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| 03:11.58 | greenlanegreb | hello, when supplying screendumps from sip debug, what information needs to be included and what should I be looking for to make sure all of the relevant information is grabbed? |
| 03:13.29 | WIMPy | The relevant information. Whatever that is for whatever you're trying to get help with. |
| 03:13.31 | WIMPy | ~ask |
| 03:13.33 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 03:15.22 | greenlanegreb | I can't make calls - I am attempting to route calls through my server (the server is inside my home network) outside of it |
| 03:18.08 | [TK]D-Fender | What point does it fail at? |
| 03:18.29 | WIMPy | What do you use to make calls? |
| 03:19.06 | greenlanegreb | well, I've just been having a look and neither Phonerlite or Portsip will connect at all atm, they did before |
| 03:19.29 | greenlanegreb | 503 error |
| 03:19.39 | greenlanegreb | according to Portsip |
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| 03:32.33 | [TK]D-Fender | Details would help... |
| 03:32.39 | [TK]D-Fender | where is the phone relative to your server? |
| 03:32.55 | [TK]D-Fender | "sip set debug on" ,- See any traffic? |
| 03:33.00 | [TK]D-Fender | Checked firewalls? |
| 03:33.02 | [TK]D-Fender | Routing? |
| 03:34.35 | MaliutaLap | passes [TK]D-Fender some multi-path routing |
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| 03:57.45 | greenlanegreb | softphone outside of internet network of the server, no traffic recognitio on SIP Debug, checked antivirus and firewall in Windows on client laptop |
| 03:57.53 | greenlanegreb | 503 - transport error |
| 03:57.58 | greenlanegreb | client side |
| 03:58.31 | greenlanegreb | internal network* |
| 04:07.44 | greenlanegreb | any idea? |
| 04:08.01 | greenlanegreb | no transports left to try |
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| 10:34.44 | Hsilamot | hi, anyone here has configured asterisk to run on a port different from 5060? trough TCP, SIP |
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| 10:43.19 | Ice_Strike2 | To setup like a Cloud Technology for hosted Asterisks |
| 10:43.36 | Ice_Strike2 | What kind of technology can that be done |
| 10:43.53 | Ice_Strike2 | vmware like of mulitple ESXi? |
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| 11:31.49 | Zetten | I think this is more dahdi/pri than asterisk, but I'm a complete novice with telephony, so maybe someone here can at least point me in the right direction |
| 11:31.49 | Zetten | What could be causing an issue where incoming calls are rejected (unknown number_ when a number is dialed off-hook (i.e. from a dialtone) but succeed when dialed on-hook (i.e. digits dialed as a block)? |
| 11:32.57 | Zetten | From my limited debugging ability it seems like PRI is receiving a truncated version of the full number in the off-hook situation, like it's trying to connect too early, e.g. _112204 instead of _11220456 |
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| 11:57.12 | steelbrain | Anyone up? |
| 11:57.25 | deranged | nope |
| 11:57.53 | steelbrain | I am quite new to asterisk, and I have just finished creating my callshop, is there a way to check the balance and proceed only if the balance is more than 0, in dialplan extensions? |
| 11:58.52 | WIMPy | Zetten: It is both a dahdi and an Asterisk question. You need to take care of overlap dialling in either of them. |
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| 12:01.03 | Zetten | WIMPy: overlap dialing is enabled in chan_dahdi.conf, with no effect |
| 12:02.19 | WIMPy | The "Immediate"s are important. If set to yes, it has to be done by Asterisk. |
| 12:02.42 | Hsilamot | anyone here has configured asterisk to run on a port different from 5060? trough TCP, SIP |
| 12:04.16 | WIMPy | ~polls |
| 12:04.18 | infobot | "Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask> |
| 12:05.34 | Hsilamot | it's not really a poll |
| 12:05.42 | *** join/#asterisk areski (~areski@80.174.128.96.dyn.user.ono.com) |
| 12:05.48 | Hsilamot | i need to know if the problem i have is a bug or a config issue |
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| 12:09.14 | Zetten | WIMPy: so how would overlaps be handled in Asterisk? the dialplan for incoming calls is pretty simple at the moment: http://pastebin.com/JN6cNw7J |
| 12:09.47 | WIMPy | In Asterisk, you use the WaitExten application. |
| 12:10.52 | WIMPy | It's probably easier to leave it to the driver. |
| 12:11.08 | Hsilamot | where should i go to find knowledge or experience on asterisk? besides the normal "google it" channels, since it seems this is not a common issue |
| 12:11.21 | WIMPy | But the last time, I tried it, I failed to find a combination that worked in all possible cases. |
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| 12:11.52 | WIMPy | Hsilamot: We're still waiting on your description of the problem. |
| 12:12.40 | Hsilamot | well, i have asterisk listening on TCP port 6506 and same UDP, with internal network 10.0.0.0/16 |
| 12:13.21 | Hsilamot | When i make a call in the intranet everything works neatly, but when the call is perfomed on the WAN side, the SIP connection gets broken and the control over the call is lost |
| 12:13.29 | Hsilamot | meaning, no hold, no transfer, no end the call |
| 12:13.50 | Hsilamot | when the remote phone ends the call, the PBX does not know it and ends if after 31 s because of RTP timeout |
| 12:13.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:14.16 | Zetten | WIMPy: thanks very much, I'll give immediate=no a try in my next test window shortly |
| 12:14.26 | Hsilamot | after debbuging the SIP packets i found that Asterisk is sending a Contact header |
| 12:14.34 | Hsilamot | with the port 5060 in it |
| 12:14.55 | Hsilamot | and if i map the port WAN:5060 to PBX:6506 everything works |
| 12:15.36 | Hsilamot | the fact is that the pbx is telling the phone to change the communication port to another one which isn't even open, or shouldn't |
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| 12:24.09 | WIMPy | Hsilamot: I might even have read about that one before. Do you use a current version? Otherwise you can always check issues.asterisk.org for bugs. |
| 12:25.29 | Hsilamot | i have not found any issues with the Contact header so far, i think it's current,, |
| 12:26.10 | Hsilamot | Asterisk 11.16.0 |
| 12:26.19 | WIMPy | Have you thought about changing to pjsip? |
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| 12:26.48 | WIMPy | See the channel topic. |
| 12:27.56 | Hsilamot | i'm using freepbx, guess that if it's an asterisk bug i should check versions,., |
| 12:28.08 | beanie | Hello, I think i've been hacked - does this evidence a successful call connection from a hacker? pbx_spool.c:402 attempt_thread: Call completed to Local/s@tc-maint |
