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| 01:05.57 | MasterSenpai | Even when the feature codes are enabled, why are they not working. I can call other phones hooked up to network, but when I try something like *98 for voicemail, it doesn't do anything and just hangs up. I also can't even dial outside, even though I am using Google Voice motif. help please |
| 01:08.03 | WIMPy | Does your phone let you? |
| 01:10.04 | MasterSenpai | Let me what? I can receive inbound calls through Google Voice motif. that is no problem. I just can't dial any number from phones other than phones that are connected to Asterisk. The phones are Nortel i2004 and they are using the unistim module and protocol |
| 01:11.06 | WIMPy | Let you dial those numbers. |
| 01:11.22 | WIMPy | Ah |
| 01:11.27 | MasterSenpai | I get I dial tone. I can dial extensions internally |
| 01:11.40 | MasterSenpai | I just cant use feature codes and outbound calls |
| 01:11.52 | MasterSenpai | like when I dial, it doesnt even make attempt. it just hangs up |
| 01:12.06 | WIMPy | has no idea about the unistim stuff. |
| 01:13.07 | MasterSenpai | Even if you don't know anything about unistim, do you know how to make sure that the google voice motif outbound settings are correct? |
| 01:14.06 | WIMPy | No, no idea abnout google either, other than that it's going to end soon anyway. |
| 01:14.25 | MasterSenpai | that sucks. i understand |
| 01:14.51 | MasterSenpai | anybody else knows how to deal with google voice motif, nortel 2004 and unistim protocol |
| 01:15.51 | WIMPy | Well, the most activity inhere is during US office hours. |
| 01:16.35 | MasterSenpai | this my second attempt to ask someone. I asked someone on Wednesday and no one said much. I will just keep trying |
| 01:17.10 | WIMPy | There probably aren't many unistim users. |
| 01:17.41 | MasterSenpai | I'm just going to keep trying anyway. better to keep trying than paying over $300 for tech support |
| 01:17.58 | WIMPy | From whom? |
| 01:18.14 | MasterSenpai | FreePBX direct support |
| 01:18.39 | MasterSenpai | you can get support directly from them but they charge you by half hour or hour and its expensive |
| 01:19.01 | WIMPy | Ok, but you know that the FreePBX part isn't supported here? |
| 01:19.25 | MasterSenpai | Yeah. My problem is the asterisk side |
| 01:19.48 | MasterSenpai | the FreePBX is working fine |
| 01:20.08 | WIMPy | As long as you can figure out how to translate the to FreePBX, ok. |
| 01:21.19 | MasterSenpai | i could be wrong though if the problem stems from how FreePBX is routing calls |
| 01:21.25 | MasterSenpai | if it isn't though |
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| 01:41.00 | rexwin_ | hi i have client sip phone incoming calls dropping at 30 seconds. i know the the sip phone is sending request for ack and sip server sends them back but it is not getting it ACK back as somehow blocked by the router connected to sip client machine. |
| 01:41.42 | rexwin_ | Anybody know how to solve this? I have tried all troubleshooting step like disable ALG but didnot work out |
| 01:52.02 | Penguin | If you can make a call from the phone to any other device on the asterisk, but not out to the world, it's a dialplan problem. |
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| 02:09.22 | rexwin_ | no i make outgoing call perfect but incoming drops in 32 seconds and it is my router dropping the packets. don't know how to resolve this so that sip phone can get ACK signal from the asterisk |
| 02:16.41 | rexwin_ | Reason: SIP;description=""ACK not received"" |
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| 03:32.11 | rexwin_ | incoming drops in 32 seconds, nat can somebody help me out? |
| 03:32.19 | rexwin_ | nat problem |
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| 04:17.01 | rexwin_ | i sad guys that this 32 seconds problem don't have a definite resolution. |
