| 00:02.24 | *** join/#asterisk b14ck (~b14ck@173.219.15.98) |
| 00:02.52 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 00:03.44 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
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| 00:11.24 | Carlos_PHX1_ | Anyone have ideas on what causes this message? |
| 00:11.25 | Carlos_PHX1_ | [Jul 22 17:10:44] WARNING[1114]: chan_sip.c:6209 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x100 (g729)(256)/0x100 (g729)(256) |
| 00:13.34 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 00:13.46 | carrar | looks like a codec missmatch to me |
| 00:14.18 | carrar | Asked to transmit frame type 4 (g711) |
| 00:14.36 | carrar | show codecs |
| 00:16.11 | Carlos_PHX1_ | The odd part is only ulaw is in the allow= in config. |
| 00:16.26 | Carlos_PHX1_ | We do have 729 licenses. |
| 00:16.53 | carrar | Then allow g729 |
| 00:18.52 | Carlos_PHX1_ | We can't do that for this case. |
| 00:20.56 | pabelanger-lap | Carlos_PHX1_: Known issue |
| 00:21.13 | Carlos_PHX1_ | ? Call still works, so we could ignore it. |
| 00:21.59 | Carlos_PHX1_ | Do you know if there's a bug open on this? |
| 00:22.10 | pabelanger-lap | Carlos_PHX1_: Depends, are you having problems? |
| 00:22.19 | pabelanger-lap | Otherwise just ignore the warning |
| 00:22.22 | Carlos_PHX1_ | Yes, but not sure if they are related. |
| 00:22.43 | pabelanger-lap | Carlos_PHX1_: What is your problem? |
| 00:23.28 | Carlos_PHX1_ | We have phones that don't ring when the server thinks it's ringing. Cisco 79x0 phones only. |
| 00:24.34 | pabelanger-lap | pb your debug log |
| 00:25.01 | Carlos_PHX1_ | sip debug? |
| 00:25.10 | pabelanger-lap | ~collectdebug |
| 00:25.11 | infobot | i guess collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 00:28.15 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 00:28.21 | Carlos_PHX1_ | While I get that, is this a 1.6.2 specific issue or with other versions also? |
| 00:32.16 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 00:32.29 | pabelanger-lap | I doubt the warning is related to your problem your having. Post your debug log |
| 00:33.09 | Carlos_PHX1_ | Will do, getting set up to reproduce, it's a very specific set of circumstances. |
| 00:33.36 | Carlos_PHX1_ | Only with calls from a 1.4 server also running Vicidial. |
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| 00:39.56 | Carlos_PHX1_ | pabelanger-lap: Collected 2mb in a few seconds. Impressive. |
| 00:42.05 | Carlos_PHX1_ | pabelanger-lap: http://televolve.pastebin.com/7FGwSBwb |
| 00:47.31 | *** join/#asterisk [Outcast] (~anonymous@24-183-176-121.dhcp.oxfr.ma.charter.com) |
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| 01:09.20 | Carlos_PHX1_ | pabelanger-lap: The Cisco phone issue is because Cisco phones simply can't work with realtime peers, they require qualify=yes. That fixed the issue. The scrolling warning is sure annoying however, and continues. |
| 01:14.14 | *** join/#asterisk suge (~deebo@unaffiliated/suge) |
| 01:14.23 | suge | can anyone recommend a good IAX2 provider? |
| 01:15.30 | xheliox | Teliax isn't horrid. |
| 01:16.23 | suge | I haven't used * since Nufone was around |
| 01:16.31 | suge | I'll check them out, thanks |
| 01:17.02 | *** join/#asterisk zzlane (~zzlane@c-67-175-44-82.hsd1.il.comcast.net) |
| 01:17.45 | suge | pay as you go, up to 10 channels.. is that what you recommend? |
| 01:19.31 | xheliox | Sure. |
| 01:20.26 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
| 01:20.56 | pabelanger-lap | Carlos_PHX1_: always a good idea to use qualify=yes |
| 01:24.53 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:27.11 | Carlos_PHX1_ | pabelanger-lap: Sure, but not possible with realtime peers. |
| 01:31.13 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 01:32.36 | pabelanger-lap | Carlos_PHX1_: it is possible. You must have be caching realtime |
| 01:38.03 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
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| 01:44.41 | nny | I am testing a SIP connection to a VoIP provider here. They have supposedly given me the proper credentials, but when I try to dial out the line, it just times out eventually. I have done a sip debug, but a little rusty on spotting any smoking guns. http://pastebin.org/412646 What's the best way to see if the remote end is the one causing the call to not go through? |
| 01:44.53 | *** join/#asterisk coppice (~chatzilla@m121-202-65-244.smartone-vodafone.com) |
| 01:44.55 | Carlos_PHX1_ | pabelanger-lap, Right, true, but we ran into another problem then, and I can't remember what that was. K Fleming told us to stop caching to resolve that one. |
| 01:45.47 | nny | I am testing with a phone connected remotely on the same interface of the router that the asterisk box is trying to connect to the VoIP provider with, so at least I know SIP can transverse the firewall and allows me to connect in/out |
| 01:47.04 | pabelanger-lap | nny: Your problem is your NAT |
| 01:47.08 | pabelanger-lap | ~sipnat |
| 01:47.08 | infobot | sipnat is, like, 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 |
| 01:47.14 | pabelanger-lap | nny: ^^^ |
| 01:47.49 | nny | pabelanger-lap: this is a new box for me to work on, wondering if they need extern-ip set in sip.conf |
| 01:48.00 | nny | pabelanger-lap: you see the issue in the pastebin though? |
| 01:49.20 | pabelanger-lap | nny: Yes, your problem is your telling the telco to contact you at <sip:10001@10.0.0.5>, at private non-route-able address. Read the document about SIP-NAT. |
| 01:49.39 | pabelanger-lap | nny: So, the problem is a routing issue |
| 01:49.55 | nny | generally speaking I don't try to setup asterisk connections to other phones behind a NAT, or rather, at least the asterisk box has a WAN interface directly. Yes I understand that would be an issue. I don't have the opportunity to set this in [general], can I specify it per peer? |
| 01:50.16 | nny | (this = externip, localnet, etc) |
| 01:51.50 | nny | pabelanger-lap: ^^^ |
| 01:52.43 | pabelanger-lap | ~sipnat |
| 01:52.44 | infobot | sipnat is, like, 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 |
| 01:52.55 | pabelanger-lap | nny: I already told you how to fix it ^^^ |
| 01:53.13 | nny | pabelanger-lap: yeah got that, i'll finf out if I can set the proper sip.conf variables per peer vs. general from another source. Thanks |
| 01:55.12 | nny | hmm doesn't look like I can, I'll have to get clearance to change the [general] sip.conf parameters, there are other factors involved here. (Ex: the box has an interface only for one provider, that isn |
| 01:55.17 | nny | isn't NAT. |
| 01:55.55 | pabelanger-lap | nny: All the information is located in configs/sip.conf.sample in the source folder. externip / localnet are global settings |
| 01:56.08 | nny | pabelanger-lap: hmm you don't read much do you |
| 01:56.15 | nny | pabelanger-lap: but thanks ;) |
| 01:56.43 | pabelanger-lap | <PROTECTED> |
| 01:56.55 | nny | hit yourself harder |
| 01:56.58 | pabelanger-lap | if you read the first link I sent you, it would have also told you |
| 01:57.04 | nny | i haev those links |
| 01:57.15 | nny | do you read what I type |
| 01:57.29 | nny | hmm doesn't look like I can, I'll have to get clearance to change the [general] sip.conf parameters, there are other factors involved here. (Ex: the box has an interface only for one provider, that isn |
| 01:57.29 | nny | (9:54:55 PM) nny: isn't NAT. |
| 01:58.01 | nny | so I don't have permission/ proper time to change [general]. Only testing with a couple of peers right now |
| 01:58.42 | pabelanger-lap | So, you have access to an Asterisk box but cannot change settings? |
| 01:59.08 | nny | Not to general while it's in production |
| 01:59.15 | nny | Or rather, I choose not to |
| 02:00.37 | nny | to [general] in sip.conf. I said that the box has another VoIP "trunk" that isn't behind NAT connected to it. I don't want to start telling asterisk it's external-ip is something else in [general] without testing what would affect that first. |
| 02:01.09 | pabelanger-lap | nny: nat=no in [general], then set nat=yes for your peers. Problem solved |
| 02:02.16 | nny | yeah nat=no is already set in general, and each peer defines wether or not it is behind nat. I assume that nat=no for the one non nat connection also works |
| 02:04.24 | pabelanger-lap | nny: Correct. externip will only be used if you set nat=yes, so your safe with your existing peers have it disabled |
| 02:04.59 | nny | ok thanks, I guess it's as good of a time as any to try |
| 02:07.31 | nny | pabelanger-lap: does this look better http://pastebin.org/412678 ? |
| 02:08.17 | nny | just wondering, issue remains, but at least i'll have the proper sip [general] setting and can dig further |
| 02:08.35 | *** join/#asterisk teknon (~teknon@c-98-219-39-208.hsd1.ga.comcast.net) |
| 02:10.37 | nny | the test setup is test phone -> NAT <---> NAT <--> asterisk <---> NAT <---> VoIP Provider |
| 02:10.52 | nny | waaay abnormal for how I like to normally do it |
| 02:12.06 | nny | I prefer test phone --> NAT --> asterisk --> VoIP Provider, but I am semi-following the example given by d-fender's sip guide.. this doesn't rule out the provider itself is having an issue, just digging as deep as I can on my end. |
| 02:13.08 | pabelanger-lap | nny: Yes, your SIP messages look better but still seem to have a routing issue. |
| 02:13.20 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:13.32 | pabelanger-lap | you can use Originate from the CLI to try your test. |
| 02:13.43 | pabelanger-lap | this will remove your phone from the test. |
| 02:14.34 | *** join/#asterisk lost_soul (~shawn@cpe-67-241-66-112.twcny.res.rr.com) |
| 02:16.00 | nny | never have used Originate before.. in 1.6 would it be "channel originate SIP/18992060001@Voip-provider" ? |
| 02:16.51 | pabelanger-lap | *CLI> core show application Originate |
| 02:17.17 | nny | k |
| 02:19.22 | pabelanger-lap | nny: 70.167.35.228 is your WAN IP for your asterisk box? |
| 02:19.32 | nny | for the network, yes |
| 02:19.56 | pabelanger-lap | and 8.14.80.33 is the provider? |
| 02:20.02 | nny | yes |
| 02:21.42 | pabelanger-lap | Any your firewall is forwarding port 5060 udp to your local IP? |
| 02:22.33 | nny | ACCEPT udp -- anywhere anywhere udp dpt:sip |
| 02:22.39 | nny | er |
| 02:22.43 | nny | that's not a forward one sec |
| 02:23.34 | pabelanger-lap | Do you have access to your firewall / router on your WAN? |
| 02:23.58 | nny | it's basically shorewall/iptables on a fedora box, but yes |