| 12:29.48 | WIMPy | beanie: That call was set up using a call file. And we don't know what that extension would do. |
| 12:30.46 | [TK]D-Fender | beanie: that is automatic stuff for FreePBX's scheduler. |
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| 12:31.20 | [TK]D-Fender | beanie: And remember: just because you're paranoid doesn't mean people aren't still out to get you.... |
| 12:31.47 | [TK]D-Fender | beanie: tcmaint = Time Conditions Maintenance. |
| 12:32.07 | beanie | thanks [TK]D-Fender :-p is there a file where I can check for any successful dodgy connections by ip so I can add them to my iptables ban list? |
| 12:32.28 | beanie | (either previous success or current success) |
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| 12:32.43 | WIMPy | security.log |
| 12:34.32 | beanie | thanks WIMPy where abouts is security.log |
| 12:38.54 | beanie | WIMPy, I'm using Centos, will there be one, I just issued a find request but it could not find it |
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| 12:40.37 | zekoZeko | beanie: logs are somewhere in /var/log most of the time, start looking there. |
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| 12:43.39 | eppigy | Hello |
| 12:43.42 | eppigy | I am Dave |
| 12:48.28 | beanie | jees zekoZeko there's like thousands of entries! |
| 12:48.41 | [TK]D-Fender | beanie: Are you running FreePBX's distro? |
| 12:48.48 | beanie | yep |
| 12:48.50 | beanie | :) |
| 12:49.01 | beanie | or a distro with freepbx |
| 12:49.03 | [TK]D-Fender | Then it comes with fail2ban already for this |
| 12:49.15 | [TK]D-Fender | Are you using THEIRS? |
| 12:49.25 | jkroon | hi guys, if i'm going to be tampering with voicemail database structures (eg, deleting recordings from a folder etc ...) what are the pitfalls? For example, let's say there is 5 messages in an INBOX folder, and I want to nuke number 2 - do I need to renumber the msgnum's that's larger to keep it sequential? |
| 12:49.33 | beanie | this i'm not sure of, it was a while ago I set it up [TK]D-Fender |
| 12:50.09 | beanie | [TK]D-Fender, the system's suddenly not accepting my connections |
| 12:50.40 | beanie | coming up with a transport error message from one softphone and the other softphone is having no joy, meanwhile there is a string of errors relating to codecs so goodness knows whats going on |
| 12:50.56 | beanie | I can ssh in though and i've checked my antivirus and firewall at client side |
| 12:51.39 | [TK]D-Fender | beanie: packets are making it through. Looking at firewalls & antivirus is a clear waste of time. |
| 12:51.47 | [TK]D-Fender | beanie: it is misconfigured |
| 12:52.21 | beanie | [TK]D-Fender, I have a few lines of what might be useful - about 4, should I pastebin them or paste them here? |
| 12:52.29 | [TK]D-Fender | beanie: Transport seems pretty clear-cut.... codecs won't amtter if the base call isn't accepted that far anyway. |
| 12:52.55 | [TK]D-Fender | beanie: You should have walked in here with a pastebin of the actual errors already prepared.... |
| 12:53.29 | beanie | [TK]D-Fender, going back to what you are saying about transport being clear cut |
| 12:53.40 | WIMPy | jkroon: That's what I've been wondering since VoiceMailPlayMsg was introduced. A DeleteMsg would be much more usefull. |
| 12:54.09 | jkroon | WIMPy, i'm contemplating providing my users with a web UI onto the voicemail ... |
| 12:54.30 | jkroon | the structure of that whole table seems to be pretty nasty anyway. |
| 12:54.31 | WIMPy | Maybe it's Digiums trade secret for their visual voice mail. I started such a thing as well, but scrapped it because of that question. |
| 12:54.41 | beanie | well, the first thing is that I have had problems with audio for a fair while but I have always been able to log in using a softphone, now I can't log in using softphone at all and i'm getting more codec errors that I hadn't seen before - I nor anyone else has actually done any config work on the errors since I could log in to the softphone |
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| 12:56.19 | beanie | [TK]D-Fender, http://pastebin.com/Gs4isLWP |
| 12:56.27 | jkroon | WIMPy, i have a friend that says (i'm pretty he got it somewhere else) that winners never quit, quitters never win, but those who never win, and never quit are idiots. i guess i'm bordering on the latter but so far I've mostly ended up not needing to quit. |
| 12:56.35 | jkroon | just takes longer some times than others. |
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| 12:57.26 | WIMPy | I have a strong dislike for things that work most of the time. |
| 12:57.49 | jkroon | but not always ... agreed. |
| 12:57.55 | WIMPy | You could reverse engeneer what Digium does. But you never know if it might change. |
| 12:58.10 | [TK]D-Fender | beanie: There is no verbose in there and no SIP debug. You should already know that's not good |
| 12:58.52 | [TK]D-Fender | beanie: And the call is getting accepted and playing back the kind of message FreePBX does when there is no inbound or outbound route to match whatever is being dialed. |
| 12:59.00 | jkroon | there was a point where asterisk screwed it's own vm folder structure ... eventually ended up with a vs_remove_repair_vm script that repaired the structure (specifically fixing the numbers to be sequential) |
| 12:59.21 | WIMPy | jkroon: Maybe the best solution would be to do your own VoiceMail and not use the provided splution. |
| 12:59.24 | beanie | [TK]D-Fender, i'm a bit overwhelmed and not sure what to look into further see and even how to look into those issues further |
| 12:59.39 | [TK]D-Fender | beanie: And failing at that for what is either a "sound files are missing", or "codec modules not loaded and can't translate" |
| 12:59.49 | jkroon | WIMPy, yea, possibly - but that really does seem like a LOT of work. |
| 13:00.14 | [TK]D-Fender | beanie: "core set verbose 10", "sip set debug on" <- These are the basics and you've been here tons of times. |
| 13:00.21 | jkroon | and the problem got fixed ... haven't needed that script in over 2 years ... |
| 13:00.24 | beanie | [TK]D-Fender, At the moment I can't connect in with my softphone to trigger any errors... |
| 13:00.31 | WIMPy | jkroon: Might be less than trying to figure out how to make it water tight with th existing one. |
| 13:01.17 | jkroon | WIMPy, so how do you go about building your own "sensible" voicemail? do you write yet another app_myvoicemail.c thing? Or is there components you can misuse? |
| 13:01.53 | Zetten | WIMPy: immediate=no doesn't seem to resolve my truncated number problem :( |
| 13:02.04 | WIMPy | Why make it an application? It souldn't be too hard to do it in the dilplan with the simpler applications. |
| 13:02.28 | WIMPy | You just might have to use something external to sort the messages. |
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| 13:04.41 | jkroon | that's the problem, it's not exactly simple either. i've thought about it before. but there is a lot of things to deal with ... and the existing system deals with everything i need/want, except for some sensible web ui interface. |
| 13:04.45 | WIMPy | Zetten: Did you restart Asterisk? |
| 13:04.54 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:04.58 | jkroon | even if for now I only do a "read-only" interface thing... |
| 13:05.11 | [TK]D-Fender | [09:00]beanie[TK]D-Fender, At the moment I can't connect in with my softphone to trigger any errors... <- what is that CLI output you showed me? |
| 13:05.14 | WIMPy | jkroon: Same here. |
| 13:05.38 | beanie | [TK]D-Fender, Here's me trying to connect in to Asterisk with my softphone and not getting in - http://pastebin.com/CrsDJ5Th |
| 13:05.51 | beanie | [TK]D-Fender, the stuff before was just appearing on it's own in sip set debug |
| 13:06.10 | [TK]D-Fender | beanie: there is nothing in there... there is NO SIP debug from ANYTHING at all... which I should be seeing... |
| 13:06.13 | Zetten | WIMPy: I did a full restart of asterisk, yes |
| 13:06.41 | [TK]D-Fender | beanie: I do NOT see you enabling SIP debug in there. |
| 13:07.07 | WIMPy | Zetten: Hmm. :-( Well, maybe just try the other way riond with immediate=yes and placing a WaitExten in your dialplan. |
| 13:07.42 | jkroon | does anybody know what the flag column in the voicemail table represents? |
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| 13:10.51 | beanie | [TK]D-Fender, There you go http://pastebin.com/AMWDEaZJ |
| 13:10.53 | beanie | thanks :) |
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| 13:22.42 | [TK]D-Fender | beanie: No matching peer for '510' from '199.48.164.51:5095' |
| 13:23.23 | beanie | thats a hacker |
| 13:23.28 | [TK]D-Fender | is it? |
| 13:23.41 | beanie | yep, not my client ip, not the server ip and there's noone else |
| 13:23.51 | beanie | both client and server external IP's are static |
| 13:23.52 | [TK]D-Fender | So firewall them off |
| 13:24.07 | beanie | i'm forever doing it - I'm getting bombarded with the buggers! |
| 13:24.45 | Zetten | WIMPy: Just to make sure I'm getting this right (WaitExten docs seem a bit sparse, and I'm hardly proficient at dialplans or any of this), could you please check my context? http://pastebin.com/JN6cNw7J |
| 13:24.47 | beanie | does it shed any light on my connectivity issues [TK]D-Fender |
| 13:26.59 | [TK]D-Fender | beanie: Packets are coming in from outside your LAN... so there's that... |
| 13:27.14 | [TK]D-Fender | beanie: And you still haven't told us where your softphone IS |
| 13:27.41 | beanie | [TK]D-Fender, The softphone is on my laptop which is outside of the network the server is in |
| 13:27.49 | beanie | so i'm in Coventry, Server is in Manchester |
| 13:27.53 | beanie | both are at "home" |
| 13:28.07 | beanie | home being both of my homes |
| 13:28.45 | *** join/#asterisk CustosLimen (~CustosLim@unaffiliated/cust0slim3n) |
| 13:29.02 | [TK]D-Fender | beanie: Apparently packets make it from the outside to your server from these hackers. Means that half is fine. |
| 13:29.07 | [TK]D-Fender | beanie: Guess which half is left? |
| 13:29.51 | [TK]D-Fender | beanie: [some/every]thing on your client side is at fault |
| 13:30.33 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 13:30.53 | WIMPy | Zetten: You will see in you logs where te call goes. Depending on the switch it can be the fixed part of your numbers or it could be noting which is the s extension for Asterisk. |
| 13:31.13 | beanie | [TK]D-Fender, How would you suggest I investigate further, I have checked Firewall and Antivirus at Clientside.. |
| 13:34.04 | *** join/#asterisk DivideBy0 (~DivideBy0@unaffiliated/divideby0x0) |
| 13:35.20 | [TK]D-Fender | beanie: How about their ROUTER. Check the ip/host you put for the server. Check the DNS. Check your routing. |
| 13:38.15 | *** join/#asterisk mort_gib (~mjensen@128.216.208.178.dsl.dynamic.gibconnect.com) |
| 13:40.38 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 13:45.12 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:45.16 | *** join/#asterisk cyford (~cyford@c-73-207-183-115.hsd1.ga.comcast.net) |
| 13:51.26 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:53.29 | *** join/#asterisk farmorg (~farmorg@li169-46.members.linode.com) |
| 14:03.51 | *** join/#asterisk kharwell (kharwell@nat/digium/x-rnuuhbbotfgmuorr) |
| 14:06.50 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:06.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:07.03 | *** join/#asterisk maddawg2 (~maddawg@198.50.152.249) |
| 14:07.56 | maddawg2 | hey all... first time using asterisk and i had a couple questions regarding setting up an asterisk server in a business environment |
| 14:08.17 | maddawg2 | currently our company s using asterisk 1.8 at other locations and are using SIP trunking over T1 |
| 14:08.34 | maddawg2 | I'm looking at replacing an aging avaya system with astersk |
| 14:09.02 | maddawg2 | we do have a T1 line with SIP trunking with our carrier but i was looking at something called broadvoice... does anyone ave experience with this? |
| 14:09.11 | maddawg2 | they claim to do SIP trunking on any internet connection |
| 14:09.19 | maddawg2 | we have a 100mbps symmetrical fiber line |
| 14:09.34 | maddawg2 | but it's with verizon fios and they dont do SIP Trunking |
| 14:10.23 | maddawg2 | can someone explain to me which options is best for a place with about 60 phones |
| 14:10.29 | maddawg2 | each with a DiD |
| 14:11.07 | maddawg2 | secondly if I decide to go with asterisk over broadvoice for the business.. can I use a Virtual Machine for that |
| 14:11.12 | maddawg2 | or do I need special cards? |
| 14:11.30 | maddawg2 | i'd be using linksys SPA phones and a couple polycoms |
| 14:11.46 | maddawg2 | i think SPA942s.... could be wrong about model number |
| 14:12.14 | beanie | [TK]D-Fender, Everything client side is fine |
| 14:12.22 | maddawg2 | i'd like to dump our T1 line as it's a bit pricey |
| 14:12.29 | WIMPy | You only need real hardware if you want to connect real hardware, like lines or phones. |
| 14:12.29 | beanie | i'm currently SSH'ing in Client Side |
| 14:12.50 | maddawg2 | WIMPy, was that directed at me? |
| 14:12.55 | lvlinux | maddawg2: virtual machine is fine if you don't have/need hardware. |
| 14:13.03 | WIMPy | maddawg2: yes |
| 14:13.04 | maddawg2 | well i do have phones |
| 14:13.10 | maddawg2 | they are IP based phones |
| 14:13.18 | maddawg2 | and POE switches |
| 14:13.24 | lvlinux | hardware as in T1 stuff |
| 14:13.27 | maddawg2 | oh i see |
| 14:13.33 | lvlinux | do your switches support QoS? |
| 14:13.34 | WIMPy | You only need a network connection for them. |
| 14:13.40 | maddawg2 | lvlinux, yes |
| 14:13.51 | lvlinux | then the whole thing should be good. |
| 14:13.53 | maddawg2 | also in regards to calling outside |
| 14:14.13 | maddawg2 | should i stick with T1 or do you think using an exisitng FiOS or cable connection is good |
| 14:14.25 | maddawg2 | on one site we have a 20mbps upload and 150mbps download |
| 14:14.29 | maddawg2 | other location has a 100/100 |
| 14:14.34 | maddawg2 | both with static IP |
| 14:14.59 | lvlinux | as long as your connectoin is good then it should work fine. |
| 14:15.00 | maddawg2 | right now they both also have a T1 line |
| 14:15.10 | maddawg2 | but what about QoS outside ? |
| 14:15.23 | maddawg2 | does QoS exist on the internet lol |
| 14:15.34 | lvlinux | haha yes and no, but mostly no |
| 14:15.36 | file | no. |
| 14:15.58 | maddawg2 | so it could be troublesome then with 60 phones then i imagine |
| 14:16.00 | WIMPy | I would definitely keep any real lines for as long as possible. |
| 14:16.33 | maddawg2 | cuz i do have machine i can use as a server and I happen to have an extra digium T1 card |
| 14:16.33 | lvlinux | It might be difficult to get tuned up properly, but should be fine once you get everything set. |
| 14:16.58 | maddawg2 | but i was gonna try a VM first with broadvoice and see how it performs |
| 14:17.04 | lvlinux | You can keep your T1 for now, add VoIP, and test. |
| 14:17.19 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-tzhaqjdbldepkkla) |
| 14:17.19 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:17.20 | lvlinux | Simwood offers QoS on their network. |
| 14:17.26 | maddawg2 | simwood? |
| 14:17.34 | maddawg2 | what's that? |
| 14:17.38 | lvlinux | But they are pricey. |
| 14:17.49 | lvlinux | It's a voip provider like broadvoice. |
| 14:18.03 | maddawg2 | ah ok |
| 14:18.15 | maddawg2 | yea i'm kinda just figuring out the whole billing thing |
| 14:18.25 | maddawg2 | broadvoice needs to know how many minutes we use outgoing |
| 14:18.28 | *** part/#asterisk mjordan (mjordan@nat/digium/x-opqkhjoqfvesayjl) |
| 14:18.29 | maddawg2 | so I have to calculate that |
| 14:18.39 | maddawg2 | plus the fact that i want international |
| 14:18.57 | maddawg2 | and DiDs for every phone |
| 14:19.01 | lvlinux | hmm, you maybe should check out other providers too. There are many good and reputable ones. |
| 14:19.14 | lvlinux | look at Flowroute |
| 14:19.16 | maddawg2 | nice good idea |
| 14:19.22 | maddawg2 | see this is why i came here :-) |
| 14:19.23 | zekoZeko | maddawg2: you don't need to use the same provider for outgoing calls that you use for your DIDs |
| 14:19.35 | lvlinux | yup |
| 14:19.44 | zekoZeko | maddawg2: or you can mix&match based on rates for the destination you're calling |
| 14:19.45 | maddawg2 | really zekoZeko ? |
| 14:19.52 | lvlinux | yup |
| 14:19.54 | zekoZeko | maddawg2: really. |
| 14:19.57 | maddawg2 | that sounds complicated |
| 14:19.58 | maddawg2 | lol |
| 14:20.00 | maddawg2 | but cool |
| 14:20.02 | lvlinux | not really |
| 14:20.07 | zekoZeko | it sounds flexible to me. |
| 14:20.17 | maddawg2 | good to know |
| 14:20.19 | lvlinux | just simple dialplan stuff mostly |
| 14:20.31 | lvlinux | i mean not complicated---it _is_ cool :-) |
| 14:20.34 | maddawg2 | i'm going to try a VM first i guess |
| 14:20.43 | maddawg2 | since that's easy to spin up |
| 14:20.55 | maddawg2 | and set up a VoIP carrier |
| 14:20.59 | lvlinux | go for it. |
| 14:21.28 | maddawg2 | so next... has some created an asterisk 13 virtual appliance or should i just build one from scratch? |
| 14:21.38 | maddawg2 | other locations use a ubuntu 14.04LTS box |
| 14:21.46 | maddawg2 | dedicated hardware with dedicated T1 card |
| 14:22.02 | maddawg2 | if it works out using VoIP here we may switch everyone else out over the course of the next 2 years |
| 14:22.10 | lvlinux | I would do scratch |
| 14:22.15 | lvlinux | but that's me |
| 14:22.16 | maddawg2 | perfect |
| 14:22.20 | maddawg2 | thanks for the help guys :-) |
| 14:22.32 | maddawg2 | i will try that today then :-) |
| 14:22.33 | zekoZeko | maddawg2: get some books to read, it will get you up to speed fastest. |
| 14:22.43 | [TK]D-Fender | [10:12]beaniei'm currently SSH'ing in Client Side <- doesn't prove there isn't a firewall issue against VoIP, or that you put the right IP in the first place, or that their WAN connectivity isn't filtered/buggered up somehow, etc |
| 14:22.49 | maddawg2 | books??? what are these things you're talking about? |
| 14:22.57 | maddawg2 | you mean I cant use the google? |
| 14:23.08 | [TK]D-Fender | beanie: Your server gets traffic from hackers. that means packets work on it's side. Whatever the failure is, it's on the client's side |
| 14:23.12 | zekoZeko | yes you can, but there's a buttload of outdated advice out there |
| 14:23.21 | [TK]D-Fender | beanie: Including their internet connectivity |
| 14:23.24 | beanie | [TK]D-Fender, what checks can I do to eliminate that - i'm 99% sure it's not the issue |
| 14:23.25 | maddawg2 | yea |
| 14:23.43 | beanie | internet connectivity on the client side is not the issue, the client side is my laptop that i'm in front of, the server is up north |
| 14:23.45 | maddawg2 | got any book recommendations... seems like books would be outdated pretty quickly since they dont get updated like the interwebs does |
| 14:24.00 | [TK]D-Fender | beanie: Also prove what you have firewalled on your server, and that GW inccase something DID cause it to get blocked specifically |
| 14:24.07 | lvlinux | ~book |
| 14:24.15 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:24.25 | maddawg2 | oooo |
| 14:24.26 | maddawg2 | nice |
| 14:24.27 | lvlinux | get _the_ book ---it's freee so no excuses :-) |
| 14:24.31 | beanie | I'm alright with Windows [TK]D-Fender it's just Linux I struggle with :) |
| 14:24.31 | zekoZeko | this |
| 14:24.37 | maddawg2 | oh free |
| 14:24.45 | maddawg2 | that's etter then my school books for sure |
| 14:24.45 | maddawg2 | lol |
| 14:24.51 | maddawg2 | my math book was close to $400 |
| 14:24.52 | maddawg2 | lol |
| 14:25.31 | lvlinux | haha yep i freak out whenever I go to a bookstore---the prices are insane. Thank God for used boookstores! |
| 14:26.52 | *** join/#asterisk LooserOuting (~LooserOut@x4d0a1cd7.dyn.telefonica.de) |
| 14:27.14 | zekoZeko | maddawg2: there's also Asterisk cookbook which is not as much a reference, but it can give you some ideas on what can be done... |
| 14:27.45 | lvlinux | yes that one is good too. |
| 14:28.02 | LooserOuting | Hi. Can you use res_hep with chan_sip or only with pjsip ? |
| 14:28.25 | file | only with pjsip |
| 14:28.54 | LooserOuting | thank you for the quick answer |
| 14:29.22 | maddawg2 | lol oreilly charges nearly $50 for a free book |
| 14:29.24 | maddawg2 | yikes |
| 14:29.29 | maddawg2 | for the downloaded edition |
| 14:29.58 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 14:30.09 | maddawg2 | $46.99 |
| 14:30.11 | maddawg2 | crazy |
| 14:31.02 | trurl | maddawg2: the pdf/book has been TeXed |
| 14:34.13 | beanie | [TK]D-Fender, Nothing is firewalled on the server |
| 14:34.30 | beanie | client side, i've just knocked down Windows Firewall and AVG to see if it made a difference - negative |
| 14:35.21 | [TK]D-Fender | beanie: Well unlkess you've got a buried firewall on your server side... OTHER stuff from the outside makes it in. That only leaves some piece of your client side being at fault. |
| 14:36.55 | beanie | Well, I'm pretty much convinced that this issue is server side |
| 14:37.12 | beanie | skype works fine on client side making using of voip and steady internet traffic |
| 14:37.40 | beanie | ive ruled out that it is an issue with one softphone by trying an alternative |
| 14:37.51 | [TK]D-Fender | Skype is a totally different protocol, I've heard nothing of what router they are using |
| 14:38.13 | [TK]D-Fender | Packets MAKE IT to your server from the outside already. So shit ain't arriving at the door. that is the CLIENT's problem.' |
| 14:38.26 | beanie | and I have these issues with audio codecs that I suspect have something to do with it |
| 14:38.39 | [TK]D-Fender | Codecs don't magically make packets NOT ARRIVE |
| 14:38.52 | beanie | turning to the router config - the router was set up at a time the system worked, at least to establish a connection between softphone and remote server and hasn't been changed since |
| 14:38.55 | [TK]D-Fender | It isn't arriving at all to be refused so far |
| 14:39.29 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-avmwdjmbmbaldqbx) |
| 14:39.39 | Hsilamot | note* QoS tagging can make the packets to drop |
| 14:39.46 | [TK]D-Fender | beanie: Do you understand the concept of "empiracal evidence"? Hackers can get packet packets to your server. If you can't do the same from your location... then that's the faul of where you are and what you're using. |
| 14:41.05 | [TK]D-Fender | beanie: Comparing to Skype doesn't mean a single thing when maybe their router is screwing you over on SIP. Or their provider. Or something else you've missed. I still have no proof the softphone is set up right. Or that packets make it out of the PC itself. Where's the trace? |
| 14:42.31 | beanie | could the issue be anything to do with port forwarding? |
| 14:42.45 | [TK]D-Fender | That's routing on the client side... |
| 14:43.02 | beanie | and your saying that the issue is likely to be client side? |
| 14:43.03 | [TK]D-Fender | go prove things from the very start of the chain |
| 14:43.37 | [TK]D-Fender | HACKERS ARE GETTING SIP PACKETS TO YOUR SERVER AND ***YOU*** CAN'T. THAT'S ***YOUR*** PROBLEM. |
| 14:43.46 | [TK]D-Fender | Am I somehow not abundently clear? |
| 14:44.09 | [TK]D-Fender | If packets from SOMEWHERE on the internet DO make it to your server ... and YOURS don't ,.... it's just YOU |
| 14:44.31 | [TK]D-Fender | </captainobvious> |
| 14:45.15 | [TK]D-Fender | Start getting PROOF from the start of the chain. |
| 14:45.36 | *** join/#asterisk tparcina (~tomo@212.92.200.41) |
| 14:47.29 | beanie | [TK]D-Fender, does port forwarding need to be configured on client computer or is it not necessary? |
| 14:47.39 | [TK]D-Fender | Port forwarding is INBOUND |
| 14:47.52 | [TK]D-Fender | Doesn't stop that first packet of a request from going OUT. |
| 14:48.19 | Hsilamot | beanie: what are you trying to do i'm half aware of your trouble |
| 14:48.48 | [TK]D-Fender | beanie: Go get proof. NOW. |
| 14:48.49 | beanie | [TK]D-Fender, so I can have a working Asterisk install that does it's job with the client side being able to place calls without port forwarding being set up at all on the client router? |
| 14:48.56 | [TK]D-Fender | beanie: You are wasting time for nothing |
| 14:49.20 | beanie | Hsilamot, Softphone won't connect to remote asterisk server |
| 14:49.30 | Hsilamot | beanie: that affirmation is correct |
| 14:49.41 | [TK]D-Fender | beanie: Clients don't need forwarding if the other side has a keep-alive. And that's only for the the idea of sending a call TO the client. |
| 14:49.58 | [TK]D-Fender | beanie: this has nothing to do with your client sending OUT a request to the server. |
| 14:50.04 | Hsilamot | beanie: i give my clients Physical phones which they put in their office or home and just plug it into the network and electricity and the phone does the rest |
| 14:50.42 | [TK]D-Fender | beanie: No trafdfic = your side's fault, not the server |
| 14:50.57 | beanie | OK, so specifically [TK]D-Fender what are you asking me to do, you've been very generic |
| 14:51.18 | [TK]D-Fender | beanie: Stop giving assurances, and start providing PROOF or you are wasting everyone's time. |
| 14:51.20 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 14:51.26 | [TK]D-Fender | I don't see your SOFTPHONE config. |
| 14:51.31 | [TK]D-Fender | I don't see firewall dumps on the PC |
| 14:51.37 | [TK]D-Fender | I don't see traceroutes being run |
| 14:51.42 | [TK]D-Fender | I don't see their router config |
| 14:51.47 | [TK]D-Fender | I don't have the MODEL they are using. |
| 14:51.57 | [TK]D-Fender | You have shown NOTHING <--------------- |
| 14:52.15 | beanie | That's because you are asking for new stuff now |
| 14:52.23 | beanie | some of that, yes you did. |
| 14:52.38 | [TK]D-Fender | [10:43][TK]D-Fendergo prove things from the very start of the chain |
| 14:52.40 | Hsilamot | beanie: so start by the beggining, you use TCP or UDP on your server? |
| 14:52.43 | [TK]D-Fender | [10:45][TK]D-FenderStart getting PROOF from the start of the chain. |
| 14:52.50 | [TK]D-Fender | [10:41][TK]D-Fenderbeanie: Comparing to Skype doesn't mean a single thing when maybe their router is screwing you over on SIP. Or their provider. Or something else you've missed. I still have no proof the softphone is set up right. Or that packets make it out of the PC itself. Where's the trace? |
| 14:53.04 | [TK]D-Fender | [10:37][TK]D-FenderSkype is a totally different protocol, I've heard nothing of what router they are using |
| 14:53.21 | [TK]D-Fender | [10:23][TK]D-Fenderbeanie: Also prove what you have firewalled on your server, and that GW inccase something DID cause it to get blocked specifically |
| 14:53.22 | beanie | Hsilamot, i'm not sure whether I'm using TCP or UDP? |
| 14:53.28 | [TK]D-Fender | I've asked for TONS of things |
| 14:53.43 | Hsilamot | beanie UDP it is then |
| 14:53.47 | [TK]D-Fender | is it? |
| 14:53.58 | [TK]D-Fender | I don';t see the SOFTPHONE setup to prove what IT is configured to use |
| 14:54.11 | Hsilamot | if he hasn't changed the server's default configuration it should be |
| 14:54.19 | [TK]D-Fender | Don't care about the server right now. |
| 14:54.22 | [TK]D-Fender | Sever GETS calls |
| 14:54.23 | beanie | ok :-) |
| 14:54.49 | Hsilamot | beanie how about the phone you are using? model? |
| 14:55.31 | beanie | I've used two softphones, the first is Portsip, the second is phonerlite - neither with success recently - I've got an unexplained problem that has come about - the softphones used to connect without issue although I've always had severe audio issues |
| 14:55.48 | beanie | i've not done any config changes and nor has anyone else |
| 14:56.03 | [TK]D-Fender | moves on to more productive matters |
| 14:56.06 | Hsilamot | when they did stop working? |
| 14:56.36 | beanie | i'm not sure exactly as the system has been pretty useless over the last year but I was trying to access the system last week and first become aware |
| 14:56.55 | Hsilamot | when was the last time you could access the system successfully? |
| 14:57.19 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-sawlyzznicbwpada) |
| 14:57.22 | beanie | i've never been able to make calls successfully but I was able to log in through softphone a couple of months ago |
| 14:57.28 | beanie | now I can't even do that |
| 14:57.44 | Hsilamot | have you changed any router or hardware in that time? |
| 14:57.46 | beanie | i'm getting a 503 error on Portsip - transport issue |
| 14:57.50 | beanie | no hardware changed |
| 14:58.20 | Hsilamot | the Server has a public IP or a LAN IP? |
| 14:58.24 | beanie | I do have lots of errors with codecs which [TK]D-Fender appears to feel are irelevent to this specific issue :-) |
| 14:58.28 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 14:58.43 | Hsilamot | yeah, codec error are common on the console |
| 14:58.50 | beanie | Hsilamot, just to give you an idea of the config |
| 14:59.00 | beanie | I have a server at my flat in Manchester |
| 14:59.09 | beanie | I'm currently 100 miles South at my Mum's old place |
| 14:59.14 | [TK]D-Fender | beanie: No packets are ARRIVING at your server. How can't Asterisk tell you to "get lost" if the REQUEST never even arrives? |
| 14:59.37 | Hsilamot | ok, but that locations for now, don't seem to help |
| 14:59.39 | beanie | my "client" laptop is behind a router, the server is also behind a router, both have external static IPs |
| 14:59.47 | [TK]D-Fender | beanie: No packets = 100% client-side failure |
| 14:59.51 | Hsilamot | the question is if the Server is connected directly to the Internet or has a Router in the middle |
| 15:00.01 | Hsilamot | ok |
| 15:00.02 | [TK]D-Fender | Hsilamot: that server GETS packets from the outside |
| 15:00.03 | beanie | a router in the middle (sorry I was just getting there) :-) |
| 15:00.10 | [TK]D-Fender | Hsilamot: Told you this isn't worth it. |
| 15:00.20 | Hsilamot | How about the Server's Router's Port Forwarding |
| 15:00.24 | [TK]D-Fender | Hsilamot: IT WORKS |
| 15:00.29 | beanie | none of that has been changed |
| 15:00.37 | [TK]D-Fender | Hsilamot: IT GETS calls from the outside |
| 15:00.47 | beanie | although I should double check that the router has not lost the port forwarding details |
| 15:00.48 | [TK]D-Fender | just not from his client. |
| 15:00.50 | beanie | powercut etc |
| 15:00.52 | [TK]D-Fender | 1005 client-side |
| 15:00.59 | [TK]D-Fender | 100% |
| 15:01.32 | *** join/#asterisk mjordan (mjordan@nat/digium/x-xuahzibjztepzreq) |
| 15:01.32 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:01.32 | Hsilamot | could you check the current ports being forwarded at the server's side? |
| 15:03.33 | [TK]D-Fender | Server is fine |
| 15:03.43 | [TK]D-Fender | Hacker attempts arrive <- |
| 15:03.54 | [TK]D-Fender | no packets from YOUR client = your clien'ts problem |
| 15:04.05 | [TK]D-Fender | Discussing further is a supreme waste of timee |
| 15:04.47 | farmorg | Hi all, having a problem with MixMonitor on 11.17.0. It creates the files but never writes any audio to them. Anybody seen this before? |
| 15:05.31 | beanie | Hsilamot, Sorry i'm just having a few issues with remembering how to port forward |
| 15:06.14 | Hsilamot | beanie are you unable to check it? |
| 15:08.45 | *** join/#asterisk ghoti (~paul@75.98.206.2) |
| 15:08.58 | beanie | I can if I can remember how to do the tunnelling so I can log in to the router back at home |
| 15:10.07 | Hsilamot | beanie will it take long? |
| 15:10.31 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-qwkhkhyleffyjqoj) |
| 15:10.32 | marceloamorim | guys, I paste the configs, http://pastebin.com/298HvrFP I'm getting this message on my var/log/asterisk/messages - > "[Apr 10 11:57:42] ERROR[11992] cel_odbc.c: Unable to query database columns on connection 'asterisk'. Skipping." |
| 15:11.31 | beanie | Hsilamot, Do you know how to do port forwarding on putty so I can get the router up |
| 15:11.34 | beanie | to answer your question |
| 15:11.39 | *** join/#asterisk blinky42 (~sb@c-76-124-208-67.hsd1.pa.comcast.net) |
| 15:12.22 | Hsilamot | beanie wut? can you explain what are you trying to do exactly? |
| 15:12.39 | Hsilamot | you want to open a SSH tunnel? |
| 15:13.08 | beanie | yes |
| 15:13.51 | Hsilamot | i only remember how to open a SOCKs tunnel with ssh... |
| 15:14.12 | beanie | ahhh - im a bit stuck then - it's been so long since i've done it |
| 15:14.22 | Hsilamot | ssh -L 800:remotehost.com:80 user@example.com |
| 15:14.30 | beanie | but anyway...if I can ssh remotely into my server |
| 15:14.39 | beanie | surely the port forwarding up there is working? |
| 15:14.57 | beanie | as the router is having to route my remote request to ssh in the first place.. |
| 15:15.13 | Hsilamot | you are seeing the SSH login? |
| 15:15.29 | beanie | yeah, i've got a remote terminal window open in Putty |
| 15:15.56 | Hsilamot | so your router's admin interface is a Web GUI? |
| 15:16.01 | beanie | yeah :) |
| 15:16.08 | Hsilamot | i see |
| 15:16.52 | beanie | i seem to remember doing something using 127.0.0.1 but not sure what port it is for the router GUI |
| 15:17.00 | Hsilamot | 80 |
| 15:17.01 | beanie | I would access it normally 192.168.1.254 |
| 15:17.40 | beanie | which section of putty do I input it in |
| 15:18.07 | Hsilamot | wait i have never done that |
| 15:18.16 | Hsilamot | us normal people use VPN's |
| 15:18.24 | Hsilamot | go to Connection SSH Tunnels |
| 15:18.26 | Hsilamot | in putty |
| 15:19.05 | Hsilamot | open a new putty i mean |
| 15:19.14 | *** join/#asterisk cyford (junkmail@c-73-207-183-115.hsd1.ga.comcast.net) |
| 15:20.05 | beanie | ok |
| 15:20.14 | Hsilamot | there in source port put 8080 |
| 15:20.28 | beanie | ive found tunnels i just need to know what to put in source port and eestination |
| 15:20.29 | Hsilamot | in destination put 192.168.1.254:80 |
| 15:20.32 | beanie | destination |
| 15:20.34 | [TK]D-Fender | beanie: Your server is getting traffic fine |
| 15:20.37 | beanie | ah |
| 15:20.38 | [TK]D-Fender | There is NOTHING wrong with your server |
| 15:20.43 | [TK]D-Fender | Stop wasting time looking at it |
| 15:21.09 | [TK]D-Fender | ***I HAVE SUCCESSFULLY CONTACTED YOUR SERVER PERSONALLY *** |
| 15:21.15 | [TK]D-Fender | It gets MY packets |
| 15:21.20 | [TK]D-Fender | Your CLIENT side is the problem |
| 15:25.09 | *** join/#asterisk c0rnoTa (~c0rnoTa@109.188.127.12) |
| 15:25.22 | *** part/#asterisk c0rnoTa (~c0rnoTa@109.188.127.12) |
| 15:27.17 | beanie | Hsilamot, I'm not sure whether i'm connected to my local router or remote one |
| 15:27.30 | beanie | they both are the same manufacturer |
| 15:27.38 | Hsilamot | how about seeing the Status page? |
| 15:27.42 | beanie | i've accessed via 127.0.0.1:8080 |
| 15:27.54 | [TK]D-Fender | beanie: *I* connect fine. You server is fine. Stop wasting everyone's time |
| 15:27.55 | Hsilamot | you should be connected to the remote one |
| 15:28.07 | beanie | status tells me nothing to diffrentiate |
| 15:28.15 | Hsilamot | MAC address of the router |
| 15:30.53 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 15:31.38 | marceloamorim | yeah, I couldn't find anything that I understand and help me remove this error |
| 15:34.49 | beanie | Hsilamot, right, after a lot of faffing |
| 15:34.55 | beanie | port forward is set up on the remote router still |
| 15:35.05 | Hsilamot | which ports? |
| 15:35.31 | beanie | UDP is set to forward on port range from 5059 - 5061 |
| 15:35.45 | Hsilamot | only thoose ones? |
| 15:36.05 | beanie | it's forwarding under a separate rule on Port Range 1000-2001 |
| 15:36.15 | beanie | and 4568 - 4570 |
| 15:36.26 | Hsilamot | 1000-2001? not 10000-20001? |
| 15:36.40 | beanie | the second :) |
| 15:36.45 | beanie | sorry missed a '0' |
| 15:36.45 | Hsilamot | ok |
| 15:36.55 | [TK]D-Fender | Doesn't matter |
| 15:36.56 | Hsilamot | how about the configuration on the phone you are trying to connect? |
| 15:37.08 | beanie | it's not very complex leaving me not much by way of options |
| 15:37.14 | beanie | Portsip has always just worked as it's set up |
| 15:37.28 | [TK]D-Fender | No excuse not to prove settings |
| 15:37.35 | [TK]D-Fender | or any of the rest of that side |
| 15:37.39 | [TK]D-Fender | Server = fine |
| 15:37.42 | [TK]D-Fender | That is proven. Twice |
| 15:37.55 | Hsilamot | can you provide the current settings? |
| 15:38.28 | beanie | it's got the range of ports for audio RTP Channel as 5060 - 5060 |
| 15:38.47 | Hsilamot | who's? |
| 15:38.57 | beanie | portsip |
| 15:39.09 | beanie | softphon |
| 15:39.11 | beanie | e |
| 15:39.29 | Hsilamot | can you take a screenshot? |
| 15:39.29 | beanie | but phonerlite is configured differently out of the box, I have not changed the settings for it |
| 15:39.33 | beanie | yeah sure :-) |
| 15:39.41 | [TK]D-Fender | only asked a dozen times |
| 15:41.02 | beanie | snag.gy is going slow |
| 15:41.38 | beanie | http://snag.gy/2utmy.jpg |
| 15:41.54 | *** join/#asterisk ChkDigit (~u388mw@74.3.144.66) |
| 15:42.37 | Hsilamot | change that to 10000 - 20000 and take a screenshot of the other tabs ? |
| 15:42.44 | beanie | http://snag.gy/RJntQ.jpg |
| 15:43.10 | Synthase_ | So you're expecting SIP on 5060-61 at the server but have the phone expecting RTP on that range instead? |
| 15:43.42 | Synthase_ | Look at your phone config again. |
| 15:44.42 | beanie | Hsilamot, http://snag.gy/RJntQ.jpg |
| 15:45.10 | Hsilamot | the opther tabs? |
| 15:45.25 | beanie | http://snag.gy/lahmz.jpg |
| 15:45.58 | Hsilamot | device? |
| 15:46.09 | beanie | http://snag.gy/dKX08.jpg |
| 15:46.39 | Hsilamot | change the RTP port range and try again |
| 15:47.09 | beanie | Hsilamot, I have done, still the same issue, bear in mind that Phonerlite also cannot connect |
| 15:47.15 | beanie | different softphone |
| 15:47.34 | beanie | http://snag.gy/dejxD.jpg want to see any more tabs? :-) |
| 15:47.55 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:48.07 | Hsilamot | i need to see the one with connection details |
| 15:48.26 | beanie | There's the router - http://snag.gy/inAR8.jpg |
| 15:48.50 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 15:49.20 | Synthase_ | What happened to this: 10:33:26 <beanie> UDP is set to forward on port range from 5059 - 5061 |
| 15:49.28 | Hsilamot | when you add the account to the softphone what do you have in that tab? |
| 15:49.40 | Hsilamot | Synthase_: it seems he has already changed it |
| 15:49.42 | beanie | That's the second Router bit - http://snag.gy/XJAYw.jpg |
| 15:49.50 | beanie | Hsilamot, nothing in that tab |
| 15:49.57 | beanie | i'll show you as best as I can with another screenshot |
| 15:49.59 | Hsilamot | can you show me? |
| 15:50.25 | Synthase_ | Sure, but everything needs to play nice. The Asterisk config is unknown, so if he's listening on 5060, what's the point? |
| 15:50.51 | Hsilamot | Synthase_ the server receives calls from the outside |
| 15:51.13 | beanie | http://snag.gy/nju7b.jpg |
| 15:51.35 | Synthase_ | Set to register? SIP debug. |
| 15:51.45 | beanie | ignore the "logout" option - the softphone isn't greatly designed, it's not actually logged in |
| 15:51.52 | Hsilamot | ahm |
| 15:51.55 | Synthase_ | No point poking in the dark, find the problem first. |
| 15:51.56 | Hsilamot | beanie.... |
| 15:52.06 | Hsilamot | can you please verify the IP address of your server? |
| 15:52.09 | Hsilamot | 545? |
| 15:52.26 | beanie | I just changed the ip address as I was publishing it |
| 15:52.28 | beanie | the ip is deff correct |
| 15:52.38 | beanie | it's static |
| 15:52.58 | Hsilamot | what's in advanced |
| 15:53.46 | beanie | advanced where? |
| 15:53.59 | Hsilamot | bottom left of that config window |
| 15:55.02 | beanie | in the softphone? Do you mean MISC |
| 15:55.35 | Hsilamot | no |
| 15:55.38 | Hsilamot | tha last screenshot |
| 15:55.41 | Hsilamot | had an advanced button |
| 15:55.52 | Hsilamot | goes to attend the door |
| 15:57.00 | beanie | Hsilamot, http://snag.gy/tFFce.jpg |
| 16:05.09 | beanie | in Phonerlite, how do I display the debug information |
| 16:08.11 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 16:14.25 | [TK]D-Fender | grabs some popcorn |
| 16:16.18 | Hsilamot | do you have access to your pbx server? |
| 16:16.31 | eppigy | lol |
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| 16:19.52 | Hsilamot | destination is your server's local address and port |
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| 16:31.26 | Hsilamot | do you see "TEST STRING" on the logs' |
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| 17:07.40 | marceloamorim | yeah, I tried other things here and nothing =( |
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| 17:29.16 | beanie | hello, i'm still testing about my previous issue, does anybody know of a softphone in fedora that works well |
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| 17:48.59 | marceloamorim | http://pastebin.com/298HvrFP could you guys put me on the right path =) |
| 17:50.01 | *** join/#asterisk qakhan (~qakhan@50-204-254-12-static.hfc.comcastbusiness.net) |
| 17:50.04 | F-G0z | Hello guys. I'd like to display a html page containing callers name for example. I think I'll use PHP. I'm really a newbie. How to proceed with Apache? |
| 17:50.15 | [TK]D-Fender | marceloamorim: I don't see any error message in there anywhere... |
| 17:50.40 | [TK]D-Fender | F-G0z: We don't do Apache support here. Try in #apache |
| 17:50.47 | marceloamorim | [Apr 10 11:57:42] ERROR[11992] cel_odbc.c: Unable to query database columns on connection 'asterisk'. Skipping. |
| 17:51.05 | *** join/#asterisk CeBe (~CeBe@port-92-200-46-125.dynamic.qsc.de) |
| 17:51.22 | [TK]D-Fender | marceloamorim: Where are the * odbc configs? |
| 17:51.49 | marceloamorim | one sec, I'll paste bin |
| 17:51.53 | robmal | I'm pretty sure you need to give full path to .so in odbcinst.ini |
| 17:52.27 | [TK]D-Fender | robmal: I'd suspect the same... |
| 17:53.01 | [TK]D-Fender | marceloamorim: You show yourself directly connecting to MariaDB, but not via ODBC there as well |
| 17:56.10 | *** join/#asterisk sparetire (~sparetire@unaffiliated/sparetire) |
| 17:56.45 | F-G0z | [TK]D-Fender: In fact my issue is not really related to Apache. It's just whether AGI can be used with any php script? It's not really clear to me. |
| 17:57.11 | marceloamorim | when I use the odbc show all the connected yes isn't mean that I try connect via odbc? |
| 17:57.33 | [TK]D-Fender | your PHP script has to respect the use of the interfaces AGI uses. What it does beyond that does not matter. |
| 17:57.48 | [TK]D-Fender | F-G0z: And that'd be a PHP issue, not an Apache one at all. |
| 17:58.31 | robmal | marceloamorim: isql -v dsn username password |
| 17:58.32 | [TK]D-Fender | marceloamorim: Please show what was requested |
| 17:58.55 | [TK]D-Fender | marceloamorim: robmal just gave you the syntx for part of it. |
| 17:59.46 | marceloamorim | [IM002][unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 17:59.46 | marceloamorim | [ISQL]ERROR: Could not SQLConnect nice |
| 17:59.57 | marceloamorim | I could go from there |
| 17:59.59 | marceloamorim | thx guys |
| 18:00.59 | marceloamorim | actually, was my syntax error haha sorry |
| 18:01.34 | marceloamorim | I could connect using this syntax |
| 18:04.20 | qakhan | i have call transfer problem, when i use t option in Dial() it does not work. but when i use t with m option then call transfer works. |
| 18:04.55 | qakhan | exten => _30XX,1,Dial(SIP/${EXTEN},25,t) does not work |
| 18:05.08 | qakhan | exten => _30XX,1,Dial(SIP/${EXTEN},25,tm) works |
| 18:06.17 | qakhan | here is config |
| 18:06.18 | qakhan | <PROTECTED> |
| 18:06.26 | qakhan | http://pastebin.com/jCpD2uwH works |
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| 18:37.35 | qakhan | any update? |
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| 20:28.37 | robmal | Any heavy polycom users who would like to test my little support webapp? |
| 20:33.54 | robmal | Also, is there any way to find out which position in the menu is the ScreenCapture feature? |
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| 22:49.06 | JeffC_NN | How can I achieve Auto Gain Correction (or any volume leveling) on G711 ulaw or siren14? |
| 22:49.46 | JeffC_NN | oops, g722 not siren14 |
| 22:51.40 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Function_AGC |
| 22:52.28 | JeffC_NN | I thought Function_AGC only worked on speex? |
| 22:53.28 | [TK]D-Fender | is that .. a question? |
| 22:53.51 | JeffC_NN | have you experienced it working on non-speex codecs? |
| 22:53.57 | [TK]D-Fender | I'm not sure what there is to "think" about there. It says very explicitly what it does |
| 22:54.34 | [TK]D-Fender | " It is primarily intended for use with analog lines," <- NOthing about this looks vaguely like speex |
| 22:54.59 | JeffC_NN | I know voip.info is officially denounced, but I saw this there and it concerned me: http://www.voip-info.org/wiki/view/Asterisk+func+speex |
| 22:56.11 | JeffC_NN | I'll begin testing with AGC on various codecs. Thanks for your help! |
| 22:56.59 | [TK]D-Fender | Created by: JustRumours, Last modification: Thu 30 of Apr, 2009 (14:30 UTC) |
| 22:57.19 | [TK]D-Fender | 5 year old crappy page on a questionable resource. |
| 22:57.33 | JeffC_NN | heh. very true |
| 22:57.37 | [TK]D-Fender | "JustRumours" |
| 22:57.55 | [TK]D-Fender | Alreays refer to the OFFICIAL Wiki and docs |
| 22:57.59 | [TK]D-Fender | always* |
| 22:58.01 | JeffC_NN | very specific (albeit old) rumors, heh |
| 22:58.26 | JeffC_NN | Just wish the official ones were a little..... more descriptive. oh well. If i'm not contributing I shouldn't complain |
| 22:59.02 | JeffC_NN | Thanks again for your help! |
| 23:00.32 | [TK]D-Fender | <[TK]D-Fender> https://wiki.asterisk.org/wiki/display/AST/Function_AGC <- very descriptive |
| 23:00.55 | [TK]D-Fender | " It is primarily intended for use with analog lines, but could be useful for other channels as well." <- not just "work, but implied as "useful" |
| 23:02.44 | JeffC_NN | [Apr 10 16:02:21] ERROR[23920][C-00000bd1]: pbx.c:4390 ast_func_write: Function AGC not registered |
| 23:03.12 | JeffC_NN | Asterisk 11.8.1 built by root @ Asterisk on a x86_64 running Linux on 2015-02-21 19:01:08 UTC |
| 23:03.45 | JeffC_NN | maybe I missed it in menuselect? |
| 23:04.56 | JeffC_NN | I do see that PITCH_SHIFT and VOLUME are registered though. I'll check my menuselect.... |
| 23:06.49 | JeffC_NN | hahaha!! I can't install "func_speex" since speex isn't installed. Looks like AGC and "noise reduction" depend on speex and speex_preprocess |
| 23:07.27 | *** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de) |
| 23:08.53 | [TK]D-Fender | So depends on CODE included with it... but whose use isn't related to actual use of the codec in your channel; |
| 23:09.15 | JeffC_NN | yep. confusing, huh! |
| 23:14.23 | *** join/#asterisk elguero (~miguel323@2001:470:1f06:12c4::2) |
| 23:16.29 | JeffC_NN | I glanced at the installation pages of the Asterisk book, and I didn't see this covered, so I'm wondering if I can run `make install` with Asterisk running and then restart the service? or will `make install` fail and I should stop it first? |
| 23:17.12 | [TK]D-Fender | Should work fine |
| 23:17.36 | JeffC_NN | yay! Thx for the tip. Linux rocks! :) |
| 23:18.13 | [TK]D-Fender | I prefer to think of this as Lazy Asterisk Coding ;) |
| 23:18.49 | JeffC_NN | convenient laziness! |
| 23:20.07 | JeffC_NN | In my ignorance of the inner workings of Channel Functions I optimistically ran 'core reload', hoping I could avoid an interruption to calls, but alas it didn't work, heh. |
| 23:26.56 | *** join/#asterisk Navion (~Navion@rivendell/users/navion) |
| 23:28.58 | [TK]D-Fender | No way any good can come from that :) |
| 23:29.07 | [TK]D-Fender | I said you should be able to INSTALL over it |
| 23:29.23 | [TK]D-Fender | Then you'd still just manually load the "new" modules that weren't currently running |
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