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| 00:14.40 | KingDavidNYC | hi, can anybody please give me a hand installing kamailio please? |
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| 00:20.06 | darkdrgn2k | hey all, whtas the current status of google voice and asterisk |
| 00:20.28 | KingDavidNYC | can somebody please give me a hand installing kamailio? |
| 00:23.38 | carrar | opensips offers classes |
| 00:24.45 | KingDavidNYC | the entire installation stops because it can't find a docbook2x |
| 00:25.44 | carrar | is it installed? |
| 00:27.38 | KingDavidNYC | no, it is compiled, but make install stops |
| 00:28.51 | KingDavidNYC | it wants to find a program named doc2book2x-man, and I can not either find the program or make it skip |
| 00:32.42 | psilikon | cmd |
| 00:33.09 | KingDavidNYC | cmd? |
| 00:33.22 | psilikon | mozilla |
| 00:33.53 | KingDavidNYC | psilikon: are trying to tell me something? |
| 00:35.27 | SunTsu | KingDavidNYC: why do you come to #asterisk in search for someone to help you install OpenSER? |
| 00:36.55 | KingDavidNYC | SunTsu: There's got to be a reason right? 1) no freaking no one soul at #openser or #kamailio for the last 2 days |
| 00:37.53 | KingDavidNYC | SunTsu: 2) Kamilio/openser related to asterisk in the sense that there might be somebody in the asterisk forum who might have ever used openser |
| 00:38.07 | KingDavidNYC | SunTsu: Does this satisfies your curiosity? |
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| 02:17.04 | *** join/#asterisk DEMNVT (~Adium@220-245-156-82.static.tpgi.com.au) |
| 02:24.12 | DEMNVT | Hey guys. Our PABX got hacked on the weekend and the guy sent 500+ international calls through our PABX. I'm having trouble tracking down how he did it. We have Fail2Ban in place which didn't pick up anything and it looks like the calls were made using an extension that doesn't even exist on the system (based on the records in the cdr). So it doesn't look like it was a straight forward attack (ie. Figure out the details of an extension, r |
| 02:26.54 | DEMNVT | Getting some wierd messages from channel.c in the full log which are a bit scary. Stuff like: |
| 02:27.01 | DEMNVT | channel.c: Avoiding initial deadlock for 'IAX2/faktortel-35' |
| 02:27.03 | DEMNVT | and |
| 02:27.18 | DEMNVT | channel.c: Got a FRAME_CONTROL (-1) frame on channel IAX2/faktortel-35 |
| 02:28.22 | DEMNVT | Faktortel is our IAX trunk. Is is possible that they somehow hijacked our connection to Faktortel and used that to start feeding calls through our server? |
| 02:32.45 | tzanger | DEMNVT: almost *all* attacks of this nature take advantage of either easily-guessed extensions or through bad configuration of default contexts |
| 02:33.15 | tzanger | DEMNVT: what is the default SIP extension, and what happens if you take any SIP phone with nosuchuser/oogabooga as a login and try to place a call? |
| 02:35.06 | drumkilla | those specific log messages are not indicitive of a problem. They happen under normal circumstances. |
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| 02:43.58 | DEMNVT | tzanger: Thanks. It never would have crossed my mind that our PABX was that wide open. I just did what you said and was able to call internal extensions |
| 02:44.26 | DEMNVT | and it actually allowed me to place a call out via our trunk (although it failed) |
| 02:44.38 | DEMNVT | that's bad! Checking default contexts etc now |
| 02:44.58 | DEMNVT | drumkilla: Looks like I was at the completely wrong end of the paranoia stick :p Thanks. |
| 02:45.03 | tzanger | DEMNVT: one thing I do on ALL my installs is actually create a [default] context and that context has something along the line of exten => _*,1,NoOp(Log this attempt to ${EXTEN}), _*,n,Hangup |
| 02:46.14 | tzanger | Mind you I never see much trying to come in the default extension... 99.99% (yes I have tracked that many attempts) of my received attacks are dictionary attacks and fail2ban takes care of them |
| 02:47.41 | WIMPy | _* is pretty unlikely to get matched. |
| 02:48.25 | p3nguin | Only when you press * on the keypad and are in the default context. |
| 02:48.27 | tzanger | WIMPy: something like. :-) |
| 02:48.31 | tzanger | _.* perhaps |
| 02:48.33 | tzanger | let me check |