| 04:35.08 | rexwin_ | there is nothing but crap for this issue. it is router problem and has nothing to do with the sip server. don't know why all crap on the internet wants to do change with the pbx server? |
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| 09:11.22 | McNopZ | Hi Guys, is it possible in anyway for a Agent to disable Wrap-Up-Time, after a end call - can't find anything :-( |
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| 13:32.48 | D30 | hi everyone, i have a question regarding this |
| 13:32.49 | D30 | Unable to create translator path for (g729) to (ulaw) on IAX2 |
| 13:33.00 | D30 | what does it mean actually??? im kind of confuse :( |
| 13:36.04 | D30 | anyone experienced this issue ^ ? |
| 13:37.52 | WIMPy | Have you run out of licences? |
| 13:39.40 | MaliutaLap | Do you have a g729 licence to begin with? |
| 13:39.58 | MaliutaLap | is the licence file in the right location? |
| 13:41.14 | D30 | hi WIMPy MaliutaLap , im sorry if this caused you trouble but i actually disregarded this in the trunk config allow=g729 |
| 13:41.25 | D30 | i changed it to allow=ulaw |
| 13:41.35 | D30 | now it finally worked. |
| 13:42.06 | D30 | i dont know that g729 requires licenses :( |
| 13:42.36 | aamu11 | yup u require to pay.... |
| 13:42.53 | D30 | ahhh |
| 13:43.24 | D30 | anyways thank u for that tip... |
| 13:43.34 | aamu11 | :) |
| 13:44.47 | MaliutaLap | Well unless it's just for passthrough ... but you can get the licence from the digium site, and dowload the codec file aswell |
| 13:45.24 | MaliutaLap | the licence ties to the network conf of the machine, and can only be moved 3 times |
| 13:46.05 | D30 | thank you MaliutaLap |
| 13:46.15 | MaliutaLap | NP |
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| 20:25.04 | Flash_G0rd0n | Hello guys,I'd like to know what is the easiest way to replace/alter the "To:" sip header in INVITE message? |
| 20:31.33 | wdoekes | Flash_G0rd0n: there is some Dial(SIP/... magic for that |
| 20:31.45 | wdoekes | exclamation mark perhaps |
| 20:32.20 | WIMPy | o.O |
| 20:32.54 | wdoekes | <PROTECTED> |
| 20:32.56 | wdoekes | <PROTECTED> |
| 20:32.56 | wdoekes | <PROTECTED> |
| 20:32.57 | wdoekes | ... |
| 20:33.03 | wdoekes | <PROTECTED> |
| 20:33.03 | wdoekes | <PROTECTED> |
| 20:36.41 | Flash_G0rd0n | @wdoekes thank you |
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| 21:47.53 | beano_ | Morning all. Can anyone point me in the right direction to set up asterisk to forward all numbers coming in on one number out on another SIP trunk? Ta |
| 21:48.54 | WIMPy | Dial() |
| 21:51.43 | beano_ | Thanks, that looks like the one. |
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| 22:24.22 | ghost75 | sip trunk :< |
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| 23:39.08 | socomm | When working with VoIP, are there great benefits of using gigabit versus megabit NICs? |
| 23:39.14 | socomm | For sip end-points. |
| 23:40.59 | WIMPy | You can talk 10 times faster. |
| 23:55.30 | SpeedEvil | Do you have over one billion users? |
| 23:55.35 | beano_ | Ok, stuck on some stuff in extensions_additional.conf. In my ext-did section I want to pass the inbound number in the 4th param to Dial(). Can I pick up the inbound number in a variable or some sort? Thx |
| 23:56.03 | SpeedEvil | Once you get below perhaps 25% of the maximum capacity of the link - there is no meaningful change whatsoever. |
| 23:57.02 | WIMPy | beano_: Sounds like a question for #freepbx |
| 23:57.50 | beano_ | OK, is the extensions handling not part of asterisk then? |
| 23:58.43 | beano_ | Sorry, I'm a bit confused about which is what. Have been bending my head around freeswitch for the last while, and since I'm actuall a Cisco guy, I'm good and confused by now. |
| 23:59.40 | WIMPy | Yes it is, but the configuration is FreePBX. |
| 23:59.50 | beano_ | Cool, thanks |