30.36 | Katty | i wonder if it works like vnc or rdp |
| 00:30.57 | WIMPy | has seen that before. |
| 00:31.09 | WIMPy | That wasn't smart. |
| 00:36.21 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:42.11 | WIMPy | Isn't SCCP like that anyway? |
| 00:42.35 | Penguin | Not smart? |
| 00:42.42 | *** join/#asterisk andrewyager (~andrewyag@8-104-141-114.static-dsl.realworld.net.au) |
| 00:43.03 | WIMPy | No, what I have seen was not smart at all. |
| 00:43.32 | WIMPy | Or SCCP? |
| 00:46.42 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 00:54.39 | *** join/#asterisk SGjunior (~sgjunior@out-on-180.wireless.telus.com) |
| 00:55.22 | *** join/#asterisk dongola7 (~dongola7@pool-71-178-179-231.washdc.fios.verizon.net) |
| 00:57.15 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 01:02.32 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:13.23 | *** join/#asterisk jpsharp (~jsharp@c-76-101-138-247.hsd1.fl.comcast.net) |
| 01:14.07 | jpsharp | I need to webstream the results of a chanspy operation. Suggestions on the best way to go about it? |
| 01:14.34 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-zrhbpllbvsdhuhkj) |
| 01:14.34 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 01:14.48 | Penguin | I don't even know what that means. |
| 01:16.51 | jpsharp | I need to monitor a channel via chanspy but make that live audio via a web browser rather than a ip phone. |
| 01:17.31 | Penguin | Now that is a concept that I can understand. |
| 01:18.57 | jpsharp | It made sense in my head :) |
| 01:19.17 | igcewieling | SCCP = Skinny ??? Control Protocol |
| 01:19.35 | igcewieling | Lightweight, maybe sort of like MGCP. |
| 01:19.47 | Penguin | I was trying to figure out how the exit status of ChanSpy() was going to be useful in a stream of media over the internet. I couldn't figure it out. |
| 01:19.51 | igcewieling | If Cisco had opened up the protocol SIP may have never taken off. |
| 01:20.14 | jpsharp | Skinny Call Control Protocol |
| 01:20.25 | Penguin | ~sccp |
| 01:20.25 | infobot | methinks sccp is Proprietary protocol used between Cisco Call Manager and Cisco VOIP phones. Also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in SS7 protocol suite in layer 4, provides end-to-end routing for TCAP messages to their proper database. |
| 01:20.56 | Penguin | *shrug* |
| 01:21.14 | igcewieling | Penguin: thanks! I wonder why people call it Skinny |
| 01:21.31 | igcewieling | ah, that part is about SS7, nevermind |
| 01:21.41 | Penguin | Yeah, two separate things there. |
| 01:21.45 | jpsharp | SIP had a 3 year head start on SCCP. |
| 01:22.08 | igcewieling | jpsharp: I imagine MGCP did too and we know how that turned out. |
| 01:22.25 | igcewieling | jpsharp: think "softphone in a browser" and you'll be part way there. |
| 01:23.06 | jpsharp | I was trying to avoid that. |
| 01:25.14 | igcewieling | you could run a console softphone to dial into chan spy, then encode the output into something you could play in a browser then stream it using something like shoutcast. |
| 01:25.42 | igcewieling | When you say "browser" "audio" and "asterisk" you should also say "overly complicated and not usually worth it" |
| 01:25.47 | jpsharp | Though I might have a rube goldbergish way of going it. Use AMI to kick off a mixmonitor process that writes to a fifo/pipe and read back out of that pipe later. |
| 01:26.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:28.09 | jpsharp | This should earn me a "Congressional Medal of Ugly" or a spot on thedailywtf.com |
| 01:30.21 | igcewieling | jpsharp: I recently wrote something to send 8-bit ascii over DTMF.. 8-) |
| 01:30.25 | WIMPy | Well, wH.323 was much more popular until SIP took over because people thought it was easier, when it acutally only was simpler. Now we see ASN.1 coded SIP messages. How sensible that turned out... |
| 01:30.41 | igcewieling | so, I'm familiar with rube golberging. |
| 01:31.13 | igcewieling | WIMPy: the nice thing about sccp/mgcp is the "smarts" is in the PBX not in the endpoint. |
| 01:31.27 | WIMPy | You could call yourself Ruber Goldcomm :-) |
| 01:31.37 | Penguin | infobot: No, sccp is <reply>SCCP is Skinny Client Control Protocol, a Cisco proprietary protocol for call signaling. Used between Cisco Call Manager and VoIP phones, also supported by some other vendors. Also Signaling Connection Control Part (SCCP), a routing protocol in the SS7 protocol suite (in layer 4), which provides end-to-end routing for TCAP messages to their proper database. |
| 01:31.37 | infobot | Penguin: okay |
| 01:32.26 | jpsharp | I used to think that "H323 was written by telco guys with no concept of IP" and "SIP was written by IP guys with no concept of telco". I still stand by the H323 part, but SIP was developed by guys who could spell IP and that was about it. |
| 01:33.01 | WIMPy | very much doubts that SIP will ever become usefull. |
| 01:33.04 | jpsharp | grumbles about SIP and NAT. |
| 01:33.35 | WIMPy | Well, that's the part it has in common with H.323, jut that H.323 will notice the failure. |
| 01:34.03 | *** join/#asterisk hjf (hjf@unaffiliated/hjf) |
| 01:35.20 | WIMPy | I found this last week. That's finally a somehow relistic text. Especially the link to the SIP myths. |
| 01:35.25 | WIMPy | http://www.packetizer.com/ipmc/h323_vs_sip/ |
| 01:35.29 | hjf | hi... a little offtopic but, has anyone tried android softphones? I set up csipsimple in 2 phones and connected it to my asterisk server. i called a phone and, while it connected instantly, the voice had terrible lag. greater than 1 second |
| 01:35.48 | [TK]D-Fender | hjf: And on another client? |
| 01:36.13 | WIMPy | Actually I think IAX was rater well made. A shame that Digium seem to think otherwise. |
| 01:37.06 | hjf | [TK]D-Fender: actually i don't have other clients... only these 2 phones |
| 01:37.22 | hjf | well 3 phones really. i tried with HTC One X, htc sensation and motorola milestone |
| 01:37.33 | [TK]D-Fender | no, that's HARDWARE. |
| 01:37.39 | [TK]D-Fender | I asked about SOFTWARE |
| 01:37.48 | [TK]D-Fender | 3cx, Android's native SIP client, etc |
| 01:38.04 | hjf | oh ... lol sorry |
| 01:38.08 | hjf | i'll try |
| 01:38.22 | [TK]D-Fender | hjf: And you never told us what thhe phone was connected via.... |
| 01:38.54 | hjf | wi-fi |
| 01:39.02 | hjf | both to the same network |
| 01:39.13 | [TK]D-Fender | ok, change clients |
| 01:39.13 | hjf | i had no idea android had a built in sip client |
| 01:40.12 | [TK]D-Fender | IIRC it got added in 2.4 |
| 01:40.48 | hjf | mine doesn't seem to have it |
| 01:40.49 | Penguin | It's in my 2.3 |
| 01:40.55 | Penguin | gingerbread |
| 01:40.58 | hjf | i'm using cyanogenmod though |
| 01:41.01 | hjf | android 4.2.2 |
| 01:41.16 | Penguin | But I like CSipSimple better than the native SIP client. |
| 01:41.34 | hjf | i thought it could be codec-related maybe |
| 01:42.21 | [TK]D-Fender | nope |
| 01:43.00 | hjf | oooooooooohh there it is... in the phone application itself. lol... i'll remove csipsimple for a bit |
| 01:50.24 | [TK]D-Fender | ok, I'm out for a while |
| 02:01.47 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 02:03.20 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 02:17.28 | *** join/#asterisk tengulre (~tengulre@171.221.149.52) |
| 02:26.00 | hjf | okay... i think i got it |
| 02:26.03 | hjf | i installed x-lite |
| 02:26.07 | hjf | much less latency |
| 02:26.44 | ChannelZ | igcewieling: Yes (re: nat=yes) and that bit turns out to be a bonafide bug, and seems fixed in 11.5.1 (I was on 11.4.0) |
| 02:27.09 | igcewieling | ChannelZ: it was fixed in 11.4 and 11.5 and 11.5.1 and and and |
| 02:27.36 | ChannelZ | But it still seems like some other mechanisms aren't working. Playing with Blink, and it is giving out the LAN IP. But whatever I do I can't get asterisk to ignore it. |
| 02:27.47 | igcewieling | I've basically given up upgrading for a while. |
| 02:28.42 | ChannelZ | Like, my understanding is that comedia should send RTP back to the IP/port it received from, but that's not happening. |
| 02:28.57 | igcewieling | At some point it will REALLY be fixed. |
| 02:28.58 | ChannelZ | Asterisk is blissfully sending RTP to 192.168.1.5 |
| 02:29.17 | igcewieling | ChannelZ: that is a classic symptom of a bad localnet |
| 02:29.37 | ChannelZ | Yeah I even fudged my localnet too though and it still won't do it. |
| 02:30.46 | ChannelZ | Previously it was 192.168.0.0/16 but I restricted it to 192.168.2.0/24 (where all the actual phones are on its LAN) but the 192.168.1.5 from my home machine seems to still get treated as LAN (I guess?) |
| 02:32.41 | ChannelZ | Gonna mess with it a bit more in a little bit to make sure I'm not doing something painfully wrong |
| 02:34.10 | j4jackj | Does freetring commit toll fraud? |
| 02:35.49 | j4jackj | heh |
| 02:35.54 | j4jackj | did you sip reload? |
| 02:37.26 | hjf | very strange. when i dial the voicemail extension from my android phone, the audio is terrible. completely choppy, can't hear a single word |
| 02:37.35 | hjf | but from x-lite (wired connection) it works fine |
| 02:37.52 | hjf | the server is asterisk 1.8 inside a freebsd jail, fwiw |
| 02:37.57 | Penguin | Which codec did you set up for the android? |
| 02:38.03 | j4jackj | GSM? G722? |
| 02:38.12 | Kobaz | sounds like bad transcoding or something |
| 02:38.16 | hjf | well i installed csipsimple and the codec pack |
| 02:38.24 | hjf | and it lets me choose: |
| 02:38.30 | Penguin | You don't really need the codec pack. |
| 02:38.43 | Penguin | Without the codec pack, you still have most every codec you could want to use in asterisk. |
| 02:38.45 | ChannelZ | Have you played with the 'clock speed' in the advanced settings? |
| 02:38.56 | hjf | pcma 8khz, mpeg4 generic, silk 24k and pcmu 8khz |
| 02:38.58 | Penguin | So I'm asking which codec you chose, not which codecs you installed on the phone. |
| 02:39.29 | hjf | Penguin: which codec in asterisk? |
| 02:39.48 | Penguin | Which codec. We're talking about asterisk and your phone. |
| 02:40.03 | Penguin | There is one codec that they use together. Which one is it? |
| 02:40.21 | hjf | how do i tell? csipsimple doesn't tell me and i didn't set any in asterisk |
| 02:40.47 | ChannelZ | In-call you can hit the little i button and see |
| 02:40.58 | ChannelZ | I think it's an i in a circle. Phone's in the other room. |
| 02:41.14 | ChannelZ | Or just turn off all codecs but one in CSip :) |
| 02:41.30 | Penguin | You can also check with "sip show channels" on the asterisk CLI. |
| 02:42.15 | Penguin | (only during a call, of course) |
| 02:42.48 | Penguin | Most people who use asterisk know what codec(s) they configure for a given peer. |
| 02:43.22 | hjf | [Aug 30 02:42:40] NOTICE[18291]: channel.c:4237 __ast_read: Dropping incompatible voice frame on SIP/hjf-0000000c of format ulaw since our native format has changed to 0x2 (gsm) |
| 02:43.34 | Kobaz | sounds like your problem |
| 02:43.57 | Penguin | I say most because some, such as j4jackj, do allow=all and have no clue what codec is actually being used. |
| 02:44.19 | hjf | guilty... |
| 02:44.25 | j4jackj | Yes. But I tell everyone to 'prefer' G722 by 'putting it at the top of the list' |
| 02:44.34 | Penguin | disallow all, then allow ONLY THE CODEC(s) that you want the device to use. |
| 02:44.43 | Kobaz | +1 |
| 02:44.45 | j4jackj | I want all codecs to be usable... |
| 02:44.49 | Penguin | g722 isn't always what should be preferred. |
| 02:44.59 | j4jackj | It is on broadband |
| 02:45.06 | Kobaz | all codecs? even ones you may not have a license for? |
| 02:45.07 | Penguin | And if you allow all, you don't get to choose which one is preferred. |
| 02:45.13 | hjf | what's the best codec for "i dont care about bandwidth i only want it to be as fast as DECT"? |
| 02:45.18 | igcewieling | " Dropping incompatible voice frame" is generally a harmless message and happens on the first frame when transcoding |
| 02:45.19 | j4jackj | I don't USE the ones I'm not licensed for |
| 02:45.29 | Kobaz | j4jackj: then that's not all codecs |
| 02:45.31 | Penguin | Then why offer them in SDP? |
| 02:45.31 | j4jackj | hjf: G711 |
| 02:45.54 | j4jackj | OK. Do I 'disallow=g729' after allow=all? |
| 02:46.03 | Penguin | You could use ulaw or g722 if bandwidth is not a problem. |
| 02:46.17 | Penguin | No, you disallow all and then allow only the ones you want to have available. |
| 02:46.18 | j4jackj | By allowing all codecs I avoid compatibility problems. |
| 02:46.25 | j4jackj | Mostly. |
| 02:46.26 | Kobaz | not necessarily |
| 02:46.27 | j4jackj | :D |
| 02:47.05 | j4jackj | MOST compat issues. |
| 02:47.20 | j4jackj | As in when you dial in to the server |
| 02:47.21 | igcewieling | I use G722 for as much as I can, ulaw for the rest, and g729 when required. |
| 02:47.28 | hjf | g.711 is ulaw/alaw? |
| 02:47.29 | j4jackj | rather than dial direct a person |
| 02:47.35 | j4jackj | g711 is alaw |
| 02:47.37 | j4jackj | and ulaw |
| 02:47.43 | Penguin | disallow=all \ allow=g722 \ allow=ulaw \ allow=ilbc |
| 02:47.53 | j4jackj | I don't like iLBC |
| 02:48.15 | Kobaz | this conversation is getting rather silly |
| 02:48.16 | j4jackj | I've heard it sounds like Mr. Roboto. Or was that LPC10? |
| 02:48.18 | Penguin | At least now you understand how the preference list works. |
| 02:48.42 | Kobaz | "i want to use all codecs"... no i'm not using g721... no i don't like ilbc |
| 02:48.58 | igcewieling | hjf: yes |
| 02:49.03 | j4jackj | heh |
| 02:49.21 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 02:49.21 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 02:50.14 | j4jackj | hi newtonr |