| 02:48.45 | p3nguin | What's the point of the extension, anyway? |
| 02:48.58 | p3nguin | To create load on the system? |
| 02:49.08 | WIMPy | p3nguin: Press * and then send. |
| 02:49.39 | tzanger | yeah I'm mixing up my regex and extension matching |
| 02:50.46 | p3nguin | ANd what is the point of the extension in the first place? |
| 02:51.27 | WIMPy | And who says that guest go to 'default'? |
| 02:51.48 | WIMPy | (for the unwanted kind of "guest") |
| 02:52.10 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:52.49 | p3nguin | I finally got around to changing my anonymous context to misc-calls and left default to asterisk's internal usage. |
| 02:53.08 | p3nguin | Yes, it took me long enough. |
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| 02:55.45 | p3nguin | I guess it's another one of those secret PBX configuration things. |
| 02:56.10 | DEMNVT | tzanger: Thanks man. Looks like you hit the nail on the head. The default context had iaxout included in it... so straight out to our trunk with no auth... awesome. Now to find the man that coded the dial plan and slap him in the back of the head. |
| 02:58.05 | WIMPy | wonders what breaks without that include... |
| 02:58.27 | p3nguin | I wouldn't worry about it. I would just wait until it breaks and fix it when it does. |
| 02:58.57 | WIMPy | Sure |
| 02:59.18 | WIMPy | It's just that I have a feeling that it could have been a thoughtless fix for something else. |
| 03:00.21 | tzanger | WIMPy: default in sip.conf is to have context=default |
| 03:00.35 | tzanger | you can change that to whatever you want of course but most problems occur when you don't know these little things or understand what they mean |
| 03:00.41 | p3nguin | There should really be nothing or almost nothing in the default context, and the context assigned in the general section of sip.conf (or another channel driver conf) should have only what you want anonymous callers to have access to... or don't allowguest at all. |
| 03:01.02 | tzanger | actually I have my default context accept my phone number or "sip.mixdown.ca" as valid extensions and route to the same place as my normal incoming SIP DID |
| 03:01.12 | tzanger | p3nguin: agreed, which is exactly what I do |
| 03:01.25 | tzanger | and maybe sip.conf has changed since I've played with it |
| 03:01.25 | WIMPy | tzanger: Exactely, why it might happen that someone changes te default context to phones or something in that area. |
| 03:01.34 | tzanger | yep |
| 03:01.44 | p3nguin | Best practice would be for you to change that general context to another name and leave [default] empty. |
| 03:02.06 | tzanger | there are a few different possibilities but it all comes down to "where do calls go when they come from unknown sip clients?" |
| 03:02.15 | tzanger | p3nguin: no, that is not best practise |
| 03:02.18 | tzanger | best practise is to log that shit |
| 03:02.44 | tzanger | same end result though -- unauthorized clients don't get through. |
| 03:02.50 | p3nguin | Log it if you want, but still don't use the default context. |
| 03:03.11 | tzanger | I added my number and sip.mixdown.ca because I specifically wanted unauthenticated access to those numbers |
| 03:03.33 | tzanger | p3nguin: I prefer to leave it named default. the name is sound, but the contents of it should be well-defined. I am not a fan of empty contexts |
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| 03:10.55 | DEMNVT | Not sure if it will break anything in our case. It doesn't look like it. I don't think we have anything in the default context so I'll just change it back to that and see if anything breaks. |
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| 03:16.04 | DEMNVT | The guy who built the system had a little network set up for his family on the server also. The default context was pointing to the context for that. |
| 03:17.04 | p3nguin | Anyone here using VMware Infrastructure Client 2.5.0 that can tell me how I can save my page layout/settings? Every time I disconnect and reconnect to the server, the panes change sizes back to original dimensions rather than staying the way I put them during the previous session. |
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| 04:23.46 | DEMNVT | Thanks again for your help guys. default context now set up and writing nasties to the logs. :) Take care. |