| 02:24.45 | pabelanger-lap | ok, then enable tcpdump on your WAN interface and filter for 5060 UDP and you SIP provider. Then, see if you get a response to the INVITE message from Asterisk |
| 02:24.58 | pabelanger-lap | Something is dropping the packets |
| 02:25.37 | pabelanger-lap | Either the firewall, or a bad route |
| 02:26.54 | nny | 21:26:28.905006 IP wsip-70-167-35-228.ks.ks.cox.net.sip > 8.14.80.33.sip: SIP, length: 846 |
| 02:27.00 | nny | so it's hitting eth0 |
| 02:27.14 | nny | but when i tell it to check for src 8.14.80.33 I get nothing |
| 02:28.06 | *** join/#asterisk iamy_china (~iamy_chin@221.221.167.158) |
| 02:28.07 | nny | so maybe the remote end is bunk. I could test this with a know sip trunk that works and check for shennaigans on their end |
| 02:29.29 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 02:31.45 | pabelanger-lap | nny: pass -n to tcpdump, it disables domain names. |
| 02:32.26 | nny | 21:32:11.073461 IP 70.167.35.228.sip > 8.14.80.33.sip: SIP, length: 846 |
| 02:32.26 | pabelanger-lap | nny: But yes, if your not getting a response, I would confirm with the provider they can see your response |
| 02:32.50 | nny | ok yeah, going to setup a test connection to another provider, verify I can dial out of it and call it a night. Thanks.. sorry to be a hard ass lol |
| 02:33.10 | pabelanger-lap | np |
| 02:35.47 | *** join/#asterisk lost_soul (~shawn@cpe-67-241-66-112.twcny.res.rr.com) |
| 02:37.57 | nny | ok yeah, works with another provider, that's useful |
| 02:51.48 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
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| 03:58.31 | *** join/#asterisk kliwon (~irfan@unaffiliated/kliwon) |
| 03:58.47 | thansen | anyone around who can help me debug a SIP/2.0 407 Proxy Authentication Required issue? |
| 03:58.50 | *** join/#asterisk Tond (~tond@CPE00226ba7ac06-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 04:00.36 | *** part/#asterisk kliwon (~irfan@unaffiliated/kliwon) |
| 04:00.48 | Tond | Hi, I am running asterisk as root and when I execute a System command (in my case a bash script) from my dialplan it is like it never happend. I run the script manually and it works fine, but from asterisk it won't. It doens't return any error and shows the command as ran in the asterisk CLI also. Not sure how to go about debuging this one |
| 04:02.22 | Tond | My SYSTEMSTATUS variable shows sucess also! It is so weird |
| 04:06.43 | b14ck | sup everyone |
| 04:06.59 | b14ck | Tond, what user is 'asterisk' running as? |
| 04:07.12 | b14ck | It is likely that whichever user that is, doesn't have access to run your Asterisk script. |
| 04:07.21 | b14ck | You can check the user/group with ls -la /path/to/script |
| 04:07.41 | Tond | b14ck, it is runnign as root |
| 04:07.48 | b14ck | Tond, are you sure? |
| 04:07.59 | Tond | yes, i do ps aufx | grep asterisk |
| 04:08.02 | Tond | and shows as root |
| 04:08.03 | b14ck | Tond, also, what does your System() command look like? |
| 04:08.21 | b14ck | (exactly as it is) |
| 04:08.29 | Tond | b14ck> exten => 1,4,System(/test/test.sh) |
| 04:08.34 | b14ck | Do this: |
| 04:08.38 | b14ck | chmod +x /test/test.sh |
| 04:08.40 | b14ck | Then try again. |
| 04:08.46 | Tond | k |
| 04:08.49 | b14ck | Also: in your script, the first line should say: |
| 04:08.51 | b14ck | #!/bin/bash |
| 04:09.00 | b14ck | So that UNIX knows how to execute it. |
| 04:09.12 | Tond | ya it does have it |
| 04:10.21 | Tond | Nope still the same |
| 04:10.56 | b14ck | try: chmod u+x /test/test.sh |
| 04:11.00 | b14ck | It should work. |
| 04:11.06 | Tond | it is weird, because i can reload asterisk using the system command, so i know it is executing stuff and also I get "SUCCESS" result after executing the system command |
| 04:11.06 | b14ck | The next step would be do something like: |
| 04:11.07 | boodu | bye |
| 04:11.17 | b14ck | System(echo "test" > /tmp/test.txt) |
| 04:11.20 | b14ck | and then do: |
| 04:11.22 | b14ck | ls -la /tmp/test.txt |
| 04:11.27 | b14ck | and see if the output is as expected |
| 04:11.38 | Tond | chmod u+x /test/test.sh didn't work either |
| 04:12.03 | *** join/#asterisk Ngupiel (~Harun@219.83.35.61) |
| 04:12.10 | Tond | k |
| 04:12.32 | Ngupiel | hi there |
| 04:12.42 | Ngupiel | please help me |
| 04:12.43 | Ngupiel | :) |
| 04:12.58 | Ngupiel | how to config sip to h323 |
| 04:13.20 | Tond | b14ck> ls -la /tmp/test.txt |
| 04:13.20 | Tond | -rw-r--r-- 1 root root 5 Jul 22 23:12 /tmp/test.txt |
| 04:13.35 | b14ck | Hrm. |
| 04:13.40 | b14ck | So it is running as root. |
| 04:13.42 | b14ck | Tond, do this: |
| 04:13.42 | Tond | weird huh? |
| 04:13.50 | b14ck | chown root:root /test/test.sh |
| 04:14.00 | Tond | ic an even write to the asterisk log folder |
| 04:14.03 | b14ck | Or maybe: chown -R root:root /test/ |
| 04:14.13 | b14ck | (if you dont mind changing *all* permissions in the /test/ folder) |
| 04:14.19 | Tond | not at all |
| 04:14.30 | Tond | did it, should i try again? |
| 04:14.46 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 04:15.11 | b14ck | Yah. |
| 04:15.13 | b14ck | Try that. |
| 04:15.20 | Tond | b14ck> the same |
| 04:15.23 | *** join/#asterisk kliwon (~kliwon@unaffiliated/kliwon) |
| 04:15.26 | Tond | b14ck> Didn't work |
| 04:15.29 | b14ck | Tond, can you post your entire bash script to pastie.org? |
| 04:15.40 | Tond | yes |
| 04:15.40 | *** join/#asterisk guilhermebr (~Guilherme@200.175.244.93.dynamic.dialup.gvt.net.br) |
| 04:15.51 | b14ck | Tond, also, paste the output of `which bash` |
| 04:15.51 | Tond | it is two lines only anyways |
| 04:15.52 | rbd_ | hi guys... compiling asterisk 1.6.2.9 on ubuntu with dahdi drivers loaded, dahdi package instanlled, and /usr/lib/dahdi/ present with header files...still, asterisk is not compiling a chan_dahdi.so file...any ideas? |
| 04:16.31 | b14ck | rbd_, can you paste your error to pastie.org? |
| 04:16.32 | Tond | b14ck> http://pastie.org/1056493 |
| 04:16.38 | b14ck | It's probably that you are missing a dependency. |
| 04:16.57 | rbd_ | b14ck: I'm not getting an error, per se. it just is silently not compiling it |
| 04:17.05 | b14ck | rbd_, AH |
| 04:17.08 | b14ck | *Tond |
| 04:17.16 | b14ck | Tond, it's your script that is the problem. |
| 04:17.20 | b14ck | You have to do like: |
| 04:17.32 | b14ck | ls -l / > /test/tttttttttttttttttttttemp.txt |
| 04:17.38 | b14ck | When asterisk runs the script, the current directory will NOT be /test/ |
| 04:17.43 | b14ck | It is a /tmp/ directory |
| 04:17.50 | b14ck | So your file is being written to a random location. |
| 04:18.01 | Tond | b14ck> oh |
| 04:18.04 | Tond | b14ck> let me try that |
| 04:18.49 | Tond | b14ck> u r absolutly right dude! |
| 04:18.54 | b14ck | :) |
| 04:19.09 | b14ck | I wrote the guide on that stuff. |
| 04:19.20 | Tond | b14ck> i had created that script to test i originaly was trying to run a php file |
| 04:19.27 | b14ck | ah |
| 04:19.33 | Goshen | I am just getting back into Asterisk - I set it up and had it working when I only had one employee and that was silly so I quit using it, but now I have 10 and need to set it up again, I am looking at AsteriskNOW, any suggestions for a different starting point on a package?, I am also looking for suggestions for ip phones |
| 04:19.36 | Tond | b14ck> Oh cool.. Thanks a lot dude! |
| 04:19.48 | Tond | b14ck> let me go try the php and see how that works out for me |
| 04:19.59 | b14ck | Tond, sure. Just make sure you do: chmod +x blah |
| 04:20.06 | b14ck | Because it needs to be executable by *nix |
| 04:20.21 | b14ck | Goshen, polycom or aastra phones are the norm. |
| 04:20.30 | b14ck | They are solid, reliable, and easy to get drivers for. |
| 04:20.40 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
| 04:20.41 | Tond | b14ck> ok, and just to check should i run it as System(/usr/bin/php /test/test.php Hello World) ? |
| 04:20.45 | b14ck | And I'd recommend using elastix, just because it has a really nice UI and is super quick to configure / setup. |
| 04:21.03 | b14ck | Tond, just add #!/usr/bin/php to the top of your script file |
| 04:21.04 | b14ck | Then do: |
| 04:21.10 | b14ck | System(/path/to/script.php) |
| 04:21.23 | Tond | b14ck> Ok, tnx |
| 04:21.24 | b14ck | or System(/path/to/script.php arg1 arg2 etc) |
| 04:21.55 | Goshen | b14ck, thanks! it helps having a starting point |
| 04:22.06 | b14ck | sre |
| 04:22.14 | Tond | b14ck> Thanks dude! |
| 04:22.16 | b14ck | Goshen, if you want to start with Asterisk again, from the beginning... |
| 04:22.30 | b14ck | Goshen, I wrote a really nice tutorial series on setting up asterisk from scratch, which walks a total beginner through. |
| 04:22.53 | Goshen | like this? http://www.ksl.com/index.php?nid=218&ad=11190725&cat=&lpid=&search=polycom |
| 04:23.15 | b14ck | Goshen, yuep! |
| 04:23.25 | b14ck | Goshen, http://neverfear.org/blog/view/80/Transparent_Telephony_Part_1_An_Introduction |
| 04:23.27 | Goshen | Sandy is where I am, is that a good deal? |
| 04:23.29 | b14ck | You can find parts 2 & 3 on google. |
| 04:23.33 | Goshen | thanks for the link |
| 04:25.16 | *** part/#asterisk iamy_china (~iamy_chin@221.221.167.158) |
| 04:25.44 | rbd_ | solved the chan_dahdi issue, was missing libtonezone-dev |
| 04:28.06 | Goshen | are grand stream budge tone 100 worth having? |
| 04:29.05 | b14ck | Goshen, nope. |
| 04:29.13 | b14ck | Don't buy them, they're cheap, but they are notorious for sucking. |
| 04:29.25 | b14ck | problems installing, falling apart, poor audio quality, etc. |
| 04:29.40 | Goshen | don't want that, our calls are too important |
| 04:30.04 | b14ck | You're best off with polycom or aastra then. |
| 04:35.12 | Goshen | This looks good- http://provo.craigslist.org/sys/1853729963.html |
| 04:38.50 | Goshen | What card would you suggest for 4 PSTN lines? |
| 04:39.23 | Goshen | I have a old modem that worked on my old system that takes one line |
| 04:50.39 | Goshen | b14ck, your pictures are not working - http://projectb14ck.org/wp-content/uploads/2010/02/flowroute_dashboard.png |
| 04:51.12 | b14ck | Goshen, oh yah--I have to fix those, I updated the website a while back and didn't move them over. They aren't particuarly useful though, just a screenshot of the flowroute webpage. |
| 04:51.57 | Goshen | I found both polycom and aastra used locally, I think I will pick one of each up to play with and see which one I like |
| 04:52.48 | *** join/#asterisk dpisites (~cheng@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 04:53.18 | Goshen | wow $1.39 a month for a DID? nice! |
| 04:59.14 | Goshen | b14ck, Does Flowroute allow you to set outbound caller ID? |
