| 00:00.29 | Letoric | the phone's dial plan is the default sip.cfg provided by polycom |
| 00:00.33 | Letoric | so it's VERY basic |
| 00:00.35 | Letoric | no modifications at all |
| 00:00.39 | golikwid|mac | so like if you have *xx|9[2-9]xxxxxx|+* in your dialplan and you try to dial *80100 it will only send the *80 |
| 00:01.13 | Letoric | but the same phone sends * correctly on the old phone system, also asterisk |
| 00:01.20 | golikwid|mac | so make one more specific to your needs |
| 00:01.29 | golikwid|mac | hm |
| 00:01.30 | Letoric | it seems like it has to be something with my contexts, but I don't understand what exactly, it is |
| 00:01.59 | golikwid|mac | can you dial an outbound call with it |
| 00:02.12 | Letoric | yep |
| 00:02.27 | golikwid|mac | are you using 3 digit internal dialing? |
| 00:02.30 | golikwid|mac | [2-9]11|0T|011xxx.T|91[2-9]xxxxxxxxx|[1-8]xx |
| 00:02.31 | Letoric | can dial in, recieve calls, dial other extensions, dial voicemail |
| 00:02.46 | golikwid|mac | from what i can tell this is the default dialpan for polycom |
| 00:02.57 | Letoric | we are mostly 4 digit extensions, however, we have some 3 digit things like 500 for voicemail access |
| 00:03.08 | golikwid|mac | hm |
| 00:03.34 | golikwid|mac | i like using the *something for features... |
| 00:03.38 | golikwid|mac | but thats what im used to |
| 00:03.39 | Letoric | me too |
| 00:03.40 | golikwid|mac | hm |
| 00:03.49 | Letoric | and that's one of the reasons I'm concerned with it not working |
| 00:03.57 | golikwid|mac | [2-9]11|0T|011xxx.T|91[2-9]xxxxxxxxx|[1-8]xxx |
| 00:04.02 | golikwid|mac | so that's for 4 digit |
| 00:04.06 | golikwid|mac | see the extra x |
| 00:04.15 | golikwid|mac | maybe thats whats going on |
| 00:04.37 | golikwid|mac | but i cant really tell without seeing your dialplan from your phone |
| 00:04.49 | Letoric | well I can happily send it to you |
| 00:05.00 | Letoric | it's plain vanilla as provided from polycom |
| 00:05.02 | golikwid|mac | if this is your first go at it why not jsut use asterisknow |
| 00:05.07 | golikwid|mac | do it |
| 00:05.10 | Letoric | asterisknow is VERY limited |
| 00:05.15 | golikwid|mac | it is |
| 00:05.19 | Letoric | at least it was when I tried to use it |
| 00:05.26 | golikwid|mac | what are you doing that it doesnt do |
| 00:05.29 | golikwid|mac | ? |
| 00:05.49 | golikwid|mac | there's an app for that |
| 00:05.53 | Letoric | well, I tried using it with freepbx, and it was cumbersome to set up groups and such |
| 00:06.08 | golikwid|mac | ring groups |
| 00:06.09 | golikwid|mac | ? |
| 00:06.14 | Letoric | ie, if I wanted our helpdesk line to ring 3 phones for 20 seconds, then 5 phones for 20, then roll over to a cell phone |
| 00:06.29 | golikwid|mac | yea it easily does that... |
| 00:06.35 | golikwid|mac | i do that with trixbox |
| 00:06.44 | golikwid|mac | and lets face it trixbox sucks |
| 00:06.46 | golikwid|mac | lol |
| 00:06.56 | Letoric | didn't seem to at first glance, and I just felt more comfortable with the command line/manual editing of the files |
| 00:07.04 | Letoric | the web interface seemed cool but highly limiting |
| 00:07.18 | golikwid|mac | it is cool did you see the frog? |
| 00:07.23 | golikwid|mac | anyway were off ocurse |
| 00:07.26 | Letoric | anyhow, I'll grab that sip.cfg and extensions.conf and send them to you, sec |
| 00:07.27 | golikwid|mac | dialplan |
| 00:07.40 | golikwid|mac | just the dialplan mate |
| 00:07.41 | golikwid|mac | one line |
| 00:07.45 | golikwid|mac | i dont want it all |
| 00:07.53 | Letoric | heh ok |
| 00:07.55 | Letoric | lemme find it then |
| 00:08.40 | golikwid|mac | plus there is always extensions_custom.conf... |
| 00:08.41 | golikwid|mac | hm |
| 00:09.25 | Letoric | <dialplan dialplan.impossibleMatchHandling="0" dialplan.removeEndOfDial="1" dialplan.applyToUserSend="1" dialplan.applyToRemoteDialing="0" dialplan.applyToUserDial="1" dialplan.applyToCallListDial="0" dialplan.applyToDirectoryDial="0" dialplan.applyToTelUriDial="1"> |
| 00:09.25 | Letoric | <PROTECTED> |
| 00:09.25 | Letoric | <PROTECTED> |
| 00:09.25 | Letoric | <PROTECTED> |
| 00:09.25 | Letoric | <PROTECTED> |
| 00:09.39 | golikwid|mac | ok first of all thats alot |
| 00:09.47 | golikwid|mac | and your gonna piss off the doc |
| 00:10.01 | golikwid|mac | second thats not the default dialplan |
| 00:10.21 | Letoric | oh? |
| 00:10.24 | Letoric | that's the one from sip.cfg |
| 00:10.32 | Letoric | you want the extensions.conf dialplan? |
| 00:10.37 | golikwid|mac | no |
| 00:12.16 | golikwid|mac | hm |
| 00:12.46 | golikwid|mac | i assume that the T sends the dialpan terminator |
| 00:12.47 | golikwid|mac | ? |
| 00:12.54 | Letoric | unsure |
| 00:13.05 | Letoric | I didn't mess with that at all, it worked before, seemed like it would continue working |
| 00:14.26 | golikwid|mac | why dont you take the T's out |
| 00:14.32 | golikwid|mac | dont know what they do |
| 00:14.59 | golikwid|mac | also your not using a 9 to get an outside line? |
| 00:15.14 | Letoric | nope |
| 00:15.18 | golikwid|mac | why? |
| 00:15.19 | Letoric | just straight dialing |
| 00:15.37 | Letoric | people in the office complained about having to dial 9, so the prior guy gave in and took it out |
| 00:15.44 | Letoric | once you give something, it's hard to take it away |
| 00:15.57 | Letoric | so would be .... challenging....to get them to dial 9 again ;) |
| 00:16.11 | golikwid|mac | so what happens when you want to dial 9591111 and you end up dialing extension 9591 |
| 00:16.43 | golikwid|mac | take a broom out of the closet and break it in half anyone caught not dialing a 9 get a smack on the hand |
| 00:17.14 | Letoric | we're a small group - 10ish now, and unlikely to break 100 anytime in the next 5 years. Using extensions 4101+ - unlikely we'll break that barrier anytime soon, so hard to make that argument |
| 00:17.32 | golikwid|mac | also why do you have the [0-1] is someone hoping for an operator assisted call |
| 00:18.08 | golikwid|mac | oh there is no local prefix in your area that matches that |
| 00:18.16 | golikwid|mac | no 410-xxxx |
| 00:18.35 | Letoric | nope |
| 00:18.58 | Letoric | dallas area. 214/469/817/972 and a couple others that I forget |
| 00:19.55 | golikwid|mac | [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|[2-9]xxx |
| 00:19.57 | golikwid|mac | try that |
| 00:20.10 | golikwid|mac | in fact |
| 00:20.11 | golikwid|mac | wait |
| 00:20.21 | golikwid|mac | do all of your extensions start with 4? |
| 00:20.33 | Letoric | no |
| 00:20.46 | golikwid|mac | what do they start with |
| 00:20.59 | golikwid|mac | i like to make narrow dialplans personally... |
| 00:21.03 | golikwid|mac | not sure what others do |
| 00:21.05 | Letoric | before we go changing the phone dial plan, can you help me to understand how the phone dial plan works on the old system, but not on the new, with the same phone dial plan? |
| 00:21.16 | golikwid|mac | no |
| 00:21.21 | Letoric | heh |
| 00:21.25 | Letoric | okey dokey |
| 00:21.56 | golikwid|mac | make a back up you can always go back |
| 00:22.05 | golikwid|mac | but we are following a problem |
| 00:22.18 | golikwid|mac | cant get caught up on anything but following it from the begining out |
| 00:22.30 | golikwid|mac | signal flow |
| 00:22.45 | golikwid|mac | what are your extension |
| 00:23.18 | Letoric | mostly 41xx but we also use 500 for voicemail, and some other random 500's for making new system recordings |
| 00:23.19 | golikwid|mac | and feature codes |
| 00:23.26 | Letoric | default feature codes |
| 00:23.29 | golikwid|mac | ok so |
| 00:23.35 | Letoric | 9xxx for conferences |
| 00:23.48 | golikwid|mac | [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|41xx|5xx|9xxx |
| 00:23.50 | Letoric | 8 for quick menu access to troubleshoot |
| 00:24.01 | golikwid|mac | just 8 |
| 00:24.08 | Letoric | yeah, it's easy ;) |
| 00:24.13 | golikwid|mac | [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|[2-9]xxx|8 |
| 00:24.33 | Letoric | ok, so what does that give us? |
| 00:24.35 | golikwid|mac | opps lol |
| 00:24.39 | golikwid|mac | [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|41xx|5xx|9xxx|8 |
| 00:24.44 | golikwid|mac | idk |
| 00:24.47 | Letoric | lol |
| 00:24.50 | golikwid|mac | but it looks pretty |
| 00:24.58 | golikwid|mac | and whats the worst that could happen |
| 00:25.26 | golikwid|mac | are you aftraid it will stop working...wait it already doesnt |
| 00:25.29 | tessier | Ah...anyone ever had Linux's ip_nat_sip and ip_conntrack_sip mess up their phones? |
| 00:25.45 | tessier | Perhaps specifically in shorewall? |
| 00:25.47 | Letoric | no, not afraid it will stop working, but I generally try to understand the logic behind things |
| 00:25.52 | Letoric | to better myself |
| 00:26.04 | golikwid|mac | ok |
| 00:26.18 | golikwid|mac | well dialplans are kinda not that hard to figure out... |
| 00:26.34 | Letoric | ok, so I'm not that good at them yet |
| 00:26.36 | Letoric | apparently |
| 00:26.51 | golikwid|mac | as soon as your ohone matches something between the | and the next | it sends it |
| 00:27.20 | Letoric | ok, so there are no * in the original dial plan for the phone |
| 00:27.26 | Letoric | so why is it sending that immediately? |
| 00:27.34 | golikwid|mac | because i said so |
| 00:28.12 | golikwid|mac | idk |
| 00:28.18 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 00:28.32 | golikwid|mac | please tell me you have updated the dialplan and restarted the phone |
| 00:28.35 | golikwid|mac | ;) |
| 00:28.43 | golikwid|mac | let me try it on mine |
| 00:28.47 | golikwid|mac | hang on its over there |
| 00:29.20 | *** join/#asterisk hesco (~hesco@c-76-17-99-144.hsd1.ga.comcast.net) |
| 00:29.53 | golikwid|mac | omg im on pins and needles |
| 00:29.56 | hesco | just built a new asterisk server, but its CLI offers me no sip commands. What might I have missed here, please? |
| 00:29.56 | golikwid|mac | did it work |
| 00:30.05 | golikwid|mac | core |
| 00:30.08 | Letoric | haven't done it yet, in process goliwid |
| 00:30.30 | golikwid|mac | hesco: are you trying core |
| 00:32.47 | hesco | I was not, but just did and did not find it there either. |
| 00:33.49 | [TK]D-Fender | hesco: did chan_sip even LOAD? |
| 00:34.20 | hesco | checking now |
| 00:34.30 | golikwid|mac | help core |
| 00:34.58 | hesco | sorry, how exactly would I check for that? |
| 00:35.02 | Letoric | still rebooting phone, it rebooted, thought it needed to update things since sip.cfg had been changed, rebooted again, and is now rebooting again |
| 00:35.06 | Letoric | I don't think it likes it |
| 00:35.34 | [TK]D-Fender | hesco: "module unload chan_sip.so" |
| 00:37.12 | golikwid|mac | hm |
| 00:38.11 | golikwid|mac | Letoric: so now the phone wont start? |
| 00:38.41 | Letoric | yeah, seems unhappy |
| 00:38.44 | Letoric | going back to what I had |
| 00:38.48 | golikwid|mac | hm |
| 00:39.07 | Letoric | maybe I made a typo, who knows |
| 00:39.15 | Letoric | changing a dial plan shouldn't choke the phone |
| 00:39.34 | golikwid|mac | why dont you use the web interface |
| 00:39.44 | golikwid|mac | change just the dialplan there |
| 00:40.08 | Letoric | I'll look at that when it comes back up |
| 00:40.11 | [TK]D-Fender | ~polycomwebconfig |
| 00:40.11 | infobot | [~polycomwebconfig] People configuring Polycom phones via the web interface should be dragged out and shot. Survivors should be shot AGAIN. |
| 00:40.13 | Letoric | still wish you'd look at my dial plan though ;p |
| 00:40.21 | Letoric | you would probably see my error more quickly |
| 00:40.41 | golikwid|mac | people who use web interfaces are hot |
| 00:40.41 | Letoric | I just don't understand how the phone dial plan which is the same on both systems, would cause the issue I'm having |
| 00:41.10 | golikwid|mac | this is technology since when is it supposed to make any sense |
| 00:41.41 | Letoric | heh, it doesn't always make sense, but frequently, logic can be applied if thought through thoroughly |
| 00:41.51 | golikwid|mac | never |
| 00:42.00 | Letoric | sure hope this phone comes back up, it's being stubborn, might have to power it down |
| 00:42.01 | hesco | Thanks [TK]D-Fender. That was the ticket. Would not unload it, but it did successfully load it. Not sure why it did not do so at first. |
| 00:42.05 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 00:42.37 | hesco | Now it is showing the extension, but not as registered, yet. But at least I can access some clues as to why now. |
| 00:42.57 | Letoric | gah |
| 00:42.58 | Letoric | rebooting again |
| 00:45.13 | Letoric | think sip.cfg didn't get overwritten for some reason, replacing it again |
| 00:47.04 | Letoric | ok phone is alive again |
| 00:47.24 | golikwid|mac | woohoo |
| 00:48.03 | golikwid|mac | mouth to earpiece resesitation |
| 00:48.19 | [TK]D-Fender | hesco: Perhaps you should look at what modules DID load and see what's missing. |
| 00:48.30 | pabelanger | you never go mouth to earpiece |
| 00:49.24 | Letoric | applied the new digit map via the web interface |
| 00:49.41 | Letoric | same issue |
| 00:49.46 | golikwid|mac | oh |
| 00:49.53 | golikwid|mac | wow so we wasted alot of time with that |
| 00:50.05 | Letoric | go figure ;) |
| 00:50.12 | golikwid|mac | idk why you didnt start with the asterisk dialplan |
| 00:50.25 | golikwid|mac | ok so moving on |
| 00:50.27 | Letoric | yeah me either! Who would have thought of that! |
| 00:50.35 | golikwid|mac | are all the other phones are in the same context? |
| 00:51.10 | golikwid|mac | have you looked at their dialplans they dont have any dialplan jujitsu or anything right |
| 00:51.34 | Letoric | only 2 phones on the system right now |
| 00:51.40 | Letoric | and yes, they are both in the same context |
| 00:51.40 | golikwid|mac | oh |
| 00:51.48 | Letoric | not moving everything until tomorrow at noon |
| 00:51.56 | golikwid|mac | so its not the context than |
| 00:51.57 | golikwid|mac | is it |
| 00:52.03 | Letoric | it could be.. |
| 00:52.09 | Letoric | the phones on the new system don't work |
| 00:52.11 | Letoric | for dialing * |
| 00:52.21 | Letoric | unless I explicitly add only * to the extension |
| 00:52.22 | golikwid|mac | wait |
| 00:52.25 | jamko | letoric - check features.conf |
| 00:52.44 | jamko | wild guess from left field.. : ) |
| 00:52.57 | Letoric | what about it? |
| 00:53.02 | Letoric | it definitely has things that use * |
| 00:53.12 | jamko | yes it does. |
| 00:53.14 | Letoric | but I can't use them, since as soon as I press *, it sends it to asterisk |
| 00:53.23 | golikwid|mac | hm |
| 00:53.25 | jamko | retarded phone. |
| 00:53.27 | Letoric | and asterisk rejects with 'rejected because extension not found' |
| 00:53.28 | Letoric | heh |
| 00:53.32 | golikwid|mac | try adding them to the dialplan |
| 00:53.49 | golikwid|mac | just for shits and giggles |
| 00:53.55 | golikwid|mac | [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|41xx|5xx|9xxx|8|*xx |
| 00:55.09 | Letoric | gimme a sec, phone is rebooting as it resets to defaults |
| 00:55.46 | golikwid|mac | the problem is definatly the phone |
| 00:55.49 | golikwid|mac | o doubt |
| 00:55.52 | golikwid|mac | no doubt |
| 00:56.02 | golikwid|mac | asterisk cannot process anything that it is not sent |
| 00:56.20 | golikwid|mac | so you need to stop the phone from sending the * |
| 00:56.30 | Letoric | ok |
| 00:56.55 | Letoric | still not logical since it works ok on the old system with same dialplan |
| 00:56.57 | golikwid|mac | unless im wrong |
| 00:57.02 | Letoric | but I'm testing when it comes back up |
| 00:57.03 | golikwid|mac | omg |
| 00:57.04 | golikwid|mac | ogm |
| 00:57.04 | golikwid|mac | omg |
| 00:57.25 | golikwid|mac | sorry my keyboard got stuck |
| 00:57.29 | jamko | yup. phone dial plan.. load the factory default config files. |
| 00:57.39 | jamko | lol |
| 00:58.17 | Letoric | jamko - I'm saying the old system has the same phone dial plan |
| 00:58.20 | Letoric | not the same extensions.conf |
| 00:58.36 | golikwid|mac | omg change the record |
| 00:58.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.72) |
| 00:58.46 | jamko | letoric: i know |
| 00:59.08 | Letoric | dialplan on the phone is updated, it's rebooting now |
| 00:59.09 | jamko | but obviously someone has hacked your phone and altered the dialplan to send * |
| 00:59.13 | Letoric | hehe |
| 00:59.32 | Letoric | I just reset it local and device defaults a few minutes ago, had same issue |
| 00:59.45 | Letoric | now trying with the [2-9]11||1[2-9]xx[2-9]xxxxxx|[2-9]xxxxxx|41xx|5xx|9xxx|8|*xx suggestion that goliwid made |
| 00:59.54 | jamko | maybe your send button is stuck. |
| 01:00.07 | Letoric | as an aside, resetting local and device settings did NOT seem to change the dialplan |
| 01:00.10 | golikwid|mac | that would be an awsome conclusion |
| 01:00.20 | Letoric | still sends * immediately |
| 01:00.24 | golikwid|mac | i hope its coke in the send button |
| 01:00.29 | Letoric | y'all are nuts |
| 01:00.35 | jamko | all the way, im all in on the coke |
| 01:00.38 | Letoric | I am really trying to get this resolved |
| 01:00.38 | golikwid|mac | are you restarting after you chance the dialplan |
| 01:00.41 | Letoric | yeah |
| 01:00.45 | golikwid|mac | hu |
| 01:00.46 | golikwid|mac | hm |
| 01:00.50 | Letoric | the web interface says it will, ut it never does |
| 01:00.53 | Letoric | so I manually did it |
| 01:01.16 | golikwid|mac | hm |
| 01:01.20 | golikwid|mac | ok new approach |
| 01:01.30 | golikwid|mac | change all yoru featurecodes to not use the * |
| 01:01.37 | jamko | yay! features.. |
| 01:01.42 | Letoric | lol |
| 01:01.44 | Letoric | ok, seriously |
| 01:01.47 | Letoric | you're killin me |
| 01:01.47 | golikwid|mac | is so smart |
| 01:01.50 | jamko | or maybe do you have a second phone?? |
| 01:02.14 | Letoric | I have tested with 2 dif phones |
| 01:02.30 | jamko | wow.. and you're going live tomorrow? |
| 01:02.41 | jamko | I can only imagine what happens after you fix this problem... |
| 01:03.17 | golikwid|mac | feature code dialing is so cliche |
| 01:03.19 | golikwid|mac | everyone does it |
| 01:03.26 | golikwid|mac | tell them your phone system is unique |
| 01:03.29 | Letoric | I have backups of the files |
| 01:03.32 | golikwid|mac | and that makes it more valuable |
| 01:03.41 | Letoric | not worried about go live, I'm stable with the one issue |
| 01:03.46 | jamko | LMFAO |
| 01:03.48 | Letoric | just would like to resolve that issue before go live |
| 01:04.38 | jamko | where did you buy the polycoms? |
| 01:04.47 | Letoric | hrm |
| 01:04.54 | Letoric | now it's working on the 'other' test phone |
| 01:05.00 | golikwid|mac | discount bargan used phones superstore why? |
| 01:05.04 | Letoric | let's see how the original test phone comes back after format |
| 01:05.15 | golikwid|mac | hm |
| 01:05.18 | Letoric | nope |
| 01:05.19 | Letoric | it isn't |
| 01:05.20 | Letoric | I lied |
| 01:05.21 | Letoric | lol |
| 01:05.25 | golikwid|mac | liar! |
| 01:05.29 | Letoric | that's the phone I was changing the whole time LMAO |
| 01:05.34 | golikwid|mac | 401+6 |
| 01:05.34 | golikwid|mac | 3. |
| 01:05.35 | Letoric | I was altering it's dial plan |
| 01:05.40 | jamko | omg |
| 01:05.41 | Letoric | not the phone we were using hahaha |
| 01:05.54 | jamko | and i was so sure on the coke. |
| 01:06.00 | jamko | DAMN IT!!!\ |
| 01:06.00 | Letoric | hahaha |
| 01:06.01 | golikwid|mac | sorry spilled some drink on my keyboard and panicked |
| 01:06.10 | Letoric | ok, so now I have to add more ** to the dial plan |
| 01:06.20 | Letoric | since we do *4101 |
| 01:06.27 | golikwid|mac | omg |
