| 00:31.46 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 00:40.31 | *** join/#asterisk citec (~citec@187.58.109.76) |
| 00:40.53 | citec | hi |
| 00:41.19 | citec | is there a wat to specify the number of digits to wait in an extension, by a variable, like this: |
| 00:41.54 | citec | exten => _X{${numExpected}},1,NoOp() |
| 00:41.55 | citec | ??? |
| 00:42.46 | *** join/#asterisk RaNa (~WinNT@174.134.251.95) |
| 00:43.19 | citec | I mean "any way" to do this |
| 00:43.21 | citec | sorry |
| 00:44.46 | ruied | don't think so... It seems that the sip phone send all digits at once, than they will be matched (all at once) or not in your dialplan... |
| 00:45.10 | citec | ruied, aha, but if I put: |
| 00:45.15 | citec | exten _XX,1,.... |
| 00:45.21 | citec | it waits just for 2 |
| 00:45.31 | citec | and if I put XXX it waits for 3 |
| 00:45.34 | ruied | yes |
| 00:45.45 | citec | so I dont think the ip phone send everything together |
| 00:45.56 | citec | cuz just when I press 2 digits, if I have XX it continues |
| 00:46.40 | ruied | the ip phone sends all at once. Once it sends, the number will be 'matched' or not in you'r dialplan... |
| 00:46.41 | citec | may be I should use the read function instead |
| 00:47.07 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 00:47.35 | RaNa | We have a 1800 number and I want to forward it directly to an extension on the pbx box instead of forwarding to a number and wasting minutes. How can I do this? The reason im asking is cause the 800 number provider has an option to forward it to an extension instead of a number. |
| 00:49.04 | citec | RaNa, configure it on your provider to redirect to your800number@yourAsteriskIP, then create a sip peer to allow incoming calls from your provider (maybe by ip authentication), then, in your dialplan, put a extension like this: |
| 00:49.29 | citec | exten => your800number,1,Dial(SIP/youripIPPhone) |
| 00:49.32 | citec | that should work |
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| 00:49.40 | citec | I have many DID numbers like tha |
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| 02:26.02 | volga629 | Asterisk ldap trying find any number which I trying to dial out in extension ldap and after just crush |
| 02:26.15 | volga629 | crash |
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| 02:46.36 | *** join/#asterisk Micc (~Micc@50.46.232.86) |
| 02:47.29 | Micc | I just got an IP7000 polycom conference phone, but I can't get it to use anything other than ulaw. What config file do I need to edit in the /Config directory to enable different codecs? |
| 02:54.23 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 02:56.42 | saliak | I'm creating a .call file to initiate an outgoing call. If it completes successfully, it connects to an extension that executes shell commands to indicate it. I'm having trouble figuring out how to handle the case when the call cannot be completed. Is there a good way to do that? ideally, have it go to another extension? |
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| 02:57.48 | WIMPy | Call out via a local channel and use the dialplan. |
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| 03:00.48 | Micc | My IP7000, only request pcmu and pcma, all my other polycom's support g722 at least. I can't find where to tell the ip7000 to use g722 and siren14 |
| 03:04.40 | saliak | WIMPy: hrm. how to do that with a call file? or some interface that can be invoked from a script? |
| 03:05.21 | WIMPy | I just told you: Use a local channel. |
| 03:09.35 | saliak | WIMPy: oh, i see. so use what would be a local extension, and dial something that takes me through the dialplan.? like, Channel: 100/default/91231231234 (and have the _9. context do the dialing/handling of error/etc.? |
| 03:11.05 | WIMPy | That's not a valid channel. |
| 03:11.16 | WIMPy | local/extension@context |
| 03:17.56 | saliak | WIMPy: k, thank. does the pbx try multiple calls at once, or one at a time? |
| 03:18.28 | WIMPy | If you dump multiple call files? |