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| 04:48.29 | *** join/#asterisk mcreedjr (62107eb1@gateway/web/freenode/ip.98.16.126.177) |
| 04:49.51 | mcreedjr | I have a TE110P connected to a Time Warner Cisco voice gateway and am having issues with the PRI going down after a short amount of time. PRI status goes to Status: Provisioned, Down, Active. |
| 04:50.22 | mcreedjr | Using dahdi 2.4.1 and libpri-1.4.12-beta3 |
| 04:50.42 | mcreedjr | Rebooting the server brings the PRI back in to service for an undetermined, but short, amount of time. |
| 04:51.32 | mcreedjr | Never lose layer 1, so i'm not thinking its a clocking issue... it's always L2 that I lose. any ideas? |
| 05:02.45 | shapr | pri debug? |
| 05:05.53 | mcreedjr | This is a PRI debug captured while the D channel actually appears to go down: http://pastebin.com/ZT0PkGFD |
| 05:07.32 | shapr | Already had a line test from the telco? |
| 05:08.34 | mcreedjr | I've opened several tickets with them. The voice gateway is actually installed in the same rack. They hand off the PRI via a 5' T1 crossover. |
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| 05:08.49 | mcreedjr | Their physical side is running clean. No clock slips, etc. |
| 05:10.46 | shapr | Is your system.conf pastebin'd somewhere? |
| 05:12.10 | mcreedjr | http://pastebin.com/Fs0ywbeH |
| 05:16.54 | shapr | mcreedjr: Nothing obviously wrong |
| 05:17.09 | mcreedjr | Yeah, thats my issue, too. |
| 05:17.29 | shapr | Do you have an analog card in this system also? |
| 05:17.49 | mcreedjr | I was on an older version of libpri, zaptel and asterisk when the issue began. It was having problems on that software stack, too. Replaced the TE110 with another. Same issues. |
| 05:18.00 | mcreedjr | I do have an analog card in the system, too. |
| 05:18.05 | p3nguin | Whoa. Did pastebin.com change its look today? |
| 05:18.28 | shapr | If changing hardware and software did not fix the issue, it surely sounds like it's a telco issue. |
| 05:18.31 | mcreedjr | With the older software stack, the PRI will stay in service, but drops calls randomly through the day. |
| 05:19.49 | shapr | Any idea what changed on the telco side when the issue appeared? |
| 05:20.06 | mcreedjr | Yeah, the circuit was installed :) |
| 05:20.33 | shapr | When did you buy the TE110? |
| 05:20.41 | mcreedjr | The service has only been installed for about 6 months. They've had on and off problems with it since the start. Seem to have gotten markedly worse. |
| 05:20.58 | mcreedjr | The TE110 has been around for a while. Sold it to them out of my used inventory when they got the PRI. |
| 05:21.11 | mcreedjr | Both TE110s that I've tried are the same approximate age. |
| 05:21.27 | mcreedjr | I find it odd that both cards exhibit the same exact problems. |
| 05:22.04 | mcreedjr | I honestly have not changed out the T1 crossover yet. Was ruling it out because I wasn't having L1 problems. My card is never in red alarm. Would think it would be if it were a cabling problem. |
| 05:22.31 | shapr | sounds right to me |
| 05:26.27 | mcreedjr | Yeah.. I'm at a loss. |
| 05:27.04 | mcreedjr | And I agree the telco is one of the few things that has remained constant throughout all of these problems, but unless I have specific proof of what the problem is we all know how likely the telco is to do anything about it. |
| 05:27.41 | *** join/#asterisk doneir (var@gateway/shell/xzibition.com/x-rdhjybdbsissmrwe) |
| 05:28.27 | doneir | i've been looking online, but can't seem to find out where one can chaneg the sensitivity of the silence detection used by the Record() application |
| 05:29.35 | doneir | i've also noticed that if silence is the only recorded data, the Record() application will truncate the recorded data to 8016 bytes (alaw, e1) |
| 05:30.10 | doneir | so either i need to tweak the silence detection sensitivity, or change the silence behaviour |
| 05:31.15 | doneir | as a quick fix I would like to change the sensitivity, then look at creating a possible patch for the silence behaviour. Would anyone happen to know where one can change the sensitivity of the silence detection for Record()? |
| 05:31.50 | doneir | this is using 1.6.2.17 btw |