| 01:06.34 | golikwid|mac | *4xxx |
| 01:06.50 | Letoric | so what happens if somebody tries to dial 469 |
| 01:06.55 | Letoric | wont' that 4xxx screw me? |
| 01:07.21 | tessier | Now that's odd....my office ip is .173 but asterisk says I am registered from .174. How on earth could that be? |
| 01:07.31 | tessier | I wonder if my phones are telling it that somehow... |
| 01:07.41 | jamko | no |
| 01:07.50 | jamko | thats your gateway |
| 01:08.21 | golikwid|mac | 4[2-5]xxx|4[7-9}xxx| |
| 01:08.23 | golikwid|mac | maybe |
| 01:08.31 | *** join/#asterisk MmixX (MmixX@unaffiliated/mmixx) |
| 01:08.32 | tessier | jamko: You talking to me? |
| 01:08.39 | jamko | i guess i was |
| 01:08.47 | jamko | didn't think so, but yes i was |
| 01:09.16 | *** join/#asterisk coppice (~chatzilla@m121-202-9-120.smartone-vodafone.com) |
| 01:09.28 | tessier | My local network gateway is 192.168.3.1. That is my firewall. It has an external IP of 76.199.182.173 |
| 01:09.37 | tessier | sip show peers in asterisk says the phone registered as 76.199.182.174 |
| 01:09.38 | jamko | tessier: is .174 the gateway for the device holding your public ip? |
| 01:09.46 | *** join/#asterisk turt1e (~pbarros@adsl-92-210-39.asm.bellsouth.net) |
| 01:09.58 | tessier | jamko: Ah, yes it is. |
| 01:10.00 | jamko | usually your gateway is one number above your highest static |
| 01:10.09 | tessier | But why would asterisk ever be seeing that? |
| 01:10.10 | jamko | ahhhh... one in the hole... |
| 01:10.14 | jamko | screw the coke |
| 01:10.31 | jamko | because |
| 01:10.38 | jamko | that is the number broadcast. |
| 01:10.49 | jamko | it is the last number asterisk can see. |
| 01:10.54 | jamko | in frot of you. |
| 01:10.58 | jamko | *front |
| 01:11.31 | tessier | The asterisk server is on a totally different network in a datacenter 30 miles and 10 net hops away. |
| 01:11.43 | tessier | How would it ever know my DSL line broadcast address? |
| 01:11.59 | jamko | because it routes traffic through it. |
| 01:12.06 | jamko | your static block is not "really" public |
| 01:12.24 | jamko | its a group of statics behind your isp router/gateway/modem |
| 01:12.37 | jamko | which is controlled by a config file in your isps router/gateway/modem |
| 01:12.40 | tessier | Are they NATing me out their .174 again?! |
| 01:12.40 | Letoric | is there a proper way to edit sip.cfg? |
| 01:12.49 | golikwid|mac | is his ip causing his phone to dial *? |
| 01:12.49 | Letoric | it seems when I use vi, it complains about line too long |
| 01:13.22 | tessier | jamko: When I go to whatismyip.com it says my IP is 76.199.182.173 |
| 01:13.29 | golikwid|mac | omg |
| 01:13.35 | golikwid|mac | he displayed his ip |
| 01:13.39 | golikwid|mac | omg my heart stopped |
| 01:13.46 | jamko | AAAAAAAAAAAAAAAAAHHHHHHHHHHHHHHHHHHHH~~~~~~~ RUN AWAY!!!!!!!!!!!! |
| 01:13.51 | jamko | LMFAO |
| 01:13.55 | golikwid|mac | well there is only 221 people in here |
| 01:14.01 | golikwid|mac | im sure they are all fine |
| 01:14.19 | jamko | tessier: it's like this |
| 01:14.56 | jamko | your browser traffic is routed differently than your sip traffic, obviously... |
| 01:15.09 | jamko | a tcp stack shows different data than udp |
| 01:15.14 | jamko | sooooo |
| 01:15.26 | jamko | if for example |
| 01:15.58 | jamko | you also had 76.199.182.172 , and you were to ping it from 76.199.182.173 .. |
| 01:16.25 | jamko | that packet should not go any further than your gateway to get to it.. It would never hit an outside DNS request. |
| 01:17.30 | jamko | You are behind a barrier controlled by an isp on a power trip with the bgp protocol. |
| 01:18.32 | Letoric | guys, can anybody tell me how to properly edit the sip.cfg file? When I edit it with VI, it's tanking the file, and the phone |
| 01:19.00 | Letoric | it is truncating the file |
| 01:19.29 | jamko | tessier: however, normally I see the static ip in asterisk, not the gateway .. :) |
| 01:20.14 | jamko | letoric: don't use vi |
| 01:20.32 | jamko | letoric: use something with a gui for right now (gasp) |
| 01:20.51 | jamko | download it to a windows machine or something (HUGE GASP) |
| 01:20.54 | Letoric | well, when I use Windows based text editors they screw up the file for unix |
| 01:20.55 | Letoric | heh |
| 01:21.00 | Letoric | I guess I can try notepad |
| 01:21.05 | Letoric | I know wordpad screws it up |
| 01:21.28 | jamko | notepad fo sho |
| 01:21.42 | jamko | or maybe try nano or pico... |
| 01:22.03 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 01:23.04 | *** join/#asterisk russ (~russ@206.29.188.230) |
| 01:24.07 | Letoric | looks like notepad did it ok |
| 01:24.12 | Letoric | weird, VI is such a powerful editor :/ |
| 01:24.24 | golikwid|mac | shutters...windows ewww |
| 01:24.45 | Letoric | I'm native Windows guy, working with *nix as secondary |
| 01:24.53 | Letoric | ceo is a hardcore solaris guy for 15+ years |
| 01:25.59 | jamko | vi is a pain in the ass.. (ssshhh) |
| 01:26.30 | drmessano | Anyone recommend any free trade publications for Telephony? Would like something in print to go with my normal "bathroom reading" |
| 01:26.48 | Letoric | golikwid, you were right, but it still makes no sense ;) |
| 01:26.56 | Letoric | thanks you two, for the help |
| 01:27.14 | jamko | awesome. |
| 01:27.19 | golikwid|mac | wait |
| 01:27.22 | golikwid|mac | what |
| 01:27.25 | golikwid|mac | i was right! |
| 01:27.26 | golikwid|mac | yes |
| 01:27.30 | jamko | golikwid: you go girl!! |
| 01:27.39 | golikwid|mac | hang on let me screen capture this and print it out |
| 01:27.43 | golikwid|mac | girll? |
| 01:27.44 | golikwid|mac | hm |
| 01:27.55 | drmessano | Just because he's a Mac user doesn't mean you need to call him a "girl" |
| 01:27.57 | golikwid|mac | im not super psyciced about that |
| 01:27.58 | drmessano | Be nice, peeps |
| 01:28.45 | Letoric | hehe |
| 01:28.45 | jamko | yes dr.. dr. dr. dr.. |
| 01:29.19 | jamko | ooooooooo sick burn... |
| 01:29.50 | jamko | whos this pickle f*cker? |
| 01:35.05 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-252-100-238.stny.res.rr.com) |
| 01:35.36 | *** join/#asterisk russ (~russ@206.29.188.230) |
| 01:36.25 | Letoric | welp, I finally get to go home, thanks again folks. Have a great night |
| 01:46.25 | golikwid|mac | http://www.ronpaul.com/ |
| 02:07.05 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 02:07.30 | *** join/#asterisk rue_mohr (~rue@24.207.119.38) |
| 02:07.44 | rue_mohr | so we installed our first aastralink pro 160 today |
| 02:08.25 | rue_mohr | had to rewire the office, so I have to go back tommorow for touch ups and trainign |
| 02:10.19 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 02:17.13 | *** join/#asterisk b0gatyr (~b0gatyr@adsl-10-92-8.mia.bellsouth.net) |
| 02:23.52 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 02:29.45 | raden | WTF would you want to run a astra link ? |
| 02:31.42 | tessier | ah-hah |
| 02:31.56 | tessier | http://pastebin.ca/1920752 Some of my phones are using random ports as expected. |
| 02:32.10 | tessier | Some are stuck using 5060 which of course causes conflict. |
| 02:32.47 | ChannelZ | not if theyre on different ips |
| 02:33.24 | tessier | ChannelZ: I don't want to assign a dozen IPs to the firewall and map them all to phones. I want the phones to just use a random source IP like they usually do. |
| 02:33.44 | tessier | Some of the phones are doing so. I need to figure out why these others (same kind of phone) are not... |
| 02:35.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 02:36.15 | *** join/#asterisk Jouva (jouva@fluffy.moufette.com) |
| 02:39.02 | Jouva | So I'm new to asterisk. Getting the hang of it, but stuck in 2 places. #1 I'm trying to get a "please wait while I try that extension" sound played after a WaitExten() and not sure how to go about that. #2 I setup code in exten "t" to loop back and ask for an extention again, but that doesn't seem to take place at all. |
| 02:39.28 | golikwid|mac | playback |
| 02:39.42 | Jouva | Yes but where? :) |
| 02:39.43 | ChannelZ | Need to see your dialplan and probably some console output |
| 02:39.45 | ChannelZ | ~pb |
| 02:39.47 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 02:39.54 | Jouva | sure |
| 02:40.31 | ChannelZ | WaitExten just pauses and hopes to match an extension in the same context as the caller mashes buttons |
| 02:40.45 | Jouva | This is 1.4.21 apparently, cause debian loves to live in the past ;) |
| 02:40.57 | ChannelZ | So maybe your extension patterns are wrong, or something else is going on. |
| 02:41.00 | Jouva | ChannelZ: Right. That part works. As does invalid extentions with exten i |
| 02:41.58 | ChannelZ | Oh, I misread. |
| 02:42.16 | ChannelZ | I thought you were GETTING a 'please wait' and nothing was happening, I see... |
| 02:44.11 | ChannelZ | Anyway yes you'll have to Playback the desired sound on each extension before it Dials, then. |
| 02:44.41 | Jouva | Ahh ok |
| 02:45.03 | Jouva | oh derp that would make too much sense haha |
| 02:45.18 | ChannelZ | As for your 't' not working, hafta see how your dialplan is constructed |
| 02:45.49 | ChannelZ | Are you doing WaitExten(3) or something? |
| 02:46.02 | Jouva | 10 for now, but yes essentially |
| 02:46.30 | ChannelZ | ok |
| 02:46.30 | Jouva | oh I did find issues |
| 02:47.01 | ChannelZ | So what does it do? Just sit there forever doing nothing, or..? |
| 02:47.28 | Jouva | How can I Goto() another extention with a labeled line? |
| 02:47.54 | Jouva | Cause in 't', I want to jump to a specific line and not the first one, but I don't want to give everything a priority number |
| 02:48.16 | ChannelZ | use a priority label where you want to jump to.. |
| 02:48.17 | Jouva | but it's back in s, so it can try and wait for an extention again but actually time out after 3 times |
| 02:48.35 | Jouva | Right. I have one, but how do I write the Goto() line? |
| 02:48.40 | ChannelZ | like exten => 444,n(fart),Whatever |
| 02:48.50 | Jouva | Cause I can't use the label itself since it's in another extention |
| 02:49.10 | ChannelZ | Then you can Goto(context,444,fart) |
| 02:49.50 | ChannelZ | or just Goto(fart) if it's in the same context and extension |
| 02:49.58 | Jouva | Ahhh cause this documentation seemed to mention that format was a "priority" and separately showed a named priority |
| 02:50.40 | ChannelZ | The priority still exists, whether you number it or use 'n' as I did (for 'next') but you attach a label to it |
| 02:51.27 | ChannelZ | so you could Goto(s,whatever) if 'whatever' was your priority label in extension 's' of the same context |
| 02:52.42 | ChannelZ | core show application Goto |
| 02:53.18 | *** join/#asterisk russ (~russ@206.29.188.182) |
| 02:53.21 | ChannelZ | It can take from 1 to 3 arguments depending on how specific you need to be. Required going right-to-left |
| 02:53.52 | *** join/#asterisk adolfomaltez (~taro@190.87.100.41) |
| 02:56.05 | Jouva | http://pastebin.ca/1920760 <-- dialplan |
| 02:56.15 | *** join/#asterisk seanjohn (~admin@173.50.101.14) |
| 02:57.09 | ChannelZ | You have to number the first priorty 1, you can't start with n |
| 02:57.22 | ChannelZ | That's why your 't' doesn't work |
| 02:57.28 | Jouva | gahhhhhh I didn't realize I removed the 1 |
| 02:57.39 | seanjohn | channelz means that you have to start with a number as the priority unless you make bookmarks |
| 02:58.04 | ChannelZ | well yes but you can't call them ALL 'n' |
| 02:58.35 | ChannelZ | I'm not even sure what that does. Does it just not load that extension? ('dialplan show' should say) |
| 02:58.37 | seanjohn | exten => s,1, could be called by context,exten and exten => s,n(bookmark) could be started by context,exten,bookmark |
| 02:58.43 | Jouva | I know what he meant ;) I just didn't realize when I was re-working the timeout exten that I forgot to mark the first one as priority "1" :P |
| 02:58.48 | Jouva | Cause I rewrote it :P |
| 03:00.02 | seanjohn | for security, none of mine are starting points with 1 except for the default context |
| 03:00.06 | Jouva | nope |
| 03:00.08 | Jouva | that's not it either |
| 03:00.15 | Jouva | I restarted asterisk even |
| 03:00.23 | seanjohn | what's wrong jouva? |
| 03:00.25 | Jouva | <PROTECTED> |
| 03:00.34 | seanjohn | the line above that |
| 03:00.37 | ChannelZ | Interesting. It just numbers it starting from the priorty of the previously parsed extension. |
| 03:00.52 | seanjohn | it numbers it consecutively, channelz |
| 03:00.56 | Jouva | <PROTECTED> |
| 03:01.02 | Jouva | was just before that |
| 03:01.07 | ChannelZ | I believe that's what I said. |
| 03:01.20 | Jouva | (well there was one unrelated line... another client pinging the server) |
| 03:01.28 | seanjohn | that's something to do with how you compiled it; I have had errors and bugs on anything but waitexten |
| 03:01.44 | seanjohn | make sure you "core set verbose 6" |
| 03:01.59 | seanjohn | try it again |
| 03:02.18 | seanjohn | are you around Fender? |
| 03:03.07 | seanjohn | Jouva, you don't have to do waitexten, try background and using silence files |
| 03:03.16 | jamko | tessier: they map to random ports because of your misconfigured NAT and sip.conf.. You need to specify port=5061 ; port=5062 etc for each peer in sip.conf, and then setup your nat to forward traffic for each of those ports to the respective UA which needs to have that port specified in it's configuration. This is the RIGHT way to do it. Anything else will get you ill results. |
| 03:03.48 | jamko | And then once that fun is done, you can start separating the RTP ports for each UA as well. |
| 03:04.06 | Jouva | ...jamko, I was just gonna ask a question and you might have already given me the answer haha |
| 03:04.22 | seanjohn | I have a question I don't know if digium missed or what. For all log files, except queue_log and even_log, you can specify them right? or is there a way to specify them? |
| 03:04.35 | jamko | My other car is a crystal ball : ) |
| 03:04.54 | Jouva | My asterisk server is behind my NAT device, has several ports open (I'll go look throught he list again). Asterisk connects as friend to gizmo5. Google Voice redirects its calls to gizmo5. Anything going to an extension (which are all behind the same NAT device) gets one way audio |
| 03:05.58 | tessier | jamko: The phone manual says: When symmetric UDP is enabled, the IP phone |
| 03:06.01 | tessier | generates and listens for UDP messages using port 5060.If symmetric |
| 03:06.04 | tessier | UDP signaling is disabled, the phone sends from random ports but it |
| 03:06.06 | tessier | listens on the configured SIP local port. |
| 03:06.21 | ChannelZ | Jouva: Do a 'dialplan show' and make sure your 't' extension is showing up |
| 03:06.28 | Jouva | ChannelZ: sure |
| 03:06.53 | Jouva | yes it is |
| 03:07.19 | jamko | tessier: this is not a phone issue. This is a SIP and Asterisk issue. |
| 03:07.30 | ChannelZ | That's strange |
| 03:07.32 | jamko | If you were using IAX it would be a different story. |
| 03:07.40 | jamko | but you are using SIP, no? |
| 03:08.49 | tessier | I am using SIP |
| 03:09.14 | jamko | tessier, your phone manual would make sense if you were using a public ipv4 address for every phone. |
| 03:09.24 | jamko | maybe |
| 03:09.57 | tessier | There would be no need to disable symmetric UDP if they all had public ipv4 addresses |
| 03:10.02 | jamko | but how can you expect your NAT to let traffic through and direct to devices behind it, through ports which are not open? |
| 03:10.24 | tessier | It is stateful and tracks sessions subject to a timeout. |
| 03:10.31 | Jouva | hmmm |
| 03:10.45 | jamko | ok, and what about when Asterisk wants in from the outside and there is no stateful session? |
| 03:10.47 | tessier | In the case of UDP session is defined somewhat more loosely than with TCP. |
| 03:10.48 | Jouva | ChannelZ |
| 03:11.04 | Jouva | extension 't' IS correct for timeouts on WaitExten() right? |
| 03:11.05 | ChannelZ | yah |
| 03:11.06 | tessier | jamko: The phones reregister periodically to keep the channel open. |
| 03:11.21 | jamko | stateful will not work in your scenario, period. This is why you are having problems. |
| 03:11.22 | seanjohn | tressier, you can't use asterisk behind a NAT without UNCONDITIONAL port forwarding of 5060:5064 and 10000:20000, all UDP |
| 03:11.48 | ChannelZ | Jouva: I have pretty much the same setup, it works fine |
| 03:11.51 | jamko | You need to get a more controlled setup. You are having a free for all and that won |
| 03:11.52 | Jouva | My only guess is that maybe this feature was not in this version? Was it put in after 1.4.21? |
| 03:11.53 | jamko | wont work |
| 03:11.57 | tessier | seanjohn: I have forwarded that for the astersik server itself. |
| 03:12.03 | seanjohn | for iax2, which is better for nat traversal, you have to foward 4569 |
| 03:12.08 | ChannelZ | No the 't' extension has been around for a long time |
| 03:12.11 | tessier | seanjohn: I am talking phones behind NAT. |
| 03:12.27 | seanjohn | from what I understand, you're having problems too |
| 03:12.30 | tessier | seanjohn: I have phones in other locations which are working perfectly. |
| 03:12.37 | jamko | seanjohn, he has asterisk behind one NAT, and the phones behind another. |
| 03:12.42 | tessier | seanjohn: I am having problems with this one particular office. |
| 03:12.57 | seanjohn | jamko, he hasn't forwarded ports at one of the locations |
| 03:12.59 | ChannelZ | Jouva: try removing 'priorityjumping=no' or commenting it out for the hell of it. Maybe there's some bug in that particular version, I have no idea. |
| 03:13.01 | tessier | I have a one to one nat with the public to private IP directly mapped and the appropriate ports forwarded. |
| 03:13.21 | tessier | I haven't forwarded ports at any of the locations except into the asterisk server itself. The other locations all work fine. |
| 03:13.30 | jamko | tessier, for one device that's fine |
| 03:13.31 | tessier | sip.conf has nat=1 etc. |
| 03:13.41 | Jouva | I should be listening in on this convo about the nat stuff too |
| 03:13.44 | jamko | unfortunatley you have more than one device at the location which is not working. |
| 03:13.51 | jamko | and you need to lock down your setup. |
| 03:13.52 | tessier | jamko: For many devices. I have several locations with two devices and on a different phone system I have nearly 30. All behind the same NAT. |
| 03:14.20 | seanjohn | tessier, to use port 5060 on an ip, that 5060 can only be forwarded to one device. Each device from then on must forward to 5061, 5062, and so on |
| 03:14.32 | Jouva | ChannelZ: A BUG? In an OLD version that Debian uses? Noooooooo... ;) |
| 03:14.36 | jamko | finally got an AMEN |
| 03:14.45 | tessier | seanjohn: Most of my phones are originating their SIP sessions on a random IP. Those phones work. |
| 03:15.06 | jamko | tessier, you are experiencing a small miracle at those other sites. |
| 03:15.18 | seanjohn | in fact, EACH physical port has to have its own port number to itself, which means if your device has two FXO ports on it, you must use 5060 and 5061 for that one device |
| 03:15.19 | tessier | http://pastebin.ca/1920752 for example |
| 03:15.21 | ChannelZ | Jouva: Up until a few months ago I ran 1.4.2 also |
| 03:15.35 | tessier | jamko: I've been getting lucky every time for nearly 7 years? |
| 03:15.42 | jamko | tessier, you are killing yourself over something that has a system in place to control. |
| 03:15.56 | seanjohn | to have the adapters behind the nat, you must foward a UDP port for every FXO port |
| 03:15.59 | jamko | apparently you have been, because your config is wrong, period. |
| 03:16.52 | seanjohn | the asterisk is fine behind a nat with one public ip |
| 03:17.02 | jamko | lets just say your setup would work... Are the firmware versions all the same on every firewall? I don't believe you are even using identical firewalls at each location. |
| 03:17.12 | seanjohn | the phones can't receive a ringing notify by sip or any origination from asterisk |
| 03:17.22 | tessier | jamko: Linux/Shorewall at every location. Not all necessarily the same kernel/shorewall version. |
| 03:17.38 | jamko | well that could be your problem, if your problem wasn't an incorrect configuation. |
| 03:18.09 | tessier | seanjohn: The phone opens a hole in the firewall from a random numbered port to port 5060 and keeps it alive via qualify= in sip.conf. Replies go back the same way. |
| 03:18.25 | seanjohn | jamko: instead of debating with him, show him by helping him fix the setup and then he'll realize what was wrong |
| 03:18.41 | seanjohn | how many FXO ports do you have at this location, tessler? |
| 03:18.48 | seanjohn | the one having problems? |
| 03:18.48 | jamko | Mark Spencer wrote the IAX protocol because SIP does NOT work smoothly over NAT, and all the extra configuration that goes into it. If your config was correct, IAX would have been a complete farse. |
| 03:19.10 | tessier | seanjohn: FXO like pots? None. SIP phones? 7 |
| 03:19.19 | jamko | seanjohn: I did show him the setup. |
| 03:19.25 | jamko | seanjohn, refer to my book above. |
| 03:19.31 | seanjohn | how many configured accounts do you have on each phone? |
| 03:19.38 | tessier | seanjohn: One |
| 03:19.51 | seanjohn | all of these phones are behind the same public ip? |
| 03:19.54 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-43-239.mia.bellsouth.net) |
| 03:19.54 | Jouva | Oh hey here's a weird question... why is it I'm seeing "Really destroying... @10.x.x.x" when this network isn't a 10.x.x.x based network? |
| 03:20.02 | tessier | seanjohn: Yes |
| 03:20.26 | seanjohn | forward udp/5060 to udp/5066, one to each phone |
| 03:20.32 | *** join/#asterisk Mhaddog__ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 03:20.38 | jamko | seanjohn: he won't do it. |
| 03:21.22 | tessier | seanjohn: So I have to give all of the phones static IPs also then right? |
| 03:21.27 | seanjohn | forward 10000:11000 to one phone, 11000:12000 to another phone, and so on |
| 03:21.30 | ChannelZ | Jouva: whatever SIP packet it's destroying came FROM a device with 10.x.x.x probably? |
| 03:21.41 | seanjohn | set the RTP ports for the ones you forwarded on that phone |
| 03:21.56 | jamko | Did I not say that with: You need to specify port=5061 ; port=5062 etc for each peer in sip.conf, and then setup your nat to forward traffic for each of those ports to the respective UA which needs to have that port specified in it's configuration. |
| 03:22.05 | Jouva | just seems odd that it'd still come through as 10.x.x.x |
| 03:22.07 | ChannelZ | Jouva: pastebin your 'dialplan show' if your 't' is still not working |
| 03:22.23 | seanjohn | jamko, i'm getting the phones and that part set, not asterisk yet |
| 03:22.30 | seanjohn | his firewall on that end |
| 03:22.35 | jamko | seanjohn, read the entire post.. |
| 03:22.59 | seanjohn | you need to do more than port=5061, you need to set the rtp in asterisk for each peer |
| 03:23.03 | jamko | at the end, I clearly state the nat needs to be forwarded to the respective UA based on the sip port assigned to it. |
| 03:23.44 | seanjohn | whatever rtp range you forwarded for the phones is what you'll put in sip.conf under that peer's section |
| 03:24.22 | jamko | jouva: yes, weird |
| 03:25.36 | jamko | seanjohn: Yes I am aware of the RTP, as stated earlier in this conversation. |
| 03:26.23 | seanjohn | the last statement from me was for tessier |
| 03:26.58 | tessier | seanjohn: Ok, I've mapped all of that to one phone. Now I need to set the phone and asterisk to use those ports. |
| 03:27.49 | seanjohn | "all of that" to one phone or each phone. Each phone only really needs 10 rtp ports (conferencing and 3way) |
| 03:28.39 | tessier | right |
| 03:28.49 | seanjohn | phone1 should have these ports: 5060,10000:10010, phone2=5061,10011:10020 |
| 03:28.53 | tessier | all of that to each phone |
| 03:29.19 | Jouva | seanjohn: how do you do the RTP setup per phone? |
| 03:29.30 | seanjohn | its in the config of the phone Jouva |
| 03:29.36 | seanjohn | and in sip.conf |
| 03:29.40 | russ | hmmm |
| 03:29.42 | Jouva | err I mean in sip.conf |
| 03:29.55 | seanjohn | its not just sip.conf, the phone has to be configured too |
| 03:30.17 | Jouva | I'm aware ;) |
| 03:30.19 | Jouva | I can do that part |
| 03:30.25 | russ | teliax isn't sending me dnid information |
| 03:30.26 | Jouva | I just am not sure how to set it up in sip.conf |
| 03:30.39 | seanjohn | however, the phone could be left at default but asterisk only needs to send 10 ports to each phone and not try to send the same port to more than one phone |
| 03:30.59 | russ | I have the DNIS box checked... |
| 03:31.21 | seanjohn | with the phone at default, it will accept 10000:20000 but the range forwaded through th nat needs to be in sip.conf for that extension |
| 03:31.41 | Jouva | seanjohn: And how do I set this in sip.conf for example? |
| 03:33.53 | seanjohn | i forgot; its the same that you used to set the default range in asterisk |
| 03:34.01 | seanjohn | i'm trying to find it |
| 03:34.33 | Jouva | ohhh so rtpstart and rtpend? |
| 03:34.46 | tessier | seanjohn: How do I set the rtp port range per client in sip.conf? |
| 03:34.53 | ChannelZ | that's not in sip.conf |
| 03:34.57 | tessier | I presume SIP port is just port= |
| 03:35.03 | Jouva | right |
| 03:35.16 | tessier | Can I do per-client configs in rtp.conf? |
| 03:35.20 | ChannelZ | The phone requests what RTP port it wants * to send its stream to |
| 03:35.28 | tessier | ah |
| 03:35.35 | tessier | So I don't have to do anything regarding that in asterisk. That makes sense. |
| 03:35.44 | ChannelZ | * requests of the phone what port to send its stream to * based on rtp.conf |
| 03:36.19 | *** join/#asterisk Whtsup (~sssi@WimaxUser3760-62.wateen.net) |
| 03:36.21 | Whtsup | hello |
| 03:36.28 | Whtsup | make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' |
| 03:36.28 | Whtsup | make: *** [all] Error 2 |
| 03:36.38 | Whtsup | i m getting this error when i m compiling dahdi |
| 03:36.40 | Whtsup | :S |
| 03:36.41 | ChannelZ | the port in sip.conf is for telling * what port of the phone to send SIP messages to. |
| 03:36.58 | Whtsup | You do not appear to have the sources for the 2.6.28.7 kernel installed. |
| 03:36.58 | Whtsup | make[1]: *** [modules] Error 1 |
| 03:37.15 | pabelanger | Whtsup: Then install the kernel source |
| 03:37.16 | Whtsup | getting this error when im installing dahdi |
| 03:37.20 | ChannelZ | Whtsup: It just said: You do not appear to have the sources for the kernel installed |
| 03:37.31 | Whtsup | which source |
| 03:37.32 | Whtsup | i need |
| 03:37.34 | Whtsup | i dont know |
| 03:37.40 | Whtsup | newbie |
| 03:37.41 | pabelanger | The Linux kernel |
| 03:37.44 | seanjohn | the other way is to set nat=yes for the peer and the client behind the nat should send a packet out to open up the nat port |
| 03:37.55 | ChannelZ | linux-headers, kernel-headers, who knows... depends on your distro if you're using packages |
| 03:38.09 | seanjohn | with setting the rtp ports, its fail safe and the firewall cant get in the way of the client |
| 03:38.09 | Whtsup | i m using centos |
| 03:38.24 | p3nguin | (1821.27) <raden> I can get 30 - 35 MB burst sometimes <-- negatory on the 35 MegaBytes. |
| 03:38.41 | pabelanger | $ yum install kernel-devel |
| 03:39.26 | seanjohn | rtpstart=10000 ; first port to use |
| 03:39.26 | seanjohn | rtpend=10100 |
| 03:39.41 | seanjohn | put those under the extensions section in sip.conf |
| 03:40.28 | raden | p3nguin, why negatory ? |
| 03:40.37 | Whtsup | already installed kernel-devel |
| 03:40.41 | Jouva | ChannelZ: Riddle me this... asterisk server (.101 SIP port 5060) behind NAT. Soft phone (.100 SIP port 5061) also behind same NAT. Asterisk server connects to Gizmo5. Google Voice forwards calls to the Gizmo5 account. Ports TCP & UDP 5060, 2000, 2727, 4520, 4569, and range 30000-31000 (defined in rtp.conf) forwarded to .101, 5061 to .100 |
| 03:40.50 | p3nguin | Ultra60 provides 60 Megabit per second rates. That's 7.5 MB/s file transfer rate. |
| 03:40.56 | seanjohn | for example, [201] rtpstart=100000 rtpend=10010 [202] rtpstart=10011 rtpend=10020 |
| 03:41.11 | seanjohn | * each space is a new line |
| 03:41.30 | Jouva | Calls onto GV that go through gizmo5 onto my network that reach my extension at .100 have one way audio |
| 03:41.38 | pabelanger | seanjohn: rtpstart and rtpend don't belong in sip.conf |
| 03:41.49 | Jouva | Yes the ports are overkill but I figured for now, overkill is better than not enough |
| 03:41.49 | raden | p3nguin, I mean like advertised speed in |
| 03:42.04 | p3nguin | Charter MAX provides better rates than your speedtest illustrated. |
| 03:42.04 | raden | best kbps i have ever had was 4300 |
| 03:42.08 | pabelanger | they are not peer specific. rtp.conf |
| 03:42.18 | seanjohn | but they will work when used to define for each extension. You should put rtpstart= and rtpend= in rtp.conf for the DEFAULT |
| 03:42.22 | tessier | seanjohn: Ok, I have it set up as you described. Something is still amiss. |
| 03:42.23 | p3nguin | MAX is 25 Mbps service. |
| 03:42.34 | seanjohn | once you set the default, you can set individual |
| 03:42.35 | tessier | seanjohn: The first phone I have setup is using 5060 for SIP |
| 03:42.36 | p3nguin | half the price of Ultra60. |
| 03:42.43 | tessier | And 10001-11000 for rtp |
| 03:42.46 | *** join/#asterisk GameGamer43|Mac (~GameGamer@65.27.76.78) |
| 03:42.50 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:42.58 | ChannelZ | Jouva: one way audio which way? |
| 03:43.04 | seanjohn | tessier, that's too many ports but ok |
| 03:43.17 | seanjohn | tessier, try nat=route or nat=yes in sip.conf |
| 03:43.20 | pabelanger | tessier: PB your sip.conf file |
| 03:43.26 | raden | p3nguin, i pay $29.95 a month locked in for 2 years so i dont complain |
| 03:43.29 | Jouva | ChannelZ: The call from a landline to GV that gets forwarded does not hear anything. |
| 03:43.37 | raden | i was going to bed happy with 1.5 MBPS dsl for that same price |
| 03:43.42 | tessier | seanjohn: Already have nat=yes in sip.conf |
| 03:43.46 | raden | and my wifi 1 mb / 1 mb ran 34 |
| 03:43.51 | ChannelZ | Jouva: so your outgoing (from asterisk) doesn't work |
| 03:44.01 | Jouva | right |
| 03:44.03 | p3nguin | Yeah, you can't get Ultra60 for that cheap. |
| 03:44.22 | Jouva | HOWEVER |
| 03:44.33 | tessier | pabelanger: http://pastebin.ca/1920785 |
| 03:44.40 | Jouva | The Background() and Playback() all work |
| 03:44.43 | raden | p3nguin, it was a customer switch over new service special with contract |
| 03:44.49 | raden | after 2 years it $69.95 |
| 03:44.56 | raden | but for 2 years im happy ;) |
| 03:45.00 | Jouva | Maybe my softphone is not configured properly |
| 03:45.04 | p3nguin | They lied to you. You have MAX, not Ultra60. |
| 03:45.29 | p3nguin | Which is still not bad. |
| 03:45.30 | seanjohn | find the first phone's section in sip.conf and set nat=yes rtpstart=10001 rtpend=10011 |
| 03:45.37 | Jouva | seanjohn |
| 03:45.53 | Jouva | FYI I have yet to read anything on the net about using rtpstart and rtpend in sip.conf |
| 03:46.00 | Jouva | And many people in here keep saying "no it doesn't go in there" |
| 03:46.10 | seanjohn | it will work in there jouva |
| 03:46.15 | tessier | It can't even register so my problem isn't rtp yet... |
| 03:46.38 | Jouva | ummm |
| 03:46.40 | pabelanger | And rtpstart and rtpend don't belong in sip.conf, they only exist in rtp.conf |
| 03:46.47 | Jouva | seanjohn |
| 03:46.53 | seanjohn | the default goes in rtp.conf and to change for each extension put it in sip.conf under the extension |
| 03:46.54 | tessier | I still need nat=yes even though I am mapping the ports in the device and firewall etc? |
| 03:47.02 | Jouva | The O'Reilly book only makes ONE mentioning of rtpstart |
| 03:47.04 | raden | p3nguin, LOL still a good price i think :) |
| 03:47.07 | Jouva | and it's in rtp.conf |
| 03:47.13 | seanjohn | tessier: keeping nat=yes doesn't make a difference for something on the same network |
| 03:47.28 | p3nguin | Yeah, you're getting it for half price. It's hard to beat that. |
| 03:47.30 | raden | whats the issue ? |
| 03:47.45 | seanjohn | he has asterisk behind a nat and the phones behind another nat |
| 03:47.46 | pabelanger | seanjohn: That is incorrect |
| 03:47.49 | tessier | seanjohn: These aren't on the same network. |
| 03:48.07 | seanjohn | i know that |
| 03:48.17 | seanjohn | leaving nat=yes won't hurt anything |
| 03:48.23 | tessier | ok |
| 03:48.35 | Jouva | seanjohn |
| 03:48.54 | Jouva | Can you show us a valid example posted somewhere that shows using rtpstart in sip.conf? |
| 03:48.59 | Jouva | Cause I can't find one |
| 03:49.02 | pabelanger | tessier: What is your problem again? |
| 03:49.10 | seanjohn | Jouva, my own configuration works |
| 03:49.13 | Jouva | seanjohn |
| 03:49.15 | Jouva | Can you show us a valid example posted somewhere that shows using rtpstart in sip.conf? |
| 03:49.17 | Jouva | Cause I can't find one |
| 03:49.25 | ChannelZ | There's no need to set an rtp range in sip.conf, even if you could, which you can't |
| 03:49.33 | seanjohn | Jouva, just try it. |
| 03:49.37 | Jouva | Once again |
| 03:49.38 | Jouva | seanjohn |
| 03:49.40 | Jouva | Can you show us a valid example posted somewhere that shows using rtpstart in sip.conf? |
| 03:49.42 | Jouva | Cause I can't find one |
| 03:50.17 | seanjohn | mod_sip.so is linked to read the two config files, which means one config from one file will work in the other file |
| 03:50.54 | seanjohn | this is how I found out how to do it; just try it. would you wait for someone else to show you before figuring out how to do something? |
| 03:51.28 | pabelanger | Jouva: It does no exist |
| 03:51.40 | Jouva | that's what I'm trying to imply ;) |
| 03:52.01 | seanjohn | ok, so because it doesn't exist, it won't work? horse shit |
| 03:52.14 | ChannelZ | hahahahaahah |
| 03:52.22 | pabelanger | seanjohn: $ grep rtpstart channels/chan_sip.c |
| 03:52.35 | Jouva | seanjohn |
| 03:52.56 | tessier | pabelanger: The problem is that we recently moved our asterisk box from the local office with 7 local phones and about 7 remote phones where everything worked fine to the datacenter where it would have better bandwidth. |
| 03:53.01 | Jouva | How about I just put banana=yummy in my config? if it doesn't exist it'd have just as much chance as working |
| 03:53.19 | tessier | pabelanger: At the office it was behind a one to one nat. Local phones worked and remote phones worked. |
| 03:53.55 | tessier | pabelanger: After the move the remote phones continue to work perfectly. But now the phones at the office do not work properly. Only one seems to be able to work at a time. |
| 03:54.28 | seanjohn | chan_rtp.so does not exist |
| 03:54.49 | seanjohn | it is from the same module, chan_sip.so |
| 03:54.49 | pabelanger | tessier: Ok, well I'm about to log for the night, do you have SIP trace you can PB of the not working phones? |
| 03:55.00 | tessier | pabelanger: Clearly some sort of NAT issue although I'm not sure what. Everyone is telling me to forward a bunch of ports to each individual phone. I've had this same setup working before and didn't have to do that as long as the phones used randomized source sip ports and qualify= to hold the nat hole open. But right now I am giving it a try their way. |
| 03:55.03 | ChannelZ | I set my rtprange to 8000-9000 in rtp.conf. I (supposedly) set a peer to use 8100-8150. Sip debug says it chose port 8606. |
| 03:55.32 | tessier | pabelanger: Yes. It seems what is happening is that the phones are sending the invite to asterisk, it asks for auth, the phone never gets the auth. Just a sec while I PB.... |
| 03:56.29 | seanjohn | Jouva, learn how to program, not just use what other people program. Maybe then you'll understand whether one config would possible work, documented or not, in another config. If the same module reads two different files, the variables get set and it doesn't matter which file it comes from |
| 03:56.40 | Jouva | ... |
| 03:56.41 | Jouva | wow |
| 03:56.45 | Jouva | just |
| 03:56.46 | Jouva | wow |
| 03:56.49 | Jouva | Yes that's right |
| 03:56.55 | pabelanger | tessier: If you are getting one-way audio, then yes it could be an RTP issue, however lets see a SIP trace first |
| 03:57.07 | Jouva | Cause you know, one module will parse two separate files the exact same way |
| 03:57.15 | ChannelZ | seanjohn, I just tried your config and it doesn't work. Or maybe my unicorn died |
| 03:57.15 | tessier | pabelanger: No one way audio. Just no registration at all. Working on that pb... |
| 03:57.18 | Jouva | Can't have two sets of rules for 2 sets of files in a single module! |
| 03:57.29 | pabelanger | tessier: Then your issue is not RTP :) |
| 03:57.36 | Jouva | seanjohn |
| 03:57.43 | Jouva | Pastebin your sip.conf |
| 03:57.51 | tessier | pabelanger: I didn't mean to say it was. I thought I said NAT...sorry if I mistyped |
| 03:57.52 | pabelanger | BTW: where is the source for mod_sip.so? |
| 03:58.03 | Jouva | http://svn.digium.com/svn/asterisk/branches/1.2/channels/chan_sip.c |
| 03:58.06 | seanjohn | stop being a troll jouva and just try it yourself instead of being ignorant |
| 03:58.24 | *** join/#asterisk MrHanMan (~MrHanMan@c-75-64-49-164.hsd1.ms.comcast.net) |
| 03:58.32 | ChannelZ | I guess my voice stopped working. *I* tried it. It didn't do anything. |
| 03:58.33 | Jouva | seanjohn you have multiple people in the channel saying you're wrong |
| 03:58.37 | seanjohn | you act like i'm saying you can take things from iax.conf and put them in sip.conf |
| 03:58.38 | Jouva | I'm trying to put humor behind it |
| 03:58.44 | Jouva | No |
| 03:58.52 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 03:59.09 | tessier | pabelanger: http://pastebin.ca/1920791 Here ya go |
| 03:59.17 | Jouva | I'm saying that chan_sip.so MIGHT just have specifif options for rtp.conf AND specific options for sip.conf |
| 03:59.24 | Jouva | specific even |
| 03:59.26 | Corydon76-dig | yawns |
| 03:59.29 | seanjohn | ok, explain this. If chan_rtp.c doesn't exist and it only deals with sip, WHERE does it come from |
| 03:59.40 | MrHanMan | Has anyone gotten a Cisco 7925g wifi phone to work with Asterisk? |
| 03:59.49 | Jouva | seanjohn: I'm saying that chan_sip.so MIGHT just have specific options for rtp.conf AND specific options for sip.conf |
| 03:59.53 | Jouva | Now do us a favor |
| 03:59.59 | Jouva | Pastebin your sip.conf |
| 04:00.16 | seanjohn | now you act like pastebin is a legal source |
| 04:00.27 | ChannelZ | This is a waste of oxygen |
| 04:00.28 | pabelanger | tessier: Problem is with your Aastra 9133i phone. Contact: 703 <sip:00085D18B35D@192.168.3.197:5060;transport=udp> |
| 04:00.30 | seanjohn | just because its in my conf doesn't mean it works |
| 04:00.33 | Jouva | stop being a troll seanjohn and just try it yourself instead of being ignorant |
| 04:00.33 | seanjohn | try it yourself |
| 04:00.36 | pabelanger | tessier: routing issue |
| 04:01.34 | tessier | pabelanger: How so? The private IP? |
| 04:01.40 | pabelanger | tessier: Do you have access to 76.199.182.173 firewall? |
| 04:01.54 | tessier | pabelanger: Yes |
| 04:02.02 | Jouva | ummm |
| 04:02.05 | Jouva | hey seanjohn? |
| 04:02.13 | seanjohn | just type to me |
| 04:02.16 | pabelanger | tessier: Fire up tcpdump and see if you get the request back from asterisk |
| 04:02.20 | seanjohn | fuckin troll |
| 04:02.28 | tessier | pabelanger: ok |
| 04:02.32 | Jouva | actually n/m I know why it didn't reload rtp.conf when I did sip reload |
| 04:02.45 | seanjohn | jouva |
| 04:02.57 | Jouva | [Aug 20 00:01:13] NOTICE[6811]: cdr.c:1373 do_reload: CDR simple logging enabled. |
| 04:02.58 | Jouva | <PROTECTED> |
| 04:02.58 | seanjohn | this is annoying to call someone and then type on another line |
| 04:03.14 | Jouva | so what's this about sip loading rtp.conf? |
| 04:03.49 | pabelanger | tessier: But yes, the private IP in the contact header is the issue. |
| 04:04.16 | pabelanger | ~sipnat |
| 04:04.17 | infobot | i heard sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:04.30 | pabelanger | tessier: ^^ not sure if you have see it |
| 04:04.36 | tessier | pabelanger: I have. Many times. |
| 04:04.43 | tessier | First time about 6 years ago. :) |
| 04:04.46 | tessier | It sucks every time. |
| 04:05.05 | tessier | pabelanger: Isn't asterisk with nat=yes supposed to ignore or rewrite that and use the sending ip instead? |
| 04:05.27 | seanjohn | 201/201 192.168.15.13 D N A 5061 OK (11 ms) |
| 04:05.36 | seanjohn | that extension has rtpstart= and rtpend in it |
| 04:05.49 | ChannelZ | so what |
| 04:05.57 | Jouva | that doesn't mean it's DOING anything |
| 04:06.01 | seanjohn | the 9 consecutive calls made all stayed within the range I put as the rtpstart and rtpend |
| 04:06.03 | pabelanger | seanjohn: *CLI> sip show peer 201 |
| 04:06.25 | Jouva | seanjohn fyi |
| 04:06.46 | Jouva | To eat my own words, I literally put in banana=yummy in my sip.conf in an extension |
| 04:06.53 | Jouva | The extension still works |
| 04:06.59 | seanjohn | jouva fyi |
| 04:07.10 | ChannelZ | Jouva: just give it up, it doesn't matter |
| 04:07.10 | seanjohn | for keeping typing to me and not just addressing me, ignored |
| 04:07.17 | Jouva | HAHAHAHA |
| 04:07.20 | Jouva | wow |
| 04:07.25 | tessier | seanjohn: You could prove it with a pointer to the source that parses and uses it. |
| 04:07.34 | Jouva | Oh fucking wow |
| 04:07.38 | Jouva | So anyway |
| 04:07.53 | Jouva | ChannelZ: So how should I go about debugging my issue PROPERLY? |
| 04:07.59 | tessier | pabelanger: Any other suggestions? |
| 04:08.02 | seanjohn | tessier: netstat works just fine |
| 04:08.32 | ChannelZ | In this mess I don't even remember what your issue was |
| 04:08.47 | Jouva | Oh actually I do wanna try and fix the nat thing first |
| 04:08.53 | seanjohn | tessier |
| 04:09.00 | Jouva | Then I'll poke at the WaitExten() thing |
| 04:09.02 | seanjohn | how annoying is it for me to keep doing this |
| 04:09.04 | seanjohn | tessier |
| 04:09.09 | seanjohn | isnt this annoying |
| 04:09.14 | Jouva | Somebody suggested doing Background() with silence |
| 04:09.16 | tessier | ? |
| 04:09.23 | ChannelZ | Yeah guess who |
| 04:09.28 | seanjohn | tessier: jova can't just type everything on one line |
| 04:09.34 | tessier | heh |
| 04:09.55 | Jouva | tessier: Just ignore him cause he's pissed that 4 or 5 people keep telling him rtpstart doesn't do shit in sip.conf but he INSISTS it does |
| 04:11.10 | tessier | I just want to get this fixed. :| |
| 04:12.10 | pabelanger | tessier: Check the settings on your phone, I'm certain there maybe a configuration issue. |
| 04:12.21 | pabelanger | Will check back in the morning |
| 04:12.51 | ChannelZ | has to go take out the trash |
| 04:12.53 | tessier | pabelanger: Yeah, I've been going through the phone literally all day. Thanks. Hope to have it fixed by tomorrow! :) |
| 04:13.15 | MrHanMan | Has anyone gotten a Cisco 7925g wifi phone to work with Asterisk? Can anyone suggest where I should start? I'm fairly new at this. |
| 04:13.26 | Jouva | Speaking of rtp configuration, even though the RTP ports are being forwarded to my server, do I need to make sure my softphone's RTP ports match up in that range as well? |
| 04:14.18 | ChannelZ | RE: the ports in rtp.conf specify what range asterisk will pick a port from to request the peer send THEIR audio stream to. |
| 04:14.52 | jamko | jouva, you need to try making an isolated rang of rtp ports for each UA, in the device config for the UA, and forward those ports to them in the NAT. |
| 04:14.53 | ChannelZ | The peer then has a range of its own that it requests asterisk send it's audio stream back to. |
| 04:15.42 | ChannelZ | If your asterisk is behind a firewall, then you have to forward the range in rtp.conf through it to your * box. The same would be true of the peer if it is behind a firewall, which you may or may not have control of |
| 04:15.43 | Jouva | mk |
| 04:16.12 | jamko | specifying the rtp port range for each UA in sip.conf, I believe is overkill, because UA do not register their RTP. They set that up during the invite, and asterisk then knows which ports to send back to. |
| 04:16.32 | *** join/#asterisk v1s (~v1s@202.84.107.67) |
| 04:16.35 | ChannelZ | not again |
| 04:17.03 | jamko | The sip port however is a different story, and should be specified individually in sip.conf as well as in the UA. |
| 04:17.08 | Jouva | Right |
| 04:17.38 | tessier | ugh...I think I'm just about burned out for tonight. I am going to take a couple of these phones home and see if they will work properly from there. Other people have working phones at other locations which is what really confuses me. |
| 04:18.05 | tessier | I'll bring in a couple of my phones from home tomorrow and see if they work properly from here. |
| 04:19.26 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 04:19.27 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 04:20.50 | Jouva | I'm trying to see where this damn softphone lets you even define the RTP ports |
| 04:21.57 | *** join/#asterisk hariom (~hariom@122.169.91.249) |
| 04:22.09 | hariom | I read: In asterisk 1.6 the '#' key must be pressed to stop recording. If you simply hung up the recording is lost until k option is used to keep the recording if channel hangs up. |
| 04:22.16 | hariom | WHat is this k option? |
| 04:22.26 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 04:23.27 | jamko | Jouva: It might not.. In fact I bet it doesn't |
| 04:23.44 | v1s | can some one tell me if this should work? http://pastebin.com/rkbftk6P because its not. Thanks |
| 04:24.05 | v1s | it works if the first number is called |
| 04:24.11 | *** join/#asterisk Bendbanks (~bendbanks@eth222.qld.adsl.internode.on.net) |
| 04:24.14 | v1s | but if I call the second number it only gets to answer |
| 04:24.44 | raden | holy flipping $h1T |
| 04:24.58 | raden | why is google chrome on linux so much faster than firefox ? |
| 04:25.17 | jamko | vs1, you have the last exten defined as n |
| 04:25.31 | jamko | for the priority... I can't see why you would do that |
| 04:25.54 | jamko | nevermind |
| 04:26.43 | v1s | its 2 different numbers |
| 04:27.06 | jamko | I don't see why the first number would even work. |
| 04:27.33 | jamko | Asterisk matches exact first and then down, but will only do so if a priority starts at 1 |
| 04:28.01 | v1s | doesnt _X. match any number ? |
| 04:28.15 | v1s | I have it start at x with priority 1 |
| 04:28.30 | v1s | the lines with x I want for either number that is called |
| 04:28.34 | jamko | yes, but then I would think it would die when it filters down.. I could be wrong though. |
| 04:28.46 | jamko | It's an interesting way of doing it. |
| 04:29.00 | v1s | thats what it seems like its doing if it goes to the 2nd number |
| 04:29.04 | v1s | but works on the first |
| 04:29.21 | v1s | let me try swaping them see what happens the other way i didnt try that |
| 04:29.55 | ChannelZ | hariom: core show application record |
| 04:31.10 | v1s | ok so it works only till the first in the order |
| 04:31.14 | hariom | Channelz: Thanks |
| 04:31.29 | v1s | if I put 1111 first it works that extension if I call it |
| 04:31.30 | jamko | yea because it hits that first number and then dies because nothing is wild |
| 04:31.45 | jamko | hmmmm |
| 04:32.08 | v1s | so i should maybe put a goto extention after answer ? |
| 04:32.15 | jamko | So if you put 1111 first, it filters to a 206 number below it? |
| 04:33.30 | v1s | if I have the 2065551111 lines first and call that number it will run through it if I have the 2065552222 and call it it will not work. if I change it around and put the 2065552222 number first it will work but then the 2065551111 will not work |
| 04:33.54 | jamko | yea, got that.. but what did you do with the 1111? |
| 04:34.07 | v1s | put it under the 2222 |
| 04:34.11 | jamko | oh |
| 04:34.11 | v1s | just changed the order is all |
| 04:34.13 | jamko | nevermind |
| 04:34.24 | jamko | I thought you just put 1111 and it worked. |
| 04:34.55 | jamko | yea so it can't get past the first number it hits, because it does not match and dies. |
| 04:35.14 | jamko | So maybe you can make the last 4 wild |
| 04:36.20 | jamko | well that won't work either.. |
| 04:37.19 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 04:37.36 | v1s | think I have todo goto |
| 04:37.42 | ChannelZ | yes you do |
| 04:37.43 | jamko | Yea, you will need goto if |
| 04:37.55 | jamko | yay agreement. |
| 04:38.17 | v1s | would macro work |
| 04:38.43 | sawgood | When running the CLI command, "sip show peers", and I get this output |
| 04:38.47 | sawgood | 121/121 172.16.50.10 D N A 5060 OK (61 ms) |
| 04:38.51 | ChannelZ | _X. matches anything and takes precedence over everything else. I'm not sure if it's by design or accident that after it runs through all those priorities it finds the matching extension of the original wildcard match at an ascending priority.. |
| 04:39.02 | sawgood | why does 121 change to 121/121 |
| 04:39.14 | sawgood | other peers do not have 120/120 (just 120) |
| 04:39.19 | jamko | because it is registered |
| 04:39.22 | ChannelZ | Name/username |
| 04:39.58 | sawgood | some sip peers (phones / extensions) are not registered, but they have the 121/121 entry |
| 04:40.09 | sawgood | 111/111 (Unspecified) D N A 0 UNKNOWN |
| 04:40.12 | jamko | sawgood: do you have qualify on for all peers? |
| 04:40.14 | sawgood | 111 is an example |
| 04:40.26 | sawgood | jamko: Yes, I believe I do |
| 04:41.30 | jamko | hmmm.. My box shows registered extensions doubled up, and unregistered single. Do you have any issues with these phones ringing etc? |
| 04:41.55 | sawgood | jamko: no not really on this box, but with other boxes its become a concern |
| 04:42.28 | sawgood | mostly because the quailfyfreq= statement was not added to sip.conf (but thanks to the help of this channel) I overcame that a few months back |
| 04:42.41 | jamko | are the other boxes showing the unregistered sip peers doubled up or no? |
| 04:43.23 | sawgood | no, they are not doubled up |
| 04:43.37 | jamko | You know now that I think about it, I had this issue once.. I think rebooting the box fixed it. |
| 04:43.58 | jamko | Or maybe not, I will check on that box and see if it still does it. |
| 04:43.59 | sawgood | cool |
| 04:44.11 | jamko | one sec.. gotta log in. |
| 04:46.03 | jamko | sawgood: it's not happening anymore on that box, and the only change I have made to it since then is rebooting. |
| 04:46.12 | sawgood | got it ... thanks |
| 04:47.08 | jamko | np |
| 04:50.26 | Jouva | Is there anything I might be doing wrong here for NAT setup? http://www.pastebin.ca/1920811 |
| 04:50.46 | rue_mohr | raden, cause we cant find a second person smart enough to administer custom rolloed systems |
| 04:50.54 | Jouva | Only thing I can't get to work is outside calls coming in hearing voice from extensions |
| 04:51.23 | raden | rue_mohr, lol |
| 04:51.31 | raden | thgat boxx is more of a headache than asterisk |
| 04:51.36 | raden | and I love aastra |
| 04:51.51 | sawgood | Jouva: I think the NAT/SIP settings you have should be in the file sip_nat.conf not sip.conf |
| 04:51.59 | rue_mohr | really? tell me the issues you have seen... |
| 04:52.03 | sawgood | I like Aastra stuff too |
| 04:52.23 | Jouva | I didn't even have a sip_nat.conf |
| 04:52.33 | Jouva | This is 1.4.21.2 |
| 04:52.34 | sawgood | You have to create the file then ... |
| 04:52.52 | rue_mohr | I want to know cause I'm trying to replace the keyed systems we otherwise sell with them |
| 04:54.59 | [TK]D-Fender | sawgood: NO |
| 04:55.18 | ChannelZ | this is more insanity |
| 04:55.25 | sawgood | [TK]D-Fender: hi! |
| 04:55.27 | jamko | jouva: what brand firewall are you using |
| 04:55.37 | Jouva | I'm behind a DD-WRT based Linksys |
| 04:55.54 | [TK]D-Fender | Jouva: Right file as it is. You problem is that your REGISTER statement must come after everything ELSE in [general]. All those settings you put below it get IGNORED <--- |
| 04:56.11 | Jouva | ohhhhhhhhh |
| 04:57.08 | [TK]D-Fender | [00:51]<sawgood>Jouva: I think the NAT/SIP settings you have should be in the file sip_nat.conf not sip.conf <-- this is FreePBX crap which only exsts because of how THEY decided to format things to make merging manual stuff easier. |
| 04:57.28 | sawgood | Personally, I put my registration statements outside of [general] right below the last entry |
| 04:57.36 | sawgood | never knew that was the reason for it though! |
| 04:58.39 | ChannelZ | I would have never guessed |
| 04:58.40 | *** join/#asterisk russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 04:58.47 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 04:59.16 | [TK]D-Fender | sawgood: HappyGUILand is 2 doors to your left... |
| 04:59.38 | sawgood | [TK]D-Fender: I like using both ... |
| 04:59.40 | Jouva | argh |
| 05:00.07 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 05:00.12 | sawgood | Connected to Asterisk C.2.0.3 (sx00iRC1-2008-08-11 1.2.0.3) currently running on asterisk (pid = 343) |
| 05:00.33 | sawgood | technically, what does Digium call the version of Asterisk which runs on the AA50 like appliances? |
| 05:00.33 | Jouva | blorgh |
| 05:00.45 | Jouva | I can't get this to work... grrr |
| 05:00.48 | sawgood | Is it still called "Asterisk Business Edition", or something else? |
| 05:01.04 | [TK]D-Fender | Jouva: Woul help if you showed us the PROBLEM.... |
| 05:01.08 | [TK]D-Fender | sawgood: Yes |
| 05:01.49 | Jouva | One way audio on calls from the outside that go to an extension. Several ports forwarded, lemme gather them all |
| 05:02.09 | russ | I think teliax added CALLTOKEN suuport to IAX today |
| 05:02.27 | Jouva | But yeah asterisk can speak to the caller, but once it's transfered, it's one way |
| 05:02.43 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 05:03.18 | sawgood | You might need to turn OFF SIP ALG on your edge device to overcome this |
| 05:05.06 | Jouva | asterisk server has 5060, 2000, 2727, 4520, 4569, on TCP and UDP forwarded to it. Softphone Extension has 5061 forwarded |
| 05:05.50 | Jouva | Server has 30000 to 30099 forwarded for RTP and softphone has 30100 to 30199 (might be too big but whatever for now) |
| 05:05.53 | [TK]D-Fender | ~sipnat |
| 05:05.54 | infobot | from memory, sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 05:05.59 | [TK]D-Fender | Jouva: ^^^ Read it again. CLOSELY |
| 05:06.03 | Jouva | kay |
| 05:08.51 | jamko | jouva: you are behind a nat but are not specifying canreinvite=no , which I believe defaults to yes. |
| 05:08.58 | Jouva | ohhh |
| 05:09.08 | [TK]D-Fender | ... |
| 05:09.14 | Jouva | well it gets specified in my config for sipphone account |
| 05:09.14 | [TK]D-Fender | Or you could just hand him the answer... |
| 05:09.48 | Jouva | Does it HAVE to be in general as well? |
| 05:09.48 | [TK]D-Fender | Jouva: Did you custom configure rtp.conf? |
| 05:10.02 | Jouva | Yes. It's a much smaller port range now |
| 05:10.08 | Jouva | Or is this a big no no |
| 05:10.12 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 05:10.21 | [TK]D-Fender | Jouva: As long as it matches. The * samples aren't in that range at all |
| 05:10.32 | jamko | you only need 4 rtp ports per call, maybe less. |
| 05:10.32 | Jouva | right |
| 05:10.35 | jamko | i cant remember. |
| 05:10.44 | Jouva | rtpstart=30000 |
| 05:10.44 | Jouva | rtpend=30099 |
| 05:10.52 | [TK]D-Fender | Jouva: pastebin the SIP debug of your registration and call attempts from * CLI |
| 05:11.05 | Jouva | just a moment |
| 05:11.38 | jamko | jouva i don't see canreinvite=no for your peer. |
| 05:12.12 | Jouva | 101 isn't my peer I posted that earlier |
| 05:12.53 | *** join/#asterisk fofware (~fabian@186.108.159.129) |
| 05:13.23 | russ | does rdnis and/or dnid/dnis information get passed from mobile networks to voip providers? |
| 05:13.49 | Jouva | oh I never did paste that |
| 05:15.13 | Jouva | full sip.conf: http://www.pastebin.ca/1920825 |
| 05:16.25 | Jouva | And the debug log from a fresh start and the incoming call: http://www.pastebin.ca/1920827 |
| 05:16.38 | [TK]D-Fender | Jouva: [sipphone.com] <-- needs to be nat=no <-- |
| 05:16.44 | Jouva | ok |
| 05:17.05 | [TK]D-Fender | Jouva: and that isn't SIP DEBUG. |
| 05:17.16 | Jouva | err derp sorry |
| 05:17.18 | [TK]D-Fender | Jouva: "help sip" <- check the syntax for your version |
| 05:17.20 | jamko | jouva: why don't you use externip instead of externhost... should work but DNS, oh DNS |
| 05:17.34 | Jouva | jamko: dynamic IP |
| 05:17.43 | [TK]D-Fender | Jouva: You also are using externhost without externrefresh. Fix this |
| 05:17.48 | Jouva | mk |
| 05:18.39 | Jouva | nat=no didn't fix it so far |
| 05:19.14 | jamko | yea but the problem isn't with 101 is it? |
| 05:19.15 | Jouva | should I remove it from general since it's in the extension sections and the peer? |
| 05:19.26 | Jouva | it is in a sense. |
| 05:19.45 | [TK]D-Fender | Jouva: only your PROVIDER entry should be nat=no |
| 05:19.48 | Jouva | 101 and 102 can talk fine. but a call coming from sipphone to talk to 101 can't hear 101 |
| 05:20.04 | Jouva | [TK]D-Fender: Right but should I remove nat=yes from general? |
| 05:20.13 | Jouva | cause it's in 101 and 102 |
| 05:20.18 | Jouva | as yes |
| 05:20.27 | [TK]D-Fender | Jouva: NO, because YOU are behind NAT. THAY are not. |
| 05:20.33 | jamko | and I don't see disallow allow for your codecs for each peer or general. |
| 05:20.34 | [TK]D-Fender | THEY* |
| 05:20.41 | p3nguin | The nat=yes in general is for when YOUR system is behind NAT. |
| 05:20.58 | Jouva | Which it is so I'll leave that as is |
| 05:21.31 | [TK]D-Fender | Jouva: fix your codecs as well now |
| 05:21.49 | [TK]D-Fender | (just because) |
| 05:21.56 | [TK]D-Fender | that doesn't cause 1-way audio however |
| 05:22.10 | [TK]D-Fender | Jouva: and make sure canreinvite=no is in [general] |
| 05:22.24 | Jouva | which it is |
| 05:22.29 | Jouva | but ONLY there? |
| 05:22.36 | Jouva | it's in there AND the sipphone.com section |
| 05:22.39 | jamko | where is your nat= for your sip phone |
| 05:22.51 | Jouva | last line in each section |
| 05:23.00 | p3nguin | If it is there, then it should be good for any peer who doesn't specifically say canreinvite=yes. |
| 05:23.36 | jamko | correct.. nat causes one way audio.. PERIOD. |
| 05:23.52 | Jouva | right |
| 05:24.32 | Jouva | I have linphone setup for SIP 5061, audio RTP 30101 |
| 05:24.53 | jamko | and where is the rest of the rtp range. |
| 05:24.53 | Jouva | and 5061 and the 30100-30199 range is forwarded to this PC |
| 05:25.02 | Jouva | hmmm? |
| 05:25.22 | Jouva | actually should my PC's RTP port fall in the RTP range of rtp.conf? |
| 05:25.33 | Jouva | or is that ONLY for the server |
| 05:25.49 | jamko | rtp.conf controls * rtp range.. |
| 05:26.07 | jamko | your UA can be on whatever rtp you specify.. that is setup in the invite. |
| 05:26.21 | jamko | specify on the UA that is, not in * |
| 05:26.43 | Jouva | Sorry I'm mildly confused in how you stated that |
| 05:27.20 | jamko | yea that was garbage. |
| 05:28.22 | Jouva | I've got a sufficient amount of network experience. I'm not a network engineer but I know enough to know general NAT setup stuff. I am very new to asterisk (this is maybe my 2nd day) so some terminology will be new to me :P |
| 05:28.49 | jamko | UA = phone or softphone etc. |
| 05:29.20 | Jouva | Ok. I kinda figured that in context with discussions of others. |
| 05:29.39 | Jouva | But you said rtp.conf controls * rtp range |
| 05:29.42 | Jouva | oh |
| 05:29.43 | jamko | Basically you rtp.conf controls asterisk's rtp range for itself. The rtp range used by your softphone is controlled by the softphone. |
| 05:29.43 | Jouva | derp |
| 05:29.44 | Jouva | ffff |
| 05:29.50 | [TK]D-Fender | Keep it up and look at the CALL in detail. |
| 05:29.55 | [TK]D-Fender | Checkout time... |
| 05:30.02 | Jouva | I'm thinking * as some sort of wildcard, not the name of the software ;) |
| 05:30.25 | jamko | So you need to figure out what rtp range the softphone wants. |
| 05:30.35 | Jouva | right, it asks for a specific port |
| 05:30.40 | jamko | just one? |
| 05:30.48 | jamko | it needs at least 4 to function. |
| 05:31.02 | jamko | or 2 maybe. |
| 05:31.05 | Jouva | surprisingly it's 1 rtp port listed... well 2 if you count the video one :P |
| 05:31.13 | Jouva | but I'm not caring about video |
| 05:31.20 | jamko | wow that's very odd. |
| 05:31.22 | Jouva | it asks for SIP, audio rtp and video rtp |
| 05:31.34 | jamko | can you specify it as a range? |
| 05:31.40 | jamko | like 10000-20000 |
| 05:31.49 | jamko | or does it max out on characters? |
| 05:31.58 | Jouva | maybe but uhh... it's got a spin box |
| 05:32.01 | Jouva | implying it wants a number |
| 05:32.06 | Jouva | oh |
| 05:32.19 | Jouva | and if I put in non numeric characters it removes them when I tab out of the box |
| 05:32.26 | Jouva | and everything after |
| 05:32.29 | jamko | what softphone is this |
| 05:32.33 | Jouva | linphone |
| 05:32.41 | jamko | is it on a computer or android? |
| 05:32.45 | Jouva | PC |
| 05:32.54 | jamko | I have it on my android and the call quality is horrifying. |
| 05:33.09 | jamko | Do you have your PC firewall on? |
| 05:33.11 | Jouva | I've tried sip communicator but it won't let me set the rtp ports |
| 05:33.13 | Jouva | Nope |
| 05:33.18 | jamko | I use x-lite |
| 05:33.21 | jamko | for testing etc. |
| 05:33.29 | Jouva | x-lite wouldn't let me set SIP port |
| 05:33.32 | Jouva | But I guess that won't matter |
| 05:33.38 | Jouva | But I tried it and it still failed |
| 05:33.49 | Jouva | But you folks suggested new things to setup |
| 05:34.03 | Jouva | So lemme try x-lite again |
| 05:34.26 | jamko | only thing with xlite is make sure you specify the domain.. asterisk.com should work fine. |
| 05:40.29 | Jouva | just did a sanity check too... made sure the mic was still working by calling my android phone from this PC. And I can safely say I'm still sane... I think ;) |
| 05:40.45 | Jouva | But no, x-lite doesn't even wanna work |
| 05:43.21 | *** join/#asterisk Tim_Toady (~moi@77.49.105.80.dsl.dyn.forthnet.gr) |
| 05:44.17 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 05:45.07 | tessier | UGH |
| 05:45.10 | tessier | How utterly frustrating |
| 05:45.17 | tessier | I brought two of the phones from the office back to my place. |
| 05:45.53 | tessier | Plug them in. No special forwards or anything. They Just Work. |
| 05:47.07 | tessier | hmm...I didn't try both of these from the office though. Just one of them. The other I grabbed off a desk on my way out forgetting that it hadn't been tested yet. |
| 05:47.29 | tessier | 00085D18B34A/00085D18B34A 68.15.4.17 D N 1024 OK (100 ms) |
| 05:47.35 | tessier | Interesting choice of ports it has there. |
| 05:47.59 | tessier | Now I have 4 IP phones here. All working. Although two each go to different SIP servers. |
| 05:51.55 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 05:59.12 | *** join/#asterisk hopper75 (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 06:04.06 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 06:08.09 | *** join/#asterisk mboehn (mathias@mboehn.alfa-skk.pr0jectX.net) |
| 06:11.50 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 06:12.12 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 06:12.56 | *** join/#asterisk the_weard (~mitch@196.212.100.148) |
| 06:15.56 | *** join/#asterisk pinoyskull (~pinoyskul@124.6.182.55) |
| 06:25.05 | *** join/#asterisk _omer (~omer@119.158.28.150) |
| 06:25.25 | _omer | hello |
| 06:26.46 | ChannelZ | ohell |
| 06:26.47 | _omer | Asterisk prompts the total number of callers in the queue .... is it possible to make it play total number of callers ahead to me ? for example if there are 4 callers in the queue including myself....then it should prompt me the number 3 .... |
| 06:28.29 | *** join/#asterisk gamedna_ (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 06:29.29 | _omer | any suggestions ? |
| 06:31.22 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 06:37.08 | tessier | What does it mean when the manual says the default value for sip registrar port in an Aastra 9133i phone is 0? Yet the phone always seems to end up going with 5060 because it successfully registers. |
| 06:45.30 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 06:45.51 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:47.28 | ChannelZ | dunno, there is no port 0 anyway. |
| 06:49.09 | xheliox | 0 probably means default, which is likely 5060 |
| 06:50.19 | xheliox | kicks ChannelZ in the head |
| 06:50.28 | xheliox | What's up pal? |
| 06:50.42 | ChannelZ | Oh, just bleeding. |
| 06:51.00 | xheliox | That sucks. |
| 06:51.01 | ChannelZ | _omer: doesn't announce-position say the callers position in the queue? |
| 06:51.04 | xheliox | Salt for your wounds? |
| 06:51.07 | ChannelZ | Well, you kicked me in the head. |
| 06:51.15 | xheliox | Yes, yes I did. |
| 06:52.58 | xheliox | you're in Golden, CO? |
| 06:53.05 | xheliox | All I'd do is gamble 24/7. |
| 06:53.17 | ChannelZ | yes |
| 06:53.30 | ChannelZ | But the only gambling down here is going to Taco Bell at 1am |
| 06:54.14 | xheliox | aren't you near Blackhawk? |
| 06:54.55 | xheliox | Colorado geography fail. |
| 06:54.58 | xheliox | Carry on. |
| 06:55.35 | ChannelZ | it's like 20mins away maybe |
| 06:57.12 | xheliox | nods |
| 07:01.50 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
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| 07:19.05 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 07:19.39 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 07:20.54 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 07:25.13 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 07:30.13 | _omer | ChannelZ: It works... but my client want asterisk to announce the total callers ahead to the current caller...for example if My Number in queue is 4 then Asterisk should play "there are 3 callers waiting in the queue" but asterisk says "There are 4 callers waiting in the queue" .... |
| 07:30.29 | _omer | this bullshit is required by my client :-/ |
| 07:30.57 | ChannelZ | Sorry I don't use queues but from all the descriptions it seems like it already does this. |
| 07:31.42 | ChannelZ | Although I guess you're saying you want it to subtract 1 and announce that |
| 07:32.00 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 07:32.27 | elliot98 | which packages are needed to compile asterisk? ncruses, header files...etc? |
| 07:32.55 | ChannelZ | yes |
| 07:33.20 | ChannelZ | gcc... |
| 07:33.42 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:33.44 | elliot98 | is there is list somewhere in the docs? |
| 07:34.08 | elliot98 | I am getting some sort of C++ sanity check error when I run ./configure |
| 07:34.48 | SirLouen | <PROTECTED> |
| 07:34.50 | SirLouen | http://10000horas.com/asterisk/2010/08/09/manos-a-la-obra-con-asterisk/ |
| 07:35.01 | SirLouen | use google translate |
| 07:35.10 | ChannelZ | ./configure --help should show you in a roundabout way other dependencies |
| 07:35.12 | elliot98 | thanks |
| 07:35.30 | elliot98 | on an Ubuntu system, btw |
| 07:35.52 | ChannelZ | me too |
| 07:37.29 | elliot98 | I don't want to use the Ubuntu Asterisk package |
| 07:37.29 | elliot98 | but rather compile from source |
| 07:37.30 | ChannelZ | I don't |
| 07:37.41 | elliot98 | do you know which dependencies you installed, then? |
| 07:37.49 | ChannelZ | not really |
| 07:37.56 | ChannelZ | because I have tons of other crap on it from building other things |
| 07:38.11 | SirLouen | elliot98 you don't have to just install those packages and you will be able to compile all four, asterisk, asterisk-addons, dahdi linux complete and libpri |
| 07:38.22 | ChannelZ | pastebin your output, it should be semi-obvious what you're getting hung up on |
| 07:38.22 | SirLouen | have you read the page? |
| 07:38.41 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 07:38.42 | _omer | <ChannelZ> Although I guess you're saying you want it to subtract 1 and announce that <------ correct |
| 07:38.52 | ChannelZ | Hack the source |
| 07:39.05 | _omer | it is HECK! to hack the source ;) |
| 07:39.23 | _omer | but I think I have to |
| 07:41.35 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
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| 07:46.01 | EmleyMoor | has some finger-pointing to do - but fortunately not at Asterisk |
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| 08:25.21 | xheliox | Google Trends makes me want to cry. |
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| 08:38.13 | v1s | When i use goto where is the hangup in the original context or the one I went to? |
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| 09:15.55 | hrhrhr | chaps |
| 09:16.06 | hrhrhr | if i send a call to a second asterisk box with a prefix of 8 |
| 09:16.33 | hrhrhr | slot it into a context the other end, match the prefix and dial a local zap channel with _8 |
| 09:16.36 | hrhrhr | should that work? |
| 09:16.46 | hrhrhr | using iax as the interbox transport |
| 09:17.34 | hrhrhr | ultimately, i have n sip channels and when they are all in use, i want it to fail over to the second box and dial sip or zap that end |
| 09:17.51 | hrhrhr | not sure which bit of documentation i should be looking at for this |
| 09:19.44 | ruyo | Dialing from an * box, from a SIP phone, from a analog phone, etc, is exacly the same thing. |
| 09:20.22 | ruyo | The same way you make rules for a phone to make a call, you make for the * box. |
| 09:22.22 | hrhrhr | ok but i get request 'thespecificnumber@internal' does not exist |
| 09:22.38 | hrhrhr | i cant seem to match it with anything but the exact number |
| 09:22.47 | hrhrhr | i am obviously being a noob... |
| 09:23.02 | ruyo | Are you using _8 or _8.? |
| 09:23.09 | hrhrhr | the latter |
| 09:23.45 | ruyo | Is thespecificnumber prefixed with an 8? |
| 09:23.50 | hrhrhr | yes |
| 09:24.14 | ruyo | Can you show me the error and that part of the dialplan? |
| 09:24.22 | hrhrhr | ok |
| 09:27.50 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 09:31.04 | hrhrhr | http://pastebin.ca/1920976 |
| 09:31.14 | hrhrhr | i think i can see the problem now but not sure why it is happening |
| 09:31.23 | hrhrhr | it is shaving off the 8 prefix outbound |
| 09:31.36 | hrhrhr | i am not applying :1 to that outbound context tho |
| 09:33.21 | ruyo | In the source PBX you are showing [outgoing] and the call is being made from [internal]. |
| 09:33.37 | ruyo | Check if you don't have a rule in [internal] with ${EXTEN:1} |
| 09:34.16 | hrhrhr | i do |
| 09:34.20 | hrhrhr | but it's for a 9 prefix |
| 09:35.05 | ruyo | Is [outgoing] being included in [internal]? |
| 09:35.13 | hrhrhr | yes |
| 09:35.59 | ruyo | Do a "dialplan reload". |
| 09:36.18 | ruyo | Maybe you changed it and didn't reload the dialplan. :P |
| 09:37.44 | *** join/#asterisk _zoom_ (~user@41.218.38.218) |
| 09:37.49 | _zoom_ | hey |
| 09:38.03 | _zoom_ | does astbill work postgres |
| 09:38.05 | _zoom_ | ? |
| 09:38.06 | ruyo | If you have the same includes in the source box as in the target, see if you have some exten that can match in [iax2forward] |
| 09:39.11 | hrhrhr | ruyo: that command made me realise i had a typo just before globals |
| 09:39.13 | hrhrhr | it's working now |
| 09:39.14 | hrhrhr | cheers! |
| 09:40.12 | ruyo | :> |
| 09:42.32 | elliot98 | SirLouen: thanks! I looked at the asterisk install page |
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| 11:07.14 | hrhrhr | video in asterisk... Unknown RTP codec 126 |
| 11:07.29 | hrhrhr | xlite seems to support h.263 and h263+ |
| 11:07.35 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 11:08.12 | hrhrhr | i have allow 261,263,263p under the extension in sip.conf |
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| 11:17.49 | wubbla | hoi! |
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| 12:01.34 | dlynes | Is asterisk still extremely space sensitive in dialplan code? |
| 12:02.49 | dlynes | I'm encountering an issue whereby exten => _s-.,1,GotoIf($["${ARG6}"=""]?unavail:voicemail) is different from exten => _s-.,1,GotoIf($["${ARG6}"=""] ? unavail : voicemail) |
| 12:03.15 | dlynes | The first one will make it go to n(voicemail); the second one will make it go to n( voicemail) |
| 12:06.00 | drmessano | If that's what youre seeing, leave out the spaces |
| 12:07.21 | *** join/#asterisk rishikesh (~Rishikesh@117.242.156.66) |
| 12:07.29 | rishikesh | hi |
| 12:08.07 | rishikesh | i would like to setup a extension on which i can listen to mp3 or streaming audio |
| 12:08.38 | rishikesh | anybody plz help me how to do this on asterisk? |
| 12:09.22 | SiNGLer | to play audio you can use Playback() |
| 12:09.45 | SiNGLer | for streaming, if I remember correctly where was example in voip-info.org |
| 12:09.49 | rishikesh | i want to setup extension no. say 301 |
| 12:10.17 | rishikesh | when i dial 301, every extension can hear audio or streaming audio |
| 12:11.08 | v1s | rishikesh: think u can just have it got to moh and have it stream something? |
| 12:11.39 | rishikesh | how to do that moh and source as streaming |
| 12:11.57 | rishikesh | how do i setup moh to be available as extension no or dial no? |
| 12:15.34 | v1s | http://www.hurdman.net/mirror/voip-info/wiki/view/Asterisk+config+musiconhold.html |
| 12:15.39 | v1s | that tells u how to do it |
| 12:19.39 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 12:20.20 | *** join/#asterisk wikii (~wiki.mir@host-172-net-105-160-119.mobilinkinfinity.net.pk) |
| 12:23.49 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:24.32 | wikii | my outbound call hangups whenever i try to send fax please see my PRI debug ::http://pastebin.ca/1921367 please help |
| 12:24.45 | rishikesh | ok, thanks |
| 12:29.22 | *** part/#asterisk skyion (~bradc@siza.thusa.net) |
| 12:29.36 | SiNGLer | wikii: ISDN cause 28: invalid number format (address incomplete) |
| 12:29.48 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:29.56 | SiNGLer | check your dialed number, your Dial() command is suspicious |
| 12:31.13 | wikii | i have checkd that number it dials from sip phne |
| 12:31.17 | wikii | no issues |
| 12:33.02 | [TK]D-Fender | wikii: Show us BOTH |
| 12:33.08 | SiNGLer | ZAP/g0/732XXXXXXX, is number X'ed? or is it dialed like this? |
| 12:33.25 | wikii | X ed |
| 12:34.31 | SiNGLer | then show call from sip phone |
| 12:34.40 | wikii | <[TK]D-Fender>ok |
| 12:34.47 | wikii | ok singler |
| 12:34.52 | SiNGLer | [TK]D-Fender: first call: 15:24:27) wikii: my outbound call hangups whenever i try to send fax please see my PRI debug ::http://pastebin.ca/1921367 please help |
| 12:35.40 | [TK]D-Fender | wikii: Do NOT mask the numbers |
| 12:39.40 | wikii | ok |
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| 12:43.22 | garymc | wikii are you using an anologue card for your faxes |
| 12:43.43 | [TK]D-Fender | BAI BAI |
| 12:44.02 | [TK]D-Fender | garymc: Missing the big print as usual... |
| 12:50.35 | garymc | who me? |
| 12:51.30 | garymc | oh ok just read up |
| 12:51.32 | garymc | :) |
| 12:52.31 | garymc | hey TK my head was burnt with this PoE all phone where powering up but one or two were not booting. So changing the connection to the port for a new one an they worked |
| 12:52.33 | garymc | :) |
| 12:53.00 | *** join/#asterisk BANSAL (~bansal@117.199.120.18) |
| 12:55.07 | [TK]D-Fender | garymc: What model again? |
| 12:56.07 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
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| 12:58.55 | [TK]D-Fender | garymc: I know it was a 7000 series... |
| 13:01.49 | *** join/#asterisk iCEBrkr (~icebrkr@cyberdyne.org) |
| 13:02.13 | benedict | I have a question: I get an error if i dial out and so i want to know: Is there a tracing tool which traces an asterisk call and shows me on which layer the error occurs? |
| 13:11.24 | jamko | benedict.. Asterisk has debug features built in.... sip set debug from the cli |
| 13:11.43 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 13:12.01 | jamko | benedict: you could also use tcpdump, and pull the .pcap into wireshark.. |
| 13:12.29 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:12.41 | benedict | Its not over SIP! Its ISDN via HFC Card! Sorry that i didn`t mentioned it :( |
| 13:13.31 | WIMPy | benedict: What channeldriver? |
| 13:14.25 | benedict | i use dahdi |
| 13:15.28 | WIMPy | Hmm. I think pri debug doesn't work with bri, does it? |
| 13:16.18 | *** join/#asterisk radic (~radic@178.2.223.164) |
| 13:16.39 | benedict | It does, but i would like to use another tool, if sth like this exists, because i dont like pri debug^^ |
| 13:17.23 | WIMPy | You could send it to a file and use some other analyzer on it. |
| 13:18.43 | benedict | i sent it with 'pri set debug file' to a file, what analyzers do exist? |
| 13:20.06 | WIMPy | I use tracI. pimped version of the original i4l stuff. |
| 13:20.19 | garymc | [TK]D-Fender : They are FSM7326p Switches |
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| 13:21.51 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 13:22.07 | benedict | okay, sounds good, where can i get it and how do i use it? =) |
| 13:23.04 | WIMPy | http://www.yeti.dk/lsoft/ but it currently needs some fixing for asterisk output. |
| 13:23.54 | benedict | does the patch shown on the site fix it? |
| 13:24.23 | WIMPy | What patch? And no it's not fixed, yet. |
| 13:25.59 | benedict | oh i thougth there is a patch on the site but i, ahhrg forget it^^ |
| 13:27.43 | benedict | ok i compiled it, how do i have to use it? |
| 13:28.26 | WIMPy | Check the config file. |
| 13:31.17 | benedict | ok, thx |
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| 13:54.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:01.26 | benedict | hi everybody! I get an error if i want to dial out (cause 34 - Circuit/channel congestion). I am using asterisk 1.6.2.10 and dahdi 2.3.0. The complete error is posted here http://paste.debian.net/84579/ and my configs here http://paste.debian.net/84578/ can anybody help me please? |
| 14:01.39 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 14:03.22 | *** part/#asterisk jayvee (~jayvee@azaroth.sunriseroad.net) |
| 14:03.31 | WIMPy | dahdi show status |
| 14:04.14 | benedict | it says |
| 14:04.37 | benedict | this http://paste.debian.net/84581/ |
| 14:04.58 | benedict | IRQ 0 ? Is that normal? |
| 14:05.19 | WIMPy | Probably not. |
| 14:05.38 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:05.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:05.38 | WIMPy | Do you find it in /proc/interrupts? |
| 14:05.54 | benedict | i find this 21: 16256535 Phys-irq-level vzaphfc |
| 14:06.19 | Chainsaw | Level-triggered IRQs? |
| 14:06.25 | Chainsaw | Sorry, what vintage is this mainboard? Pentium III? |
| 14:06.45 | hrhrhr | cat /proc/cpu |
| 14:06.47 | benedict | no^^ |
| 14:07.09 | WIMPy | And what dahdi version? |
| 14:07.09 | benedict | its an quad core |
| 14:07.13 | benedict | Q6600 |
| 14:07.26 | benedict | 2.3.0 |
| 14:07.27 | benedict | but |
| 14:07.35 | benedict | i have to say that it is a domU |
| 14:07.52 | benedict | the asterisk server, i passed the hfc card through |
| 14:07.53 | Chainsaw | benedict: I don't support virtualised Asterisk. Ask others. |
| 14:08.16 | Chainsaw | runs away from the horrible ideas |
| 14:09.25 | benedict | perhabs you have an idea for this case even you do not support virtualised Asterisk? ;) |
| 14:09.34 | *** join/#asterisk haighn (~haighn@82.196.42.132) |
| 14:09.55 | benedict | can i tell asterisk or dahdi what interrupt it has to use? |
| 14:10.50 | WIMPy | doesn't know vzaphfc, but probably not. |
| 14:11.18 | benedict | damn :( |
| 14:13.04 | benedict | if the irq of proc/interrupts and dahdi show status is identical it should work, am i right? |
| 14:13.36 | WIMPy | It might. |
| 14:14.17 | benedict | mhh lets see if i can do anything on it |
| 14:17.59 | hrhrhr | anyone actually got video working over * ? |
| 14:18.03 | benedict | ok i looked on another system and there dahdi status also says 0 and it works, although the card runs on irq 0 |
| 14:18.09 | hrhrhr | it crashes xlite as soon as i start video |
| 14:18.13 | hrhrhr | 1.4 and 1.6 |
| 14:18.34 | *** part/#asterisk haighn (~haighn@82.196.42.132) |
| 14:18.57 | hrhrhr | benedict: have you had it working on native hardware? |
| 14:19.11 | hrhrhr | i suspect that may be your next course of action |
| 14:19.46 | benedict | xes it works in another system |
| 14:19.57 | benedict | *yes |
| 14:22.05 | hrhrhr | is 1.6 even stable? |
| 14:22.15 | hrhrhr | it doesn't give the same errors as 1.4 |
| 14:22.20 | hrhrhr | sits there... doing nothing |
| 14:22.39 | hrhrhr | https://issues.asterisk.org/view.php?id=16753 |
| 14:22.45 | hrhrhr | much like that problem i had this week |
| 14:22.51 | shapr | Does anyone have a bash AGI script that does SIGHUP handling? |
| 14:23.50 | [TK]D-Fender | [10:18]<hrhrhr>it crashes xlite as soon as i start video <--- this sounds like an X-Lite problem, not an * one |
| 14:24.03 | hrhrhr | does the same for 1.4 and 1.6 |
| 14:24.12 | hrhrhr | except on 1.4, it gives a codec error first |
| 14:24.12 | festr_ | hi, can DAHDI sniff T1 to pcap? |
| 14:24.18 | festr_ | like sangoma? |
| 14:24.43 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:25.33 | WIMPy | festr_: Take a look at the issue tracker. There has just been something about passive sniffing. |
| 14:26.10 | *** join/#asterisk TigerKing (~Mathijs@195-240-31-32.ip.telfort.nl) |
| 14:26.39 | hrhrhr | [Aug 20 15:25:37] NOTICE[18741]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '192.168.1.98' (xlite client) |
| 14:27.59 | festr_ | https://issues.asterisk.org/view.php?id=16831 |
| 14:33.18 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 14:33.51 | hrhrhr | what other clients can i test video calls in... |
| 14:43.26 | *** join/#asterisk allstatehosting (~allstateh@adsl-068-016-118-195.sip.bct.bellsouth.net) |
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| 14:46.20 | Chainsaw | hrhrhr: Ekiga. |
| 14:47.49 | Kobaz | what's a quick way to check if an exten/context exists |
| 14:48.07 | Kobaz | I usually use ChanIsAvail(Local/exten@context).. but the behavior seems to have changed in 1.6.2 |
| 14:48.21 | hrhrhr | Chainsaw: n1 |
| 14:49.07 | *** join/#asterisk megalomano (~samus@38.124.169.126) |
| 14:49.10 | hrhrhr | would dialplan show do it? |
| 14:49.16 | Kobaz | no |
| 14:49.21 | Kobaz | i need a function/app |
| 14:49.51 | megalomano | hi , someone can help me Or explain the way to install g729 codec |
| 14:50.01 | Kobaz | oh nice |
| 14:50.02 | Kobaz | DIALPLAN_EXISTS |
| 14:50.49 | Kobaz | thanks |
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| 14:54.43 | [TK]D-Fender | megalomano: www.digium.com <- instructions are there |
| 14:55.57 | Naikrovek | jeepers qdoba would you just open already gosh |
| 14:55.59 | Naikrovek | is hungry |
| 14:56.28 | Naikrovek | i prefer chipotle but it's an hour away, qdoba is right there *points* |
| 14:56.45 | garymc | Both sound nice to me |
| 14:56.56 | garymc | I had a ham cheese wrap for my lunch. So boring |
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| 15:03.26 | SirLouen | <PROTECTED> |
| 15:03.46 | SirLouen | i'm having a big issue with a tdm410p fxs module, anyone can help me out? |
| 15:03.56 | SirLouen | this is the problem the most complete I could: http://forums.digium.com/viewtopic.php?f=1&t=75054&p=147823 |
| 15:07.07 | Chainsaw | SirLouen: " DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM CARD??" |
| 15:07.11 | Chainsaw | SirLouen: So, did you? |
| 15:08.28 | Chainsaw | (I'm glad it's in capitals now, it probably needs to be a blinking marquee though) |
| 15:09.59 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:11.55 | Chainsaw | SirLouen: While this is a very one-sided conversation and your input is severely lacking, I do not see a voltage rail measurement for 5V (red & black), only for 12V (yellow & black) which is likely to be irrelevant. |
| 15:11.56 | SirLouen | Chainsaw sure |
| 15:12.09 | SirLouen | well that was also done |
| 15:12.12 | SirLouen | the red and black |
| 15:12.15 | SirLouen | 5V |
| 15:12.18 | SirLouen | was perfect |
| 15:12.26 | Chainsaw | I have never seen a perfect 5V rail. |
| 15:12.39 | SirLouen | you know, 5V in the voltimeter :) |
| 15:12.49 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:12.56 | SirLouen | but only took the capture of the 12V one |
| 15:13.08 | SirLouen | since was the first to checkout |
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| 15:13.24 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 15:13.25 | Chainsaw | SirLouen: Okay, if you are certain that you have 5V & 12V supplied on the molex and that the FXO works on every slot on the TDM410... then it sounds like you may have a faulty FXS module. |
| 15:13.38 | Chainsaw | SirLouen: You need to contact whoever sold you that module. |
| 15:16.36 | SirLouen | Chainsaw i see |
| 15:16.43 | SirLouen | i was believing so, probably is dead |
| 15:18.34 | *** join/#asterisk neurosys (~neurosys@adsl-072-151-195-039.sip.mia.bellsouth.net) |
| 15:19.39 | [TK]D-Fender | SirLouen: Unplug All the lines from the card. check each jack with a phone one by one as the order may not be as expected |
| 15:19.57 | [TK]D-Fender | SirLouen: then change the POSITION of the module within the card one by one. Retest as before. |
| 15:20.13 | hesco | Can anyone point me to a general phone registration troubleshooting checklist or flowchart, please? |
| 15:22.09 | *** join/#asterisk bsaxon_ (~bsaxon@12.107.149.61) |
| 15:22.48 | SirLouen | [TK]D-Fender done that |
| 15:22.50 | SirLouen | nothing |
| 15:23.33 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 15:23.57 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 15:24.04 | [TK]D-Fender | SirLouen: Single phone testing all jacks, only FXS module on the card testing each slot on the card, and jack on the back (16-tests) worth? |
| 15:24.09 | ghenry | is astricon in Washington DC? |
| 15:24.27 | ghenry | I'm from the UK and checking out flights? What airport should i look for |
| 15:24.27 | SirLouen | [TK]D-Fender not that much |
| 15:24.34 | SirLouen | i've checked like 3 channels |
| 15:24.38 | SirLouen | 1-2 and 4 |
| 15:24.42 | SirLouen | same error |
| 15:24.49 | SirLouen | by the way, is not about the phone |
| 15:24.56 | SirLouen | is about the kernel |
| 15:25.00 | SirLouen | shows an error |
| 15:25.06 | SirLouen | even with no phone connected |
| 15:25.28 | SirLouen | i believe that with that error active it wont power any phone ever |
| 15:26.06 | [TK]D-Fender | SirLouen: where do I see this error? |
| 15:31.25 | chazzam | ghenry: http://www.astricon.net/hotelTravel.aspx |
| 15:31.54 | chazzam | the bottom half of the information there is about flights |
| 15:36.11 | SirLouen | [TK]D-Fender i see this problem in the kernel boot process |
| 15:36.14 | SirLouen | dmesg |
| 15:36.21 | SirLouen | also in /var/log/messages |
| 15:36.34 | [TK]D-Fender | [11:26]<[TK]D-Fender>SirLouen: where do I see this error? |
| 15:37.03 | SirLouen | http://forums.digium.com/viewtopic.php?f=1&t=75054&p=147823 |
| 15:37.05 | SirLouen | hyere |
| 15:38.01 | [TK]D-Fender | DID YOU REMEMBER TO PLUG IN THE HD POWER CABLE TO THE TDM CARD?? wctdm24xxp 0000:05:08.0: Unable to do INITIAL ProSLIC powerup on |
| 15:38.14 | [TK]D-Fender | SirLouen: Maybe your molex is no good. Change it for another |
| 15:38.55 | [TK]D-Fender | SirLouen: Inspect the card carefully as well |
| 15:43.06 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:44.20 | hesco | sip show peers gives me: 21 (Unspecified) D N A 5060 UNKNOWN |
| 15:45.06 | hesco | I've checked and reset the password twice. Any ideas how I should proceed debugging this so my phone registers to the server? |
| 15:46.50 | p3nguin | Turn off the ACL for now. |
| 15:47.05 | p3nguin | Re-apply it after you get things working. |
| 15:52.12 | SirLouen | <[TK]D-Fender> SirLouen: Maybe your molex is no good. Change it for another... the molex? |
| 15:52.21 | SirLouen | i've changed the power supply!! |
| 15:52.25 | SirLouen | and same problem |
| 15:52.28 | SirLouen | i believe the module is broken |
| 15:52.32 | SirLouen | the FXS module |
| 15:52.59 | Kobaz | anyone familiar with audiocodes mediapack boxen |
| 15:53.12 | Kobaz | i'm getting some echo from time to time |
| 15:53.22 | Kobaz | i can't really find any settings to adjust the echo canceller |
| 16:02.01 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 16:03.07 | coppice | why would you expect to find any settings? |
| 16:06.59 | ghenry | thanks chazzam |
| 16:07.35 | ruyo | SirLouen, have you tried using only one module at a time? |
| 16:13.34 | chazzam | ghenry: no problem, enjoy the trip! |
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| 17:07.13 | jamko | Is it not possible to use the 'n' priority with realtime extensions? |
| 17:08.28 | [TK]D-Fender | JamNo |
| 17:09.18 | jamko | thanks |
| 17:09.29 | Corydon76-dig | There's lots of things that you cannot do with realtime extensions |
| 17:09.49 | Corydon76-dig | including pretty much anything |
| 17:09.50 | Kobaz | realtime is kinda crappy |
| 17:10.03 | Corydon76-dig | Realtime isn't. Realtime extensions are |
| 17:10.06 | Kobaz | static-realtime is where it's at |
| 17:10.23 | Corydon76-dig | Realtime voicemail is one of the best |
| 17:13.30 | *** part/#asterisk pyite (~dschreibe@unaffiliated/pyite) |
| 17:15.19 | bmoraca_work | realtime SIP and realtime voicemail work very well |
| 17:15.30 | jamko | @corydon76-dig / Kobaz: So referencing, sip, extensions, voicemail, queues, which would you argue fits best with Realtime, and which fits better with Static Realtime? |
| 17:15.32 | bmoraca_work | realtime extensions are useful for temporarily superceding normal dialplan, i've found |
| 17:15.47 | Kobaz | jamko: in every case, static-realtimr |
| 17:16.01 | bougyman | we run everything realtime (mod_xml_curl) |
| 17:16.08 | bougyman | er woops, wrong chan. |
| 17:17.01 | Kobaz | jamko: i've found realtime to be essentially useless in all cases |
| 17:17.09 | Kobaz | but that's just me |
| 17:17.11 | bmoraca_work | you're doing it wrong, then |
| 17:18.00 | Corydon76-dig | bougyman: known as res_config_curl here |
| 17:18.59 | Corydon76-dig | bougyman: so why, if you aren't running Asterisk, do you hang out here? |
| 17:19.15 | bougyman | Corydon76-dig: i still have two asterisk boxes. |
| 17:20.42 | bougyman | til I can replace orderlyq with a new reporting backend, i'll have to keep em |
| 17:20.58 | bougyman | unless orderly ports their stuff to a new platform. |
| 17:22.31 | jamko | So with Static Reatime, do the config files actually get imported into the MySQL DB? ie: > mysql asterisk < extensions.conf ? |
| 17:24.58 | Corydon76-dig | jamko: yes |
| 17:25.53 | jamko | Corydon76-dig: I would like to have multiple * boxes feeding off the same central DB, and not have to update .conf files on every machine, every time, for every change. So I would just load the .conf files into mysql one time for each change when using Static RealTime? |
| 17:25.59 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-43-239.mia.bellsouth.net) |
| 17:26.29 | bougyman | that or use something like res_config_curl |
| 17:26.35 | Corydon76-dig | jamko: plus do a reload on each machine |
| 17:26.54 | Corydon76-dig | bougyman: that's orthogonal |
| 17:26.54 | *** join/#asterisk ruyo (~psantos@a81-84-220-57.cpe.netcabo.pt) |
| 17:27.07 | jamko | well that's fine.. Don't mind the reload .. Thanks for the tips.. |
| 17:27.40 | ChannelZ | Corydon76-dig: here's some insanity from a guy last night.. rtpstart and rtpend have no relevance in sip.conf do they? |
| 17:27.47 | Corydon76-dig | res_config_curl is simply an abstraction layer between Asterisk and the database |
| 17:28.10 | Corydon76-dig | ChannelZ: not in that config file, no |
| 17:28.18 | jamko | omg... |
| 17:28.22 | Corydon76-dig | They have relevance to the sip layer |
| 17:28.48 | ChannelZ | Right, but you can't set port ranges per peer... he seemed to think you could and would not hear anything otherwise |
| 17:28.56 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 17:29.30 | ChannelZ | (IE he insisted he was right) |
| 17:29.30 | jamko | Both of those guys were lost causes. |
| 17:30.01 | jamko | One guy had magic nat... you can't debate someone with fairy dust. |
| 17:30.04 | Corydon76-dig | ChannelZ: He can insist whatever he wants. The question is whether he can show logs to prove that it works |
| 17:30.44 | ChannelZ | I found it particularly amusing when he kept saying "just try it" and when I did and proved nothing worked, he conveniently ignored me twice. |
| 17:30.58 | Corydon76-dig | Yeah, I saw that |
| 17:31.19 | ChannelZ | And then to top it off he cursed someone out telling them to 'learn how to program' or something as his defense that it did work. |
| 17:31.58 | ChannelZ | Anyways.. just checking :) I was beginning to wonder if *I* was mad |
| 17:33.02 | Corydon76-dig | Some users try to get things implemented by insisting that it should work a particular way, then try to beat the developers into submission into making it work that way |
| 17:33.39 | Corydon76-dig | The easy way to determine that for you is to grep rtpstart from channels/chan_sip.c |
| 17:33.47 | Corydon76-dig | If it ain't there, it ain't a keyword |
| 17:33.53 | ChannelZ | Yeah we tried that, he wasn't having any of it. |
| 17:34.26 | jamko | I think he missed the boat on rtp not being a registration method. |
| 17:34.46 | ChannelZ | Yah. Hitting the moonshine a little too hard I guess. |
| 17:34.50 | Corydon76-dig | The one saving grace is when he claims that to a potential employer, and the employer realizes that he's a complete charlatan |
| 17:35.28 | jamko | I use magic nat. |
| 17:36.01 | ChannelZ | My Asterisk is powered by unicorn tears |
| 17:36.03 | xheliox | Corydon76-dig: Unfortunately there are a lot of employers who wouldn't know the difference until 2 weeks later and nothing works properly. |
| 17:36.22 | xheliox | ChannelZ: Dare we ask how you make them cry? |
| 17:36.26 | Corydon76-dig | xheliox: yes, and that's where the rubber hits the road |
| 17:36.45 | ChannelZ | It's best if you don't know. |
| 17:36.58 | jamko | chan_unicorn.so |
| 17:37.15 | Corydon76-dig | res_unicorn.so |
| 17:37.32 | xheliox | Is it Charlie The Unicorn? |
| 17:37.35 | v1s | what kind of phone or equipment do u need to send/recv sms from * ? |
| 17:37.36 | jamko | realtime unicorn, magic nat, smoking the competition. |
| 17:37.40 | xheliox | Come to magic mountain, Charlie.. |
| 17:38.09 | Corydon76-dig | v1s: either a connection to a telco which supports it on a line or an SMS modem |
| 17:38.40 | Corydon76-dig | v1s: British Telecom is the only provider I know that supports it on a line |
| 17:38.57 | v1s | sorry I should have phrased my question better. I mean for client side of it. |
| 17:39.29 | Corydon76-dig | v1s: Doesn't matter. You need to be on a network which supports sending or receiving those messages |
| 17:39.52 | Corydon76-dig | excuse me, they're called "GSM modem" or "CDMA modem", depending upon the carrier |
| 17:40.45 | v1s | is there way for snom phones or any other voip client or phone to send recv that you know? |
| 17:40.58 | bougyman | snom yes. |
| 17:40.58 | v1s | if I the network supports it ? |
| 17:41.03 | bougyman | you could make an xml app to do it. |
| 17:41.18 | Corydon76-dig | v1s: relayed via an Asterisk server, if you have the right hardware, yes |
| 17:41.41 | Corydon76-dig | bougyman: WTF is with you and XML |
| 17:41.44 | v1s | thanks ;) |
| 17:41.53 | bougyman | Corydon76-dig: snoms have an xml browser for apps. |
| 17:41.53 | Corydon76-dig | XML has utterly NOTHING to do with SMS |
| 17:42.06 | bougyman | you can make xml apps for stuff like this (there are some already made) |
| 17:42.11 | bougyman | adhearsion has some, iirc. |
| 17:42.34 | Corydon76-dig | Yes, but you still need a provider to relay the messages, and a web app provided by the carrier is not a good option |
| 17:42.52 | Corydon76-dig | especially when you're running a service |
| 17:43.08 | bougyman | eventually it has to hit a provider, sure. |
| 17:43.11 | v1s | what if you where going between to different * boxes :) |
| 17:43.14 | bougyman | i don't think the middleware affects that. |
| 17:44.09 | Corydon76-dig | Uh, the GSM modem is the middleware. If you ain't got it, your app won't do crap |
| 17:44.16 | Corydon76-dig | v1s: that's called text messaging |
| 17:44.18 | bougyman | i prefer sms through IM (backed by ejabber), in our env. |
| 17:45.06 | Corydon76-dig | v1s: probably need to use a separate server. Asterisk does not support sending text messages outside of an active call at this time |
| 17:46.38 | Corydon76-dig | To be clear, SMS != text messaging. SMS is a particular technology that requires the use of a telecom network. Text messaging does not necessarily need to go through a telecom network |
| 17:47.30 | v1s | right now I am using the chan_datacard with 4 modems so was just trying to think of something useful I could do with sms since theres I have unlimited sms on each modem |
| 17:49.04 | Corydon76-dig | v1s: I've used them before to trigger calls from the US to Saudi Arabia and give reverse dialtone when answered |
| 17:49.57 | v1s | yes that is kind of what I am doing ;) |
| 17:50.12 | v1s | but just using missed call |
| 17:50.26 | v1s | not sms |
| 17:50.47 | Corydon76-dig | Rate arbitrage was an interesting business |
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| 18:06.14 | a_nonamiss | I have a question on incoming settings using an elastix (2.0) server that I'm trying to set up with an existing SIP trunk. |
| 18:06.24 | a_nonamiss | Any help would be greatly appreciated. |
| 18:06.47 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-kwnztqvzqvruykbv) |
| 18:07.15 | shido6 | can I use sox to help build an .srt ringtone file for a linksys spa942 ? |