| 02:50.20 | newtonr | j4jackj, yo! |
| 02:50.33 | j4jackj | having a discussion bout codecs |
| 02:50.50 | igcewieling | Kobaz: you'll fine he is not very logical |
| 02:51.02 | Penguin | or sensible. |
| 02:51.02 | hjf | jitter 0.000 1.439 2.875 1.125 |
| 02:51.05 | j4jackj | I prefer g722, allow u-/a-law, and disallow most others |
| 02:51.06 | hjf | what the hell is that |
| 02:51.09 | hjf | it doesnt sound good? |
| 02:51.14 | igcewieling | Penguin: /me hugs /ignore |
| 02:51.21 | j4jackj | kills igcewieling |
| 02:51.28 | j4jackj | with a pointy Asterisk |
| 02:51.30 | Kobaz | hey no killing |
| 02:51.32 | igcewieling | hjf: you can look up the meaning of jitter on google |
| 02:51.36 | Penguin | Setting allow=all followed by allow=g722 and allow=ulaw is not disallowing most others. |
| 02:51.43 | Penguin | It's not disallowing anything, as a matter of fact. |
| 02:51.45 | j4jackj | I now disallow=all |
| 02:51.56 | hjf | igcewieling: i know what jitter is... but i mean is that milliseconds? or seconds |
| 02:52.04 | j4jackj | seconds |
| 02:52.10 | j4jackj | it so seems |
| 02:52.12 | Kobaz | okay have fun kids, i need to deploy some stuffers before friday morning |
| 02:52.12 | hjf | i notice that, if i talk continuously, there's no delay |
| 02:52.15 | Penguin | (2147.43) <Penguin> disallow=all \ allow=g722 \ allow=ulaw \ allow=ilbc <-------- this is how a preference list is to be created. |
| 02:52.17 | igcewieling | hjf: I'd have to check the asterisk docs, I assume ms |
| 02:52.36 | hjf | if there's a silence, there's a high delay after i start talking again |
| 02:52.42 | j4jackj | heh |
| 02:52.48 | j4jackj | silence detection? |
| 02:52.50 | igcewieling | hjf: turn off VAD / CND |
| 02:52.58 | hjf | say, 200-300ms if no silence, up to like >1s if silence |
| 02:52.59 | igcewieling | ..er VAD / CNG |
| 02:53.11 | j4jackj | ... |
| 02:53.27 | igcewieling | hjf: you are beginning to understand why softphones on cell phones are not all that popular. |
| 02:53.34 | hjf | igcewieling: :) |
| 02:53.53 | j4jackj | igcewieling: yaou are beginning to understand why I use Asterisk |
| 02:54.02 | hjf | it could be the beta firmware on my access points though |
| 02:54.12 | hjf | i deployed some ubiquiti unifi 3.something |
| 02:54.13 | Penguin | I use my softphone on my mobile phone, but only because I can't be bothered to sit still all the time and still want to receive important calls. |
| 02:54.16 | hjf | with zero-handoff |
| 02:54.26 | igcewieling | I had hoped to stop using cell voice and use all VoIP on my phone. Didn't happen. |
| 02:54.31 | hjf | and the zero-handoff feature is flaky |
| 02:55.13 | igcewieling | 3G back channel seldom has enough bandwidth for ulaw or g722 and g729 just sucks. |
| 02:55.55 | j4jackj | I am led to believe 3/4G have enough for G722 |
| 02:56.08 | j4jackj | They have at least 100k either way |
| 02:56.23 | Penguin | You must have never used 3G. |
| 02:56.49 | j4jackj | heh :D |
| 02:56.53 | j4jackj | i believe i have |
| 02:57.04 | hjf | alaw without VAD is passable |
| 02:57.14 | igcewieling | I used to use it all the time, months at a time actually |
| 02:57.29 | hjf | dect is just crazy though |
| 02:57.29 | igcewieling | When I was at my cabin it was 3G or satellite |
| 02:57.31 | j4jackj | had to use my mum's precious internet minutes, she got a $400 phone bill with my dialing up all the time |
| 02:57.33 | hjf | what's the latency for dect? |
| 02:57.49 | igcewieling | hjf: have you checked http://en.wikipedia.org/wiki/Digital_Enhanced_Cordless_Telecommunications |
| 02:57.51 | j4jackj | DECT is about <150ms I think |
| 02:58.08 | hjf | i have a cordless dect that goes crazy sometimes, and it feeds back |
| 02:58.21 | hjf | when on an internal call |
| 02:58.26 | j4jackj | that would be implosive- |
| 02:59.06 | Penguin | At many times throughout the day, 3G is super slow. It's impossible to put gsm across it clearly at times. |
| 02:59.29 | j4jackj | heh |
| 02:59.50 | hjf | i tried to hook up my credit card terminal to a voip adapter |
| 02:59.52 | hjf | didn't work... |
| 02:59.55 | j4jackj | 3g gsm in the UK is not quite as bad. |
| 03:00.14 | hjf | it connects at 1200bps and my voip provider doesn't allow fast codec |
| 03:00.16 | j4jackj | s/3g gsm/3G/ |
| 03:00.20 | igcewieling | Penguin: I've had good luck with G729 from a bandwidth standpoint. |
| 03:00.28 | j4jackj | And GSM is OK |
| 03:00.29 | igcewieling | ALMOST makes me want to guy a 4G smartphone. |
| 03:00.29 | j4jackj | :D |
| 03:00.31 | Penguin | I haven't been in the UK to check the speeds of 3G. |
| 03:00.46 | j4jackj | Penguin: well I lived there almost all my life |
| 03:00.54 | j4jackj | They're just rolling out 4G |
| 03:00.55 | Penguin | But in several areas of the US, it sucks... sometimes. |
| 03:01.01 | hjf | here in argentina i can get 4mbps over 3G |
| 03:01.08 | hjf | ...at 3AM or so |
| 03:01.09 | j4jackj | hjf: heehee |
| 03:01.14 | igcewieling | hjf: not the upstream you don't |
| 03:01.16 | hjf | during the day i can't even send SMS |
| 03:01.24 | igcewieling | with voip you are limited by the slowest direction |
| 03:01.28 | j4jackj | yup |
| 03:01.33 | j4jackj | tried to explain that to Alex_Bkash |
| 03:01.39 | j4jackj | who is in fact a black hat |
| 03:01.50 | Penguin | I have a black hat. |
| 03:01.51 | hjf | yeah speedtest.net reports i think 500k up |
| 03:01.58 | Penguin | I have several black hats, actually. |
| 03:02.05 | j4jackj | you know what i mean |
| 03:02.45 | Penguin | Some plain, some with agency acronyms, some with company logos. |
| 03:03.12 | j4jackj | I mean someone engaging in illegallities |
| 03:03.39 | Penguin | We allow criminals here. We just don't discuss illegal activities here. |
| 03:04.15 | hjf | guys |
| 03:04.19 | igcewieling | Penguin: Much like the USA government |
| 03:04.26 | hjf | i don't want to appear like a badass but |
| 03:04.52 | hjf | i do loiter in places where "NO LOITERING" is painted on walls |
| 03:05.00 | Penguin | You're going to HELL! |
| 03:05.14 | hjf | danger is my middle name, yes |
| 03:08.03 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 03:09.12 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 03:09.15 | hjf | i was reading that DECT has the ability to send data, but that was never really developed |
| 03:09.32 | hjf | it could have been DECT instead of wi-fi |
| 03:09.44 | hjf | that's interesting |
| 03:10.07 | WIMPy | It has been sold, but at 32 or 64 kbps it's hardly an interesting option. |
| 03:10.15 | hjf | and...sad. i mean it would be awesome to have a single house-wide wireless network for voice and data |
| 03:10.40 | hjf | what's 4G like? |
| 03:10.45 | Penguin | I think we already have that. |
| 03:10.50 | Penguin | It's called wifi. |
| 03:10.59 | hjf | Penguin: except wifi sucks at QoSing |
| 03:11.11 | hjf | i read about the femtocells but i don't udnerstand how that works |
| 03:11.13 | Penguin | That's possible, I guess. |
| 03:11.50 | hjf | is a femtocell something you get like a wi-fi modem, to get your own 4G at home? |
| 03:11.51 | WIMPy | wifi for voice? You haven't tried that, have you? |
| 03:12.22 | WIMPy | cell=base station |
| 03:19.20 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 03:19.27 | igcewieling | Sometimes I wonder how our customers manage to tie their shoelaces. One of them just sent us a list of 15,000 people and their debts |
| 03:19.35 | igcewieling | I presume by mistake |
| 03:19.54 | Penguin | Velco |
| 03:19.57 | Penguin | Velcro |
| 03:20.32 | igcewieling | Penguin: of course! silly me. |
| 03:20.50 | igcewieling | thankfully MY name is not in the list |
| 03:20.55 | Penguin | hahaha |
| 03:21.01 | Penguin | Is mine on there? |
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| 03:21.46 | igcewieling | Yup! $43,000 in debt! |
| 03:21.56 | Penguin | Probably not. My only debt is a monthly credit card bill. |
| 03:22.06 | Penguin | The limit isn't nearly that much. |
| 03:22.14 | igcewieling | You live in Jersey City, right? |
| 03:22.20 | igcewieling | 8-) |
| 03:22.33 | Penguin | Nope, must be the other Robert. |
| 03:22.49 | igcewieling | oh robert, I was looking at Mr Peng |
| 03:22.56 | Penguin | D'oh! |
| 03:23.24 | Penguin | Ah well, it was an honest mistake. |
| 03:24.59 | igcewieling | The last name was not Uin, but I thought it might have been your maiden name. 8-) |
| 03:25.13 | Penguin | :/ |
| 03:25.16 | Penguin | /: |
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| 05:53.10 | ChannelZ | ~book |
| 05:53.10 | 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 |
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| 07:51.09 | j4jackj | I am getting SPITted at by such cryptic people as: 172.245.13.2 |
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| 08:16.46 | *** join/#asterisk Formater (d91169f6@gateway/web/freenode/ip.217.17.105.246) |
| 08:16.57 | Formater | hi all :) |
| 08:19.08 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 08:20.29 | Formater | i started using 11.5.1 after 1.8.x, it's connecting to mysql via odbc, and i have strange sql errors: You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near... with tcpdump i captured the mysql query, and it is this: SELECT * FROM siptrunks WHERE name LIKE '%' ESCAPE '\\' AND callbackextension LIKE ? ESCAPE '\\' ORDER BY name.. and tried, it's not vali |
| 08:22.02 | Formater | i found, that the 'ESCAPE' words are added because ast_odbc_backslash_is_escape() |
| 08:22.57 | BeeBuu | anyone help me please. my context in extensions.conf doesn't work. |
| 08:23.17 | BeeBuu | even no "dialplan reload" command |
| 08:23.28 | BeeBuu | what's wrong? |
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| 08:32.06 | j4jackj | I HATE DIALUP :D |
| 08:33.31 | Formater | BeeBuu: what you get when you execute dialplan reload ? |
| 08:33.56 | BeeBuu | No such command 'dialplan reload' |
| 08:34.29 | j4jackj | eek= |
| 08:34.31 | j4jackj | :D |
| 08:37.43 | BeeBuu | what's wrong? |
| 08:38.03 | Formater | BeeBuu: asterisk version? |
| 08:38.43 | Formater | BeeBuu: try: module load pbx_config.so |
| 08:38.49 | Formater | and then dialplan reload to see if that helped |
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| 08:44.14 | BeeBuu | Formater: Yes! It's work! |
| 08:44.24 | BeeBuu | Tons of thanks. |
| 08:45.17 | Formater | you're welcome :) |
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| 09:18.59 | danfromuk | Hi. Whats the best way to get asterisk to ring forever? Is it just Ringing and then Wait(99999) ? |
| 09:19.01 | danfromuk | Its for dealing with a nuisance caller. |
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| 09:28.17 | tomdm | timeout is an option in the dialcommand: Dial(type/identifier, timeout, options, URL) |
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| 11:13.45 | PLMg | hey, I need the following: Inbound > extension, if not answerd in 5 seconds > queues. Don't know how to set the 5 second part. Does anyone have any idea? |
| 11:14.56 | PLMg | using freebpx :) |
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| 11:27.00 | kaldemar | PLMg: ask in #freepbx |
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| 12:59.44 | igcewieling | tomdm: without the extra spaces, yes |
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| 13:03.01 | igcewieling | Trying to help someone and then discovering they use FreePBX is like taking someone home from the bar and realizing they are not the gender you thought they are. |
| 13:03.15 | Greenlight | +1 for that analogy |
| 13:03.29 | Greenlight | wonders if igewieling is speaking from experience |
| 13:03.36 | WIMPy | LOL |
| 13:03.46 | WIMPy | Double LOL |
| 13:03.52 | igcewieling | Greenlight: LOL! I'd just shrug and keep going. |
| 13:03.58 | Greenlight | hahaha |
| 13:05.00 | bacobart | what's so bad about freepbx? (honest question) |
| 13:05.06 | igcewieling | ~freepbx |
| 13:05.06 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:05.07 | Greenlight | ~freepnx |
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| 13:07.28 | jmetro | So there are tons of ways to calculate bandwidth requirements |
| 13:07.32 | jmetro | what about processing/ram requirements? |
| 13:07.43 | Greenlight | That's a little more subjective |
| 13:07.59 | jmetro | I know it depends a lot on media conversion, features installed, etc |
| 13:08.09 | Greenlight | You can guesstimate depending on what you're going to be doing |
| 13:08.21 | jmetro | but lets just say 1 DID to 150 endpoints |
| 13:08.23 | Greenlight | And what hardware other people are using for similar tasks |
| 13:08.39 | Greenlight | How many simultanous calls? Call recording? IVR's? |
| 13:09.20 | igcewieling | jmetro: core show translation recalc 10 |
| 13:09.55 | WIMPy | That doesn't work any more. |
| 13:09.59 | jmetro | call recording done via port mirroring on the switch, so asterisk not involved. IVR's would be a yes, i'm trying to find an equation that I could use to figure out how much processing/ram would be on 50 calls vs 100 calls |
| 13:10.02 | Greenlight | It's *still* broke. |
| 13:10.32 | jmetro | so i could say, at 50 calls we'd be at 25%, and 75 calls ...etc, scalable for math's sake |
| 13:10.37 | Greenlight | If you're not call recording, and not transcoding, then at 100 calls I'd not be worried. |
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| 13:11.11 | igcewieling | jmetro: if you ever figure it out let us know. People have wanted something like that since the dawn of Asterisk. Some people say it is impossible. |
| 13:11.55 | jmetro | It shouldnt be too hard to come up with numbers, like saying.. Call recording adds 10% per call |
| 13:12.10 | Greenlight | I've a small VM that handles 400 simultenous calls with RTP, no problems. 2gb ram 4 cores. |
| 13:12.23 | jmetro | each call would be Xkb ram and X draw on processing |
| 13:12.31 | Greenlight | jmetro: It doesn't work like that |
| 13:12.31 | igcewieling | jmetro: but call recording would add different amounts depending on your hardware |
| 13:12.35 | Greenlight | It's not linera |
| 13:12.37 | Greenlight | linear |
| 13:13.01 | WIMPy | It also depends on he speed of your HD, if you're using RAID and if you're crypting. |
| 13:13.02 | jmetro | hm, hardware efficiency would play a part, but thats where "YMMV" comes into play. |
| 13:13.40 | Greenlight | It's not like "normal" server specing. You really want to over privision. It's often very time senstive and stuff starts to fall down at high loads |
| 13:13.45 | igcewieling | jmetro: then call recording adds as much overhead as writing to the disk. unless you have to transcode, or cousrse. |
| 13:14.23 | jmetro | igcewieling: The theory being that everything is capable of being measured mathematically, we just have to figure out in how we need to modify our variables |
| 13:14.24 | igcewieling | jmetro: The general method most people use is to throw hardware at the problem until it goes away. |
| 13:14.26 | Greenlight | And I've no idea where that 10% figure is coming from. You'd lost p2p bridging, and take RTP to the asterisk core for a start, and thats WAY more than 10% |
| 13:14.53 | igcewieling | jmetro: The universe could in theory be described mathamatically. that doesn't mean we are able to do so at this time. |
| 13:15.06 | Greenlight | Indeed - there's too many variables and too many unknowns. |
| 13:15.24 | igcewieling | Greenlight: or less then 10% if he was not doing p2p bridging anywa |
| 13:15.38 | jmetro | 10% was a random number |
| 13:15.42 | Greenlight | Oh |
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| 13:15.54 | Greenlight | Well if you're just going to put random numbers in here and there... well... |
| 13:16.27 | Greenlight | I guess you can make a formula to give a "safe" spec, to an extent |
| 13:16.45 | jmetro | it was just an example of trying to find an equation <.< like "if [call recording] proc = proc * 1.10" |
| 13:16.45 | Greenlight | Based on a large enough sameple of peoples usage |
| 13:16.49 | igcewieling | My general rule of thumb is max 200 calls under normal situations, max 400 calls in case of emergency |
| 13:17.12 | Greenlight | I found 11 scales much better than earlier ones |
| 13:17.27 | igcewieling | these days, unless you are doing transcoding or writing to disk, you'll have plenty of CPU |
| 13:17.34 | WIMPy | The type of fs would make a difference and potentially even the amount of sapce used. |
| 13:17.46 | jmetro | im imagining having my typical Asterisk VM be split in half |
| 13:18.29 | WIMPy | I'm not so sure about the CPU thing. My call delivery delay issue does have a lot CPU in it. |
| 13:18.31 | Greenlight | Things really start to get intersting at the 500-1000 simultanous calls levels |
| 13:18.57 | Greenlight | You really don't want network drivers with bad interrupt handling for example |
| 13:19.31 | Greenlight | The newer linux kernels, and proper server hardware that's linux certified seem much better in this respect |
| 13:19.39 | WIMPy | If you're worried about interrupot handling, never install dahdi. |
| 13:19.47 | Greenlight | Yea :) |
| 13:19.58 | Greenlight | DAHDI is only 1000/sec though isn;t it? |
| 13:20.13 | WIMPy | That's not the issue. |
| 13:20.29 | Greenlight | When the CPU is spending more than 50% if it's time inside the interrupt handler it is! |
| 13:20.34 | WIMPy | It does all th DSP and meetme mixing in the interrupt handler. |
| 13:20.39 | Greenlight | ooohhh |
| 13:20.40 | Greenlight | lol |
| 13:20.44 | Greenlight | That's nasty |
| 13:20.53 | WIMPy | Extremely so. |
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| 13:21.11 | Greenlight | That;s like exacty what NOT to do with an interrupt handler |
| 13:21.20 | WIMPy | Actually that is a good argument for "must never be used on a production system" :-( |
| 13:21.26 | Greenlight | Mind, guess it means it gets whatever it wnats from the CPU |
| 13:21.51 | WIMPy | It means there's a point where it simply falls over. |
| 13:21.56 | Greenlight | Yea, I could imagine |
| 13:21.56 | igcewieling | WIMPy: Are you SURE. I thought Digium changed that when Sangoma was kicking their ass with their cards? |
| 13:22.22 | WIMPy | Yes, I ran in to that issue not that long ago. |
| 13:22.22 | igcewieling | around the same time Digium released their commercial fax. coppice would know |
| 13:22.51 | *** part/#asterisk bbourdage (~BBourdage@50-196-242-73-static.hfc.comcastbusiness.net) |
| 13:23.33 | igcewieling | WIMPy: assuming you mean inside the card driver and not the dahdi core, it would explain a LOT of things about why Sangoma worked so much better in the past. |
| 13:24.06 | igcewieling | file |
| 13:24.08 | WIMPy | It was when the old HWEC was no longer supported and it used SWEC and the thing completely exploded. |
| 13:24.11 | igcewieling | Qwell: |
| 13:24.34 | Qwell | O.o |
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| 13:24.52 | igcewieling | Qwell: any comments on WIMPy's claims? |
| 13:25.07 | igcewieling | (9:20:34 AM) WIMPy: It [dahdi] does all th DSP and meetme mixing in the interrupt handler. |
| 13:25.08 | WIMPy | I try to stear clear of hardware that uses dahdi. Less features, but mor reliable. |
| 13:25.34 | igcewieling | WIMPy: your phobia of the PSTN is well known. |
| 13:25.42 | Greenlight | I've a 4 port card in a server that was originally there to give timing. Dahdi's not even installed on that box anymore. |
| 13:26.07 | Greenlight | 4port e1 with hwec no less |
| 13:26.29 | WIMPy | No, I like PSTN. It used to work a lot better than the voip shit. |
| 13:27.01 | igcewieling | I love PRIs. |
| 13:27.25 | igcewieling | So simple, so clean, so easy (usually) to diagnose |
| 13:27.37 | WIMPy | yes |
| 13:27.45 | Greenlight | I got a response on my odd PRI call failure issue eventually. |
| 13:27.50 | jmetro | i <3 voip |
| 13:28.00 | Greenlight | You know the one I pb a trace, and it was cutting off mid call with "Temp failure" from the network side |
| 13:28.21 | igcewieling | jmetro: VoIP is complicated, dirty, and hell to troubleshoot. |
| 13:28.22 | WIMPy | remembers |
| 13:28.48 | Greenlight | This particular client is calling customers of the network "EE" here in the UK. Specifically ones currenly on 3G but in a newly avaialbe 4G area. |
| 13:28.57 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 13:28.59 | jmetro | igcewieling: <.< Not true. |
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| 13:29.21 | igcewieling | jmetro: I guess you've never had one-way audio for no reason whatsoever |
| 13:29.22 | Greenlight | Apparently the 4G upgrades have made some issues, and customers still using 3G are getting a lot of disconnects when using LTE |
| 13:29.43 | jmetro | igcewieling: we VPN all our clients and have our system setup and yes, we never have that issue anymore :3 |
| 13:29.47 | Greenlight | So it's actually the calls dropping at the 4G mast |
| 13:29.58 | igcewieling | every day we get a few reports of dropped calls and one-way audio on our VoIP platform. We almost never get such reports on PRIs |
| 13:30.17 | Penguin | jmetro: What do you use for the client-side VPN gateway? |
| 13:30.34 | WIMPy | Greenlight: I wonder why that's not a fully digital PSTN connection then. |
| 13:30.45 | jmetro | we install a router + static IP's |
| 13:30.50 | igcewieling | When you send in a pcap to the VoIP carrier it never does any good. when you send a Q.931 cap to the telco they usually say "the issue should be resolved now" |
| 13:31.07 | Greenlight | WIMPy: WHat makes you think it's not? |
| 13:31.14 | igcewieling | jmetro: there is no NAT involved in these problem calls |
| 13:31.53 | WIMPy | Greenlight: IIRC you got the location "network beyond interworking point" |
| 13:31.58 | Greenlight | Yea |
| 13:32.08 | Greenlight | That means it's analogue at some point ? |
| 13:32.09 | jmetro | igcewieling: from what i remember, one way audio was a big problem in 1.8 which might be the issue you get |
| 13:32.34 | WIMPy | Greenlight: Or voip. |
| 13:32.35 | igcewieling | jmetro: not the case here. |
| 13:32.42 | WIMPy | Non ISDN |
| 13:32.46 | Greenlight | Gotcha |
| 13:33.01 | igcewieling | Ast 1.8 <-> Ast 11 or Adtran <-> Ast 11, doesn't matter |
| 13:33.17 | Greenlight | It was intersting that once they put a trace on, they could actually trace calls to specific mobile towers |
| 13:34.01 | Greenlight | For instance they traced two calls which took different routes to the same tower in Manchester, and failed at that tower |
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| 13:34.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:35.27 | igcewieling | Greenlight: A perfect example of what can be done when the NSA helps other countries. 8-| |
| 13:35.47 | Greenlight | admires igewieling's tin foil hat |
| 13:36.13 | WIMPy | is oretty sure GCHQ helps NSA more than vice versa. |
| 13:36.20 | mjordan | It's not paranoia if they're really out to get you |
| 13:36.23 | mjordan | <_< |
| 13:36.36 | igcewieling | WIMPy: only for stuff inside the USA |
| 13:36.44 | jmetro | the NSA is doign some scary things, including spying on USA's allies. |
| 13:37.06 | igcewieling | NSA spies on UK people, the GCHQ spies on USA people, everyone wins (except for the people) |
| 13:37.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:37.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:37.26 | Greenlight | tor usage increased by 100% in August ^^ |
| 13:37.37 | igcewieling | Greenlight: I saw that. |
| 13:37.57 | SuperNull | my silk road purchaes will still come in late. .. lol jk |
| 13:38.06 | Greenlight | hahaha |
| 13:39.02 | jmetro | I wish Lavabit was still open, that was a sad closure |
| 13:40.32 | igcewieling | I host most of my stuff in switzerland. won't keep the spooks from getting it if the want, but I hope it will be at least a little bit of a hassle to get my data |
| 13:41.20 | igcewieling | campradio.us don't click on the men pics link, it contains stuff no straight man should ever see. |
| 13:41.20 | WIMPy | No. It makes it easier. |
| 13:41.35 | WIMPy | They always concentrate on international traffic. |
| 13:41.38 | igcewieling | WIMPy: asking the swiss is easier than asking Verizon |
| 13:41.59 | igcewieling | WIMPy: in that case I hope they waste their CPU resources on me |
| 13:42.51 | WIMPy | I think that can be taken for granted. |
| 13:47.43 | *** join/#asterisk serafie (~erin@nat/digium/x-tvabtgubbqnxjbxn) |
| 13:55.29 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 14:00.40 | WIMPy | BTW: I haven't seen anyhing about it for some time. Die dahdi also still have that "glare" feature? |
| 14:00.54 | WIMPy | s/Die/does/ |
| 14:02.26 | *** join/#asterisk saxa (~saxa@201-50-67-77.user.veloxzone.com.br) |
| 14:03.03 | saxa | WARNING[2225]: chan_dahdi.c:7876 handle_alarms: Detected alarm on channel 3: Red Alarm |
| 14:03.10 | saxa | what this red alarm means ? |
| 14:03.19 | saxa | my line with the analog phone works ok |
| 14:03.25 | WIMPy | No connection |
| 14:03.49 | saxa | each entering call rings 2 times an then * stops rininging the sip phones |
| 14:03.52 | saxa | hi WIMPy |
| 14:04.01 | saxa | how you mean no connection ? |
| 14:04.10 | saxa | from line to * ? |
| 14:04.23 | WIMPy | yes |
| 14:04.34 | saxa | ok this is then due to a disconnected line |
| 14:04.40 | WIMPy | For analog that should be battery power. |
| 14:04.45 | WIMPy | Analog is evil. |
| 14:05.37 | saxa | its strange as it was working up to now |
| 14:06.00 | saxa | now everytime i want to place a call from my sip phone to the analog line i get no response error |
| 14:06.29 | saxa | and entering calls , ring twice then stops, and then ring twice again and then stops |
| 14:06.46 | saxa | but on the calling phone its as it is calling all the time |
| 14:06.57 | saxa | any ideas on what this could be ? |
| 14:10.25 | *** join/#asterisk cb122 (~cb122@c-24-98-48-45.hsd1.ga.comcast.net) |
| 14:13.38 | WIMPy | The only way to debug analog stuff that I know is to find an analog phone and try to use that and see what happens. |
| 14:13.41 | saxa | http://pastebin.com/kK29Khbf |
| 14:13.56 | saxa | i think this is all normal as it was always |
| 14:14.00 | saxa | no errors here |
| 14:14.27 | saxa | analog phone works on all 3 lines i have |
| 14:16.46 | [TK]D-Fender | That isn't "what happens" |
| 14:17.10 | hjf | i was reading that TCP SIP could save battery on phones. i tried enabling tcp on sip.conf (set transport=tcp,udp) and in my phone. on UDP i can register OK, but with TCP it gets stuck in "registering" |
| 14:17.55 | WIMPy | hjfL Did you open the tcp port? |
| 14:18.14 | hjf | WIMPy: no firewall in my LAN |
| 14:18.33 | WIMPy | And in sip.conf? |
| 14:18.42 | [TK]D-Fender | hjf: Perhaps you should show us your config. and the call debug.... |
| 14:18.54 | saxa | http://pastebin.com/NJBkJtkJ |
| 14:18.55 | hjf | weird, i switched to UDP then back to TCP and it registered |
| 14:19.06 | hjf | but i try to place a call and i get 408 Timeout |
| 14:19.28 | saxa | I called all my 3 lines, the DAHDI! and DAHDI2 work, DAHDI3 starts rinining it rings 2 times and then it stops |
| 14:20.25 | saxa | and so on goes, but on the calling phone it hears as it was ringing normally |
| 14:20.47 | saxa | could it be the TDM FXO module at fault ? |
| 14:20.59 | saxa | will try to switch the line to dahdi2 |
| 14:21.55 | saxa | this line has a filter for my internet also |
| 14:22.16 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 14:22.21 | saxa | ok it definetly is something wrong on dahdi3 |
| 14:22.21 | WIMPy | Have you tried turning it off and on again? |
| 14:22.35 | saxa | i put that line to dahdi2 and it works ok |
| 14:22.47 | saxa | just switched cables on the back |
| 14:22.59 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 14:23.05 | saxa | yes WIMPy i restarted dahdi many times |
| 14:23.18 | saxa | and asterisk too |
| 14:23.21 | WIMPy | That's not the same. |
| 14:23.32 | saxa | how do i switch it off ? |
| 14:23.36 | saxa | on command line ? |
| 14:23.42 | WIMPy | Switch off the whole box. |
| 14:23.52 | saxa | oh yeah I did that too |
| 14:23.55 | WIMPy | Not just reboot. |
| 14:23.59 | WIMPy | ok |
| 14:24.03 | saxa | halt |
| 14:24.13 | saxa | i halted the system 2 or 3 times |
| 14:24.16 | WIMPy | Well, I guess you have a good chance of a fried module then. |
| 14:24.21 | saxa | as i had a failing HDD |
| 14:24.36 | saxa | ok, have a spare one i think |
| 14:24.46 | saxa | will try to change it |
| 14:24.53 | saxa | i hope i still have it |
| 14:25.00 | saxa | thanks |
| 14:25.10 | *** join/#asterisk bbourdage (~BBourdage@50-196-242-73-static.hfc.comcastbusiness.net) |
| 14:25.14 | WIMPy | Maybe you should also check if the module sits firmly on the board. |
| 14:25.19 | saxa | but as a question, does internet signal can interefere ? |
| 14:25.42 | saxa | i have a filter on that line for adsl |
| 14:25.45 | WIMPy | The splitter is supposed to take care of that. |
| 14:25.49 | saxa | ye |
| 14:26.13 | saxa | so it should not interfere. |
| 14:26.21 | saxa | ok was just a doubt i had |
| 14:26.38 | *** join/#asterisk Defraz (~Defraz@209.141.122.71) |
| 14:26.38 | saxa | as this dahdi3 was creating problems already some time ago |
| 14:26.52 | WIMPy | I really doubt it would cause trouble even without the splitter. The frequencies are far off. |
| 14:27.01 | saxa | ok thx |
| 14:27.12 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-svemdtdxwccbajra) |
| 14:27.13 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:27.23 | saxa | good, will try to unmount the tdm and check the dahdi3 |
| 14:27.30 | saxa | or substitute it |
| 14:27.41 | *** part/#asterisk bbourdage (~BBourdage@50-196-242-73-static.hfc.comcastbusiness.net) |
| 14:27.53 | *** join/#asterisk bbourdage (~BBourdage@50-196-242-73-static.hfc.comcastbusiness.net) |
| 14:27.54 | saxa | goes |
| 14:28.24 | *** join/#asterisk Formater (d91169f6@gateway/web/freenode/ip.217.17.105.246) |
| 14:29.59 | hjf | http://pastebin.com/7pBn8cAa |
| 14:30.02 | hjf | thats my sip.conf |
| 14:32.00 | Formater | hi all :) i'm using asterisk 1.8.11-cert10, and lots of times, calls start to stuck in ringing state... like this: SIP/193.37.150.146-0 33664449517@terminat Ring (full list: http://pastebin.com/FyiHmkGD), and the other stupid thing, is that is says 7-800 activel calls, and there should be normally only 50-80 |
| 14:32.02 | Formater | any idea? |
| 14:33.07 | [TK]D-Fender | I'd probably be looking at your dialplan... |
| 14:33.14 | [TK]D-Fender | And "core show channels concise" |
| 14:35.03 | Greenlight | Formater: Sounds like calls are not getting "closed off" properly. Perhaps a networl or NAT issue. Try enabling session timers. |
| 14:35.16 | [TK]D-Fender | it is |
| 14:35.21 | jmetro | or add same => n, hangups :3 |
| 14:35.23 | [TK]D-Fender | don't touch SIP here |
| 14:35.30 | [TK]D-Fender | LOOK at the dialplan |
| 14:35.37 | [TK]D-Fender | don't go randomly changing things |
| 14:36.43 | Formater | sec, checking :) |
| 14:36.57 | tenspeed705 | jmetro, makes a good point |
| 14:37.33 | jmetro | I used to get a bunch of autodestructs that were the same kind of thing. I terminated every single point of my dialplan with same => n,hangup and fixed it with no inadvertant hangups. |
| 14:37.58 | Formater | so, i have exten => _.,n,Hangup |
| 14:38.13 | Formater | also: |
| 14:38.14 | Formater | exten => _.,n(was_error),Verbose(2,${ERROR}) |
| 14:38.19 | Formater | exten => _.,n,Congestion |
| 14:38.22 | Formater | exten => _.,n,Hangup |
| 14:38.28 | Formater | and i have thsi: |
| 14:38.28 | Formater | exten => h,1,Verbose(2, Inside hangup) |
| 14:38.38 | Formater | should i add Hangup to this "h" as well? |
| 14:38.38 | Penguin | _. is usually not the right pattern to use. |
| 14:38.46 | Formater | i know, but i need it :) |
| 14:38.51 | Penguin | Probably not. |
| 14:39.09 | WIMPy | Is _. your only extension? |
| 14:39.27 | jmetro | Formater: please put your stuff in a pastebin, i cant read it in the chat like that. |
| 14:39.31 | [TK]D-Fender | Formater: NEVER use _. as a pattern |
| 14:39.42 | [TK]D-Fender | Formater: that is your problem, it is MATHCING "h" |
| 14:40.02 | [TK]D-Fender | Formater: And calling an app that is causing it to seize |
| 14:40.07 | Penguin | It matches everything, including things it should not match ever. |
| 14:40.28 | WIMPy | thinks that special extensions shouldn't be extensions. |
| 14:40.43 | Formater | okay.. so you say, if I change it to _X. |
| 14:40.45 | Formater | it will solve my issue? |
| 14:40.47 | Formater | :) |
| 14:40.50 | Penguin | _X. would be better; at least it would require the extension starts with a number and it at least two characters long. |
| 14:41.00 | [TK]D-Fender | yes, but that is still an excessively wide pattern |
| 14:41.06 | jmetro | please put it in a pastebin so i can fix your code for you |
| 14:41.12 | Formater | sec |
| 14:41.17 | [TK]D-Fender | jmetro: He can fix it for himself |
| 14:41.34 | tenspeed705 | won't learn if you do it for him |
| 14:41.54 | Formater | http://pastebin.com/6byx9QvY |
| 14:41.55 | jmetro | I know, but i fix it for myself to make it pretty and feel satisfied about it. |
| 14:42.26 | Formater | so, _X. is not always good for me, because sometimes my clients dial with +131231212, and it not catches it |
| 14:42.33 | tenspeed705 | lol. Formater should read this http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 14:42.45 | Penguin | Create a second extension that is _+X. then. |
| 14:42.50 | jmetro | Formater: a leading + can be stripped by using the variable options |
| 14:42.52 | jmetro | or _X. |
| 14:43.10 | jmetro | erm, _+X |
| 14:43.12 | Formater | i know that... just i'm really curious if this _. created my Ring-state issue |
| 14:43.17 | jmetro | yes |
| 14:43.23 | Formater | yeah, beer for everyone :) |
| 14:43.43 | Formater | and it's normal, that it working fine for 15-30 mins, and only then this issue arises ? |
| 14:43.54 | jmetro | _. means it will match on things like "h" or "s" or "i" or "t" |
| 14:44.01 | jmetro | "h" happens to be the extension asterisk goes to when you hangup |
| 14:44.09 | [TK]D-Fender | Formater: it arises as the calls END |
| 14:44.29 | WIMPy | Hmm. |
| 14:44.34 | [TK]D-Fender | Formater: And loop back through dialplan that isn't valid for a call in that state |
| 14:45.04 | WIMPy | There is an explicit h,1 it would go to. The h extension is just missing a Hangup(). |
| 14:45.19 | jmetro | you might have less issues if it was at the absolute bottom and you had no fallthrough, but its still a HORRIBLE idea to have it |
| 14:45.27 | Formater | ok, i tought, if asterisk sees no new command, it hangs up the line without me telling Hangup :) |
| 14:45.36 | jmetro | incorrect, it times out |
| 14:45.43 | WIMPy | It does. |
| 14:45.45 | Penguin | _. is also h, so whatever _.,2, is will be h's 2 also. |
| 14:46.14 | WIMPy | But if you have another extension with more priorites, it might continue there. That's exactely what happened to you. |
| 14:46.27 | Formater | thank you guys! :) |
| 14:46.27 | jmetro | ^ asterisk doesnt like the same priority defined twice |
| 14:46.30 | Penguin | s/might/will/ |
| 14:47.02 | Formater | i'm moving back and forth between asterisk 1.8, 11.5, kamailio and freeswitch, and sometimes get lost :) |
| 14:47.14 | WIMPy | Depends on the patterns. In this case it will. |
| 14:48.06 | tenspeed705 | having same priority defined twice has also been known to cause complete random outcomes. 3-4 of the same in the mix...ouch |
| 14:48.13 | Formater | how many simultaenous calls you can handle without transcoding ? with 1.8.x, I reach around 200-400... will this get better on 11.5.1? |
| 14:48.27 | *** join/#asterisk Nickinator (~Nickinato@123-243-142-239.static.tpgi.com.au) |
| 14:48.34 | Penguin | What happens when you reach that many calls? |
| 14:48.35 | [TK]D-Fender | Formater: Depends what you're doing with those calls |
| 14:48.36 | Penguin | fire? |
| 14:48.56 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:48.56 | Formater | hm, started to drop calls |
| 14:49.14 | Formater | there will big delays in executing Dial, and reaching termination provider |
| 14:49.34 | Formater | *there will=there was |
| 14:49.55 | WIMPy | has big delays executing Dial() with only a single call. |
| 14:50.26 | Greenlight | Formater: 11.5 seems to handle a lot more calls |
| 14:50.36 | *** join/#asterisk saxa (~saxa@201-50-67-77.user.veloxzone.com.br) |
| 14:50.36 | Greenlight | But again, that depends what you're doing with each call |
| 14:50.46 | saxa | ok, my TDM was fired |
| 14:50.50 | Formater | good :) call comes in, i check if the client has enoguh money, get routing, and dial |
| 14:51.03 | Formater | it comes in in SIP or IAX and goes out on SIP |
| 14:51.08 | Formater | without transcoding |
| 14:51.10 | Greenlight | So, like a billing gateway? |
| 14:51.16 | Formater | 4 cores, 4 gigs of ram |
| 14:51.28 | Formater | yep, something like that, termination for call centers |
| 14:51.33 | Formater | they use us as 'termiantion providers' |
| 14:51.35 | Greenlight | I've a similar box, similar specs, quite happily sitting on 400 sim calls without breaking a sweat |
| 14:52.30 | Formater | Greenlight: on 11.5 ? |
| 14:52.48 | Greenlight | Yes |
| 14:53.04 | Formater | super :) |
| 14:53.06 | Greenlight | In fact i think it has less ram than yours |
| 14:53.41 | Formater | currently having 5 asterisk 1.8.x on virtual servers, 1 astersik 11.5 on psyical server and one 11.5.1 on virtual server |
| 14:53.54 | Formater | just had no time to test the performance of 11.5 |
| 14:54.22 | *** join/#asterisk wltjr (~wltjr@unaffiliated/wltjr) |
| 14:54.31 | Formater | what do you think, is there performance differnce running asterisk in virtualized vmware environment versus on real machines? |
| 14:55.15 | wltjr | anyone know why ReceiveFAX fails to write/create the tiff file, and/or is app_fax needed? I only have res_fax and seems you can't build app_fax if you have res_fax |
| 14:55.50 | Greenlight | We use ESXi so it's a close to "real" hardware as we can get, and really haven't had any issues with performance. |
| 14:56.02 | Formater | Greenlight: tnx :) |
| 14:56.04 | Greenlight | I'd be a little nervious with too much timing senstive stuff though |
| 14:56.21 | Penguin | wltjr: Did you install res_fax_digium too? |
| 14:56.21 | Greenlight | We still use "real" hardware for anything complex |
| 14:57.02 | Penguin | You'll need both for it to work. |
| 14:57.11 | wltjr | Penguin: doesn't look like it, its the * package on gentoo, not sure if its building what I need, seems pieces are missing |
| 14:57.23 | Qwell | ~faxforasterisk |
| 14:57.23 | infobot | it has been said that faxforasterisk is Digium's commercial Fax For Asterisk module is available at http://www.digium.com/en/products/software/faxforasterisk.php |
| 14:57.37 | wltjr | Penguin: thanks for the pointer, will look to see why I do not have that res and build it |
| 14:57.45 | Penguin | wltjr: res_fax_digium is the licensed part of ffa, so I can understand why it is missing. |
| 14:57.56 | Penguin | You can get one free license from digium.com. |
| 14:58.24 | wltjr | Penguin: ah ok ty some change I didn't stay current on, still miss old rx/tx |
| 14:58.26 | coppice | If you are using the gentoo packages spandsp should be in there providing the FAX engine |
| 14:58.51 | wltjr | coppice: its providing some but not all, I have ReceiveFax command but its not working, nothing written to file |
| 14:58.52 | Penguin | wltjr: Remove ffa (res_fax and res_fax_digium) and use app_fax if you prefer that. |
| 14:59.34 | wltjr | Penguin: I was looking into that, but I could not even select it with menuselect, even when I unselected res_fax, maybe needed to unselect res_fax_spandsp, I don't care just want to get faxes receiving or being written to a file |
| 14:59.57 | Penguin | I personally use ffa and like it just fine. |
| 15:00.20 | wltjr | stupid Cisco ata failed miserably had to bring back an analog line for outbound faxes, till we get fiber internet with sla |
| 15:00.58 | [TK]D-Fender | wltjr: Which ATA? |
| 15:02.05 | hjf | hm |
| 15:02.07 | wltjr | [TK]D-Fender: spa112, but not sure I blame that, there has been lots of problems with latency and jitter with the Comcast SMB connection, 100/10 via cable modem, sucks arse |
| 15:02.20 | hjf | i enabled sip debug to the console |
| 15:02.32 | hjf | if i register with tcp i see the sip messages just fine |
| 15:02.43 | [TK]D-Fender | wltjr: And you are using T.38 clearly.. right? |
| 15:02.49 | hjf | but then i try to unreg and i don't see anything else |
| 15:02.56 | cb122 | anyone using monitoring to push alerts to growl? |
| 15:03.06 | wltjr | [TK]D-Fender: yes, it seems to work just not reliably, faxes fail and have to be resent repeatedly, staff hates me |
| 15:03.21 | hjf | even when placing a call, it times out (servr sees no data). tcpdump shows m phone sends data, and netstat shows the TCP connection is open |
| 15:03.39 | coppice | wltjr: is the spa112 firmware up to date? the early versions were terrible |
| 15:03.43 | wltjr | [TK]D-Fender: pretty sure line issues, we had problems with credit cards as well, till I replaced the terminals with ip cc terminals |
| 15:03.56 | wltjr | coppice: its up to date except for latest which removed some feature I am not sure I am even using |
| 15:04.31 | wltjr | I was going to open a ticket with Cisco, but knowing we have line quality issues, till thats resolved really can't blame Cisco or the ata |
| 15:04.46 | wltjr | I even tried diff fax modes, 9600 baud etc, no diff |
| 15:04.58 | [TK]D-Fender | wltjr: Doesn't sound like a functional T.38... |
| 15:05.10 | [TK]D-Fender | wltjr: And how does that relate to your use of those dialplan apps? |
| 15:05.16 | [TK]D-Fender | wltjr: Also, what version are you running? |
| 15:05.32 | wltjr | [TK]D-Fender: well again with latency and jitter on the line, voice calls were effected, it must wreak havoc on fax |
| 15:05.58 | wltjr | [TK]D-Fender: asterisk 11.5 |
| 15:06.30 | coppice | if you are using T.38 you should have a fair amount of tolerance of latency and jitter |
| 15:07.30 | wltjr | I am pretty sure I am using t38, t38pt_udptl=yes is set in sip.conf, not sure I have any logs atm to reference |
| 15:09.10 | wltjr | it could be me not having the spa 112 configured properly and/or diaplan issues there was some oddness with reinvites etc at first, but I am not going to mess with that till we get the internet connection replaced, and have quality there, for now just need to receive fax so I can email it via bash script |
| 15:09.51 | wltjr | modified gentoo ebuild and have app_fax now instead of the res_fax, will try that, but probably still go the ffa route, * fax is still confusing mess... |
| 15:11.43 | wltjr | coppice: spa112 is running 1.3.2, not the 1.3.2(XU) that had srtp removed, probably no diff to 1.3.2 |
| 15:12.14 | Qwell | They removed srtp? |
| 15:12.32 | wltjr | Qwell: yes Firmware for SPA112 and SPA122, Version 1.3.2(XU) [SRTP Removed] |
| 15:12.47 | wltjr | Qwell: but the other 1.3.2 still has it, no clue as Cisco |
| 15:12.51 | Qwell | maybe export stuff |
| 15:12.52 | Qwell | shrugs |
| 15:12.53 | Penguin | They probably tried to make the image smaller or something silly. |
| 15:14.29 | wltjr | they do dumb stuff like the spa501g without screen, do not buy I did and regret it, trying to get client to replace, its basically a 16 line phone not 8, each line can get 2 calls, and with no display no way to control what call is placed on hold when you have 2 on one line really stupid stuff |
| 15:14.49 | wltjr | if you disable call waiting, the phone goes from 16 lines to 1 line, and they consider that a feature... |
| 15:15.08 | wltjr | problem with all SPA phones, inherited problem from Sipura days |
| 15:15.38 | WIMPy | Yes. Extremely annoying. |
| 15:15.47 | wltjr | bottom line don't buy! |
| 15:16.17 | wltjr | I went through all kind of stuff and got it reported to them, ticket filed, but it ended up as a feature request not bug report, fing Cisco... |
| 15:17.00 | wltjr | app_fax doesn't seem to work any better or diff than res_fax, not on to the digigum one... |
| 15:18.25 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 15:19.19 | *** join/#asterisk Rumbles (~Rumbles@mail.solutiontelecom.co.uk) |
| 15:22.03 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:28.21 | wltjr | app_fax and res_fax are really confusing they seem to be the same thing, and niether works for me, creating accounts for 1 free ffa license for each client, pita... |
| 15:28.49 | *** join/#asterisk ankurgupta0001 (~ankurgupt@216-75-234-234.static.wiline.com) |
| 15:29.30 | *** join/#asterisk StaRetji (~LittleAll@178-221-235-118.dynamic.isp.telekom.rs) |
| 15:29.41 | ankurgupta0001 | I am having issues on TLS with some carriers on Asterisk 11.5 |
| 15:29.52 | coppice | I think you need three components - app_fax, res_fax and res_spandsp_fax/res_digium_fax |
| 15:29.54 | StaRetji | hello all. Is there a know issue if using asterisk realtime and trunk peer? |
| 15:30.01 | Qwell | coppice: no app_fax |
| 15:30.38 | coppice | QWell: then what is app_fax? |
| 15:30.39 | wltjr | app_fax and res_fax conflict, its one or the other, res_fax seems to have options like t38, while app_fax has more channel variables, odd stuff |
| 15:30.44 | Qwell | coppice: old and busted |
| 15:30.46 | StaRetji | I am having problem that I have client with his own asterisk connected to my server, but he can't make call no matter. I get Playing 'prepaid-enter-pin-number.gsm' |
| 15:31.05 | wltjr | Qwell: ah, good to know :) res_fax doesn't seem to much better, maybe I need to try the f option for t38 |
| 15:31.10 | StaRetji | any help is highlu appreciated |
| 15:31.12 | coppice | QWell: so the apps are actually in res_fax? |
| 15:31.16 | Qwell | coppice: correct |
| 15:31.20 | jmetro | StaRetji: are you on freepbx |
| 15:31.30 | StaRetji | asterisk/a2billing |
| 15:31.37 | StaRetji | thx jmetro, btw |
| 15:31.43 | [TK]D-Fender | StaRetji: Call is getting processed and you've configured things wrong |
| 15:32.02 | ankurgupta0001 | The server is sending a RST packet when trying to dial a number |
| 15:32.02 | [TK]D-Fender | StaRetji: This isn't #a2billing ... |
| 15:32.06 | ankurgupta0001 | any help will be highly appreciated |
| 15:32.10 | coppice | QWell: weird. does digium intend to release a complete fax module? |
| 15:32.11 | StaRetji | [TK]D-Fender: thx , I am sure I did something wrong |
| 15:32.19 | Qwell | coppice: fucked if I know. :) |
| 15:32.34 | jmetro | StaRetji: have him call you, record your console output with verbose at 10 and debug at 10. Start from the beginning of the call and go to the end, then put the entire thing into a pastebin and share it with us |
| 15:32.41 | wltjr | coppice: its a mess it seems, even the doc do not make each clear |
| 15:33.00 | wltjr | https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX_app_fax |
| 15:33.00 | drmessano | "Complete Fax Module"? |
| 15:33.05 | wltjr | https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX_res_fax |
| 15:33.11 | coppice | QWell: it seems unlikely after all this time. I have colour FAXing now :-) |
| 15:33.19 | StaRetji | jmetro: I suspect somethning with caller id, as normal sip client can make call |
| 15:33.24 | Qwell | coppice: neat. I have color email. :) |
| 15:33.29 | drmessano | lol |
| 15:33.32 | StaRetji | trunk is set properly, with ip only authentication |
| 15:33.45 | jmetro | StaRetji: Let us see the call fail from the beginning to the end. |
| 15:33.49 | coppice | drmessano: the spandsp module is the only one which supports gateway operation |
| 15:33.56 | drmessano | Oh I see |
| 15:34.07 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 15:34.33 | drmessano | Wait, hang on |
| 15:35.08 | coppice | some source code changes would be needed in res_spandsp_fax to make use of the latest spandsp with colour, T.85 compression and other stuff |
| 15:35.28 | drmessano | So gateway is a function of the fax module, and the Digium module doesn't support it? |
| 15:35.40 | coppice | right |
| 15:35.47 | drmessano | Interesting. |
| 15:35.49 | Qwell | drmessano: it requires stuff in the FAX stack that's used |
| 15:36.04 | [TK]D-Fender | Funny that * 10+supports T.38 termination & gateway..... |
| 15:36.31 | coppice | for digium, who use the commetrex stack, I think it means adding a whole new extra gateway stack |
| 15:36.33 | drmessano | So the "Gateway support" of 10+ is dependant on the Fax module itself, correct? |
| 15:36.39 | Qwell | drmessano: yes |
| 15:36.42 | drmessano | Ohhhhhh |
| 15:36.44 | wltjr | [TK]D-Fender: it doesnt work, can't get 11.5 to receive fax adn write to file, moving to ffa |
| 15:37.10 | [TK]D-Fender | wltjr: If you say so. We've seen nothing. |
| 15:37.15 | wltjr | idk why this stuff had to change rx/tx fax used to work for years for me |
| 15:37.39 | wltjr | [TK]D-Fender: I can't get it to work and not doing anything elaborate, checking config files in case something is off there, not sure if ecm is causing problems |
| 15:38.17 | wltjr | http://pastebin.com/XP3yUUT8 |
| 15:38.32 | coppice | wltjr: do you use any of the features in the old fax code which aren't there any more, like page headers? |
| 15:38.33 | wltjr | its basically just like I did for many years with rxfax, just changed commands |
| 15:38.37 | *** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net) |
| 15:38.51 | wltjr | coppice: not really, just straight pass through, write to file, convert to pdf email done |
| 15:39.22 | wltjr | the only thing of * I have had break and had to re-address is faxing, I cared less for years, but have a client who faxes allot in and out sucks |
| 15:39.47 | coppice | I'm puzzled that lots of people didn't bitch about loosing that, as numerous people used to ask for extra tricks in that code |
| 15:39.55 | wltjr | I remained using rx/tx fax for long time after it was removed, till I could not use due to symbol chagnes, etc |
| 15:40.38 | wltjr | coppice: I think many use hylfax and others, most documentation even back then bascically said rx/tx was crude, etc but worked for me, better than anything else since |
| 15:42.16 | coppice | the big thing hylafax gives you is job submission from well developed clients |
| 15:43.10 | wltjr | coppice: it looked to overkill anytime I looked into it, and to complex to setup thus avoided it, just wanted to keep things simple, which they are, but makes no sense why receivefax via res_fax or app_fax won't write to file, app_fax is failing with error app_fax.c:224 phase_e_handler: Error transmitting fax. result=48: Disconnected after permitted retries |
| 15:43.41 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:43.56 | wltjr | but Qwell said app_fax is old crap, which seems to be the case in sources, not easily built, I got it built via modifying gentoo ebuild |
| 15:44.38 | Qwell | I never called it crap. |
| 15:45.11 | wltjr | Qwell: I embellished :) |
| 15:45.33 | wltjr | ok old and busted, to me that means crap :) |
| 15:45.35 | StaRetji | jmetro: here it is mate http://pastebin.com/1mYhhvj2 |
| 15:45.47 | Qwell | wltjr: not maintained, and replaced |
| 15:46.05 | wltjr | Qwell: would be nice if docs said that, or even sources when selecting |
| 15:46.10 | Qwell | It only remains, for backwards compatibility. It doesn't need to exist anymore. |
| 15:46.10 | StaRetji | sip client made a call over their asteriks (in trunk with mine) |
| 15:46.27 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:46.28 | wltjr | Qwell: I want rx/tx fax to remain for backwards capability :) |
| 15:46.37 | jmetro | StaRetji: It looks like its hitting your A2billing AGI and the AGI is deciding they need to enter their pin number, you might want to try the A2 Billing room. |
| 15:46.46 | Qwell | wltjr: Then ask the maintainer of that module to make a new version. |
| 15:47.00 | StaRetji | is there 1? |
| 15:47.23 | jmetro | There is i believe, i think its #a2billing or something |
| 15:47.33 | StaRetji | ok, thx jmetro |
| 15:47.34 | wltjr | Qwell: or more have it removed from say voip-info.org, as most fax stuff still mentions that which has been MIA for many years now |
| 15:47.39 | jmetro | No-one here would know how to troubleshoot the A2billing AGI though. |
| 15:47.47 | Qwell | wltjr: voip-info.org is crap. |
| 15:47.50 | wltjr | Qwell: little stuff mentions ReceiveFax command |
| 15:47.50 | Qwell | and that, you can quote me on |
| 15:48.23 | wltjr | Qwell: oh when did that become the case? what is the new reference? long ago if you asked newb stuff in here, voip-info.org was the answer, shows how long I have been outside the * world |
| 15:48.34 | Qwell | Last modification: Tue 16 of Feb, 2010 |
| 15:48.39 | wltjr | spent to much time laying on the beach, heading there soon :) |
| 15:49.16 | Qwell | ~wikis |
| 15:49.16 | infobot | i guess wikis is VoIP Wiki covering FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners. For Asterisk, see ~asteriskwiki |
| 15:49.21 | Qwell | infobot: I hate you. |
| 15:49.21 | infobot | You hate you.? |
| 15:49.23 | Qwell | ~asterisk wiki |
| 15:49.23 | infobot | it has been said that asterisk wiki is http://wiki.asterisk.org/ |
| 15:49.28 | wltjr | Qwell: I wish there was some current fax reference, looking at the ffa stuff now, waiting for stupid confirm eamil from digium so I can get my license |
| 15:49.49 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:50.33 | *** part/#asterisk StaRetji (~LittleAll@178-221-235-118.dynamic.isp.telekom.rs) |
| 15:53.11 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 15:53.22 | ankurgupta0001 | I am having issues on TLS with some carriers on Asterisk 11.5,any help will be appreciated |
| 15:53.38 | Qwell | ankurgupta0001: You have said nothing at all about your problem. |
| 15:54.18 | ankurgupta0001 | Using pjsip 2.1 as the client for connecting to asterisk on iPhone. |
| 15:54.18 | ankurgupta0001 | The registration goes through fine but when trying to call it send a RST packet to the client on phone causing the connection to reset. |
| 15:54.40 | wltjr | see this looks like a fax arrived without issues, but it failed, not tiff file written or created :( http://pastebin.com/qQSb8w68 thats with res_fax |
| 15:55.50 | ankurgupta0001 | this is only happening on some carriers, everything works perfectly on Wifi |
| 15:56.33 | [TK]D-Fender | ankurgupta0001: then your carrier is the issue |
| 15:57.01 | [TK]D-Fender | ankurgupta0001: And maybe they just don't want you running SIP and getting around their usual fees for calls |
| 15:57.02 | ankurgupta0001 | the call connects once if I try 5-6 times |
| 15:57.07 | coppice | wltjr: "Fax receipt completed with status: FAILED" does sound like its without issues |
| 15:58.49 | wltjr | coppice: sure, but what no idea :) there is not much to the configuration, res_fax.conf is all defaults, and my dialplan is very basic, doesn't make sense for it to fail |
| 15:58.50 | *** join/#asterisk saliak (~saliak@ip72-195-155-142.ri.ri.cox.net) |
| 15:59.54 | wltjr | coppice: I had to add that line just to see if it failed, but I already knew it was, so its not really helping to debug |
| 16:00.16 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:00.47 | ankurgupta0001 | If the carrier would have blocked the sip calls then it should not have went through even once |
| 16:03.33 | [TK]D-Fender | ankurgupta0001: Not if they are screwing with you at random |
| 16:03.55 | [TK]D-Fender | ankurgupta0001: Either way, it if works on local Wi-Fi, then * itself isn't the problem |
| 16:03.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:05.13 | WIMPy | There could be an issue with fragmentation. |
| 16:05.23 | wltjr | tried to turn on debugging and use t38 via res_fax options d and f, but no change ReceiveFAX(${FAXFILE},df) |
| 16:05.53 | navaismo | Penguin, just FYI my rpm issue was solved running the command: depmod -ae `uname -r` |
| 16:06.30 | Penguin | I thought it might have been kernel version mismatch, but I didn't think to depmod it. |
| 16:07.05 | mjordan | wltjr: you have strictrtp enabled. |
| 16:07.10 | mjordan | wltjr: > 0x7fb594020190 -- Probation passed - setting RTP source address to 4.55.5.130:19564 |
| 16:07.19 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:07.31 | mjordan | wltjr: the RTP stack is probably blocking some of the packets. Try disabling strictrtp in rtp.conf |
| 16:07.53 | wltjr | mjordan: ah ty, did not see that default |
| 16:08.48 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 16:08.48 | mjordan | it is on by default, as - technically - someone can do something bad to you if you don't have strictrtp enabled. You can control how many RTP packets it takes to lock onto a source address using the probation parameter |
| 16:09.27 | mjordan | if you're dropping faxes or dropping too much audio, it's worth setting the probation to something a bit lower. For testing purposes, you could disable strictrtp to see if that's causing the issue. |
| 16:10.02 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:10.23 | wltjr | mjordan: turned it off, doesn't seem to be any diff |
| 16:11.28 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 16:13.10 | wltjr | I assume I put the key for ffa in the res_fax_digium.conf file, but not sure on syntax, can they make this any more of a pita... |
| 16:13.11 | mjordan | hm. Nuts. |
| 16:13.22 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:13.34 | Penguin | wltjr: You probably should have read the instructions. |
| 16:13.50 | Penguin | It's very simple if you follow the guide for it. |
| 16:13.55 | wltjr | Penguin: I am, its telling me stuff I already did just need to put the key somewhere |
| 16:14.17 | wltjr | pita registration utility wtf |
| 16:14.23 | *** join/#asterisk ipiera (~Paul@ipiera.plus.com) |
| 16:14.26 | Penguin | The instructions don't say, "put the key somewhere." The instructions do tell you exactly what to do with the key. |
| 16:16.45 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:16.48 | [TK]D-Fender | Instructions aren't overrated ... because people to lazy to read them wouldn't make the extra few clicks to enter a rating on the system either :) |
| 16:17.02 | wltjr | this is dumb you create an account on digium for the license, which you turn aroud and register with the same information, really stupid |
| 16:17.26 | wltjr | [TK]D-Fender: no its registering a license after going through a registartion process, they really make you earn the free license |
| 16:17.46 | Penguin | It's the same procedure for a paid license. It's really quite easy. |
| 16:18.06 | Penguin | You register for an account with digium. That part is not related to anything else. |
| 16:18.40 | Penguin | You "buy" a free license. They send you some information. You in turn have to register your system to license it. |
| 16:18.49 | Penguin | It. Is. Easy. |
| 16:18.58 | wltjr | not easy if you make a typo, you have to start over |
| 16:19.23 | Penguin | It's not like you have to type in the license key by hand. |
| 16:19.27 | wltjr | why I am entering my name, company info, address, etc again 2 times for 1 license makes no sense, never seen this sort of craziness before |
| 16:19.45 | wltjr | Penguin: you have to type all the other information, i am filling out a form, name, address, etc |
| 16:19.57 | Penguin | I don't remember having to do that. |
| 16:19.59 | wltjr | screwed up first time and put more than my first name in so had to start over... |
| 16:20.12 | Penguin | Or maybe I did it and it was just so trivial I didn't give it a second thought. |
| 16:20.23 | wltjr | Penguin: clearly they change the tool thus having to download vs it being in the tarball with the res_fax_digium.so |
| 16:20.29 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:20.54 | wltjr | makes no sense why they would not just email the .lic file, I get its likely machine specific so you can't move to antoher, but what a pita |
| 16:21.15 | jmetro | if its free, it kinda sounds like the license files should just be on a download page. |
| 16:21.40 | Penguin | It's a license. |
| 16:21.43 | ChannelZ | it's not a different license than a paid one.. except that you paid nothing for it. |
| 16:21.51 | ChannelZ | It's not an open license |
| 16:21.52 | Penguin | The fact that they give you one for free is irrelevant. |
| 16:23.00 | *** join/#asterisk v0lZy (~Thunderbi@84-255-194-41.static.t-2.net) |
| 16:23.07 | jmetro | unless its "free for one server paid for more" |
| 16:23.20 | ChannelZ | which it is |
| 16:23.28 | Penguin | That's how it is. It is a license, but they allow you ONE FOR FREE. |
| 16:23.33 | ChannelZ | although s/one server/one channel/ |
| 16:24.33 | Penguin | You can buy 50 of them first, then say, "Hey, I want my free one now." Then you'll have 51 licenses. |
| 16:25.30 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:25.53 | wltjr | even that is not working, f * fax damn, can't even get simple fax working with ffa |
| 16:26.22 | Penguin | FFA works quite well for me. I'd recommend anyone give it a try. |
| 16:27.01 | wltjr | Penguin: I am its failing.... |
| 16:27.05 | navaismo | recommend that too, and the digium support was awesome |
| 16:27.17 | wltjr | here is ffa failing just like other stuff was http://pastebin.com/MT4HGEke |
| 16:27.42 | wltjr | this is at my location, my * box, and no calls nothing just me trying to send a fax from faxzero.com which I used before with rx/tx fax years ago, over 5 |
| 16:29.23 | wltjr | revised and you can see the res_fax_digium.c log output, and I did noload => res_fax_spandsp.so so its definitely using ffa and failing |
| 16:29.39 | wltjr | http://pastebin.com/2fL5akGi |
| 16:30.01 | wltjr | so app_fax, res_fax, and ffa all fail... wonderful |
| 16:30.33 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:31.10 | wltjr | wonder if I get support from digium with this free license... :) |
| 16:31.41 | Penguin | Probably a little. |
| 16:32.05 | Penguin | I'm sure there's a statement addressing that on the ffa page. |
| 16:32.09 | wltjr | 24 fax attempts later, still no progress or any idea what is causing it to fail across 3 different means of receiving |
| 16:32.43 | wltjr | I am not even sure if my problems with the SPA112 are not 100% * related, clearly * has its own problems receiving faxes |
| 16:33.12 | wltjr | so much that digium had to make a commercial offering which still seems to have issues |
| 16:33.39 | Penguin | The other methods work for thousands of people, too. I just prefer to use ffa because it is so simple. |
| 16:34.12 | wltjr | sure I get it works for others but not for me, but my dialplan and stuff is not much different than others, my bash script sure, but that is not the issue, I dont even get a tiff file, or an empty file |
| 16:34.49 | wltjr | seems it should be an empty file or something, I would think permissions, but /tmp is 777, so its definitely not that, and i have * writing other stuff to the /tmp directory, so I know there are no problems there |
| 16:34.52 | ChannelZ | I just sent myself one with FFA. Albeit not over an SPA112 |
| 16:34.54 | wltjr | maybe I need to try a older version of * |
| 16:35.07 | Penguin | Which version are you using now? |
| 16:35.09 | wltjr | no spa112 here that is at clients, but no longer in use |
| 16:35.12 | wltjr | 11.5 |
| 16:35.36 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:35.43 | ChannelZ | Real fax -> PSTN -> SIP -> Asterisk FFA |
| 16:36.04 | wltjr | my setup here is Comcast cable modem, sip trunk provider is nexVortex, and terminates in *, pretty straight forward, all IP, no ata, etc |
| 16:36.17 | ChannelZ | codec? |
| 16:36.31 | wltjr | at clients I had real fax -> spa112 -> sip -> * |
| 16:37.00 | Penguin | You have to use ulaw (or alaw) or t.38 for fax to work over VoIP. |
| 16:37.06 | wltjr | ChannelZ: standard 64k one, can't recall off hand ulaw I think |
| 16:37.26 | wltjr | I always use ulaw, never bandwidth issues so rather go with the lesser issue ones |
| 16:38.00 | wltjr | Penguin: most of my stuff has been configured for years, I just switched from rx/tx fax to newer stuff, at one point it did work, now its not |
| 16:38.15 | ChannelZ | and you unloaded all that other fax junk before trying FFA? (they clash last I knew) |
| 16:38.18 | *** join/#asterisk roentgen (~irc@openvpn/community/support/roentgen) |
| 16:38.31 | wltjr | Penguin: I had faxing working long ago, it broke with stupid * changes, and its a complete mess now, at least then there was only rx/tx fax, now there is app_fax, res_fax, and ffa |
| 16:38.48 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:38.58 | ChannelZ | that was interesting.. |
| 16:39.19 | wltjr | res_fax is loaded, but I am explicitly not loading the other noload => res_fax_spandsp.so |
| 16:39.38 | wltjr | I am a gentoo guy, I trim down my *, including only having modules I am using loaded, so dial in my modules.conf per system |
| 16:39.58 | Penguin | That could be part of the problem. |
| 16:40.00 | wltjr | module show only shows the res_fax and ffa |
| 16:40.14 | wltjr | Penguin: maybe, but it usually bitches about missing stuff wanna see? |
| 16:40.40 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:40.50 | wltjr | http://pastebin.com/BxCmij26 |
| 16:41.02 | wltjr | you can see when I load stuff that needs other stuff, it fails to load those modules |
| 16:41.24 | wltjr | Penguin: if ffa, res_fax, app_fax needed a module or something I did not have loaded, there should be some error or something in logs some where |
| 16:41.45 | wltjr | everything looks like its working, but faxes just fail, no tiff file is ever created, i would think the file shoudl at least be created even if empty |
| 16:41.46 | [TK]D-Fender | Portage .... for when youère upa creek without a paddle... |
| 16:43.56 | wltjr | more like * fax is s creek with or without a paddle |
| 16:44.13 | Penguin | Up a paddle without a creek. |
| 16:44.18 | ChannelZ | Your paste earlier just shows the channel terminating for no reason. Turn on fax debug, see if it's saying something. Turn on SIP debug see if it's saying anything. |
| 16:44.41 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:44.43 | wltjr | feel like a paddle is up my butt now, hate faxing, I think it knows that so it has to give me problems all the time, why can't it just ago away... |
| 16:45.13 | wltjr | ChannelZ: I did turn on fax debugging and nothing, I am calling ReceiveFax with df optoins, not sure if that applies to ffa but did to res_fax |
| 16:45.20 | navaismo | fax must die, |
| 16:45.45 | *** join/#asterisk micols (~t@81.161.188.225) |
| 16:45.49 | wltjr | also app_fax and res_fax have different parameters/arguments so they are not really direct replacements for each other, no clue what other stuff ffa has |
| 16:46.01 | mjordan | FFA uses res_fax. |
| 16:46.07 | mjordan | app_fax should not be loaded. |
| 16:46.30 | wltjr | yes I have res_fax loaded, you can't have both app_fax and res_fax you can only build one at a time, I tried app_fax no different so went back |
| 16:46.50 | wltjr | I do not have res_fax_spandsp loaded since I have ffa loaded, but neither works, both fail for different reasons |
| 16:47.31 | wltjr | brb meeting with the browns, the paddle and creek are coming out of me... damn faxing... |
| 16:47.43 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:47.48 | ChannelZ | if fax debug said *nothing* then you have some other problem prior to FFA |
| 16:50.26 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:52.05 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:53.22 | wltjr | ChannelZ: debug made no diff, and I agree there seems to be some other problem, but its either with the sip trunk provider, possible but not to likely, or something with *, which is more what I am thinkking |
| 16:53.34 | wltjr | maybe firewall, but that should effect voice I would assume |
| 16:54.26 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:54.41 | wltjr | not seeing anything on the firewall, dropped/rejected packets are to other ports and different source ips |
| 16:54.50 | drmessano | Not if you're using udptl and don't have those ports open |
| 16:55.00 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 16:55.01 | navaismo | last time I used FFA, I changed providers need one with t.38 support using sip |
| 16:55.05 | wltjr | really makes no sense, and its like this on different machines at differnet locations, but basically all the same stuff, provider, ISP, connection, * version, config, etc |
| 16:55.46 | wltjr | nexvortex does have t38 support, I have used them for years, again fax used to work years ago, I stopped caring when it broke, so did some other clients, just this one has to have it |
| 16:56.04 | ChannelZ | I just did SIP to SIP and it worked fine even. |
| 16:56.28 | wltjr | it broke when rx/tx stopped working, and I used them long after supported, till symbols broke, I guess I could revert to some way earlier version of * and have it all working, but then likely get people connecting I do not want |
| 16:57.17 | wltjr | I am doing just what I did years ago, same dial plan, same scripts, sip trunk provider, same site sending the fax, only difference is the stuff in *, receivefax vs rxfax, the only difference |
| 16:58.53 | wltjr | worked then, doesn't now, and at clients when using the spa112, it was using t.38 udptl ports, I have udptlstart=4000 udptlend=4999 and iptables -A INPUT -p UDP -i eth1 -m multiport --dport 4000:5002 |
| 16:59.26 | wltjr | these are 64bit systems if that matters, stuff was 32bit long ago, but these are fully recompiled, not same code migrated |
| 16:59.31 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:59.47 | wltjr | I do not believe I ever had receivefax working, only rxfax long ago |
| 17:03.13 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:07.36 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:10.44 | drmessano | wltjr, do you have a udptl.conf ? |
| 17:11.04 | wltjr | drmessano: yes, I have those ports configured in that |
| 17:11.57 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:13.58 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
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| 17:23.32 | *** join/#asterisk ticklemeozmo (ticklemeoz@2600:3c03::f03c:91ff:feae:f7b1) |
| 17:24.52 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:25.48 | Katty | hi kids. |
| 17:26.01 | Penguin | It's "the moms" again. |
| 17:26.08 | Katty | hi dear. |
| 17:26.10 | Katty | hugs Penguin |
| 17:27.13 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:29.34 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:32.35 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:34.02 | *** join/#asterisk imox (~imox@91-64-148-46-dynip.superkabel.de) |
| 17:36.58 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:37.05 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 17:39.19 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:39.27 | Penguin | prods mjordan |
| 17:41.20 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:42.02 | Penguin | lights a fire under qwell |
| 17:42.19 | *** join/#asterisk evilman_home (kvirc@89-179-77-66.broadband.corbina.ru) |
| 17:43.22 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:44.35 | Penguin | throws stink bombs near the desks of file, newtonr, and putnopvut |
| 17:45.22 | apb1963 | Good morning all. One call goes through, two others do not (same phone number). By go through I mean that both sides can hear each other, whereas the "not go through" situation it apparently connects, but I do not know if the other side can hear me, however I cannot hear them. Here's the call: http://ix.io/7He |
| 17:45.44 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:46.43 | apb1963 | I don't know if the problem is on my side or theirs or what the story is.. all I know is I have to dial multiple times to get through and all I hear is silence when I don't. |
| 17:47.08 | jmetro | what if you dial another external number that you can test with [cellphone? |
| 17:47.24 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:47.33 | apb1963 | cell what now? |
| 17:48.13 | jmetro | call your cellphone and see if you can duplicate. |
| 17:48.42 | jmetro | that way you can find out if its "no audio" or "one way audio" |
| 17:48.57 | apb1963 | well I get audio when I call my own landline |
| 17:50.07 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:50.56 | *** join/#asterisk wilyjean (~william@gw1.voxtel.com) |
| 17:51.24 | jmetro | im not the greatest at looking at sip debug traffic so thats for one of the others to look at |
| 17:51.57 | Katty | hugs jmetro |
| 17:52.07 | jmetro | giggles like the pillsbury dough boy |
| 17:52.55 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:53.14 | ChannelZ | saveabunny.com - nice |
| 17:53.39 | apb1963 | it was that or chickenkiller.com |
| 17:53.46 | ChannelZ | There's nothing remarkable in the SIP debug at first glance |
| 17:54.48 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:54.51 | apb1963 | well, the only thing that stood out to me was the message about OPUS... but I didn't think that had any real effect one way or the other. |
| 17:55.00 | ChannelZ | although it seems to be using port 5060 for RTP too which seems odd |
| 17:55.07 | apb1963 | that I didn't notice |
| 17:56.28 | ChannelZ | actually both the device and asterisk seem to use the same IP? |
| 17:56.43 | apb1963 | ok here's the topology... |
| 17:56.51 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:57.05 | Penguin | Problems, problems. |
| 17:57.08 | apb1963 | both asterisk & device are on the same subnet. 192.168.0.x |
| 17:57.21 | apb1963 | both plugged into the same router |
| 17:57.31 | apb1963 | which of course has a public ip |
| 17:58.58 | Penguin | If both asterisk and the device are on the same LAN, how is the public IP address relevant? |
| 17:59.02 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:59.03 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:59.18 | ChannelZ | well unless I'm reading this wrong, the device (PhonerLite) is coming from 174.x.x.138, but that's also the IP that Asterisk is telling it to send its RTP to...? |
| 17:59.33 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:59.49 | Penguin | I think that would indicate a network misconfiguration. |
| 17:59.53 | ChannelZ | What I'm seeing either flat out doesn't make sense, or I'm missing something (entirely possible) |
| 17:59.54 | apb1963 | .138 is the router |
| 18:00.27 | ChannelZ | Is it a softphone running on the same machine as asterisk? |
| 18:00.38 | apb1963 | sofphone yes - same machine no. |
| 18:00.52 | ChannelZ | Well then that's a problem. |
| 18:00.57 | Penguin | But they are on the LAN together, in the same subnet, right? |
| 18:01.04 | apb1963 | Penguin: Yes |
| 18:01.08 | ChannelZ | I'm actually unsure why the whole thing didn't blow up or throw any other error messages |
| 18:01.11 | apb1963 | 192.168.0.x |
| 18:01.20 | Penguin | Why are you using the public address then? |
| 18:01.53 | apb1963 | Penguin: I'm not sure how to respond to that question |
| 18:02.13 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:02.15 | Penguin | Stop it. Use the private address when on the LAN. |
| 18:02.29 | apb1963 | Penguin: Please tell me how |
| 18:02.40 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:02.40 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:02.47 | ChannelZ | do you have externaddr and localnet set correctly in sip.conf? |
| 18:02.51 | Penguin | However you put in the wrong address... do that again, but use the private address. |
| 18:03.27 | apb1963 | ChannelZ: I'll doublecheck but I think so |
| 18:03.34 | ChannelZ | and/or have you forced an IP into your phonerlite app, or is it using STUN to try and determine its external address and that's where it's getting confused? |
| 18:03.47 | apb1963 | I believe it's using stun |
| 18:03.53 | Penguin | That's the problem. |
| 18:04.06 | apb1963 | I take it back |
| 18:04.07 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 18:04.09 | apb1963 | not using stun. |
| 18:05.15 | apb1963 | I use the fqdn, so no not forcing the IP |
| 18:05.25 | apb1963 | then again |
| 18:05.29 | Penguin | That's probably where the discrepancy is. |
| 18:05.41 | apb1963 | since the fqdn resolves to the public address... |
| 18:05.46 | Penguin | If you resolve the fqdn that you put in, is it a private address? |
| 18:05.51 | apb1963 | nope |
| 18:05.52 | ChannelZ | well somehow it's deciding to use your WAN IP, as is Asterisk.. |
| 18:05.52 | Penguin | That's why it's wrong. |
| 18:06.02 | Penguin | Stop using the PUBLIC address. |
| 18:06.11 | Penguin | You are on the LAN, so configure it that way. |
| 18:06.16 | apb1963 | hmmm |
| 18:06.35 | Penguin | If you must use the fqdn, override the DNS while in the LAN. |
| 18:06.47 | apb1963 | yeah I was just thinking that |
| 18:07.02 | apb1963 | but really it just moves the problem to another spot |
| 18:07.17 | apb1963 | so I might as well leave it in the easy spot rather than the hard spot |
| 18:07.29 | apb1963 | ok |
| 18:07.41 | Penguin | I have a SIP client on my Android phone and I have my fqdn programmed onto the asterisk host address. I can go inside the lan or outside and the phone always connects to the right place because when I am inside, the IP address is a local address as per local DNS records. |
| 18:08.38 | Penguin | Not all routers (along with asterisk) will deal with using the public address when not outside the LAN. |
| 18:09.37 | mjordan | man, a lot of hate goin on from Penguin |
| 18:09.48 | mjordan | :-) |
| 18:10.22 | apb1963 | so you're saying the fqdn is the private or public address? ("asterisk host address") |
| 18:10.33 | apb1963 | oh oh |
| 18:10.35 | apb1963 | nvm |
| 18:10.50 | apb1963 | just absorbed the rest of what you wrote |
| 18:10.52 | Penguin | You said the fqdn resolves to your public IP address, which is on the internet side of your router. |
| 18:11.02 | Penguin | But your device and asterisk are on a LAN together. |
| 18:11.02 | apb1963 | Right... I don't run a local DNS server |
| 18:11.23 | apb1963 | yes |
| 18:11.29 | apb1963 | yes |
| 18:11.42 | Penguin | Even plastic routers usually have a local DNS forwarder where it will allow you to override certain addresses. |
| 18:11.45 | apb1963 | i'll just hard code the private ip |
| 18:11.50 | Penguin | That's one way. |
| 18:11.59 | Penguin | But what happens when you go outside the LAN? |
| 18:12.10 | apb1963 | dunno... softphone is on a desktop pc |
| 18:12.18 | Penguin | Ah, so that won't matter. |
| 18:12.36 | Penguin | I was thinking of a mobile device. |
| 18:12.42 | apb1963 | but talk to me more about the dns forwarder... I do have forwarding setup on the router |
| 18:12.49 | [TK]D-Fender | apb1963: So your softphone and * are both on the same LAN right now? |
| 18:12.50 | ChannelZ | It's really awkward when he uses the phone on the bus. |
| 18:12.58 | apb1963 | [TK]D-Fender yes |
| 18:13.02 | [TK]D-Fender | apb1963: <--- SIP read from UDP:174.134.98.138:5060 ---> <------ |
| 18:13.07 | [TK]D-Fender | apb1963: Stop pointing it to the WAN IP |
| 18:13.08 | Penguin | I'd imagine carrying around his desktop computer gets tiresome. |
| 18:13.09 | [TK]D-Fender | ^ |
| 18:13.19 | apb1963 | yah, penquin said that |
| 18:13.20 | [TK]D-Fender | apb1963: HAIRPIN NAT FAIL |
| 18:13.32 | apb1963 | well at least now we know the technical term |
| 18:13.34 | Penguin | Someone's a little tardy to the party. |
| 18:13.54 | [TK]D-Fender | Bathroom break |
| 18:14.03 | Penguin | Hope you sprayed. |
| 18:14.21 | apb1963 | rumor has it matches work too |
| 18:14.30 | wilyjean | Hi, I'm looking for a way in order to transmit some extra data just before the end of a SIP transfer (or during hangup), like we can do it using SIP header with the INVITE message before a DIAL instruction ? To be more psecific, I want to know If I can add and send a custom reason on the call terminaison ? |
| 18:14.57 | apb1963 | Kind of interesting how it works most of the time though |
| 18:15.50 | [TK]D-Fender | wilyjean: Nope. |
| 18:18.23 | ChannelZ | Whats that idiom about a broken clock being right twice a day... |
| 18:26.19 | apb1963 | ChannelZ: Makes it hard to test |
| 18:26.28 | apb1963 | It works... but, will it work everytime? |
| 18:27.55 | Penguin | Set it up right; it will work every time. |
| 18:28.22 | apb1963 | sure.. if you know you haven't made any mistakes :) |
| 18:29.28 | apb1963 | but if you don't know what "right" is... there's no guarantee you haven't made mistakes |
| 18:31.46 | *** join/#asterisk JoshBudde (~Adium@63.138.28.186) |
| 18:32.04 | apb1963 | so it's working at the moment... private address shows up.... each side has it's own IP... no public IP in sight. |
| 18:32.25 | apb1963 | so, thank you Penguin, ChannelZ & [TK]D-Fender |
| 18:33.01 | apb1963 | and jmetro |
| 18:34.05 | ChannelZ | Yah good luck |
| 18:35.08 | apb1963 | chan_sip.c: Peer audio RTP is at port 192.168.0.100:5062 |
| 18:35.09 | ChannelZ | I wonder/hope asterisk 12 will be better in this area and able to make better decisions, being able to bind to muliple IPs |
| 18:35.52 | apb1963 | 5062 is odd for RTP.... yes? |
| 18:36.26 | ChannelZ | Odd but not unheard of |
| 18:36.53 | apb1963 | wouldn't it have to be in my rtp.conf ? |
| 18:37.07 | [TK]D-Fender | apb1963: No |
| 18:37.10 | apb1963 | I have that at 10000-20000 like normal people |
| 18:37.14 | ChannelZ | well depends on who that IP is |
| 18:37.16 | [TK]D-Fender | apb1963: THEIR port is not your system's concern |
| 18:37.30 | apb1963 | i see |
| 18:37.32 | ChannelZ | rtp.conf controls what ports you ask others to send to. |
| 18:37.44 | ChannelZ | What others ask you to send to is up to them |
| 18:38.55 | apb1963 | well then I guess I'm ok |
| 18:39.30 | apb1963 | thank you all again :) |
| 18:44.22 | apb1963 | Via UPnP PhonerLite can retrieve the public IP address of the router and can set a port forwarding automatically. The router has to support UPnP and the control via UPnP. <<<< When would I use that feature? |
| 18:45.23 | [TK]D-Fender | not while you're on the local LAN... |
| 18:48.27 | apb1963 | so if I were elsewhere.... well, i'm not sure I see the purpose of that option. I'm thinking it's so you don't have to muck with the router... but you have to turn it on, on the router so you're mucking again. either you muck with forwarding on the router or you muck with UPnP... not sure I understand the benefit. |
| 18:56.30 | ChannelZ | Unless it's not your router to be mucking with |
| 18:57.11 | ChannelZ | The hope being that UPnP was ubiquitous and enabled everwhere else - which in a lot of wireless routers and such it might be |
| 18:58.08 | ChannelZ | But UPnP has its own problems and sysadmins who are paying attention usually have turned it off |
| 19:00.04 | ChannelZ | In another who-knows-how-many-years when we're all IPv6, a lot of this NAT mess won't be such a mess. |
| 19:01.29 | apb1963 | got it. Thank you |
| 19:02.01 | *** join/#asterisk P424D0X (~kvirc@dslb-188-102-016-008.pools.arcor-ip.net) |
| 19:02.28 | apb1963 | So my TP-Link router seems to work |
| 19:02.52 | apb1963 | $20 well spent I guess. |
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| 19:57.05 | *** join/#asterisk jakent (~jakent@169.sub-70-208-153.myvzw.com) |
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| 20:00.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:06.14 | *** join/#asterisk nam3l3zz (~quassel@86-46-203-155-dynamic.b-ras1.pgs.portlaoise.eircom.net) |
| 20:08.18 | *** join/#asterisk julgr (~julgr@c-24-143-72-222.customer.broadstripe.net) |
| 20:14.44 | Penguin | drmessano: I tested the antenna and cans today. I connected the xcvr right to the antenna and had someone about 35-40 miles away adjust their power level to where I had 4 bars on my meter. I assume the bars match up to a standard S meter, where the marks are 1,3,5,7,9,+10,+30. That would mean I had S7 on him. I then connected up the cavities and put the RX cable to my radio and he transmitted again. This time it showed 3 ... |
| 20:14.50 | Penguin | ... bars, which should be 2 S units lower. |
| 20:16.21 | Penguin | I don't know if that's enough to be a problem, but it is certainly lower signal with the cavities hooked up. |
| 20:28.17 | *** join/#asterisk CeBe (~CeBe@port-92-206-237-61.dynamic.qsc.de) |
| 20:34.38 | Penguin | drmessano: Any idea what the typical loss would be when going through the cavities? |
| 20:34.40 | *** part/#asterisk saxa (~saxa@201-50-67-77.user.veloxzone.com.br) |
| 20:40.25 | *** part/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-syvllkitzlijrbdc) |
| 20:40.32 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 20:41.18 | *** join/#asterisk dpeloquin (uid13057@gateway/web/irccloud.com/x-syvllkitzlijrbdc) |
| 20:46.00 | drmessano | Penguin, insertion loss should be no more than 2.0db and typically closer to 1.0db. 3 to 7 would be 4 S-Unit (as marked), which indicates 24db of insertion loss |
| 20:46.08 | drmessano | I think you found your problem |
| 20:47.17 | drmessano | That would be one of three things.. Bad cavities, Mistuned cavities, TX and RX connected to wrong ports. |
| 20:51.00 | drmessano | Maybe not so much the last one.. you should have at least 90db of attenuation from RX to TX and you should see THAT much of a DIP, not 20-30db. I would say bad cavities or mistuned |
| 20:51.43 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:04.32 | Penguin | I've got another set of cavities that we'll take out there next week. |
| 21:08.50 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:226:221:6aff:feb8:e0b2) |
| 21:09.16 | j4jackj | Penguin: should I start a dialup BBS with a forum all about Asterisk? |
| 21:10.34 | leifmadsen | j4jackj: yes, yes you should |
| 21:10.42 | leifmadsen | and make sure you distribute it via frontdoor |
| 21:11.14 | leifmadsen | I've been meaning to find some time to setup FreeDOS and RA |
| 21:12.47 | WIMPy | With IAXmodem? |
| 21:12.56 | j4jackj | Nope, with a real modem |
| 21:12.59 | *** join/#asterisk bobbyz (~bobbyz@c-67-161-141-149.hsd1.co.comcast.net) |
| 21:13.34 | j4jackj | I want to make it a Linux Telnet+Dialin BBS, at or around 24000 baud (you read that right, 24kbaud, without a dot). It would actually be a non-Internetted IP network where the user dialing in gets 10000bits/s of Telnet and 12000bits/s to use for voice chat as he pleases, maybe leaving voicemail for other users. |
| 21:16.02 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
| 21:16.14 | j4jackj | It'll be $8 a month with a $2 surcharge every time you use dialup (called 'modem tie-up rent') and free for Telnet users who don't think that me having the phone in use between 9:00 and 10:00 AM UTC is funny. |
| 21:17.27 | WIMPy | still has two terminal adaptors with modem emulation. |
| 21:17.34 | j4jackj | I think you should have seen the hint of 'before-I-was-born-technology' there |
| 21:19.12 | j4jackj | WIMPy: did you see the hint of technology from before I was born? |
| 21:19.44 | WIMPy | No, but I guess I should have notice you're only 13 years old (or less). |
| 21:20.34 | j4jackj | Yup |
| 21:20.36 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 21:21.53 | *** join/#asterisk jakent (~jakent@63-252-49-201.ip.mcleodusa.net) |
| 21:23.19 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 21:25.21 | ChannelZ | Bring back Quantum Link! |
| 21:30.59 | *** join/#asterisk Meaw (~salem@146.185.62.88) |
| 21:31.12 | Meaw | Hi |
| 21:31.21 | Meaw | i have this issue with asterisk call hangs after 20 seconds |
| 21:31.58 | Meaw | maximum retries exceeded - no critical packet |
| 21:32.49 | wltjr | finally making progress with faxing, took a wild guess and commented out t38pt_udptl=yes and I am getting a tiff file now, still not working 100% but further than I have been, lots of output as well http://pastebin.com/af4KujSG |
| 21:33.37 | ChannelZ | Meaw: Network issues.. one side or the other is behind NAT and not working |
| 21:33.38 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:40.50 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 21:53.03 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 22:02.49 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 22:17.00 | wltjr | keeps exiting after receiving the fax, skipping the rest of the stuff to email the fax, even with ffa its showing success no error, resolution, transfer rate I have a tiff on disk, but it just bails after |
| 22:17.39 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-eiphpvfgfubatjpu) |
| 22:18.31 | [TK]D-Fender | Which is perfectly normal |
| 22:19.15 | *** join/#asterisk Dovid (~Dovid@ool-457f6ac5.dyn.optonline.net) |
| 22:20.00 | wltjr | I guess wasn't how it was with other stuff, and I think examples I have seen of receivefax show stuff after that command, reading something along the lines where someone modified the code or somethign and set a flag |
| 22:22.35 | wltjr | guess I need to switch to h |
| 22:23.51 | [TK]D-Fender | Always has, |
| 22:24.01 | wltjr | but I still see lots of examples with something after receive fax, guess they are wrong or just outdated |
| 22:24.05 | [TK]D-Fender | The face apps always hung up at the end |
| 22:24.08 | Penguin | drmessano: Do you know anything about PolyPhaser? |
| 22:24.08 | wltjr | example http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Fax_id263810.html |
| 22:24.12 | [TK]D-Fender | There was never expected to be "more" |
| 22:24.20 | [TK]D-Fender | fax* |
| 22:24.52 | wltjr | this example shows hangup after, not needed not called if it already exited http://www.teamforrest.com/156/integrating-fax-for-asterisk/ |
| 22:25.13 | wltjr | there are many, wonder if its diff behavior of app_fax vs res_fax/ffa |
| 22:26.06 | Penguin | wltjr: http://pastebin.com/6RQV9nEx |
| 22:26.32 | wltjr | [TK]D-Fender: but am seeing stuff to confirm your comments as well |
| 22:26.42 | wltjr | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg235733.html |
| 22:26.51 | wltjr | follow ups point out that after receive fax its gunna bai |
| 22:26.52 | wltjr | bail |
| 22:27.38 | [TK]D-Fender | wltjr: those are some broken samples |
| 22:27.45 | Penguin | After ReceiveFax() is done, it exits. See my pastebin. |
| 22:27.47 | wltjr | yeah ok, so I have dial plan issues, trivial compared to my day long bashing head cause of stupid option I enabled not sure why |
| 22:28.15 | wltjr | [TK]D-Fender: I kinda want to test it with app_fax, I have a feelling it might work, and that might have been incorrect behavior of app_fax |
| 22:28.34 | wltjr | I know rxfax for sure did not hang up, these dialplans are left over from those years, and zaptel, etc |
| 22:29.01 | wltjr | I had them break a few time as things changed in asterisk, how you set variables or something |
| 22:29.45 | wltjr | why I kinda hate *, almost a decade or so later, still discovering and learning stuff about it... |
| 22:30.01 | Penguin | I think app_fax is going to exit after it's done, too. |
| 22:30.58 | wltjr | Penguin: maybe, I do not recall ever having things work after rxfax, even with faxreceive, I would test but I kinda care less now, just going to address correctly in dialplan and move on finally |
| 22:31.06 | wltjr | er receivefax |
| 22:31.29 | Penguin | I've got it laid out for you in the pastebin. |
| 22:31.46 | wltjr | Penguin: yes I am stealing, I mean copying now... thanks |
| 22:35.07 | j4jackj | I want to make the weirdest dialup thing. Everyone uses Speex 8kbps to encode their voice on the client side, and it's sent as an unpacketized SMTP message to the other person, using MIME. then they get to log in to the textual interface. |
| 22:37.08 | wltjr | Penguin: out of curiosity how come you do all that in * vs say bash file or something? |
| 22:37.25 | wltjr | nothing external to have to maintain/keep track of? |
| 22:37.27 | Penguin | Because I'm using asterisk for the phone stuff. |
| 22:37.52 | wltjr | Penguin: sure but I would just say do one system call to a script and have it call tiff2pdf, remove files, etc |
| 22:38.14 | Penguin | It could just as easily be written in something like python and asterisk has to do almost nothing. |
| 22:38.48 | Penguin | Then I'd have to deal with another file. |
| 22:39.24 | wltjr | that I could see as a benefit for sure as I migrated my * install some time back and left scripts behind.. kinda why i was asking |
| 22:39.40 | wltjr | Penguin: you don't see much overhead in * vs calling script, bash/python/etc |
| 22:40.14 | Penguin | There's a saying that starts out something like "half a dozen of one thing..." |
| 22:40.38 | Penguin | I do almost everything in dial plan. |
| 22:41.56 | wltjr | Penguin: sure, just curious due to my recent dilemma, I hadn't thought about doing it all from the dialplan |
| 22:47.12 | wltjr | wonder why I have set t38pt_udptl=no in sip.conf to get faxing to work, sip trunk provider issue? |
| 22:51.06 | ChannelZ | Do they support t.38? |
| 22:52.03 | wltjr | seems to be per www.nexvortex.com/pdf_files/nexVortex-Implementation-Guide-V1.5.pdf |
| 22:52.10 | wltjr | Fax: nexVortex supports fax over IP via G.711µ passthrough and T.38. T.38 is the recommended formatfor reliability. |
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| 23:07.11 | [TK]D-Fender | Some provider force you pay for "premium" services to actually get T.38 |
| 23:07.19 | [TK]D-Fender | Don't recall if they were one of them |
| 23:07.49 | wltjr | pretty sure not they are a business provider all around, but will double check to make sure |
| 23:08.41 | navaismo | and did he showed a debug of the cli with:core set verbose 10, core set debug 10, fax set debug on, fax set t38cap on, fax show version,fax show licenses ,sip set debug on and the debug enabled in the cli from logger.conf? |
| 23:09.25 | j4jackj | I'm trying to facilitate Opus and Speex in battling it out for the title of 'best dialup music codec' |
| 23:12.56 | wltjr | I did do the set fax debug stuff in cli and nada, there was some issue, might need to tcpdump or something to find out |
| 23:14.04 | wltjr | meh finally got mine done, not to fix 2 clients... |
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| 23:22.57 | navaismo | lst time i debug a fax issue with FFA cli debug throw a txt with ~42000 lines of output jus to get the T1_TIMEOUT error and the |
| 23:22.57 | navaismo | res_fax.c:1529 receivefax_t38_init: channel 'SIP/voipms-000013cf' refused to negotiate T.38 |
| 23:22.58 | navaismo | res_fax.c:1550 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/voipms-000013cf' and T.38 negotiation failed; aborting. |
| 23:23.40 | wltjr | navaismo: I think I did the debug with spandsp not ffa |
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| 23:36.00 | UnixDev | hi, is there some way to use SIPAddHeader and actually get it to add the header when using Transfer application? It seems the SIPAddHeader succeeds in dial plan, but no headers are added to 302 redirect request |
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