| 05:34.23 | mcreedjr | shapr thank you for your time and insight. i appreciate it! |
| 05:35.06 | shapr | sure, sorry I couldn't help more |
| 05:37.39 | doneir | ah, my google fu was bad. Looks like dsp.conf might have a silencethreshold (was looking for senstivity or volume) setting |
| 05:38.59 | doneir | nope, that's more determining if a peroid of time is silence |
| 05:39.54 | *** part/#asterisk nightwalk (~null@daimon.vixel.org) |
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| 06:18.47 | CRCinAU_ | ok, sorry about disappearing... |
| 06:19.24 | CRCinAU_ | The cisco 7970 is registering via SIP with asterisk |
| 06:19.53 | CRCinAU_ | however using tshark to watch packets for the register coming from the phone, it shows [Malformed Packet] |
| 06:20.00 | CRCinAU_ | couple this with the phone not actually registering |
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| 07:31.58 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 07:31.58 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 07:32.12 | p3nguin | I would have changed it over to SCCP by now. |
| 07:35.40 | p3nguin | I'm not all that worried about my MWI not turning back on right after a call, since I know I have messages I need to deal with. |
| 07:37.31 | p3nguin | I guess I need to roll back a few revisions to see if the bug disappears. It could be a setting that I changed, such as the earlyrtp value or something equally as irrelevant. |
| 07:42.56 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 07:46.54 | p3nguin | MWI handset light : ON |
| 22:00.35 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:00.35 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3 (2011/02/28), 1.6.2.17 (2011/02/28), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 22:02.54 | p3nguin | This is going to be a repeat, but it seems you don't understand me when I've said it before. PHONES almost never need to have ports forwarded on their NATs, so stop forwarding the ports unless you absolutely need to (and you don't need to). PHONES should _NEVER_ have their NAT settings enabled in any way, so disable NAT traversal in the phone. Properly configure Asterisk for NAT, and then configure the PEER ... |
| 22:03.01 | p3nguin | ... definition for the phone to be for a NATted phone. Now that you have reduced the port range to forwrad at the NAT, you must reduce the rtp port range in rtp.conf. I'm probably not going to repeat these things for you again. I'm getting irritated that you won't follow the basic steps that everyone else accepts. |
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| 22:13.23 | jaytee | p3nguin, thanks for the advice. I worked on this over the weekend and reducing the ports had no effect. I'm going to try your advice and get rid of the NAT forwards in the fw/router for the external phone and also get rid of the NAT settings on the phone itself and try again. |
| 22:13.41 | jaytee | and sorry if I've irritated you, was not my intention. |
| 22:14.04 | *** join/#asterisk ghchowdhury (cdd3a902@gateway/web/freenode/ip.205.211.169.2) |
| 22:15.04 | p3nguin | That's just how it works. Everyone else accepts it, so I felt like you were doing your own thing in spite of my best efforts to relay the common configuration practices to you. I wanted you to be able to have the same success with Asterisk and remote phones as everyone else. |
| 22:16.20 | p3nguin | Asterisk is capable of handling the NAT for the device all by itself. But if you ignore Asterisk's abilities and configure the phone to do the NAT traversal, your success isn't nearly as likely. |
| 22:19.50 | p3nguin | Don't overlook the part about the rtp ports. If you limit your rtp ports at Asterisk's router/firewall/NAT device, Asterisk can tell the peers that it can use all the ports when they could in reality be blocked. You must match the port range in rtp.conf with the port range you're forwarding at the firewall. Well, actually you could forward a larger range than what you configure in rtp.conf without a problem, but ... |
| 22:19.57 | p3nguin | ... you have to forward AT LEAST the same range. |
| 22:20.16 | jaytee | so now I've got no forwarding setup on the fw/router for the external phone and nothing in the NAT section of the Network page in the Polycom config. |