| 04:59.51 | b14ck | Goshen, yeh |
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| 05:37.22 | p3nguin | $1.39 a month for a DID? That's highway robbery. |
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| 05:43.57 | thansen | anyone around? looking for some help with sip problem authenticating (it appears) |
| 05:47.26 | ChannelZ | I'm asquare |
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| 06:20.00 | Alton35 | p3nguin, where do you get dids for less? sheesh, I'm with teliax and I think they're good. |
| 06:25.43 | ChannelZ | That's in line with Vitelity, Flowroute.. I think voip.ms is like $.40 cheaper but |
| 06:30.24 | *** join/#asterisk iscsi (~light@78.108.73.46) |
| 06:31.42 | p3nguin | Yep, I pay $0.99 at VoIP.ms. |
| 06:31.49 | xheliox | Alton35: Teliax's pay as you go DIDs are quite the rip off. |
| 06:32.21 | Alton35 | hmm, their service is solid though |
| 06:32.23 | xheliox | p3nguin: Ditto, I think some areas are $1.30-ish with voip.ms.. but their inbound termination is considerably less expensive. |
| 06:32.28 | Alton35 | hell, I only have one number from them |
| 06:32.50 | xheliox | Alton35: When you have 30+, you might start to consider DID pricing a bit closer. |
| 06:32.59 | Alton35 | I guess so, |
| 06:33.08 | Alton35 | but they work? lemme see their web site. |
| 06:33.16 | xheliox | http://voip.ms |
| 06:33.41 | xheliox | I've only been using them for a few months, but they've been quite reliable. |
| 06:34.04 | Alton35 | well, my idea is to have 2 of everything anyway, so they'd be good to use first |
| 06:34.54 | Alton35 | I think it looks good, lemme search for DIDs |
| 06:35.43 | xheliox | I use Teliax, VoIP.ms, and Junction Networks.. voip.ms is the best value, Teliax is the most accomidating, and I have no real pro/con opinion on Junction. |
| 06:36.11 | Alton35 | unlimited channels? that's interesting, on their "USA DID numbers" |
| 06:36.35 | xheliox | Teliax will extend the channels if you ask them, that's no biggie. |
| 06:36.59 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 06:37.07 | Alton35 | yeah, I know, it's just interesting |
| 06:37.32 | *** join/#asterisk Ngupiel (~ngupiel@219.83.35.61) |
| 06:38.01 | Ngupiel | hi there |
| 06:38.09 | xheliox | I don't know it for a fact, but I suspect voip.ms has their channels capped at some level until you call them and tell them why you need more. It'd just be irresponsible to do anything to the contrary, imho. |
| 06:38.30 | Ngupiel | anyone help me to give me sampel config for sip to oh323 |
| 06:39.47 | Alton35 | maybe they just keep an eye on you, hard to say |
| 06:41.04 | xheliox | Perhaps, but I just can't imagine the # of channels being truly limitless by default. Someone could create an account and overload every bit of resources they have. |
| 06:41.38 | Alton35 | amazing, they have a lot more numbers then teliax for some reason |
| 06:41.45 | xheliox | Besides the point, I'm sure they're willing to live up to their as advertised product, I just don't think they'll do it by default. |
| 06:41.48 | Alton35 | I will get with them for sure. Useful information from you guys tonight. |
| 06:42.29 | xheliox | I do the majority of my outbound termination from them currently. We haven't heard a single complaint. |
| 06:42.49 | Alton35 | interesting, seems to be around 1 cent per minute in general |
| 06:43.22 | xheliox | Certainly for domestic calls, yeah. |
| 06:43.39 | Alton35 | teliax seems to be 2 |
| 06:43.52 | p3nguin | There is probably a limit of 10 channels until you ask for more. They're metered channels, so they don't really care how many you need. |
| 06:44.08 | xheliox | p3nguin: Exactly what I just said? |
| 06:44.55 | p3nguin | Um, yes, I'm talking about what you were just talking about. |
| 06:45.38 | Alton35 | well, interesting, |
| 06:45.57 | Alton35 | 1:45am here, lemme haul the baby off to sleep, he's 4 and I am a bad influence upon him ;-) |
| 06:46.01 | Alton35 | but I appreciate the advice. |
| 06:46.15 | xheliox | give him some bourbon and get back to the keyboard! |
| 06:46.25 | xheliox | that's what my Dad would have done.. |
| 06:46.27 | xheliox | ;) |
| 06:46.39 | Alton35 | hah, I'll give him a sip of mine next time |
| 06:46.44 | Alton35 | regards. |
| 06:47.09 | xheliox | goodnight. |
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| 07:02.43 | *** join/#asterisk cadey2 (~x@mail.officebroker.com) |
| 07:04.25 | cadey2 | Hi peeps, im fairly new to asterisk so I hope this isnt somthing really stupid however i cannot seem to find the answer on google. We have set up voice mail for our extensions and when the phone is logged in the phone gets the MWI message and the light flahes, however if the phone is not registered and then registers it does not pick up on any of the voice mails. what should i be looking for or at to resolve this please? |
| 07:05.22 | ChannelZ | are you setting mailbox=xxx in sip.conf (assuming these are sip phones) |
| 07:05.43 | cadey2 | yes ChannelZ we have and yes your correct its a SIP connection :) |
| 07:06.04 | cadey2 | I will just triple check that however! |
| 07:06.50 | cadey2 | mailbox=2000@default is present |
| 07:07.41 | ChannelZ | hmm |
| 07:07.57 | cadey2 | it gets the MWI when the phoen is online at the time the VM is left |
| 07:08.15 | cadey2 | just does not seem to get sent the message when it log back on |
| 07:09.06 | *** join/#asterisk joobie (~joobie@CPE-124-181-130-3.vic.bigpond.net.au) |
| 07:09.16 | ChannelZ | turn on sip debug for one of the offline devices with messages (sip set debug ip x.x.x.x) and then connect the device and let it register. |
| 07:09.42 | ChannelZ | It should send a SUBSCRIBE at some point to Asterisk, if it doesn't then that's something on the phone-side needing config. |
| 07:10.08 | cadey2 | arr ok, makes sence - its a Aastra 57i fyi |
| 07:13.29 | *** join/#asterisk stix (~stix@firewall.o4.dk) |
| 07:14.02 | ChannelZ | possibly look for a setting on the phone, "Explicit MWI Subscription" |
| 07:18.21 | *** join/#asterisk [netman] (~netman@83.54.228.245) |
| 07:19.03 | cadey2 | thank you ChanneLZ :) |
| 07:19.12 | cadey2 | Explicit MWI Subscription - is not enabled |
| 07:19.43 | ChannelZ | turn that bitch on and you might be in bid-ness |
| 07:19.49 | cadey2 | I must have been tired last night because i really did not see that in the config options :( |
| 07:20.09 | *** join/#asterisk gigirock (4d2b6739@gateway/web/freenode/ip.77.43.103.57) |
| 07:20.17 | gigirock | hi all |
| 07:20.25 | cadey2 | lol Channel, you english? |
| 07:20.43 | ChannelZ | american |
| 07:21.09 | cadey2 | really, bit late your side of the pond isnt it... or early haha |
| 07:21.21 | ChannelZ | 1:20a |
| 07:21.36 | gigirock | i have an international network and I want to implement some simple multi videoconference, i tested something but openmcu doesn't work for example |
| 07:21.52 | gigirock | I have some chances with asterisk ? |
| 07:22.25 | ChannelZ | gigirock: pretty sure no |
| 07:22.41 | gigirock | :( |
| 07:22.46 | ChannelZ | I think Asterisk can tell two ends where to send their video stream to if supported but that's about it. |
| 07:22.55 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 07:23.16 | gigirock | is there some sw other than openmcu to realize an mcu ? |
| 07:24.56 | ChannelZ | Dunno. Multiple video conferencing is not easy (computationally and bandwidth expensive) |
| 07:26.25 | gigirock | Dunno bandwidth is a number .... if you know how much u can use , u use it |
| 07:26.57 | gigirock | note 80% of videoconf are isdn based on 128k channel |
| 07:29.35 | cadey2 | CHannelZ : that works :) |
| 07:29.49 | gigirock | IMHO Then about video quality is not a problem, 80% communication are about a video-phone use..., a lot of software send only changed pixel of the next frame..... |
| 07:31.08 | gigirock | anyway.....nice to take a (cloud) coffee with You.....have nice week end |
| 07:31.54 | ChannelZ | cadey2: yay! |
| 07:32.00 | ChannelZ | gigirock: you too, good luck |
| 07:32.28 | gigirock | bye |
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| 08:09.49 | TehRabbitt | Quick question... I am using chan_sccp_b for two cisco phones I have... It supports transfers via the "transfer" button but for some reason it does not work.... The Asterisk CLI says something along the lines of "You must have more than 2 channels to transfer" when I hit the transfer button... does anyone have any ideas? |
| 08:11.31 | joobie | TehRabbitt, quick answer, you need to register your extension on more than one line |
| 08:11.34 | joobie | and check call-limit |
| 08:11.44 | joobie | i dont use cisco phones so i have nfi, but they are general comments that may help |
| 08:11.55 | joobie | fuk cisco |
| 08:11.57 | joobie | use polycom |
| 08:12.13 | joobie | wankers think they own all networking products and can charge extra for it |
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| 08:21.13 | TehRabbitt | joobie: Eh, I won't say fuk cisco if I get the equipment for free/reletivly cheap... sure, it's used, but it's still much more affordable than buying stuff off of ebay |
| 08:21.34 | TehRabbitt | what do you mean by register the ext to more than 1 line |
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| 08:27.49 | TehRabbitt | "2AB6D: We need 2 channels to transfer" |
| 08:27.55 | TehRabbitt | that is the error I get |
| 08:28.29 | Gugge | TehRabbitt: and you do have two calls running when you press trnasfer? |
| 08:28.38 | TehRabbitt | Yes. |
| 08:28.48 | Gugge | Then i have no idea. :) |
| 08:28.54 | TehRabbitt | it happens when I try to do a traditional transfer, or a dirtransfer |
| 08:32.26 | TehRabbitt | every time I hit transfer I get: "can't put on hold an inactive channel 500-28001D80 (Progress)" |
| 08:34.13 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 08:35.20 | garymc | anyone up for helping this newb pain in the arse? |
| 08:35.24 | garymc | me |
| 08:41.32 | TehRabbitt | any ideas of what that could mean |
| 08:48.57 | TehRabbitt | looks like the phone only has one channel open at a time |
| 08:49.02 | TehRabbitt | either incoming or outbound |
| 08:49.30 | TehRabbitt | as soon as i hit transfer and get the dialtone, it frees the channel / lets me make an outbound call... |
| 08:49.43 | TehRabbitt | but then it can't connect the new call with the old one i suppose |
| 08:49.50 | TehRabbitt | because it only has the 1 channel |
| 08:57.17 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.226) |
| 08:57.39 | qvsqvs | hi |
| 08:57.40 | qvsqvs | how can is change the default from unknown to some thing els |
| 08:58.23 | qvsqvs | i'm useing misdn |
| 08:58.43 | joobie | TehRabbitt, if you used polycom i could tell u the solution |
| 08:58.46 | joobie | but u use a shitty phone |
| 08:59.19 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
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| 09:22.23 | stix | Can you call a local program/script via the AMI? |
| 09:26.02 | hrhrhr | morning guys |
| 09:26.06 | hrhrhr | anyone used these? |
| 09:26.07 | hrhrhr | http://www.air-touch.com/rates/did.html |
| 09:26.33 | hrhrhr | am i right in thinking if i sign up with them (i'm uk based) for $12.50/month |
| 09:26.42 | hrhrhr | i could get $0.03 to singapore? |
| 09:27.45 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 09:30.06 | BarthezZ | no never heard of them hrhrhr, but what do you want to do? just call cheap to singapoure? or an DID number? |
| 09:30.43 | hrhrhr | uber cheap calls to singapore, hopefully |
| 09:30.55 | hrhrhr | we're paying £0.50/min atm |
| 09:31.13 | hrhrhr | been offered £0.10 with a new provider |
| 09:31.20 | BarthezZ | well, your looking for the wrong thing, that's an singapore(ian?) inward number |
| 09:31.22 | hrhrhr | but if i could get this working, it would be massively lower |
| 09:31.52 | BarthezZ | if it's just outgoing calls.... you could use voipbuster |
| 09:32.03 | BarthezZ | depending on your purpouse :p |
| 09:33.28 | hrhrhr | looking at it now |
| 09:33.34 | hrhrhr | it's for business use tho |
| 09:33.40 | hrhrhr | so need to work with * really |
| 09:33.46 | hrhrhr | and no pikey solutions :P |
| 09:33.52 | BarthezZ | ah oke, than you don't want voipbuster for your outgoing lines :p |
| 09:33.57 | hrhrhr | hehe |
| 09:34.28 | hrhrhr | so i'm barking up the wrong tree with air-touch? |
| 09:34.46 | hrhrhr | if they provide a sip trunk, couldn't a i make 'local' calls at those rates? |
| 09:35.05 | BarthezZ | well, it won't be the 0,03$ |
| 09:35.24 | hrhrhr | what you think it will be |
| 09:35.30 | hrhrhr | am i looking at the wrong section on their website... |
| 09:35.52 | BarthezZ | oh |
| 09:35.56 | BarthezZ | http://www.air-touch.com/rates/premiumcall00.html |
| 09:36.09 | BarthezZ | it is the 0,03$ but you were just at the wrong page :p |
| 09:36.26 | hrhrhr | cool :D |
| 09:37.00 | BarthezZ | but, they put prices in USD, you're in the UK right? |
| 09:37.09 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 09:37.31 | hrhrhr | yeh |
| 09:37.36 | hrhrhr | it's all a bit new to me |
| 09:38.09 | BarthezZ | try to find a provider in your country |
| 09:38.55 | BarthezZ | for example: if they have their sip servers in the US, and you are calling your neighbour, you will have a little extra delay for the round-trip time (UK->US back and forth) |
| 09:39.43 | hrhrhr | yeh, i appreciate that |
| 09:39.52 | hrhrhr | this will be specifically for dialling .sg numbers tho |
| 09:40.08 | BarthezZ | high volumes or incidental? |
| 09:40.21 | hrhrhr | we have an office there |
| 09:40.27 | hrhrhr | it accounts for a large amount of our call spend |
| 09:40.44 | hrhrhr | as well as the office, we ring all local sg companies too |
| 09:41.26 | BarthezZ | ah oke |
| 09:42.00 | BarthezZ | well, if there's a high volume to the office itself, don't rule out the possibility of putting a small pbx on that site |
| 09:42.22 | BarthezZ | and just connect it to your main site |
| 09:42.30 | hrhrhr | yeh, it is a possiblity |
| 09:42.42 | hrhrhr | i think a colleague tried a basic sip fone to our box |
| 09:42.47 | hrhrhr | and it was lol quality |
| 09:43.00 | BarthezZ | lol quality? it was that fun? :p |
| 09:43.07 | hrhrhr | so trying the 'premium grade' carrier route now |
| 09:43.46 | BarthezZ | well i have no idea what the internet situation is over there :p |
| 09:43.54 | hrhrhr | we get about 200ms to them |
| 09:44.01 | hrhrhr | so it should just about be doable |
| 09:44.02 | hrhrhr | i hope |
| 09:44.07 | hrhrhr | :P |
| 09:45.01 | BarthezZ | it's a little tight I think :p |
| 09:45.11 | hrhrhr | init |
| 09:48.38 | BarthezZ | but there must be more SIP-provider willing to provide cheap connectivity to singapoure :p |
| 09:49.29 | hrhrhr | there's 3 on voip-info http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential#Singapore |
| 09:49.44 | hrhrhr | my understanding is that i need to be in singapore to minimise the break out cost |
| 09:50.10 | hrhrhr | i'm beginning to wonder how sip providers are doing this |
| 09:50.19 | hrhrhr | do they contact each country and try and do what i am doing? |
| 09:51.09 | hrhrhr | we already dial out via a sip provider but i have no idea where the break out point is |
| 09:51.15 | hrhrhr | they aint come back to me with costs yet :s |
| 09:51.39 | *** join/#asterisk [netman] (~netman@83.54.228.245) |
| 09:51.42 | BarthezZ | I have no idea, think big provider break out everywhere :p |
| 09:51.52 | hrhrhr | yeh i would think so |
| 09:52.03 | hrhrhr | like a 'peering' arrangement with local companies everywhere |
| 09:52.08 | BarthezZ | or just peer with people :p |
| 09:53.00 | hrhrhr | would have to be telco specific tho i spose, not just a peering arrangement with a country? |
| 09:53.35 | BarthezZ | yeah ofc telco specific |
| 09:53.52 | BarthezZ | there isn't a point in the ocean where you can hook up a cable for "connect to singapore here" :P |
| 09:55.46 | cadey2 | Hi peeps, I would like to implement the transfer of a call from a UI on the users computer. I was thinking of pushing some dial commands to the users phones (say dial on line 2 who you want to transfer to) but then I realised I dont no what i would have to do to connect the original call to the person I jsut called on line 2... has anyone anu suggestions on how I could perform somthing like this |
| 09:57.34 | BarthezZ | cadey2: http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+Transfer |
| 10:06.07 | *** join/#asterisk addeswe (~adde@c-0fbbe255.013-16-756d651.cust.bredbandsbolaget.se) |
| 10:07.16 | addeswe | Quick question. When i dial out, i get 'Forbidden' and then a tag... What is that tag? :-) |
| 10:08.40 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 10:09.22 | qvsqvs | how can is change the default from unknown to some thing els, on incoming calles |
| 10:09.42 | tzafrir | (you're talking about chan_misdn, right?) |
| 10:10.36 | tzafrir | qvsqvs, you can test in the dialplan and set the callerid |
| 10:10.56 | *** join/#asterisk josexato (~josexato@64.76.110.198) |
| 10:11.05 | qvsqvs | but i don;t have call id on my line |
| 10:11.12 | *** part/#asterisk josexato (~josexato@64.76.110.198) |
| 10:11.17 | qvsqvs | evry call in unknown |
| 10:12.17 | qvsqvs | i want to change this "dialparties.agi: Caller ID name is 'unknown' number is 'unknown'" to "dialparties.agi: Caller ID name is 'From Telkom' number is 'unknown'" |
| 10:29.29 | cadey2 | BarthezZ : Hay thanks for the link, that redirects the call right so would that mean it would work like this... - Pickup call, call person its for, redirect call, the picked up call now rings on the new persons line ? |
| 10:30.12 | BarthezZ | you were talking about something on the pc right? |
| 10:30.28 | BarthezZ | i mean, on the phone itself you could just enable xfer of bxfer in the features.conf |
| 10:30.31 | cadey2 | Yeah, so we could make the UI transfer the call |
| 10:31.21 | cadey2 | shall i explain in more detail so it make more sence :) |
| 10:31.22 | BarthezZ | check out the docs on the AMI |
| 10:34.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 10:38.52 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
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| 10:56.09 | cadey2 | I dont seem to be having much fun today, I have set atxfer => *2 in the features config and also put ,tT in my dial yet when im on a SIP to SIP call pressing *2 does nothing ? |
| 10:56.10 | hrhrhr | BarthezZ: my provider just came back |
| 10:56.10 | hrhrhr | we're paying 2.9p/min for .sg |
| 10:56.11 | hrhrhr | making the whole thing not worth pursuing |
| 10:56.11 | hrhrhr | cheers for your help tho :D |
| 10:56.11 | BarthezZ | hehe oke |
| 10:56.11 | BarthezZ | no problem :) |
| 11:00.20 | puzzled | cadey2: you need to add something like exten => 1234,n,Set(DYNAMIC_FEATURES=bla in your dialplan where bla is from features.conf |
| 11:05.42 | cadey2 | hi puzzled, you got me puzzled :) I kind of understand what you mean I thikn, the set line needs to be after the dial line ? |
| 11:05.42 | cadey2 | the feature is atxfer right ? |
| 11:06.07 | cadey2 | sorry im new to all this and am trying to learn it : |
| 11:06.08 | cadey2 | :) |
| 11:06.14 | puzzled | cadey2: you need to add that line before the dial line |
| 11:09.30 | cadey2 | :( i fail |
| 11:09.52 | *** join/#asterisk [netman] (~netman@83.54.228.245) |
| 11:16.22 | cadey2 | i added this and it still no work :( |
| 11:16.24 | cadey2 | Set(__DYNAMIC_FEATURES=atxfer); |
| 11:16.24 | cadey2 | Dial(SIP/${exten},10,tT); |
| 11:16.38 | cadey2 | usine ael btw |
| 11:16.42 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:17.13 | puzzled | sorry I don't ael. try the mailing list |
| 11:17.51 | cadey2 | its more a less the same dude :) just dont need the 1234,1 part on the Set |
| 11:18.13 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
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| 11:30.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:36.14 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 12:08.44 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 12:08.51 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:09.25 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 12:13.36 | tuxx- | hey guys, is it possible to get more then 63 call pickup groups? Were running asterisk 1.4 |
| 12:13.37 | *** join/#asterisk ttwhy (~tekkno@p4FECFD66.dip.t-dialin.net) |
| 12:14.10 | garymc | [TK]D-Fender : could you please help me out. My sangoma cards have disapeared since updateing to Asterisk version 1.6 from 1.4 |
| 12:14.30 | garymc | im having trouble follow any setup guides again |
| 12:14.33 | garymc | :( |
| 12:15.06 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 12:15.56 | [TK]D-Fender | tuxx-: No. It's 0-63 |
| 12:16.01 | tuxx- | d'0h.:) |
| 12:16.05 | tuxx- | tnx anyway |
| 12:16.14 | [TK]D-Fender | tuxx-: What kind of scenario are you in when you'd need more than 64 groups? |
| 12:16.22 | tuxx- | big sites ;P |
| 12:17.02 | [TK]D-Fender | tuxx-: I'm scared to think what that would ahve to imply if you aren't dedicating groups to tons of individuals |
| 12:17.39 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:19.37 | *** join/#asterisk renshen (~renshen@93-63-217-144.ip29.fastwebnet.it) |
| 12:21.05 | renshen | Can you help me |
| 12:21.12 | renshen | Excuse me can you help me |
| 12:22.09 | tuxx- | ~as |
| 12:22.09 | infobot | somebody said as was the tranny. so swapping to a v8 and t5 doesn't add as much weight as you'd expect. |
| 12:22.10 | tuxx- | ~ask |
| 12:22.10 | 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. |
| 12:22.17 | renshen | tuxx |
| 12:22.42 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:22.50 | renshen | tuxx: you can illustrate me the use of the language in the sip channell |
| 12:23.34 | tuxx- | sorry, but what? language in a sip channel? can you explain in more detail? |
| 12:23.46 | renshen | the voice is |
| 12:23.52 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:23.55 | renshen | language?it |
| 12:24.00 | renshen | language=it |
| 12:28.13 | renshen | Pazzo |
| 12:28.16 | renshen | Pazzo |
| 12:28.25 | renshen | you can respond me |
| 12:30.38 | [TK]D-Fender | renshen: Ask a question that actually makes SENSE, and stop targeting everybody who walks in the door. |
| 12:31.09 | [TK]D-Fender | renshen: And Pazzo never even heard your question (which doesn't make any sense anyway). |
| 12:33.27 | renshen | language=en |
| 12:34.32 | tuxx- | :') |
| 12:34.35 | [TK]D-Fender | renshen: That isn't a question. |
| 12:34.46 | [TK]D-Fender | renshen: That is a setting in SOME config file. |
| 12:34.48 | renshen | In the definition |
| 12:35.01 | Pazzo | renshen? |
| 12:35.02 | tuxx- | any new on when 1.8 is gonna be released? it says Q2 on the website, but no real date :-( |
| 12:35.04 | [TK]D-Fender | renshen: WHAT IS THE ***QUESTION*** |
| 12:35.14 | [TK]D-Fender | tuxx-: "wHEN IT'S DONE" |
| 12:35.17 | tuxx- | hehe ;-) |
| 12:35.25 | renshen | of the sip.conf |
| 12:35.27 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 12:35.31 | renshen | of the iax.conf |
| 12:35.38 | renshen | define the language=en |
| 12:36.05 | renshen | misdn.conf:language=en |
| 12:36.19 | Pazzo | Hi [TK]D-Fender, thanks for protecting me ;-) |
| 12:36.27 | renshen | Can you help me |
| 12:36.37 | [TK]D-Fender | renshen: WHAT IS YOUR FUCKING QUESTION? |
| 12:37.02 | russellb | [TK]D-Fender: CALM THE FUCK DOWN |
| 12:37.12 | [TK]D-Fender | \o/ |
| 12:37.14 | renshen | Fender: the definition of the language=en in the sip.conf o the misdn.conf |
| 12:37.19 | russellb | :-p |
| 12:37.38 | [TK]D-Fender | renshen: it is a definition. says the language is "english" |
| 12:38.11 | renshen | Fender : example for the use |
| 12:38.16 | renshen | excuse me |
| 12:38.20 | renshen | an example for the use |
| 12:38.26 | [TK]D-Fender | renshen: What example? |
| 12:38.51 | renshen | an different betwen the language=en on language=it |
| 12:39.07 | [TK]D-Fender | renshen: It determine what language of sound files are played to the user |
| 12:39.24 | [TK]D-Fender | renshen: You don't know the difference between ENGLISH and ITALIAN? |
| 12:39.31 | renshen | Yes |
| 12:39.37 | *** join/#asterisk mbranca (~matteo@host139-217-static.224-95-b.business.telecomitalia.it) |
| 12:40.15 | [TK]D-Fender | renshen: Sorry if you don't understand what a language is then I think you have a much larger issue and might want to reconsider your very presence in a "communications" channel |
| 12:41.33 | *** join/#asterisk stefmtl (~stef@stef.istop.com) |
| 12:41.42 | Pazzo | ROFL |
| 12:43.37 | stefmtl | Hi. I am about to buy a Cisco unit AS5400XM to use with 28 T1 lines, and the seller is telling me that this unit is capable of 6-8 simultaneous calls at the same second. I have a lot of TE420 Digium card, but never heard about such limit, is there any ? |
| 12:46.07 | [TK]D-Fender | stefmtl: Do you actually need something that huge? Starting your own ITSP? |
| 12:47.24 | [TK]D-Fender | stefmtl: Limit as far as call setupteardown load? |
| 12:47.51 | stefmtl | [TK]D-Fender : Yes I need that capacity, I am about to receive over 500 calls |
| 12:48.50 | [TK]D-Fender | stefmtl: Just voice going over all that? |
| 12:48.55 | stefmtl | [TK]D-Fender : yes this is the limit the cisco seller is taling about |
| 12:49.05 | stefmtl | [TK]D-Fender : yes just voice |
| 12:49.53 | [TK]D-Fender | stefmtl: IIRC on Ciscos you have them do SIP conversion direct and jsut direct the calls that way. Lot less wiring, expense, load, and capacity on your servers |
| 12:52.36 | *** join/#asterisk iscsi (~light@sun28.ipfw.su) |
| 12:53.49 | stefmtl | ok thanks |
| 12:54.15 | [TK]D-Fender | stefmtl: What are you going to do with the calls? |
| 12:54.58 | cadey2 | Hi peeps, we really cannot seem to get Attended Transfer to work on 1.6. has anyone got any ideas why putting atxfer => *2 into the features.config and sending the Dial tT would still mean dialing *2 on the phone during a call to do nothing ? |
| 12:55.04 | cadey2 | we are using Aastra 57i |
| 12:55.32 | stefmtl | [TK]D-Fender : transfer the calls to the asterisk servers (SIP) |
| 12:55.42 | [TK]D-Fender | cadey2: Prove that DTMF works in call elsewhere first (VoiceMailMain) |
| 12:57.39 | cadey2 | Fender, do you mean if press 7 to delet works for example ? - that does work |
| 12:58.24 | *** join/#asterisk Aqituado (~aqutiado@93.167.108.90) |
| 12:58.44 | Aqituado | ChannelZ hi =) ive found my problem.... :D it is non functional RTCP in asterisk =) |
| 12:58.52 | Aqituado | https://issues.asterisk.org/view.php?id=17236 |
| 12:59.00 | Aqituado | thx for you help the other day |
| 13:01.42 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 13:09.36 | Katty | morning |
| 13:11.04 | leifmadsen | yo! |
| 13:11.12 | leifmadsen | M17236 |
| 13:11.23 | leifmadsen | what/! |
| 13:15.01 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 13:16.05 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:17.33 | *** join/#asterisk CarlFK (~juser@216.5.81.185) |
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| 13:18.43 | Katty | hugs on leifmadsen |
| 13:20.00 | CarlFK | calling in from ptsn to *, get "all circuits are busy." no one is using the phone here. how do I figure out if * is acting up or if the telco is overloaded ? |
| 13:20.35 | russellb | determine if the call is making it to asterisk |
| 13:20.45 | russellb | how you do so depends on how the call is (supposed to be) delivered to asterisk |
| 13:22.25 | carrar | mmm sippin in the sunrise |
| 13:22.55 | CarlFK | what are the choices? (first time looking at *) |
| 13:23.09 | *** join/#asterisk yonahw (~user@75.99.93.178) |
| 13:23.13 | CarlFK | I am sshed to the box |
| 13:23.41 | russellb | the choices? there are many telephony technologies asterisk supports ... |
| 13:23.48 | russellb | presumably you configured one of them to allow incoming calls, heh |
| 13:23.54 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 13:24.05 | cadey2 | [TK]D-Fender : Seems DTMF is working on voicemail however no DTFM does not seem to work on SIP TO SIP calls ? |
| 13:24.23 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:24.23 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:24.25 | CarlFK | tim did, he died last week... I have been asked to take over |
| 13:24.33 | carrar | WHAT |
| 13:24.37 | carrar | I am still alive |
| 13:24.37 | russellb | :-( |
| 13:24.39 | carrar | <- tim |
| 13:24.47 | CarlFK | heh |
| 13:24.59 | CarlFK | oh, then you should know how this is set up :) |
| 13:25.17 | carrar | Traffic in, traffic out |
| 13:25.20 | carrar | CAKE |
| 13:25.40 | CarlFK | coffee cake... coffee... maybe that will help |
| 13:26.09 | carrar | Might start with looking at the Asterisk CLI |
| 13:26.16 | carrar | asterisk -r |
| 13:26.27 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:26.33 | carrar | and reading the BOOK |
| 13:26.35 | carrar | ~book |
| 13:26.36 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:27.07 | carrar | But I am sure someone has mentioned that before |
| 13:27.22 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 13:27.46 | russellb | CarlFK: do you have phone lines hooked up? or is it all IP? |
| 13:28.36 | CarlFK | russellb: phonelines |
| 13:28.39 | CarlFK | t1 |
| 13:29.15 | russellb | the vendor of your T1 board should be able to assist you with debugging getting the card up and working |
| 13:30.16 | russellb | run ... # dahdi_hardware |
| 13:30.20 | russellb | it should tell you what board you have |
| 13:30.50 | [TK]D-Fender | [09:19]<CarlFK>calling in from ptsn to *, get "all circuits are busy." no one is using the phone here. how do I figure out if * is acting up or if the telco is overloaded ? <-- meaningless message which also indicates you are using FreePBX. Try looing at the actual CALL in detail. |
| 13:31.54 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 13:33.59 | CarlFK | russellb: -bash: dahdi_hardware: command not found |
| 13:34.13 | CarlFK | [TK]D-Fender: how do I look? |
| 13:34.57 | CarlFK | I got to: Connected to Asterisk 1.2.24... asterisk*CLI> |
| 13:37.46 | CarlFK | http://dpaste.de/RRTb/ Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 13:40.04 | pabelanger | CarlFK: Is DAHDI installed? |
| 13:40.30 | pabelanger | likely not, based on your Zap error |
| 13:41.10 | *** join/#asterisk dailylinux (~fedora@s21-00210.dsl.no.powertech.net) |
| 13:41.30 | CarlFK | pabelanger: how do check? (first time looking at * - so I have about 30 min of experience) |
| 13:42.10 | tzafrir | 1.2.24 - zaptel |
| 13:42.24 | cadey2 | Arg this is strange, DTMF works on VoiceMail and also on the Directory however it does not work on a call between extensions, so for example *2 does nothing when atxfer is set in the features and the dial is sent tT |
| 13:42.30 | pabelanger | CarlFK: Well, how did you install Asterisk? From source or repository? Install order is usually LIBPRI -> DAHDI -> Asterisk |
| 13:42.47 | tzafrir | (s/DAHDI/Zaptel) |
| 13:43.02 | pabelanger | ya. just notice 1.2.24, thanks tzafrir |
| 13:43.03 | CarlFK | pabelanger: I did not install it. |
| 13:43.26 | pabelanger | CarlFK: Ignore my statement about DAHDI, you need Zaptel. |
| 13:43.26 | tzafrir | CarlFK, what's the output of: cat /proc/zaptel/* |
| 13:43.57 | pabelanger | CarlFK: BTW, if you are just starting to use Asterisk for the first time. Why are you using 1.2.24? Thats pretty old. |
| 13:44.32 | CarlFK | tzafrir: http://dpaste.de/iaDC/ 24 lines of 1 WCT1/0/1 Clear (In use) |
| 13:45.03 | CarlFK | pabelanger: it was installed years ago by someone who died last week |
| 13:46.38 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 13:46.40 | pabelanger | CarlFK: ouch |
| 13:46.46 | wcselby | o/ |
| 13:47.17 | tzafrir | CarlFK, so basically there's a dial there of: Zap/G1/18478252642 |
| 13:47.28 | tzafrir | And this fails with "congested" |
| 13:47.51 | tzafrir | Which might as well be some variant of "wrong number" |
| 13:48.12 | [TK]D-Fender | ISDN 34 is sometimes used as a "busy" indication. pastebin your zap configs |
| 13:48.25 | russellb | CarlFK: note that in the output you posted, it says "RED" |
| 13:48.47 | russellb | that means your T1 is in red alarm. |
| 13:49.17 | russellb | check to make sure it is all plugged in. |
| 13:49.34 | russellb | your current mission is to make the red light on the T1 board turn green :-) |
| 13:49.58 | carrar | Russellb will self destruct in 30 seconds |
| 13:50.11 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 13:52.13 | russellb | *boom* |
| 13:52.22 | russellb | for real, have to go, time to head into the office |
| 13:52.36 | CarlFK | russellb: thanks - that's what was kinda hoping. someone else is talking to the t1 providor - "what is your circut ID?" um.. we will get back to you. |
| 13:52.46 | CarlFK | russellb: thanks again. |
| 13:52.51 | russellb | good luck |
| 13:57.15 | qvsqvs | hi [TK]D-Fender: how can i change an incomming call's cid from unknown |
| 13:57.55 | [TK]D-Fender | qvsqvs: "core show application set" , "core show function CALLERID" |
| 14:00.04 | qvsqvs | how do i use CALLERID in the consol |
| 14:00.38 | pabelanger | can't |
| 14:00.43 | pabelanger | extensions.conf |
| 14:01.21 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:01.39 | *** join/#asterisk ickmund (~magnus@cli-5b7ee16c.bcn.adamo.es) |
| 14:01.48 | *** join/#asterisk Natureshadow (~nik@178.5.52.123) |
| 14:02.10 | Natureshadow | hi there |
| 14:02.55 | qvsqvs | do i set it under the from-pstn context in extention.cong |
| 14:02.57 | qvsqvs | conf |
| 14:03.22 | [TK]D-Fender | qvsqvs: You set it any time before you care to actually see it |
| 14:04.12 | [TK]D-Fender | qvsqvs: And don't ask us "what context". This is YOUR dialplan. You could have contexts named [fred] and [alice] for all we know and its all meaningless.. as are teh extensions in there. |
| 14:04.44 | qvsqvs | i don;t what to change on the exstentions |
| 14:04.52 | qvsqvs | i want to change the incomming cid |
| 14:04.59 | [TK]D-Fender | qvsqvs: Set. The. CALLERID |
| 14:05.04 | [TK]D-Fender | ~book |
| 14:05.05 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:05.06 | [TK]D-Fender | ^^^^^ |
| 14:05.38 | [TK]D-Fender | qvsqvs: This is 1 dialplan app and one function to reference in there... maybe a GotoIf to actually CHECK it first... |
| 14:05.46 | [TK]D-Fender | 2 lines |
| 14:07.31 | qvsqvs | ok thx |
| 14:07.59 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
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| 14:30.47 | ariel_ | Hello everyone. quick question. Should this line just post the IP address? It's actually in 1.6.0.28 posting "0" exten => 9999,1,NoOp($[IAXPEER(CURRENTCHANNEL:ip)}) |
| 14:36.54 | wcselby | ariel_ - you line has incorrect formatting / look at your brackets, parens, colons, etc |
| 14:37.22 | leifmadsen | ya something looks off about that |
| 14:37.41 | leifmadsen | ariel_: s/:/, |
| 14:37.52 | leifmadsen | [Syntax] |
| 14:37.52 | leifmadsen | IAXPEER(peername[,item]) |
| 14:38.12 | leifmadsen | also the item is optional, the default is 'ip/ |
| 14:38.16 | leifmadsen | 'ip' |
| 14:38.33 | leifmadsen | so you should only need: NoOp(${IAXPEER(CURRENTCHANNEL)}) |
| 14:38.48 | ariel_ | ok let me try that |
| 14:40.02 | [TK]D-Fender | explicit > default |
| 14:40.22 | ariel_ | wow it works if you don't put the :ip |
| 14:40.24 | ariel_ | t/y |
| 14:41.13 | *** join/#asterisk mlarsen (~mlarsen@0x57370306.esnqu1.dynamic.dsl.tele.dk) |
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| 14:42.22 | kn0x | is res_config_odbc.conf deprecated? |
| 14:48.24 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:49.05 | pabelanger | kn0x: yes? |
| 14:49.14 | pabelanger | what version of asterisk you using |
| 15:28.16 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:28.16 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.9 (2010/06/18), 1.4.33.1 (2010/06/22), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 15:31.38 | wcselby | darth vader robbed a bank yesterday - http://www.latimes.com/features/odd-news/wpix-darth-vader-robs-banks,0,551177.story |
| 15:32.55 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:33.54 | *** part/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 15:34.48 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 15:35.34 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.113.77) |
| 15:38.13 | wcselby | better story with comments - http://www.escapistmagazine.com/forums/read/7.217358-Darth-Vader-Robs-a-Bank |
| 15:40.11 | Kobaz | heh |
| 15:40.20 | Kobaz | saw that on slashdot |
| 15:42.07 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 15:42.44 | *** join/#asterisk ChkDigit (~mike@static24-72-71-175.r.rev.accesscomm.ca) |
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| 15:46.58 | *** join/#asterisk Nwab (~Benwa@unaffiliated/benwa) |
| 15:47.34 | thansen | I'm experiencing this exactly, can someone give me some tips on how to avoid it? https://issues.asterisk.org/view.php?id=9678 |
| 15:47.58 | thansen | it was working just fine and then I restarted asterisk after a crash and this issue appeared |
| 15:52.28 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 15:53.03 | Kobaz | http://www.theonion.com/articles/kid-ready-to-start-playdating-again,17762/ |
| 15:55.28 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 15:56.45 | LemensTS | What is the resource/bandwidth difference of call coming from my voip provider to my asterisk box as SIP then to my customer as SIP vs coming from my voip provider to my asterisk box as IAX then to my customer as SIP? |
| 15:57.55 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 16:00.03 | [TK]D-Fender | LemensTS: Unless you are trunking calls to your provider, none |
| 16:00.31 | [TK]D-Fender | LemensTS: "trunk=yes" <--- |
| 16:00.40 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:01.01 | [TK]D-Fender | LemensTS: and the savings = (channels - 1) * RTP overhead |
| 16:01.52 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:06.28 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 16:06.29 | [sr] | hi |
| 16:08.16 | LemensTS | TKD-Fender: Ive mainly used SIP, had to lookup what trunk=yes was. I am just setting up the IAX, I'll try to do trunk=yes and use the IAX for G729 calls and SIP for G711 (point of me adding IAX is to seperate g711 custoemrs from g729...and I read G&29 is more efficient on IAX anyways) all that sound correct to you? |
| 16:13.47 | [TK]D-Fender | LemensTS: No. G.729 payload is the same regardless. An untrunked call wastes about the same UDP overhead between RTP & IAX. |
| 16:13.56 | *** join/#asterisk lladnar (~randall@static-71-172-94-66.nwrknj.fios.verizon.net) |
| 16:14.08 | [TK]D-Fender | LemensTS: the only savings to be had is where multiple calls to one host are trunked with IAX2 Trunk mode |
| 16:14.22 | [TK]D-Fender | LemensTS: the codec itself isn't ebtter from one transport to another |
| 16:14.48 | [TK]D-Fender | LemensTS: Its where you bundly multiple calls to save on RTP's voerhead |
| 16:14.52 | [TK]D-Fender | overhead |
| 16:16.55 | LemensTS | TKD-Fender: Thanks that clears things up for me |
| 16:21.01 | wcselby | short of writing my own application map, is there a way to change the file name of an automon recorded call? |
| 16:21.07 | wcselby | automon from features.conf |
| 16:21.36 | [TK]D-Fender | wcselby: CHANNELVARIABLES.TEX <----------- |
| 16:23.18 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 16:33.46 | [sr] | people |
| 16:33.55 | [sr] | to make two phones ring ate same time can i do |
| 16:34.09 | [sr] | exten=>mynumber,1,DIAL(SIP/999) |
| 16:34.14 | [sr] | exten=>mynumber,2,DIAL(SIP/888) |
| 16:34.19 | [TK]D-Fender | [sr]: No. |
| 16:34.23 | [sr] | Dial not DIAL |
| 16:34.38 | [TK]D-Fender | [sr]: Capitalization doesn't matter |
| 16:34.46 | [TK]D-Fender | [sr]: "core show application dial" |
| 16:34.51 | [sr] | let me see |
| 16:36.05 | [sr] | hum, exten=>mynumber,1,Dial(SIP/999)&[Dial(SIP/888)] ? |
| 16:36.28 | [sr] | braquets are wrong |
| 16:36.34 | [sr] | ]) |
| 16:37.28 | yonahw | the brackets just mean that it is optional. you need Dial(SIP/999&SIP/888) |
| 16:37.45 | [sr] | ops |
| 16:38.05 | [sr] | ok i get the point, i can use as many i want separated with & |
| 16:38.22 | yonahw | exactly |
| 16:38.31 | [sr] | didn't tested yet but the 1st one to answer get the call and the others just stop ring |
| 16:39.11 | [sr] | have to read the rest of this information |
| 16:40.41 | *** join/#asterisk oelewapperke (wapper@85-158-215-1.powered-by.benesol.be) |
| 16:40.56 | oelewapperke | if you dial multiple endpoints using "Dial", how do you know which endpoint answered ? |
| 16:44.02 | [sr] | i don't.. |
| 16:48.29 | [TK]D-Fender | oelewapperke: You see in CLI |
| 16:48.44 | [TK]D-Fender | oelewapperke: And there are channel variables set for this |
| 16:48.55 | oelewapperke | [TK]D-Fender: I'm doing this from AGI, so I'd need to get it as a response to the "dial" command |
| 16:49.13 | [TK]D-Fender | [12:48]<[TK]D-Fender>oelewapperke: And there are channel variables set for this |
| 16:51.52 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:53.49 | *** join/#asterisk sgimeno (~chatzilla@147.Red-79-153-158.dynamicIP.rima-tde.net) |
| 16:55.07 | *** join/#asterisk comradeb14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 17:03.27 | [sr] | [TK]D-Fender: i see something interesting here: If not specified, this defaults to 136 years. |
| 17:03.34 | [sr] | i wanna live 136 years :D |
| 17:03.54 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:04.27 | [TK]D-Fender | [sr]: No you don't.... |
| 17:04.34 | [sr] | hehe |
| 17:04.51 | [sr] | wall i'd love to, but only if i was always young |
| 17:05.25 | [sr] | wall=well |
| 17:05.46 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 17:15.18 | [sr] | brb |
| 17:20.40 | *** join/#asterisk vally (~vally@ip-109-90-253-92.unitymediagroup.de) |
| 17:27.43 | *** part/#asterisk Trab (~H4x0rzTra@wsip-64-58-150-178.oc.oc.cox.net) |
| 17:34.23 | *** join/#asterisk mbowie (~mbowie@99-7-126-96.lightspeed.simica.sbcglobal.net) |
| 17:34.47 | *** join/#asterisk hacim (~micah@debian/developer/micah) |
| 17:35.08 | hacim | i've got a music on hold mp3 set, and it seems to play and then stops |
| 17:35.15 | hacim | like maybe 3 seconds and then silence |
| 17:35.24 | hacim | and then suddenly it comes back, at a different point in the song |
| 17:40.16 | *** join/#asterisk hmodes (hmodes@B1-66ER.matrix.gs) |
| 17:40.34 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 17:40.56 | hmodes | so i just got a 1.6.2.10 announce and no code, yet there is .11-rc1 |
| 17:40.58 | hmodes | aroo? |
| 17:41.08 | Qwell | and no code? |
| 17:41.19 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:41.25 | hmodes | not mirrored, at least |
| 17:42.09 | hmodes | i've received the notif's before it's on downloads.asterisk.org before, but it's strange that there's a future rc1 there already |
| 17:42.53 | Qwell | you aren't looking hard enough |
| 17:44.49 | hmodes | oic |
| 17:44.54 | hmodes | that's still slightly abnormal |
| 17:44.56 | hmodes | *shrug* |
| 17:44.59 | Qwell | what is? |
| 17:45.48 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.10 (2010/07/23), 1.4.34 (2010/07/23), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 17:45.54 | hmodes | -current in /asterisk being ahead of what's just been announced |
| 17:46.06 | hmodes | i see .10 in /asterisk/releases tho' |
| 17:46.51 | hmodes | so i'll just roll with 11-rc1, I enjoy just-baked releases anyway, otherwise I wouldn't even be bringing this up :) |
| 17:47.09 | hacim | what variable controlls where voicemail greetings are stored? |
| 17:47.31 | Qwell | -current isn't newer than what was just released... |
| 17:47.49 | hacim | ah I found it |
| 17:49.55 | *** join/#asterisk lighthouse321 (5b79c6b1@gateway/web/freenode/ip.91.121.198.177) |
| 17:50.02 | lighthouse321 | hi all |
| 17:50.18 | lighthouse321 | i just installed asterisk today and was trying to get it up and working with agi php |
| 17:50.32 | hmodes | okay, you're right, different directory structure, I'm used to going by version number and update time |
| 17:50.56 | lighthouse321 | unfortunately i'm getting channel.c:3066 __ast_read: Dropping incompatible voice frame on ... of format ulaw since our native format has changed to 0x2 (gsm) |
| 17:51.02 | lighthouse321 | when i try to call an agi script |
| 17:52.45 | hmodes | sorry for the false alarm, i'll test 11-rc1 as punishment |
| 17:52.55 | lighthouse321 | exten => s,1,Answer() exten => s,n,Festival(trying php via agi) exten => s,n,AGI(try.agi) |
| 17:53.17 | lighthouse321 | try.agi: Failed to execute '/var/lib/asterisk/agi-bin/try.agi': No such file or directory |
| 17:53.23 | *** join/#asterisk mcr (~mcr@2001:4830:16ca:1:20d:60ff:fefa:7f03) |
| 17:53.27 | lighthouse321 | but the file is there and chmodded +x as well :S |
| 17:53.32 | lighthouse321 | any ideas ? |
| 17:54.09 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 17:54.37 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
| 17:54.53 | mcr | using 1.4.21, looking for a way to set the entire From: of a SIP call. It goes out right now as FOO@10.10.5.5 (Internal IP of *), but I need it to say FOO@domain.of.sip.provider. I tried CALLERID(all)=, but that only hacked the part before the @ |
| 17:54.57 | redax | hi, is there a way to get Voicemail as minimalistic as possible (especially listen back messages) or shall I go and get app_minivm a try? |
| 17:54.59 | hmodes | is happy to report a 7965 and sessiontalk on iphone work as usual |
| 17:55.09 | hmodes | bows to qwell |
| 17:55.55 | lighthouse321 | anyone here ? |
| 17:56.07 | *** join/#asterisk Fubard (~brawr@vpn.bctconsulting.com) |
| 17:57.32 | LemensTS | VOIP PROVIDER(G711/G729) <--> (G711/G729)ASTERISK <--> (G729)SIP CUSTOMER : If an incoming call comes into the voip provider to this sip customer with allow=g729 for that customer, does the voip provider signal to asterisk that a call is coming in, and then asterisk runs thru the dialplan and sees that the DID is ringing to that customer who only has G729 enabled, so it signals back to VOIP PROVIDER to make the leg fro |
| 17:58.13 | LemensTS | or does the voip provider ask asterisk what is allowed= for the voip provider trunk in sip.conf, and not even worry about who the call is going to |
| 17:58.21 | Fubard | I am having trouble putting my finger on what causes a call to go straight to voicemail when it goes to an extension that forwards to a cell phone. The voicemail box is the cell phone's. Anyone have any ideas? It seems to have started to ing on its own, last week and it worked correctly before that. |
| 17:58.56 | *** join/#asterisk Gershwin (~fake@unaffiliated/gershwin) |
| 17:59.53 | Fubard | It seems to have started on its own last week and it worked correctly before that.** |
| 18:02.18 | *** join/#asterisk b14ck (~b14ck@dsl-lfkn-207-70-143-25.consolidated.net) |
| 18:02.53 | [TK]D-Fender | Fubard: If it goes right to Cell VM, then that's a cell issue. |
| 18:03.00 | [TK]D-Fender | Fubard: Nothing to do with * |
| 18:04.53 | lighthouse321 | tk fender, any ideas on why the agi thingy doesn't see the file even though it's there ? |
| 18:06.37 | [TK]D-Fender | lighthouse321: Show me some real backup and I'll give you an opinion |
| 18:08.25 | lighthouse321 | hmmm, ok, it depends on the fact that i had added festival in front of it |
| 18:08.45 | lighthouse321 | apparently if i run the agi thingy it has to be run alone or it will fail if there are other commands before it, blah. |
| 18:08.46 | [TK]D-Fender | lighthouse321: No reason that should have any impact at all |
| 18:11.01 | lighthouse321 | hmmm, k |
| 18:11.38 | *** join/#asterisk knot (yiffstar66@unaffiliated/devemo) |
| 18:12.08 | [TK]D-Fender | mcr: fromdomain=domain.of.sip.provider |
| 18:12.28 | [TK]D-Fender | redax: No. |
| 18:13.10 | lighthouse321 | somehow it now works, even though it spits out lots of red ERROR lines with broken pipe written at the end of them, but it works, so good enough |
| 18:14.17 | [TK]D-Fender | lighthouse321: Probably for numerous AGI SNAFU's |
| 18:18.08 | Katty | peers |
| 18:23.59 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:32.49 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:33.21 | yonahw | I'm looking for hard phones which would work the plantronics cs55 without a lifter or hookswitch cable. Migrating from Toshiba which doesn't require them and don't want to purchase a whole bunch if we can avoid it. Does anyone have any suggestions for that? |
| 18:39.41 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 18:41.01 | [TK]D-Fender | yonahw: Most of Polycom's newer models 650/550/450 should support this |
| 18:41.34 | [TK]D-Fender | yonahw: I've seen specific info on the 550. The 650 is higer end, and the 450 is newer. Should all support it |
| 18:42.37 | mcr | [TK]D-Fender, thanks I think that did the trick! |
| 18:42.40 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:43.06 | [TK]D-Fender | mcr: You're welcome |
| 18:43.28 | *** join/#asterisk mcab (~mb@79.99.65.173) |
| 18:46.00 | yonahw | [TK]D-Fender: thanks, I will look into them. When I spoke to Plantronics they said that Polycom supported the hookswitch cable, which is the same cost as the lifter. I really wanted to look into the Polycom's anyway since I have only heard good things about them. |
| 18:46.08 | *** part/#asterisk mcab (~mb@79.99.65.173) |
| 18:52.57 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 18:54.13 | *** join/#asterisk mlarsen (~mlarsen@0x57370306.esnqu1.dynamic.dsl.tele.dk) |
| 19:03.49 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 19:05.42 | *** join/#asterisk exothermc (~miles@174.127.153.10) |
| 19:06.04 | exothermc | How do you setup CAS with a sangoma card and dahdi? |
| 19:14.16 | wcselby | with asterisk, is this workflow possible - http://pastebin.com/AUsxLXkq |
| 19:14.27 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:14.52 | BarthezZ | yes wcselby |
| 19:16.12 | b14ck | sup guys |
| 19:16.39 | wcselby | BarthezZ - how? |
| 19:17.41 | BarthezZ | incoming call -> atxfer (cancel when spoken to staff) -> bxfer to staff |
| 19:18.35 | wcselby | how do you bring the caller into the call between staff members then disconnect? |
| 19:18.51 | wcselby | i mean |
| 19:19.09 | wcselby | how do you bring the caller into the call so that all three of you can talk and recap, then drop out without haning up on the caller or the transferee |
| 19:19.15 | *** join/#asterisk Beltechs (~Beltechs@netblock-68-183-48-2.dslextreme.com) |
| 19:22.40 | Beltechs | helllo, using *1.6 im tring to SIP Debug, some remote extensions are having registration problems |
| 19:22.43 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
| 19:23.12 | [TK]D-Fender | Beltechs: So, 1 hour later. Anything to show us? |
| 19:23.23 | Beltechs | lol |
| 19:23.33 | Beltechs | thats funny |
| 19:23.46 | Beltechs | I dont know how to use the Debug |
| 19:24.12 | Beltechs | went asterisk -vvvr |
| 19:24.26 | Beltechs | then SIP DEBUG |
| 19:24.53 | [TK]D-Fender | Beltechs: Syntax varies. Try looking at the CLI Help |
| 19:25.43 | BarthezZ | oh sorry wcselby, you need a 2 way call |
| 19:27.01 | Beltechs | how do you keep up with it when it keeps scrolling? |
| 19:27.22 | Beltechs | core show help |
| 19:28.26 | wcselby | Beltechs |
| 19:28.30 | wcselby | sorry |
| 19:28.34 | Beltechs | yes |
| 19:28.44 | wcselby | Beltechs - are you connecting to the cli in a terminal client? |
| 19:28.49 | wcselby | such as putty? |
| 19:29.01 | Beltechs | yes Tunnelier |
| 19:29.10 | wcselby | does it have a logging feature? |
| 19:29.19 | wcselby | if you have a fast moving cli, that's what you'll probably have to do |
| 19:29.31 | wcselby | then parse the logs in your favorite text editor |
| 19:29.31 | Beltechs | i see |
| 19:29.41 | Beltechs | <PROTECTED> |
| 19:29.54 | wcselby | there are other options also, I think you can add verbose to your logger.conf log option and it may also pick up the sip debug info |
| 19:30.00 | wcselby | depending on the version of asterisk you're using |
| 19:30.17 | [TK]D-Fender | Big scroll-back buffer. Copay all. Paste |
| 19:30.20 | [TK]D-Fender | copy* |
| 19:30.56 | Beltechs | tried like 20 times is too busy |
| 19:31.13 | BarthezZ | or something like: asterisk -vvvvn -x sip debug | tee /tmp/sip.log | more :P |
| 19:31.38 | Beltechs | let me try |
| 19:31.42 | *** join/#asterisk Alagar (~Administr@122.164.42.194) |
| 19:31.52 | BarthezZ | i think |
| 19:33.06 | Beltechs | dont think it does logging |
| 19:33.21 | wcselby | [TK]D-Fender - any ideas on this workflow - http://pastebin.com/AUsxLXkq ? |
| 19:33.32 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 19:33.35 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 19:38.52 | Beltechs | Im gonna keep looking into how to use SIP Debug, but now the extension is working. Should I wait to debug the extension when it fails to register? |