| 03:19.14 | saliak | WIMPy: yeah. do i need to be able to handle the condition with that given channel being busy (say 10 calls are dumped at the same time) |
| 03:19.34 | WIMPy | Yes. It will process the call files in parallel. |
| 03:20.03 | saliak | is there a way to make it serial? |
| 03:20.09 | WIMPy | If you need more control and feedback, AMI may be the better aproach. |
| 03:20.19 | WIMPy | no |
| 03:22.50 | saliak | WIMPy: i was hoping to not have to get into AMI as i almost have this working. making an email to fax gateway. i can generate the call file and point it at an extension to send the fax. i just need to handle the condition where i have too many dumped at once. i guess once it's in my dial plan, if my outgoing channel is full, i can at least detect and report the error. |
| 03:24.08 | WIMPy | No need to worry. If it's only about an occasional short overload, you can simply rely on the retry parameter in the call file. |
| 03:24.49 | WIMPy | Although I'm not sure if that works if you call a local channel. |
| 03:25.38 | saliak | WIMPy: i'd think the local call would be fine, it's just when i issue the Dial() command, it would presumably error out when i've used my two outgoing channels<?> |
| 03:25.40 | *** join/#asterisk [FireBird] (~raul@190.142.64.85) |
| 03:26.03 | *** part/#asterisk [FireBird] (~raul@190.142.64.85) |
| 03:26.10 | WIMPy | Yes, you can check for DIALSTATUS. |
| 03:26.42 | WIMPy | BRI or two POTS? |
| 03:27.21 | WIMPy | You could also use CHANISAVAIL before trying to dial. |
| 03:28.41 | saliak | WIMPy: yeah, that should work |
| 03:31.59 | saliak | WIMPy: thanks for the help! |
| 03:34.43 | volga629 | For local extension call from ldap I added [extensions] switch => Realtime/@, but can't dial 200 to 201 localy and if I add include line under from-internal it looking any extension in ldap include that I dial out and crash the asterisk |
| 03:36.17 | WIMPy | Can you re-phrase that? I'm not sure what the issue is. |
| 03:41.02 | volga629 | To call locally between extensions, asterisk pull extension from ldap, but when I dial 200 to 201 it say can't complete as dial |
| 03:43.25 | volga629 | I added switch line with context extensions on top extensions.conf, but still no going, but when I add the same switch line under [from-internal] it starts looking in ldap any number and crash asterisk |
| 03:44.30 | volga629 | [02/Jun/2012:21:49:25 -0400] conn=264 op=263 SRCH base="ou=NetLabExtensions,dc=networklab,dc=ca" scope=2 filter="(&(objectClass=AsteriskExtension)(AstExtension=*98)(AstContext=extensions-ldap)(AstPriority=10))" attrs=ALL |
| 03:45.31 | volga629 | this I see in ldap log and it all numbers include any external number |
| 03:49.48 | volga629 | and I can't really determine what cause in dial plan for this behavior |
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| 03:50.53 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:51.07 | WIMPy | Can't comment on ldap or the crash, but the priority is always the extensions within the context, then switches and last included contexts. |
| 03:52.14 | WIMPy | So if you want includes to be searched before the switch, then move the switch to its own context that you include last. |
| 03:53.17 | volga629 | ok let me try move after from-internal |
| 03:56.15 | volga629 | Executing [110@extensions-ldap:6] Playback("SIP/snomdesk-00000005", "silence/1&cannot-complete-as-dialed&check-numberk |
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| 04:01.38 | volga629 | Ok I found the trick |
| 04:01.51 | WIMPy | Either 110 doesn;t exist anywhere or whatever pattern hit there had higher priority. |
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| 04:03.26 | volga629 | The file extensions_additional.conf was overwritten I changed :%s/from-internal/extensions-ldap and [extensions-ldap] |
| 04:03.27 | volga629 | switch => Realtime/@ |
| 04:03.27 | volga629 | include => from-internal-noxfer |
| 04:03.27 | volga629 | include => from-internal-xfer |
| 04:03.27 | volga629 | include => bad-number ; auto-generated |
| 04:03.29 | volga629 | <PROTECTED> |