| 22:20.53 | jaytee | ah, so if I'm forwarding 10000-11000 in * it should match on the fw/router Asterisk is behind. |
| 22:21.11 | p3nguin | Good. Now correctly configure the peer in sip.conf to use nat. |
| 22:21.27 | jaytee | I've got NAT=YES for that peer |
| 22:21.38 | p3nguin | If you forward 10000-11000 at the firewall, Asterisk's rtp.conf must use the same range or a smaller range. |
| 22:22.17 | p3nguin | If Asterisk is configured to use 10000-20000 but you only forward 10000-11000, Asterisk will be telling the phones that it can use ports which are being blocked at the firewall. |
| 22:22.34 | jaytee | logical |
| 22:22.41 | p3nguin | You need nat=yes, canreinvite=no, qualify=yes |
| 22:23.04 | jaytee | rtpstart=10000, rtpend=11000 |
| 22:23.12 | p3nguin | If you are using 1.8, canreinvite I guess is renamed to directmedia, so it would be directmedia=no rather than canreinvite=no. |
| 22:23.34 | klixix | Guys, i'm trying to communicate w/ a metaswitch w/ asterisk 1.4.x and using FreePBX. Incoming calls work wonderfully, however anything outbound is rejected ... based on this: http://pastebin.com/gCWwhJfY --- is it the meta, or my configuration? |
| 22:26.16 | jaytee | nat=yes, canreinvite=no and qualify=yes are all set on the peer. |
| 22:26.30 | jaytee | but I'm getting this on the console when I make a call attempt |
| 22:26.35 | jaytee | WARNING[16975]: chan_sip.c:3825 retrans_pkt: Maximum retries exceeded on transmission d183ea9f-e7b50310-ccba2efd@192.168.44.60 for seqno 1 (Critical Response) -- See doc/sip-retransmit.txt. |
| 22:27.23 | p3nguin | nat=yes tells Asterisk that the phone is behind NAT, so Asterisk will look at the correct part of the transaction to get the IP addresses for the call. canreinvite=no/directmedia=no will keep Asterisk in the media path so that the previous part works. qualify=yes is used as a nat keepalive to keep the NAT at the phone's firewall open for the phone. |
| 22:27.28 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 22:28.07 | jaytee | the 192.168.44.60 is the inside network address of the external phone. |
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| 22:28.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 22:30.31 | p3nguin | Try getting a standard sip debug for that peer, so that we can see the REGISTER attempts, and also if you want to try to make a call to the device or from the device to Asterisk. |
| 22:31.39 | jaytee | p3nguin, that was going to be my next thing. I'd done one already calling in both directions but I've got another phone on my inside network that keeps cluttering the sip debug even when no other calls are in progress so I've disabled it for now. |
| 22:32.17 | p3nguin | If you did sip set debug peer <some peer>, did the other phone still clutter the debug? |
| 22:40.40 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 22:43.41 | jaytee | p3nguin, sorry, I was afk for a few. |
| 22:45.13 | jaytee | I've got a sip debug of a call attempt from the external phone to an inside extension and a call to the external phone's extension. Noticed that now I've only got ringing when calling from the external phone to an inside extension. When I call out to the external phone it no longer rings. |
| 22:45.26 | jaytee | here's the sip debug if you feel like taking a look. |
| 22:45.29 | jaytee | http://pastebin.com/1hsEQGAu |
| 22:49.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:50.54 | tvc123 | so quick question I'm using a AEX410 with 4 lines ... one of them also goes to a fax machine is there an easy way to check if that line is active before I send a call out on it? |
| 22:53.50 | jaytee | tvc123, check out the ChanIsAvail application |
| 22:56.44 | tvc123 | jaytee: thanks! |
| 23:01.30 | *** join/#asterisk gpled (~gpled@69.1.105.42) |
| 23:02.37 | *** join/#asterisk war9407 (war@71.6.165.232) |
| 23:05.55 | p3nguin | jaytee: Is the phone registering? Meaning, if you "sip show peers" do you see that phone on the list and does it have an IP address listed for it? |
| 23:09.54 | jaytee | p3nguin, yes but since I removed the NAT info on the phone and the port forwarding on the firewall it now shows as UNREACHABLE |
| 23:10.22 | jaytee | when I dial an internal extension from it the internal phone rings but not the other way |