| 19:40.01 | *** join/#asterisk p3nguin_ (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 19:40.37 | telnettech | in a MGCP packet that contains an RSIP.... is there anyway to find out why the restart happened? |
| 19:44.46 | exothermc | [TK]D-Fender: Have you ever setup a Channel associated signaling or robbed bit signaling voice T1 before? |
| 19:45.04 | [TK]D-Fender | exothermc: Yes |
| 19:45.18 | exothermc | [TK]D-Fender: You know of any good docs? |
| 19:45.18 | [TK]D-Fender | exothermc: WIKI & sample configs show this |
| 19:45.36 | [TK]D-Fender | exothermc: Stuff hasn't changed in over half a decade |
| 19:45.42 | exothermc | [TK]D-Fender: which wiki? asterisk wiki? |
| 19:46.03 | [TK]D-Fender | ~wikis |
| 19:46.04 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:46.12 | [TK]D-Fender | there is no "asterisk wiki" |
| 19:46.32 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 19:46.43 | Kobaz | it would be nice if there was one |
| 19:47.03 | Kobaz | and it was more of a wiki... rather than having 10 year old information that doesn't get updated |
| 19:47.10 | Kobaz | i was thinking of starting one |
| 19:47.27 | exothermc | [TK]D-Fender: Ok just so I'm getting things straight. Is this the R2 stuff? |
| 19:51.02 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 19:51.26 | LemensTS | is this how it is supposed to work from voip provider, to asterisk, to customer http://pastebin.com/bpugZZ3x ? |
| 19:52.05 | LemensTS | Im trying to do g711 all the way to the customer, and g729 all the way to the customer on incoming calls |
| 19:53.33 | LemensTS | I was going to setup different a trunk for g729 with the provider, and a trunk for g711 with the provider. Then add my g729 DID's to that, and vice versa for g711...but I am getting different answers on how this works from various sources |
| 19:55.44 | telnettech | in a MGCP packet that contains an RSIP.... is there anyway to find out why the restart happened? |
| 19:56.11 | telnettech | or i should say what the reason method means |
| 19:56.37 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 19:57.52 | exothermc | [TK]D-Fender: So when I try to compile libunicall I get "testcall.c:100: error: field dtmf_tx_state has incomplete type" google doesn't want to help with that error. |
| 20:00.26 | tzafrir | exothermc, what version of Asterisk do you have? |
| 20:00.48 | exothermc | 1.6.2 |
| 20:01.00 | tzafrir | just grab libopenr2 |
| 20:01.17 | tzafrir | And use the R2 support inlcluded in chan_dahdi |
| 20:01.28 | exothermc | tzafrir: |
| 20:01.48 | tzafrir | No need to mess with the whole libunicall stack |
| 20:02.01 | [TK]D-Fender | exothermc: I've never touched R2 |
| 20:02.21 | exothermc | [TK]D-Fender: how did you do CAS without R2 stuff? |
| 20:04.23 | tzafrir | exothermc, You have E2? T1? |
| 20:04.29 | tzafrir | What device? |
| 20:04.51 | exothermc | tzafrir: ya Channel associated signaling T1s on a Sangoma A104 |
| 20:04.55 | *** join/#asterisk generalhan (~asd@about/windows/staff/generalhan) |
| 20:05.12 | tzafrir | So no, you don't need R2 |
| 20:05.23 | exothermc | tzafrir: what do I need then? |
| 20:05.32 | tzafrir | I'm not really sure what you need to set up in Sangoma |
| 20:05.47 | exothermc | Seems like everything I find is for ISDN PRI or R2. |
| 20:06.03 | tzafrir | But there's absolutely no need to R2. Nither for ISDN |
| 20:06.05 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:06.08 | [TK]D-Fender | exothermc: You are overassociating things |
| 20:06.19 | [sr] | hi again |
| 20:06.30 | [sr] | 1.6.2.10? oh lah lah |
| 20:06.31 | [sr] | :) |
| 20:06.39 | tzafrir | (well, R2 is indeed used on top of CAS in E1) |
| 20:06.51 | tzafrir | (but that's not related) |
| 20:07.12 | generalhan | hey all, we are moving offices and the new location cannot get us very good pricing on T1 lines seperate (DIA T1 + PRI) our provider is suggesting we do their FLEX lines for data and voice shared. Has anyone used one of these setups with Asterisk with good results ? |
| 20:07.16 | exothermc | tzafrir: ahh ok, so don't need R2, but still need to figure out how to setup CAS instead of ISDN PRI |
| 20:07.30 | tzafrir | exothermc, IIRC some extra switches in the Sangoma setup switch, or whatever it is called |
| 20:07.52 | generalhan | i want to be sure that i can order a pakcage with enough bandwidth to accomodate all my users internet usage if there was ever a time when all 23 Channels are being used. but i am not really sure how much bandwidth 23 simultaneous calls would eat up |
| 20:07.54 | exothermc | tzafrir: ok but the asterisk/dahdi part is the same? |
| 20:07.58 | [TK]D-Fender | generalhan: "FLEX lines" doesn't tell us exactly what that means |
| 20:08.19 | tzafrir | I'm not really sure what extra settings it sets beyoud those in dahdi and asterisk . dahdi_genconf can generate dahdi and asterisk CAS config |
| 20:08.34 | tzafrir | (look for 'CAS' in /etc/dahdi/genconf_parameters) |
| 20:08.49 | tzafrir | That said, I have a feeling that there's some Sangoma Way of doing it |
| 20:08.50 | generalhan | [TK]D-Fender: sorry. its a single data connection that has QoS managed by them, to ensure that phone calls take priority over data. essentially reducing our bandwidth for each simultaneous call |
| 20:09.09 | [TK]D-Fender | generalhan: VoIP it is. |
| 20:09.45 | [TK]D-Fender | generalhan: So far this doesn't change much for *. |
| 20:09.58 | generalhan | well it makes more sense when you put it that way |
| 20:10.28 | generalhan | they keep saying PRI and DIA T1 on one line. so thats how i kept viewing |
| 20:10.29 | [TK]D-Fender | generalhan: Usually IAD's for links like this spit out ethernet for the data and a repacked T1 signalling of some sotr (CAS/PRI, etc) |
| 20:10.47 | [TK]D-Fender | generalhan: the IAD probably spits the VoIP back out as PRI to your PBX. |
| 20:11.02 | [TK]D-Fender | generalhan: Which, sounds wasteful in that you need a card.... |
| 20:11.27 | generalhan | [TK]D-Fender: but essentially i can do the same math as normal VoIP calls to figure out they bandwidth i would need to support 23 calls |
| 20:11.56 | [TK]D-Fender | generalhan: In theory... if you're going to assume the tech it actually uses. |
| 20:12.55 | generalhan | boo, now i am worried. i want to stay with this company because i have been really happy with their PRI quality, if all of that is changing, whos to say it wont flat out suck now |
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| 20:24.53 | ofauchon | Hi, is the following codec order correct : g722,ulaw,alaw (ST2030 phones, Gigaset 470, patton SIP to PRI E1) . Thanks |
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| 20:25.14 | Kobaz | there is no 'correct' codec order |
| 20:25.29 | Kobaz | you put the codecs in the order of preference that you want them to be used |
| 20:26.55 | ofauchon | Kobaz, ok. |
| 20:27.14 | [TK]D-Fender | checkout time, BBIAB |
| 20:27.50 | ofauchon | Why NativeFormat is diffenet from WriiteFromat and Readfomat when 'core show channels' ? thx |
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| 21:26.09 | digitalml | how come in freePBX it says that my total active calls are 1 of 2 and my channels are 1 of 4 when making an outbound call. Does that mean i can only make a total of 2 outbound calls? |
| 21:26.52 | pabelanger-lap | digitalml: #freebpx |
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| 21:51.46 | Jinxed- | has anyone tried to connect a magneto phone using * |
| 21:55.18 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta1 (2010/07/23), 1.6.2.10 (2010/07/23), 1.4.34 (2010/07/23), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
| 21:55.26 | leifmadsen | Asterisk 1.8.0-beta1 is now available for testing! http://www.asterisk.org/node/51396 |
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| 21:56.14 | russellb | leifmadsen: oh snap! |
| 21:56.25 | leifmadsen | russellb: oh snap indeed! |
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| 22:12.40 | digitalml | i was reading somewhere that to use the G.729 codec licenses had to be purchased... is this accurate? |
| 22:13.34 | Qwell | digilink: It requires a license, yes. http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC |
| 22:15.12 | Jinxed- | Has anyone ever tried to connect a magneto phone using astrisk |
| 22:15.22 | Jinxed- | asterisk* |
| 22:16.49 | digitalml | Qwell: how do i know if the calls im placing are using the 6.729 codec |
| 22:17.46 | pabelanger-lap | *CLI> g729 show licenses |
| 22:17.56 | digitalml | ah im not using it |
| 22:18.23 | digitalml | are there benifits to using this g729 codec? |
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| 22:21.18 | pabelanger-lap | smaller payload? |
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| 22:26.14 | digitalml | for the basic install of asterisk, is it in any way limited to the number of calls that can be made or the number of channels? |
| 22:26.50 | Qwell | no |
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| 22:37.13 | digitalml | can anyone here recoomend a hosted asterisk provider? |
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| 22:40.31 | paulc | digitalml: I've heard link2voip.com do it but haven't used them myself. I got myself a server at ThePlanet.com and set mine own up. Depends on how much you want to pay/do with it I guess |
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| 22:45.04 | digitalml | my problem is qos, i didnt want to have to buy a $250 router |
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| 23:25.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 23:25.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta1 (2010/07/23), 1.6.2.10 (2010/07/23), 1.4.34 (2010/07/23), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
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| 23:42.27 | *** join/#asterisk NEEDINGHELP123 (Mordi@v58.sgsvr.com) |
| 23:42.46 | NEEDINGHELP123 | hi guys have a real issue, need some help project due next week |
| 23:42.53 | NEEDINGHELP123 | isue is like this: |
| 23:43.04 | xheliox | gets the popcorn |
| 23:43.31 | NEEDINGHELP123 | i am creating a call via asterisk's ooh323 module to another test server of mine, i see the answer in the asterisk CLI , but no record of the answer in the CDR |
| 23:43.35 | NEEDINGHELP123 | it still says NO ANSWER |
| 23:45.11 | NEEDINGHELP123 | some help please guys? |
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