| 04:03.38 | WIMPy | ~pb |
| 04:03.38 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 04:03.59 | volga629 | yes, sorry wrong place |
| 04:04.37 | WIMPy | Are you hand editing a GUI generated file? That's scary. |
| 04:04.44 | volga629 | http://fpaste.org/Qu7r/ |
| 04:05.57 | volga629 | trying avoid edit not from GUI, but with ldap case not really match to do about it |
| 04:07.10 | volga629 | How I can preserve the changes still don't know |
| 04:07.42 | WIMPy | Only by never touching the GUI again, I guess. |
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| 04:07.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 04:09.34 | volga629 | yes look like, and another think strange that when I dial *97 it can' find match for ldap extension |
| 04:11.44 | volga629 | I just tried dial *97 I got seg fault again hmm |
| 04:17.26 | volga629 | what default value for TRANSFER_CONTEXT ? |
| 04:23.18 | resist0r | WIMPy: haha |
| 04:23.39 | resist0r | sorry I LOL'd |
| 04:23.47 | resist0r | Gnite Phreakz |
| 04:23.49 | resist0r | PEACE |
| 04:26.13 | volga629 | WIMPy Thank you for idea and help |
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| 04:43.50 | volga629 | /]\]\\ |
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| 06:28.08 | Dovid | morning |
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| 08:11.57 | atmark | Noob question, can asterisk work without provider? |
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| 08:12.09 | atmark | Noob question, can asterisk work without voip provider* |
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| 08:17.58 | coppice | asterisk can work without any VoIP at all. that's how it started |
| 08:28.49 | atmark | last question |
| 08:29.10 | atmark | is it possible to connect two asterisk servers on different locations? |
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| 11:35.09 | gusto | since when i do have my own IP address on UMTS? |
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| 11:52.40 | gusto | hm... tsss ... ip, but ... it's somehow in a private network, i do not understand it |
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| 13:48.04 | uskerine | hi, i would like to know what it is needed to configure a "hunt group". I have 4 standard POTS lines and I would like to temporarily use them before the ISDN PRI is ready. |
| 13:48.24 | uskerine | Any advice from someone that has done that in the past before? what do I have to ask to the telephony company? |
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| 14:07.29 | WIMPy | You just configure them as a group in your chan_dahdi,conf. |
| 14:15.21 | uskerine | but the operator needs to be something, right? |
| 14:15.33 | uskerine | i mean, do I have to ask for special POTS? |
| 14:15.49 | uskerine | because so far I have 4 independent POTS lines |
| 14:16.12 | WIMPy | You need to tell them to send call to you on all lines. |
| 14:16.40 | uskerine | ok, so the "single" number must call all lines at the same time, right? |
| 14:17.22 | WIMPy | Not at the same time. But in some sequence. |
| 14:18.08 | uskerine | ok |
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| 14:54.14 | mazpe | is there a way to create ring groups with macros? or an easier way than just exten => 2000,1,Dial(SIP/1&SIP2&etc) ? |
| 15:00.47 | WIMPy | You can define them as a variable or put them in AstDB or whatever. |
| 15:03.00 | mazpe | hmm, interesting.. going over my dial plans and optimizing them. |
| 15:03.08 | mazpe | using templates ect. |
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| 15:40.48 | [sr] | hi |
| 15:41.34 | greenwolf | good morning all m:) |
| 15:41.45 | WIMPy | //////////////////////////////////////////// |
| 15:41.57 | greenwolf | wimpy: sup |
| 15:42.18 | WIMPy | oh, just tried to get rid of some dirt :-) |
| 15:43.34 | WIMPy | I guess it's time to put the keyboard in to the dish washer again. |
| 15:44.02 | WIMPy | If only I knew where I put that lingerie net... |
| 15:45.25 | slav3_kitten | WIMPy, how many of your keys still have their markings? |
| 15:47.14 | WIMPy | All. |
| 15:47.50 | WIMPy | It's a cheap printed one, but not that cheap. |
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| 15:51.56 | slav3_kitten | well i mean my toughbook keyboard i missing most it's markings as is my ibm model m |
| 15:52.08 | slav3_kitten | i always lose the ; first from doing C/C++ |
| 15:52.44 | slav3_kitten | ;{}() are the keys first to wear off, then a divot forms in the spacebar |
| 15:57.04 | WIMPy | Wow. Luckily I didn't come across such a bad keyboard so far. |
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| 15:59.12 | WIMPy | This is still the keyboard that came with the computer. That was '95 and it's still looking good. |
| 15:59.32 | slav3_kitten | i code a lot... |
| 15:59.50 | slav3_kitten | mostly embedded microcontroller stuff |
| 15:59.55 | WIMPy | used to do so as well. |
| 16:00.07 | slav3_kitten | a good bit of the reason why i got a raspberry pi |
| 16:00.08 | slav3_kitten | :D |
| 16:00.19 | slav3_kitten | i was all "it's awesome!" |
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| 16:00.43 | WIMPy | has been on ARM since '88. |
| 16:00.49 | slav3_kitten | but it think my prblem is i keep my nails a bit long so i think the keyboard gets worn away from my nails bopping into it |
| 16:01.29 | Ethos | A point of curiousity for the technically challenged. |
| 16:02.07 | Ethos | Can I install asterisk as a stand alone application with a usb phone? |
| 16:02.16 | Ethos | without any additional hardware? |
| 16:02.26 | WIMPy | There is no such thing as an USB phone. |
| 16:02.48 | WIMPy | But you can use your sound system with Asterisk. |
| 16:03.05 | slav3_kitten | Ethos, you can install asterisk on a system with no hardware and use software sip phones if you want |
| 16:03.12 | WIMPy | Probably not a great choice, however. |
| 16:03.43 | slav3_kitten | WIMPy, better then going all out like me though |
| 16:04.04 | Ethos | so if I were to compare asterisk with something like Avaya, I would be way off the mark. |
| 16:04.16 | WIMPy | Thinking back to the ARMs.. I managed to build openhorst two days ago. I think I should go and flash it. |
| 16:04.31 | WIMPy | Avaya what? |
| 16:04.45 | WIMPy | Phone? PBX? |
| 16:05.34 | Ethos | IP based telephony system. the Software is loaded on each pc and then employees connect using a usb headset. |
| 16:06.13 | Ethos | I understand that I am way over my head here. Just trying to understand what is available. |
| 16:06.19 | WIMPy | So that's jst softphones. Asterisk is for the server they connect to, not for the users. |
| 16:06.33 | Ethos | ah, I see now. |
| 16:06.53 | WIMPy | You can use Asterisk as a phone, but as said: Not a good choice. |
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| 16:07.49 | Ethos | so asterisk is, in a sense, the switch? |
| 16:08.07 | WIMPy | yes |
| 16:09.31 | slav3_kitten | Ethos, asterisk is the call manager for your entire operation, it tells thing where to go an what to do when they get there |
| 16:09.39 | Ethos | So I could set it up so that a party could call in on one number, employees could be linked to an extension, and that extension could forward to an ip bound to a specific mac address? |
| 16:10.11 | WIMPy | for example. |
| 16:10.30 | slav3_kitten | anyhow my dog died, an my hand is bleeding from digging a grave |
| 16:10.31 | WIMPy | But the IP<>MAC relationship is an OS thing, off course. |
| 16:10.36 | slav3_kitten | so i'm going to go out and get food |
| 16:10.50 | slav3_kitten | because i hurt, and i need to get out of the house for a bit |
| 16:11.02 | WIMPy | hopes there's no relation between the two. |
| 16:11.54 | slav3_kitten | relation between which bits ? |
| 16:11.58 | Ethos | Thank you for entertaining my questions. I know how frustrating it can be to work with someone who truly has no clue. |
| 16:12.09 | WIMPy | Th dog dying and you going for food. |
| 16:12.24 | slav3_kitten | only relation is 4 hours digging a grave making me hungry |
| 16:12.38 | *** part/#asterisk Ethos (~Ethos@rrcs-71-43-240-50.se.biz.rr.com) |
| 16:12.38 | slav3_kitten | Ethos, i suggest getting the book |
| 16:12.57 | slav3_kitten | well that was a day late an a dollar short |
| 16:13.00 | slav3_kitten | peace |
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| 16:24.07 | zamba | how can i get a shoutcast stream as music on hold on asterisk 1.8? |
| 16:25.04 | WIMPy | IIRC there's a module for that. |
| 16:25.09 | pabelanger | icecast? |
| 16:25.41 | zamba | the module is called icecast? |
| 16:25.42 | WIMPy | Aren't they more or less compatible? |
| 16:25.57 | zamba | WIMPy: yeah, they are |
| 16:26.09 | zamba | compatible, that is |
| 16:26.12 | zamba | more or less :) |
| 16:26.17 | zamba | icecast uses the shoutcast protocol |
| 16:26.56 | pabelanger | no, there is no module but you can pipe icecast into musiconhold.conf, though I have never tried it |
| 16:27.15 | zamba | i see there are some howtos on how to set up a dummy folder and creating a shell script, but it seems all to over-complicated and pre 2000 |
| 16:28.54 | WIMPy | Oh, right. The module was for the other way. |
| 16:29.09 | zamba | streaming to an icecast/shoutcast server from asterisk? |
| 16:29.35 | WIMPy | yes |
| 16:30.42 | zamba | interesting |
| 16:31.38 | [sr] | sorry to put myself in the midle of the conversation... |
| 16:31.59 | [sr] | and about having MOH from an audio stream source? |
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| 18:10.57 | WIMPy | Great. That thing is bricked :-( |
| 18:15.40 | Simak | Can anyone refer me to some guids on setting up Asterisk for home use? |
| 18:15.55 | Simak | Current guides--I'm finding a lot from the mid-2000s and such |
| 18:16.03 | WIMPy | ~book |
| 18:16.03 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:18.05 | Simak | Thanks :D |
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| 18:49.27 | resist0r | ~secret_knowledge |
| 18:49.33 | resist0r | darn |
| 18:55.32 | *** join/#asterisk timahvo1 (~rogue@41.90.213.29) |
| 19:14.59 | *** join/#asterisk gusto (~gusto@80.187.246.54) |
| 19:15.10 | gusto | i am getting m=audio 16092 RTP/AVP 0 8 101 |
| 19:15.12 | gusto | a=[Jun 3 21:12:48] WARNING[81361]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 2e98540c023ce5ac48d231b2181c6f61@178.40.140.247:5060 for seqno 103 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
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| 19:15.28 | gusto | and i think it may be something with ports |
| 19:15.46 | gusto | how can i check which ports is the peer using for rtp? |
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| 19:32.20 | kaldemar | gusto: that's not an issue with RTP. is your asterisk behind a NAT? |
| 19:32.59 | gusto | no |
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| 19:36.54 | gusto | what is the name of g726-32 codec? |
| 19:38.42 | gusto | hm |
| 19:38.52 | gusto | maybe jitter buffer would help? |
| 19:38.55 | wonko | ugh, how do i debug google voice inbound calls? i'm seeing nothing in the asterisk console |
| 19:44.51 | gusto | [Jun 3 21:44:41] ERROR[22267]: res_rtp_asterisk.c:1064 ast_rtcp_write_sr: RTCP SR transmission error to 192.168.10.14:16431, rtcp halted No buffer space available |
| 19:44.55 | gusto | what is that? |
| 19:52.20 | slav3_kitten | i return! |
| 19:53.29 | WIMPy | Felling better? |
| 19:54.03 | WIMPy | s/ell/eel/ |
| 19:56.26 | slav3_kitten | WIMPy, not really |
| 19:56.52 | slav3_kitten | i regret a lot of things, and can't see to cry like everyone else in the house has done |
| 19:57.48 | slav3_kitten | i just went out and dug a hole till my hands were blistered and bleeding. so now i'm sitting here with gauze covering one hand and looking over the asterisk book |
| 19:58.08 | slav3_kitten | but! while at breakfast i thought of something i wanted to ask |
| 19:58.51 | WIMPy | Not sure that's a sound combination. |
| 19:59.14 | slav3_kitten | so say i have a hypothetical dialplan, and in that dialplan i have some time of day transfers an such setup so like if it's late it won't ring a house phone but transfers it to a cell phone |
| 20:00.11 | slav3_kitten | say the cell phone never gets picked up, is asterisk capable of doing something like "enter a password to ring the house phone" and if that times out in say 5 seconds have it transfer to voicemail? |
| 20:00.32 | WIMPy | sure |
| 20:00.51 | WIMPy | You could even do it in parallel. |
| 20:00.55 | slav3_kitten | oh? |
| 20:02.27 | slav3_kitten | but yea WIMPy when shit like this happens i pick some project to over engineer the ever loving hell out of to keep my mind off the bad stuff like regrets i have |
| 20:03.02 | WIMPy | You have to do what ever works best for you. |
| 20:03.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:03.49 | slav3_kitten | so true, so current project is learning voip, goin to do it to the max |
| 20:05.14 | WIMPy | If you have some telco background that topic has a good potential to make you cry. |
| 20:05.53 | slav3_kitten | i've got zero telco background aside from setting up the wiring aspects of it |
| 20:06.07 | slav3_kitten | and the telco fun toys i made when i was a kid |
| 20:06.18 | slav3_kitten | like diy butt sets, blue boxes, etc |
| 20:08.01 | slav3_kitten | so how would i do that in parallel WIMPy? |
| 20:08.20 | WIMPy | The magic of local channels. |
| 20:08.30 | slav3_kitten | i thought the dialplan is much like basic where it just follows down the line of events |
| 20:08.54 | slav3_kitten | any other books aside from the definitive asterisk guide i should pick up an read? |
| 20:09.02 | WIMPy | They always feel like a hack to me, but you can do some quite nifty things with them. |
| 20:09.39 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 20:09.39 | WIMPy | With a local channel, you enter the dialplan again. |
| 20:09.54 | slav3_kitten | interesting... |
| 20:10.10 | WIMPy | So by Dial()ing an additional local channel, you suddenly get parallel execution. |
| 20:10.38 | slav3_kitten | well i think i know what i need to play with :D |
| 20:13.40 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
| 20:21.05 | gusto | strange... the data is passing by, but i do not hear anything |
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| 20:22.20 | gusto | i am trying to do voip over a openvpn tunnel |
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| 21:20.00 | dijib | hello all! |
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| 21:32.10 | uskerine | which package should be usedfor installing asterisk in ubuntu? |
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| 21:57.49 | zamba | uskerine: asterisk |
| 21:58.30 | zamba | :p |
| 22:05.29 | *** join/#asterisk cdahmedeh (~cdahmedeh@24-212-153-203.cable.teksavvy.com) |
| 22:05.53 | cdahmedeh | hello, i'm looking for a cheap voip sip provider. just basic local number with unlimited calling to us and canada |
| 22:11.21 | *** join/#asterisk TimeRider (~steve@92.40.188.173.threembb.co.uk) |
| 22:13.47 | jpsharp | You're not going to get "cheap" and "unlimited" at the same time. |
| 22:15.54 | WIMPy | Cheap is always expensive. |
| 22:18.29 | cdahmedeh | ok, how about something like 1000 or 2000 minutes per month? |
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| 22:35.39 | gusto | hey |
| 22:35.48 | gusto | does asterisk know g729(a)? |
| 22:36.17 | gusto | btw. i solved my problem ... it works now ... the jitter was faulty |
| 22:37.20 | jpsharp | It knows g729, but only if you purchase codec licenses. |
| 22:37.29 | gusto | ah shit |
| 22:37.34 | gusto | so i can not use it, right? |
| 22:38.04 | gusto | my ata adapter does know 729a, that seems to be somewhat less patented, however |
| 22:38.22 | jpsharp | its all licensed codecs. |
| 22:38.30 | gusto | not al |
| 22:38.31 | sruffell | "less" patented, heh |
| 22:38.33 | gusto | not all |
| 22:39.19 | jpsharp | Well, actually, you can use G729 as long as Asterisk doesn't have to play/record any audio. Straight passthrough doesn't require a codec license. |
| 22:39.58 | gusto | well, then i would be stuck in ekiga, that one does not have that codec either |
| 22:40.03 | jpsharp | So if you're using * as a dumb switch, you're fine. But the instant you want to do something like play voicemail or music on hold, you'll need a license. |
| 22:40.25 | gusto | it's not that bad |
| 22:40.32 | gusto | i can go with g726-32 |
| 22:43.46 | gusto | however ... compared to g711 the quality degrades significantly, w/o noticably less traffic |
| 22:44.18 | WIMPy | That's the joy of RTP. |
| 22:45.52 | jpsharp | I used GSM when I can, plunk down for the G729 licenses when I have to. |
| 22:46.49 | gusto | so ... i see no reason to use this g726 now ... it has bad quality and still about 64 kbps overall traffic |
| 22:47.20 | gusto | g711 goes with 80 - 90 kbps all in all, but it is understandable |
| 22:48.02 | gusto | i just tested it on a high latency tunnel - about 120 - 300 ms and radio loop from my DAB+ radio |
| 22:48.09 | gusto | it works surprisingly good |
| 22:50.03 | jpsharp | Latency isn't a problem up till about 700ms. It's jitter that screws with you. |
| 22:50.16 | gusto | yes |
| 22:50.24 | gusto | that's what i found out today |
| 22:50.38 | gusto | ehm ... yesterday .. we have a new day now |
| 22:51.16 | gusto | but now my telephone works and that is goood!!! |
| 23:00.10 | gusto | ok, now i fixed that problem that i had first ... the hostname was not right - provider already updated the info on his web, but i had it printed out and did not take the new names over |
| 23:00.18 | gusto | now everything works |
| 23:00.34 | gusto | is good |
| 23:01.06 | gusto | is proud of himself ;-) |
| 23:02.55 | WIMPy | has a feeling redboot is ignoring my kernel parameters. |
| 23:03.30 | gusto | not only redboot, there are several things that are ignoring kernel parameters ... i had issues with that some time ago too |
| 23:03.45 | WIMPy | Hmm |
| 23:04.02 | WIMPy | And do you also know a way to get paramters working? |
| 23:05.04 | gusto | no |
| 23:05.10 | gusto | it just worked some time |
| 23:05.17 | gusto | or not, i do not know any more |
| 23:05.35 | gusto | however, i am going to sleep now, i am very lucky to have done everything i wanted |
| 23:08.26 | *** join/#asterisk Zopiac (~zopiac@c-68-40-13-61.hsd1.mi.comcast.net) |
| 23:08.36 | Zopiac | sighs |
| 23:09.14 | Zopiac | I have no doubt my problems are because of my incomplete or improper setup and rather trivial but I haven't been able to work them out |
| 23:10.13 | *** join/#asterisk netman (netman@46.238.76.188.dynamic.jazztel.es) |
| 23:14.29 | WIMPy | A busybox of only 284 bytes is unlikely to do anything usefull :-( |
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| 23:27.36 | Zopiac | the only thing I know that is wrong with my setup is that ekiga (on another computer) is unable to connect to the asterisk server (in the same network) |
| 23:28.06 | Zopiac | I'm not getting any messages in the asterisk console (verbosity is 8) |
| 23:29.17 | WIMPy | 'sip set debug on' |
| 23:29.44 | Zopiac | still no messages when I re-enable the account on ekiga |
| 23:30.26 | WIMPy | Then Asterisk is not receiving anything. |
| 23:30.32 | Zopiac | booo |
| 23:33.19 | Zopiac | I have no idea why it wouldn't be receiving anything though |
| 23:34.04 | WIMPy | 'sip show settings' or hunt the traffic with tcpdump or the like. |
| 23:34.46 | Zopiac | what should I be looking for in the settings? or should I just pastebin it? |
| 23:35.14 | WIMPy | The address it's listening on. |
| 23:35.47 | jpsharp | Firewall? iptables? |
| 23:35.50 | Zopiac | UDP bindaddress is 0.0.0.0:5060 |
| 23:36.03 | Zopiac | no firewall (simple home internal network) |
| 23:36.31 | WIMPy | Loks like it's not as simple as you'd like it to be. |
| 23:36.57 | WIMPy | Are you sure there's no firewall running? |
| 23:37.11 | Zopiac | Never have installed a firewall |
| 23:37.41 | jpsharp | Doesn't mean its not there. Never assume :) |
| 23:38.09 | Zopiac | I know for a fact that arch linux does not come packaged with a firewal |
| 23:38.23 | Zopiac | ah, windows has one by default, but i allowed ekiga through |
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