| 18:07.17 | a_nonamiss | I'm able to make outgoing calls on said trunk, but not recieve incoming calls. |
| 18:08.00 | a_nonamiss | In the Elastix GUI, there are Outgoing Settings and Incoming settings. |
| 18:09.02 | a_nonamiss | There is a box titled "USER Context" When I enter my cell phone in that box (with insecure-very and context=from-trunk in the details) I can get calls from my cell phone but nowhere else. |
| 18:09.44 | a_nonamiss | On the old (trixbox) server (where the trunk is currently working) I have "from-trunk" in that field. |
| 18:10.07 | a_nonamiss | but when I enter "fron-trunk" in the elastix server, the incoming call fails. |
| 18:10.16 | a_nonamiss | "from-trunk" rather. |
| 18:11.53 | a_nonamiss | I can see in the SIP debug that the call is coming from mycellphonenumber@ip.address.of.sip.provider. |
| 18:15.49 | a_nonamiss | Oh, I left out type=user, also in the detail field. |
| 18:19.29 | [TK]D-Fender | a_nonamiss: "fails" is not a usable description. pastebin teh failed call with SIP DEBUG enabled. |
| 18:19.32 | [TK]D-Fender | ~pb |
| 18:19.33 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:19.48 | Kobaz | oh man |
| 18:19.55 | Kobaz | make sure you clean up after your children |
| 18:20.29 | Kobaz | i had an app that fork()'d and then pushed the child pid to a list... so when the parent got a hup... it would hup the children... but i never removed anything from the list |
| 18:20.47 | a_nonamiss | fails with SIP/2.0 403 Forbidden |
| 18:21.02 | Kobaz | so after weeks of running, it would have amassed thousands of no longer running pids in the child_pids list... and then it got a hup... it was hupping random programs |
| 18:21.25 | a_nonamiss | What I really would like is a pointer to documentation on what should go in there, what different values mean, etc. |
| 18:21.47 | a_nonamiss | I'm sure with proper information, I can probably figure it out. |
| 18:22.53 | *** join/#asterisk neurosys (~neurosys@adsl-072-151-195-039.sip.mia.bellsouth.net) |
| 18:24.36 | bougyman | where is sequel_orderable now? |
| 18:24.48 | bougyman | i had it as a require in an old project, can't find it on gemcutter. |
| 18:25.55 | a_nonamiss | http://pastebin.com/tyq9pDNE |
| 18:26.43 | [TK]D-Fender | a_nonamiss: pastebin your SIP setup masking ONLY passwords |
| 18:28.39 | a_nonamiss | http://pastebin.com/ziC8zgzB |
| 18:29.34 | a_nonamiss | The second bit, the from-trunk. That's the "USER context" under incoming settings in Elastix. |
| 18:30.17 | a_nonamiss | If I change it to the number I'm calling from, it works. Is there some sort of wildcard so that it'll accept all calls from my provider? |
| 18:31.18 | [TK]D-Fender | a_nonamiss: under [Citynet] add "insecure=port,invite" |
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| 18:33.55 | a_nonamiss | [TK]D-Fender, you are a brilliant man. I have been hacking at this for 4 days. Can I name my children after you? |
| 18:34.18 | a_nonamiss | Or woman, sorry. ;-) Name is pretty gender neutral. |
| 18:34.18 | [TK]D-Fender | a_nonamiss: Sure. |
| 18:34.35 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:34.48 | [TK]D-Fender | a_nonamiss: "Andrew". Safe enough. You can use "Andrea" for the girls :) |
| 18:34.56 | a_nonamiss | Heh. |
| 18:37.45 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:41.27 | *** join/#asterisk yang (yang@freenode/sponsor/cacert.assurer.yang) |
| 18:41.45 | yang | Which is a popular PBX software developed lately ? |
| 18:41.57 | russellb | o.O |
| 18:42.06 | shido6 | hrmm |
| 18:42.09 | shido6 | thats a hard one |
| 18:42.18 | shido6 | Call Manager? |
| 18:42.24 | yang | I know FreePBX, freeSWITCH, Elastix are there any others |
| 18:42.25 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 18:42.53 | p3nguin | I prefer Asterisk. |
| 18:42.59 | Qwell | I heard about this thing called "Asterisk", but never looked at it. |
| 18:43.08 | shido6 | no its called aztrix |
| 18:43.12 | yang | Yes, off course Asterisk |
| 18:43.23 | p3nguin | qwell: It's pretty nice. Very robust. I can get you a web link if you would like. |
| 18:43.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:43.35 | Qwell | p3nguin: no thanks, I like my PBX software. |
| 18:43.49 | *** join/#asterisk voxter (~voxter@dsl110.net.ubc.ca) |
| 18:43.53 | p3nguin | ;) |
| 18:44.06 | yang | I guess there aren't many other forks |
| 18:44.07 | russellb | i wrote my own PBX in bash |
| 18:44.11 | Qwell | forks? |
| 18:44.17 | russellb | it's called PBashX |
| 18:44.18 | Qwell | none of the things you've mentioned are forks of anything else. |
| 18:45.19 | Qwell | "Private Bash eXchange" sounds like an sh script blackmarket |
| 18:45.32 | yang | ;) |
| 18:45.50 | Qwell | yang: your questions are making no sense |
| 18:46.07 | yang | I was only wondering about the new products on the market |
| 18:46.13 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:46.22 | Qwell | None of the things you've mentioned are "new" by any means. |
| 18:46.25 | *** join/#asterisk neurosys (~neurosys@adsl-072-151-195-039.sip.mia.bellsouth.net) |
| 18:46.50 | yang | Those were the ones I've heard of |
| 18:48.23 | [TK]D-Fender | [14:42]<yang>I know FreePBX, freeSWITCH, Elastix are there any others <- these 3 items don't belong in that list |
| 18:48.49 | [TK]D-Fender | yang: Unlike items |
| 18:49.59 | Qwell | [TK]D-Fender: "software that is related to telephony in some way" |
| 18:50.21 | Qwell | yang: what, specifically, are you looking for? |
| 18:50.23 | russellb | this channel makes my head hurt |
| 18:50.37 | russellb | no wonder [TK]D-Fender is so mean to people :-p |
| 18:50.39 | *** kick/#asterisk [russellb!~north@pdpc/sponsor/digium/Qwell] by Qwell (you'll thank me later) |
| 18:50.39 | *** join/#asterisk russellb (~russellb@asterisk/digium-open-source-team-lead/russellb) |
| 18:50.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:50.41 | Qwell | :P |
| 18:50.53 | Deeewayne | fight fight |
| 18:50.56 | [TK]D-Fender | russellb: Hang out in #freepbx more... it'll ass-plode... |
| 18:51.02 | russellb | no thanks |
| 18:51.15 | *** mode/#asterisk [-o+b Qwell *!*north@*pdpc/sponsor/digium/Qwell] by russellb |
| 18:51.15 | *** kick/#asterisk [Qwell!~russellb@asterisk/digium-open-source-team-lead/russellb] by russellb (Qwell) |
| 18:51.36 | *** mode/#asterisk [-b *!*north@*pdpc/sponsor/digium/Qwell] by ChanServ |
| 18:51.36 | *** mode/#asterisk [-b *!*north@*pdpc/sponsor/digium/Qwell] by russellb |
| 18:51.40 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 18:51.40 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 18:51.42 | Qwell | ass |
| 18:51.47 | russellb | ^_^ |
| 18:51.49 | yang | heh |
| 18:52.34 | russellb | Qwell: you'll thank me later |
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| 19:11.13 | jeffg | other than progressinband=no and dtmf=rfc2833, are there any other settings that should always be on for a polycom IP33x? i'm getting no DTMF at all from an IP335 and three IP330s across two different asterisk installations. |
| 19:11.51 | jeffg | one asterisk is 1.6.1, the other 1.6.2 |
| 19:13.59 | russellb | jeffg: i assume you have dtmfmode=rfc2833, not dtmf=... |
| 19:14.15 | russellb | other than that, i dunno, i would check the phone's configuration to see what DTMF mode they have been configured for |
| 19:15.24 | jeffg | russellb, you're correct |
| 19:16.09 | russellb | you could also try dtmfmode=auto |
| 19:16.23 | russellb | or explicitly dtmfmode=inband |
| 19:16.27 | russellb | in case it's being sent that way |
| 19:20.26 | jeffg | russellb, the way my plcm configs stack up, it would seem i've got: tone.dtmf.rfc2833Payload="127", tone.dtmf.viaRtp="0", tone.dtmf.rfc2833Control="1" |
| 19:20.56 | jeffg | wonder if the payload setting is jacked... |
| 19:21.03 | russellb | hmm |
| 19:21.07 | jeffg | that's the setting in the distributed sip.cfg |
| 19:21.10 | russellb | viaRtp="0" looks suspicious |
| 19:21.14 | jeffg | oh yeah? |
| 19:21.20 | russellb | opens his polycom admin guide |
| 19:21.32 | jeffg | i've always thought dtmf-inband = recipe for trouble |
| 19:21.44 | russellb | for sure, just meant to see if that's what it was doing |
| 19:22.19 | russellb | i'm pretty sure you set that to 1 to use RFC2833 |
| 19:22.40 | jeffg | that's counterintuitive :) |
| 19:22.57 | jeffg | but i only play a phone guy on irc |
| 19:23.08 | russellb | well, RFC2833 == DTMF in the RTP stream ... |
| 19:23.17 | russellb | viaRtp from their admin guide: "If set to 1, encode DTMF in the active |
| 19:23.17 | russellb | RTP stream, otherwise, DTMF may be |
| 19:23.18 | russellb | encoded within the signaling protocol only |
| 19:23.18 | russellb | when the protocol offers the option. |
| 19:23.18 | russellb | " |
| 19:23.28 | russellb | oops, thought that was going to paste as one line... |
| 19:23.31 | jeffg | yeah, i think i'm reading the same guide |
| 19:23.51 | russellb | set to 1, reboot, try again! |
| 19:23.58 | russellb | and then we celebrate working DTMF! |
| 19:25.15 | jeffg | will do... i'm walking my user through this remotely, funsies |
| 19:25.18 | jeffg | russellb, thanks! |
| 19:25.24 | russellb | of course, no problem |
| 19:25.30 | russellb | you also shouldn't have to set the payload |
| 19:25.44 | russellb | but it doesn't hurt anything, so nm |
| 19:25.48 | *** join/#asterisk fofware (~fabian@186.108.159.129) |
| 19:30.16 | jeffg | yeah, the payload is set in sip.cfg... i've got a three-stage setup: sip.cfg, local.sip.cfg, and a per-phone one |
| 19:31.08 | russellb | cool - same here :-) |
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| 19:38.21 | jeffg | russellb, just got a w00t from one of the IP330 users, muchos gracias! |
| 19:39.48 | russellb | \o/ |
| 19:39.55 | russellb | jeffg: you're quite welcome |
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| 19:46.59 | Kobaz | is there a way you can make a polycom always show your extension on the display |
| 19:47.25 | Kobaz | when you're idle it shows your extension... but then when you are on the phone, or you have missed calls... the extension display goes away.. (this is for 320/330) |
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| 19:56.14 | [TK]D-Fender | Kobaz: Try by using a MicroBrowser idle screen |
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| 20:05.23 | Schreiber1337 | Noob to AGI... anyone willing to help me get my first script executing? |
| 20:07.14 | Schreiber1337 | run_agi: unable to send SIGHUP to AGI process 13224: No such process |
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| 20:14.08 | marcompile | hello, does chan_alsa supports multiple devices? |
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| 20:28.26 | tessier | When I do sip show peers it has a port column. That is the port number asterisk expects to be able to send sip messages to on the remote phone, correct? |
| 20:29.12 | jamko | tessier: that is where it will send the original invite when it has something to send to the peer. |
| 20:30.02 | tessier | jamko: Right. So that is where the peer is listening. |
| 20:30.27 | jamko | That is where it "should" be listening. Whether or not it is on that port, is in your device and nat config. |
| 20:30.42 | jamko | but |
| 20:31.11 | jamko | sometimes without a port= specified in sip.conf for the peer, it will use a random port, not specified it's its own device config. |
| 20:31.12 | marcompile | so, nobody uses chan_alsa? |
| 20:31.16 | jamko | For example |
| 20:31.40 | jamko | the device may have 5060 listed in it's gui or config file, but asterisk is showing 89776 |
| 20:31.50 | jamko | this is what causes problems with sip and nat and * |
| 20:32.12 | ChannelZ | marcompile: I played around with it once one night but that's it |
| 20:32.33 | tessier | jamko: Right. Some of my phones are showing up in asterisk as 5060. Some are showing up as random high numbered ports. |
| 20:32.39 | tessier | jamko: I'm wondering what the difference is. |
| 20:32.46 | marcompile | I think it does not support multiple soundcards (as opposed to the oss module) |
| 20:32.48 | tessier | jamko: Phones that show up as random high numbered ports work. |
| 20:32.59 | jamko | tessier: sometimes they will. |
| 20:33.02 | tessier | jamko: But it should be the local NAT that is choosing the port it is going out of. |
| 20:33.20 | tessier | jamko: The phone should not have any control over that. |
| 20:33.39 | bmoraca_work | tessier: the NAT router that the phone is behind determines what port Asterisk perceives the phone to be on |
| 20:33.54 | jamko | tessier: it is imperative that you make the ports align properly, as discussed last night. |
| 20:33.55 | ChannelZ | jamko: when you specify a port in the sip.conf for a peer, it's a lot like specifying the hostname... IE you're setting it statically. The actual phone might not agree or even be aware of what you did. |
| 20:34.36 | jamko | channelz: right, which is why you must set the phone to use the same sip port as specified in sip.conf for the peer. |
| 20:34.44 | jamko | or you WILL have problems with nat. |
| 20:34.45 | ChannelZ | yesh |
| 20:34.55 | jamko | now onto RTP |
| 20:34.55 | bmoraca_work | specifying port in sip.conf is only useful if you're statically configuring the peer. you don't need it if the host=dynamic |
| 20:35.11 | tessier | jamko: I still don't get that. And I still don't understand why the other sites here work properly without doing any port forwarding at all. |
| 20:35.20 | bmoraca_work | in fact, you don't WANT it if host=dynamic |
| 20:35.23 | tessier | I have set up a port range to forward though, as discussed last night. |
| 20:35.30 | tessier | And it still didn't work. I'm going to try it again now. |
| 20:35.49 | jamko | tessier: I think you need to start looking at a bad firmware version on your firewall. |
| 20:36.00 | jamko | and turn off all sip aware / sip alg settings in it. |
| 20:36.03 | bmoraca_work | tessier: what symptoms are you noticing? |
| 20:36.31 | tessier | jamko: It's Linux/Shorewall. And I have unloaded the sip "helper" modules. |
| 20:36.40 | jamko | tessier: first thing is you must get those sip ports to align with what comes up in sip show peers. |
| 20:36.54 | jamko | If you can't get that far, then you need to go back and find the answer to that problem. |
| 20:36.56 | tessier | bmoraca_work: Normally I can put as many SIP phone as I want behind a Linux based firewall. As long as nat=1 in sip.conf they just work. |
| 20:37.01 | bmoraca_work | jamko: no, no you don't. |
| 20:37.05 | tessier | bmoraca_work: But at this particular location only one phone can register at a time. |
| 20:37.18 | jamko | bmoraca_work: oh yes you do. |
| 20:37.19 | bmoraca_work | tessier: what type of phone? also, that's a firewall issue. |
| 20:37.23 | tessier | bmoraca_work: And it is the same firewall as I run everywhere else. But these phones are aastra phones which I don't haev much experience with. |
| 20:37.34 | tessier | bmoraca_work: I know it is a firewall issue. I'm trying to figure out how to fix it. |
| 20:37.35 | bmoraca_work | jamko: no. if host=dynamic, you do NOT want to specify the port in sip.conf. |
| 20:37.46 | jamko | bmoraca_work: are you on crack? |
| 20:38.03 | bmoraca_work | jamko: no |
| 20:38.11 | bmoraca_work | jamko: i suspect you have no idea what you're talking about. |
| 20:38.18 | bmoraca_work | tessier: what type of phone? |
| 20:38.55 | tessier | bmoraca_work: Everyone keeps telling me I have to forward a ton of ports in the firewall and configure each phone to use unique port numbers matching what's in the fw etc. But I've never had to do that before. But I'm in the middle of giving it a try now because I'm out of other ideas. These are Aastra 9133i phones. |
| 20:39.03 | jamko | bmoraca_work: whatever.. good luck. |
| 20:39.15 | bmoraca_work | tessier: you don't need to forward any ports on the NAT router the phones are behind |
| 20:39.20 | *** join/#asterisk Jouva (jouva@2001:470:8:2c3:1337:dead:beef:f00d) |
| 20:39.25 | bmoraca_work | jamko: i don't need luck. I know what I'm doing. |
| 20:39.36 | tessier | bmoraca_work: In the past I use nat=1 host=dynamic etc in sip.conf and the phones would punch a hole which would be kept open with qualify= in sip.conf and they would work. |
| 20:39.44 | jamko | bmoraca_work: actually you dont, but go ahead, this should be fun to watch. |
| 20:40.10 | tessier | bmoraca_work: Am I wrong here? Have my previous installations somehow worked by luck? |
| 20:40.18 | Jouva | Ok so i can find the reference pages for asterisk (which gives good info on dialplan stuff) but I'm looking for configuration info, and "the book" is on 1.4 not 1.6. Where can I read up on everything that goes in sip.conf? |
| 20:40.28 | jamko | bmoraca_work: maybe you are thinking of the iax protocol, but apparently sip you know nothing about. |
| 20:40.52 | chazzam | Jouva: the sample sip.conf that comes with Asterisk might be a good start |
| 20:40.55 | bmoraca_work | jamko: shut the hell up and leave the advice to the people who have done this longer than 2 weeks. |
| 20:41.12 | jamko | bmooraca_work: wow the troll can curse. |
| 20:41.28 | jamko | keep going.. I am smiling all the way. |
| 20:41.32 | bmoraca_work | tessier: nat=1 and host=dynamic is correct. |
| 20:41.39 | Jouva | chazzam: Which I do not have since I upgraded and the install wanted to completely rewrite it |
| 20:41.52 | bmoraca_work | tessier: however, depending on the phone, you may need to also tell the phone it is behind a NAT |
| 20:41.54 | Jouva | There's deprecated config lines in there |
| 20:42.10 | Jouva | But most of it is correct |
| 20:42.15 | bmoraca_work | tessier: the only port forwards you need are 5060 and your RTP range on the router in front of Asterisk |
| 20:42.39 | chazzam | Jouva: you can still look at them in the extracted source |
| 20:42.41 | bmoraca_work | tessier: you don't need ANY port forwards on the routers the phones are using. you don't need to tell the phone to use anything special. NAT, by itself, will take care of that. |
| 20:42.42 | chazzam | or online |
| 20:42.47 | tessier | bmoraca_work: Right, I've already got that. I did a one to one nat and ALLOWED those ports in the firewall. I have half a dozen other remote phones already working correct. |
| 20:42.50 | Jouva | chazzam: I used apt-get |
| 20:42.53 | Jouva | But yeah |
| 20:42.55 | tessier | bmoraca_work: That's what I thought. |
| 20:43.04 | tessier | bmoraca_work: That is how I have done it for ages. |
| 20:43.15 | jamko | wow another one using magic nat. This is amazing. |
| 20:43.27 | tessier | bmoraca_work: One odd thing here: The phones coming from this site keep registering to asterisk as coming from 5060. |
| 20:43.28 | bmoraca_work | tessier: what does a SIP debug show you in asterisk? |
| 20:43.32 | ChannelZ | A lot of it depends on your firewall and how it behaves |
| 20:44.00 | tessier | bmoraca_work: Let me find that pastebin... |
| 20:44.04 | bmoraca_work | tessier: that's a symptom of the firewall not working properly. are you sure you haven't made any static NAT mappings on that firewall? |
| 20:45.06 | bmoraca_work | jamko: if you knew anything about networking, you'd know it's not "magic", it's "design". |
| 20:45.16 | jamko | <PROTECTED> |
| 20:45.22 | tessier | bmoraca_work: http://pastebin.ca/1920791 |
| 20:45.33 | bmoraca_work | jamko: this is an issue with his particular firewall's implementation of NAT, not NAT itself. |
| 20:45.41 | jamko | oh sure.. magic nat is failing. |
| 20:45.44 | jamko | that is the issue. |
| 20:45.48 | Jouva | The asterisk website mentions an "Administrator's Guide" which I had to use the search function to find and didn't seem to have anything to do with settings |
| 20:46.13 | tessier | bmoraca_work: That's what I'm thinking. Somehow I've got something confused in this firewall. The asterisk system used to be here at this location. It had a one to one nat. But I have removed that. Otherwise there is nothing special here. Just a few port forwards for other services like ssh etc. |
| 20:46.15 | bmoraca_work | tessier: just FYI, fonality is a really shitty system. |
| 20:46.20 | tessier | bmoraca_work: I know. :( |
| 20:46.48 | tessier | bmoraca_work: Wasn't my choice. I typically support only asterisk systems I have compiled from source. |
| 20:46.52 | bmoraca_work | tessier: is the version of software running on that firewall any different than any of the other sites? |
| 20:47.35 | tessier | They are all Linux 2.6 kernels with recent versions of shorewall. This stuff has been stable for ages. I don't have access to the other site firewalls right now to compare exact version numbers though. |
| 20:47.47 | chazzam | Jouva: link? http://svn.asterisk.org/svn/asterisk/branches/1.6.2/configs/sip.conf.sample |
| 20:47.53 | jamko | bmoraca: I suppose Mark Spencer knows nothing about networking either, which is why he created the IAX protocol, as an alternative to the nightmare of nat + sip.. but please keep making yourself look like an arrogant fool.. : ) |
| 20:48.55 | bmoraca_work | tessier: retransmits mean that the packets aren't getting back from the PBX to the phone. the fact that the phone appears to register from port 5060 means one of two things: 1) a SIP ALG is running on the firewall, or 2) there is still a static mapping on the firewall |
| 20:49.48 | bmoraca_work | jamko: IAX was created for a variety of reasons. not the least of which was bandwidth consumption. its ability to run both signaling and media over one port is incidental to its benefits. |
| 20:49.56 | tessier | bmoraca_work: I understand about the retransmits. I am trying to figure out the other two things. Initially the sip nat helper modules were loaded. I unloaded them, configured it not to reload them, and have even rebooted the firewall. /sbin/lsmod |grep sip produces nothing now. |
| 20:50.49 | bmoraca_work | tessier: just for shits and giggles, set nat=no in sip.conf and re-enable the NAT helper |
| 20:50.50 | tessier | bmoraca_work: I took these phones home last night behind my Linksys WRT54GL and one still registered as 5060 and the other as 1024. |
| 20:50.53 | tessier | bmoraca_work: Very weird. |
| 20:51.17 | bmoraca_work | tessier: that's common for SOHO routers...asus routers work the same way |
| 20:51.21 | tessier | bmoraca_work: So the strange port problem moved even when I changed locations/firewalls. |
| 20:51.36 | tessier | bmoraca_work: WRT54GL runs Linux. I flashed it with dd-wrt or some such ages ago. |
| 20:51.42 | bmoraca_work | tessier: i wouldn't associate the two. most SOHO routers will register the first device translated to 5060 |
| 20:52.06 | tessier | hmm....I guess that makes sense. |
| 20:52.11 | tessier | Keep the same source port consistent. |
| 20:52.22 | tessier | For the first one anyway since it is possible to do so. |
| 20:52.24 | bmoraca_work | tessier: maybe it's the design of IPtables, then. i tend to stay away from linux-based "appliance" routers |
| 20:52.26 | Jouva | chazzam: Thanks, though I think documentation for this configuration is still a good idea, since for example seeing "Disable overlap dialing support. (Default is yes)" as the only piece of info for "allowoverlap" doesn't tell me, a new user, very much |
| 20:52.40 | Jouva | Not that I'm asking about that specific setting. Just an example |
| 20:52.48 | tessier | bmoraca_work: I do too. But it was cheap/easy for home. Everywhere else I have full blown servers running Linux/shorewall. |
| 20:53.22 | bmoraca_work | tessier: it was those "full blown servers" i was referencing :) don't really care what's at home |
| 20:54.23 | bmoraca_work | tessier: either way, i'd attempt to try and use the NAT helper. some of them actually do work...Adtran's works OK for about 10 phones. |
| 20:54.34 | *** part/#asterisk Gary_B (~IceChat7@85.211.217.168) |
| 20:55.02 | bmoraca_work | tessier: but when you do that, you need to make sure that Asterisk is set to nat=no and that the phone is not set to think it's behind a NAT (important for Cisco phones, not sure about Aastras, though) |
| 20:55.23 | tessier | bmoraca_work: I have set nat=no and done a reload. And reloaded the nat helper modules. Restarting phones now. |
| 20:55.26 | tzanger | win 15 |
| 20:55.49 | *** part/#asterisk jeffg (~jeffg@pdpc/supporter/professional/jeffg) |
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| 20:56.42 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 20:56.45 | tessier | bmoraca_work: I generally use Snom phones and have never had this problem with them. I've used some Cisco phones also but having to reflash them with what amounts to pirated SIP firware was a pain. |
| 20:56.49 | tessier | firmware |
| 20:57.24 | bmoraca_work | running 30 phones through Adtran's SIP ALG on the lowest-end Adtran router was fun...didn't work very well :) but up to about 10 phones, it worked great |
| 20:57.38 | bmoraca_work | tessier: check out the Cisco 500 series phones...they're pretty slick |
| 20:58.01 | tessier | bmoraca_work: Now not even one phone will register. |
| 20:58.18 | *** part/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 20:58.37 | bmoraca_work | tessier: you may have a corrupt firewal config. any chance of reloading it and starting from scratch? (matching version number to a working one, of course) |
| 20:58.58 | tessier | Sending to 192.168.3.197 : 5060 (non-NAT) |
| 20:59.07 | tessier | Almost looks like the nat helpers aren't doing a thing. |
| 20:59.13 | bmoraca_work | your NAT helper's broken |
| 20:59.32 | *** join/#asterisk zircote (~zircote@64.74.105.240) |
| 20:59.32 | bmoraca_work | it may be what's causing your problems |
| 21:00.01 | tessier | bmoraca_work: Corrupt firewall config? There's nothing in shorewall but a masquerade/nat line and a few port forwards for ssh etc. I have rebooted the box already just in case something was corrupt in memory. |
| 21:00.02 | bmoraca_work | even Cisco makes those mistakes...for instance, 8.23 ASA firmware has issues with SIP and NAT while 8.22 works perfectly |
| 21:00.28 | bmoraca_work | tessier: not just config, but actual program. or could be a bug. |
| 21:00.32 | tessier | Last option is to setup a VPN between this site and the phone system. But I would prefer not to have these phones configured differently. |
| 21:00.49 | tessier | bmoraca_work: It's shorewall generating iptables rules. It's possible I suppose. Would be shocking though. |
| 21:01.05 | bmoraca_work | tessier: shit happens |
| 21:01.49 | bmoraca_work | like I said, i've seen things where one version number off causes issues |
| 21:06.38 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 21:14.59 | jamko | tessier: have you tried using a STUN server with these phones? |
| 21:17.00 | Jouva | ChannelZ: Dunno if you remember from yesterday, but I was having issues with the 't' exten in my dialplan script |
| 21:17.16 | *** join/#asterisk zircote (~zircote@64.74.105.240) |
| 21:19.50 | Jouva | Just saw something very odd. I got it to work, but only when I did a sip:(hostname) with no extension from my softphone. Trying to use my landline phone to call my GV number that redirects to the Gizmo5 account that my asterisk config links up with, it fails to make use of the 't' exten and instead exits with non-zero on a non-existing priority line in the 's' exten |
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| 21:32.56 | shido6 | thats what happens Jouva when you dont use an extension |
| 21:33.08 | shido6 | so setup an 's' exten or choose a name or number to use |
| 21:33.19 | Jouva | shido6: That's what I have |
| 21:33.25 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:33.48 | shido6 | cool |
| 21:33.51 | shido6 | keep it movin' |
| 21:33.52 | shido6 | :0 |
| 21:33.56 | shido6 | :) |
| 21:35.44 | Jouva | Here's what I have: I want to prompt for an extension 3 times before it times out and hangs up. So I do a set a var (COUNT) to 3. I call Background() followed by WaitExten(10) in exten 's'. In exten 't' it decrements COUNT and goes to goodbye in exten 't' if COUNT is 0 (says goodbye, hangs up). Otherwise it goes back to the Background() call in exten 's' |
| 21:39.36 | Letoric | hello folks. Having a bit of an issue on recieving calls from a cisco call manager into asterisk |
| 21:39.54 | Letoric | it doesn't seem to be identifying the incoming call as coming from the pstn that I have in sip.conf |
| 21:40.17 | Letoric | I added the loopback address of the call manager to be on the safe side in case it was originating calls from that IP instead of the primary |
| 21:40.27 | Letoric | both peers show as registered with sip show peers |
| 21:40.31 | Letoric | Any ideas? |
| 21:42.35 | shido6 | um |
| 21:42.46 | shido6 | what does sip debug say when u make calls, Letoric ? |
| 21:43.05 | Letoric | I'll fire it up and see |
| 21:43.12 | Letoric | so far I had not had a lot of good luck on debugging sip debug |
| 21:43.15 | Letoric | but I will try ;) |
| 21:45.41 | Letoric | <--- Reliably Transmitting (no NAT) to 10.254.250.30:5060 ---> |
| 21:45.41 | Letoric | SIP/2.0 404 Not Found |
| 21:45.41 | Letoric | Via: SIP/2.0/UDP 10.254.250.30:5060;branch=z9hG4bK4C62388;received=10.254.250.30 |
| 21:45.41 | Letoric | From: <sip:9726936924@10.254.250.30>;tag=6E297BCC-DDA |
| 21:45.41 | Letoric | To: <sip:2142690746@69.94.238.61>;tag=as000003a4 |
| 21:45.52 | shido6 | dont let it intimidate you - just read it line by line - and before you know it - you'll be turning the txt to green and reading it full screen and your friends will think you are reading the matrix |
| 21:45.55 | shido6 | and use pastebin.ca |
| 21:46.03 | shido6 | btw thats a good thing, 404 not found |
| 21:46.12 | Letoric | oh ok heh |
| 21:46.17 | shido6 | no look closer - what context is it not finding your number in |
| 21:46.20 | shido6 | no = now |
| 21:46.39 | shido6 | whispers to himself... "wait for it...." |
| 21:46.46 | Letoric | I don't see the context |
| 21:46.48 | shido6 | "wait for it" |
| 21:47.01 | shido6 | how about searching for the word context in your debug |
| 21:47.24 | Letoric | it says 'cannot find context' |
| 21:47.25 | Letoric | ;p |
| 21:47.39 | shido6 | ast box is 10.254.x.x, right? |
| 21:47.46 | shido6 | what is 69.94.x.x. ? |
| 21:47.56 | Letoric | that's the ast box |
| 21:48.00 | Letoric | 10.254.x is cisco call manager |
| 21:48.46 | shido6 | turn on debug, make note of where your debug starts - make a test call - paste the entire sip debug into pastebin.ca |
| 21:49.46 | Letoric | I'm debugging to a console |
| 21:49.49 | Letoric | is there a better way? |
| 21:50.13 | shido6 | this is fine - cut & paste my friend - but paste into pastebin.ca |
| 21:51.37 | Letoric | http://pastebin.ca/1921659 |
| 21:54.41 | Letoric | http://pastebin.ca/1921662 |
| 21:54.44 | Letoric | that might be a little better |
| 21:55.04 | Letoric | still no context though |
| 21:55.11 | [TK]D-Fender | Letoric: Looking for 2142690746 in default (domain 69.94.238.61) <- GUESS |
| 21:55.29 | [TK]D-Fender | Letoric: Looking for X in Y. Guess what Y is <- |
| 21:55.37 | Letoric | yeah |
| 21:55.52 | Letoric | ok, so why is it going to default? |
| 21:55.59 | Letoric | it's supposed to be going to incoming_calls |
| 21:56.08 | Letoric | that's how the peer is configured in sip.conf |
| 21:56.15 | [TK]D-Fender | Letoric: And you cut off the invite whick would have TOLD you |
| 21:56.20 | [TK]D-Fender | Which* |
| 21:58.00 | p3nguin | app_voicemail.c:3547 make_email_file: Sox failed to reencode |
| 21:58.02 | p3nguin | An error occurred during file processing (have you installed support for all sox file formats?) |
| 21:58.05 | p3nguin | Here's a new problem I see after my last upgrade... |
| 21:58.16 | p3nguin | Does this mean in the make menuconfig menu or what? |
| 21:59.01 | p3nguin | Of all the upgrades I have done, I have never encountered this until the upgrade to 1.4.34. |
| 21:59.25 | shido6 | Letoric: show us the way you have your peer configured - removing any passwords and use pastebin again :) |
| 22:00.07 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 22:01.30 | [TK]D-Fender | shido6: no need. Look at the CALL first |
| 22:02.27 | Letoric | D-Fender: You were correct that it's going to default - WHY is what I need to figure out now |
| 22:03.17 | [TK]D-Fender | ~wmmfpb |
| 22:03.18 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 22:03.21 | [TK]D-Fender | :D |
| 22:04.23 | russellb | you don't have enough #$@#$(*#$ in front of "ING" |
| 22:04.31 | russellb | 1 short. |
| 22:04.45 | Letoric | http://pastebin.ca/1921667 |
| 22:05.09 | Qwell | russellb: you assume it's supposed to be a 4 letter word. |
| 22:05.12 | Letoric | pstn3 has no problem coming into the right context |
| 22:05.18 | shido6 | wow |
| 22:05.21 | Letoric | it has the same configuration |
| 22:05.28 | shido6 | ~wmmfpb |
| 22:05.29 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 22:05.29 | russellb | Qwell: yes, i do. |
| 22:05.34 | shido6 | lol |
| 22:07.16 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 22:09.46 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 22:10.11 | Letoric | so, any thoughts on why it's using the default context instead of the one I set? |
| 22:16.13 | [TK]D-Fender | Letoric: Where's the real pastebin that ISN'T cut off at that crucial point for us to SHOW YOU? |
| 22:16.59 | Letoric | what crucial point? it shows the template and the peer |
| 22:17.02 | Letoric | what am I missing? |
| 22:17.08 | Letoric | did you need the general options from the file? |
| 22:17.50 | Letoric | we're running 1.6.2.7, so there is no username anymore, and you told me leave out the password |
| 22:20.13 | [TK]D-Fender | Lethe crucial part of the SIP DEBUG of your CALL |
| 22:20.56 | Letoric | oh I thought you already saw that, thats' where you told me I was going into the default context |
| 22:21.04 | Letoric | so you need another debug? |
| 22:21.11 | Jouva | So I'm poking around with MeetMe() in 1.6.2.9, which is giving the "unable to open pseudo device" message. What's the module that I need to double check that loaded and how would I check it? |
| 22:21.26 | [TK]D-Fender | Letoric: I need a COMPLETE call. |
| 22:21.40 | Jouva | Some posts told me zaptel, but something else said that 1.6 uses dahdi, but I am not sure what to check for in the logs |
| 22:21.43 | Letoric | ok |
| 22:22.37 | [TK]D-Fender | Jouva: Did you install and configure DAHDI? |
| 22:24.04 | Jouva | Welp, debian seems to say dahdi is installed. I'll have to look into this in a moment. Being dragged away for a moment for food :P |
| 22:29.08 | *** join/#asterisk CrashSys (~james@65344hfc124.tampabay.res.rr.com) |
| 22:29.21 | CrashSys | Does anyone know if Asterisk 1.4.27.1 can use Dahdi v.2.3.0? |
| 22:30.41 | [TK]D-Fender | CrashSys: Probably |
| 22:33.00 | CrashSys | Hmmm, for some reason dahdi show status isn't listing dahdi dummy |
| 22:33.53 | hardwire | people |
| 22:34.05 | hardwire | have you ever seen an SPA-941 lose it's config due to power outage? |
| 22:34.10 | hardwire | or power surge? |
| 22:34.25 | hardwire | I have two that suddenly decided they needed to be factory reset due to an outage |
| 22:39.13 | TJNII | I've never seen it, but I can believe it. |
| 22:39.58 | TJNII | Especially if it uses NAND flash |
| 22:53.57 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 22:55.33 | MrHanMan | Has anyone gotten a Cisco 7925g wifi phone to work with Asterisk? Can anyone suggest where I should start? I'm fairly new at this. |
| 23:03.25 | [TK]D-Fender | MrHanMan: have you gone and looked at the admin guides for it? |
| 23:07.13 | TJNII | Documentation is for pussies. |
| 23:08.26 | TJNII | Real admins just cut blindly through. Sure, you'll probably wind up scratched, bleeding, and missing a show on the other end, but you did it yourself! |
| 23:08.34 | TJNII | s/show/shoe |
| 23:10.45 | *** join/#asterisk nibbier (~sven@g230242226.adsl.alicedsl.de) |
| 23:12.29 | nibbier | hi. i have a mobile device connected to my asterisk (ancient version, 1.2.21) - and this asterisk is constantly sending packets to my mobile, even without any calls or such going on. can this somehow be limited/prevented, to save battery? |
| 23:13.27 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 23:13.35 | hardwire | nibbier: have you determined what type of packets? |
| 23:14.22 | nibbier | hardwire, well, i eneded the sip client on my mobile, and i still see packets orignationg from port 5060 udp going to my client |
| 23:14.46 | hardwire | nibbier: is qualify on for that peer? |
| 23:15.50 | Jouva | ok so now that I am back... |
| 23:16.08 | Jouva | [TK]D-Fender: how do I go about checking if DAHDI is working? |
| 23:16.11 | nibbier | hardwire, yes, it is. should i switch it off? |
| 23:17.47 | nibbier | hardwire, googled it, thanks for the hint |
| 23:20.05 | Jouva | hmmm |
| 23:20.43 | Jouva | No DAHDI found. Unable to open /dev/dahdi/ctl: No such file or directory |
| 23:20.47 | [TK]D-Fender | Jouva: Stop * and initialize it... |
| 23:22.25 | Jouva | How do I initialize it though. Debian says the dahdi package is installed |
| 23:25.05 | [TK]D-Fender | .... |
| 23:25.15 | [TK]D-Fender | Jouva: dahdi_cfg -vvvv |
| 23:25.59 | Jouva | Thanks for helping me, but FYI, this is my 3rd day on Asterisk and I'm not finding the best documentation on everything. It's all scattered. |
| 23:26.16 | Jouva | So sorry if I don't know everything and I'm just trying to learn stuff |
| 23:26.36 | TJNII | Jouva: See /etc/init.d/ |
| 23:26.52 | TJNII | That is something you should know as a *nix admin, anyways |
| 23:27.14 | Jouva | I know THAT much :P But how am I supposed to be aware that dahdi is a service? |
| 23:27.36 | TJNII | You mean started at boot and whatnot? |
| 23:27.43 | nibbier | hardwire, i changed the qualify to "no" - but still get tons of packets. ehre is one of them: http://nopaste.info/e41928e53d.html |
| 23:27.49 | [TK]D-Fender | Jouva: Moving on. What is the result? |
| 23:28.00 | Jouva | is 3 lines too much to paste? |
| 23:28.22 | TJNII | Cusp. |
| 23:29.22 | Jouva | http://pastebin.ca/1921710 |
| 23:29.55 | Jouva | Also, /etc/init.d/dahdi start says: FATAL: Module dahdi not found. |
| 23:30.35 | [TK]D-Fender | Jouva: There is an init.d script to start the service... |
| 23:30.39 | TJNII | modprobe dahdi |
| 23:30.53 | Jouva | TJNII: Same result as dahdi start |
| 23:31.01 | [TK]D-Fender | Jouva: That should initialize the drivers including dahdi_dummy which is likely what you need |
| 23:31.06 | Jouva | so I guess the kernel module doesn't exist |
| 23:31.57 | TJNII | wanders off |
| 23:33.35 | Jouva | Trying to see how to install the kernel module in Debian. Looks like the main dahdi package is installed but the kernel module wasn't. Seems kinda silly for Debian to do that if you ask me, but there's probably some reason for it |
| 23:34.34 | Jouva | But yeah, please don't say "you should know about /etc/init.d" when my knowledge about dahdi is that it's simply an asterisk module and not a kernel module or a system service |
| 23:35.06 | Jouva | To me it was as much a system service as vi is a system service |
| 23:40.42 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 23:55.24 | jamko | Just to confirm, is it true that odbc must be used for storage of voicemail messages in a mysql db? Using * 1.6.2.10 |