| 23:10.47 | klixix | sigh |
| 23:10.48 | jaytee | and the IP address of the phone is the outside static IP address of the fw/router |
| 23:12.05 | jaytee | I think it's the piece of crap Netgear fw/router on the network the external phone is on. |
| 23:17.11 | p3nguin | jaytee: It's possible that the router appliance is at fault. There is a small percentage of cases where the phone side needs port forwarding to work. You can always add the port forwarding back to the phone side and see what changes. As long as you don't configure NAT on the phone and you do let Asterisk handle the NATting of the phone, it shouldn't ruin anything. It's not working all the way right now, so all it ... |
| 23:17.17 | p3nguin | ... can do is put you into that small percentage of cases. |
| 23:18.06 | *** part/#asterisk gpled (~gpled@69.1.105.42) |
| 23:18.10 | jaytee | p3nguin, understood. I'll try re-adding the forwarding on the Netgear appliance. |
| 23:18.52 | jaytee | if I still get no joy I'll try another new phone out of the box without any NAT settings as I've read that if you set it once it sometimes "sticks" unless you reflash the phone. |
| 23:19.37 | jaytee | I'm also considering testing with a phone set to a static IP plugged directly on the cable modem to eliminate the Netgear. |
| 23:19.37 | p3nguin | It's really annoying if you have to do port forwarding to phones because that requires that you misconfigure any additional phones you add to that network to use different ports, since you can only forward a given port to a single device. |
| 23:20.19 | jaytee | if only Polycom would build in IAX support :-( |
| 23:20.29 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 23:21.10 | p3nguin | I expect to be able to put as many phones as I want onto a LAN and they all speak to a remote Asterisk exactly like the next one does. |
| 23:32.46 | jaytee | p3nguin, I readded the port forwarding on the Netgear appliance and I also rebooted the phone. It registers but shows Unreachable in the Status column when I do a "sip show peers" |
| 23:36.25 | jaytee | well, it's two hours after quittin time so I'm going to try again tomorrow. |
| 23:36.35 | jaytee | p3nguin, thanks for your help. |
| 23:38.19 | p3nguin | jaytee: If the port forward doesn't make it show that it is registered, that brings you back out of the small percentage which need port forwarding... but it might put you into another category: those who have routers that do not work with SIP *at all*. |
| 23:39.30 | jaytee | well, not sure if that's the case because at one point when I had the NAT info set on the phone it would register and not show unreachable and I could dial and get ringing in both directions, but no audio. |
| 23:39.55 | jaytee | I'm going to try another phone set to static directly on my cable modem bypassing the Netgear. |
| 23:40.27 | IsUp | jaytee: if you have audio problem, probably you need to forward RTP ports too |
| 23:40.33 | jaytee | but I'll have to wait to do that tomorrow. need dinner. |
| 23:41.17 | jaytee | I made sure I forwarded the RTP ports 10000-11000 on both fw/routers. The one my * server is behind and the one the phone is on. |
| 23:41.22 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:41.25 | IsUp | jaytee: i had the same problem with Cisco Pix firewall. as i far know, ive set localnet= nat= and externip= on sip.conf |
| 23:41.28 | jaytee | and that matches the range in RTP |
| 23:42.07 | IsUp | also, if your phone supports, you can try using a STUN server. ie stun.xten.net |
| 23:42.30 | jaytee | I've got externip and localnet set in sip.conf. Calls to and from Flowroute to my * server and internal extensions work fine. just this one external phone. |
| 23:42.42 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 23:43.39 | jaytee | but I'm going to try to tackle this tomorrow with a fresh brain. |
| 23:43.52 | jaytee | be back later |
| 23:43.55 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 23:47.28 | p3nguin | stun.xten.net is a free-to-use STUN server? |
| 23:48.51 | IsUp | p3nguin: yes its public |
| 23:49.54 | IsUp | p3nguin: also there are many public servers listed @ http://www.voip-info.org/wiki/view/STUN |
| 23:55.59 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |