| 00:17.25 | *** join/#asterisk dorphalsig (be543448@gateway/web/freenode/ip.190.84.52.72) |
| 00:21.55 | dorphalsig | Hello, I am looking @ a cdr report and I see some of the status in calls that go to a queue is 'Busy', when would I get that? When the Queue is full? |
| 00:24.37 | dorphalsig | looks at the ball of hay tumble down the packed but quiet room |
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| 03:06.08 | uskerine | hi, i have setup a test environment: asterisk server and three softphones connected through WiFi directly to server LAN: one pc with zoiper and two android phones with sipdroid. As said everything same LAN. |
| 03:07.36 | uskerine | When I call from zoiper to one android everything is fine |
| 03:07.56 | uskerine | if i call from that android to zoiper, audio only works in onw way |
| 03:08.11 | uskerine | with the other android audio only works in one way, no matter who makes the call |
| 03:08.19 | uskerine | any ideas? |
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| 03:14.40 | uskerine | i am reading that sipdroid is not very reliable, any other recommended SIP softclient? |
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| 05:13.02 | dijib | anybody with issues? |
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| 05:43.50 | ChannelZ | always |
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| 05:49.08 | ChannelZ | wow.. lots of spam calls today |
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| 11:33.26 | rjvvliet | anybody a sugestion? asterisk started from console (asterisk -c) works,but when started from init is runs but cannot connect to CLI(asterisk -r) asterisk.ctl is missing. |
| 11:33.48 | rjvvliet | i'ts my first time this happens, no clue where to start. OS is CentOS |
| 11:36.50 | rjvvliet | even using safe_masterisk it is working, only not when using /usr/init.d/asterisk start or service asterisk start, then no CLI |
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| 12:07.40 | ujjain | I have 2 questions that are probably not that smart. |
| 12:07.41 | ujjain | 1) Is it possible to send both text messages and fax messages via SIP? |
| 12:07.46 | ujjain | 2) Can Asterisk somehow send fax messages via Voipbuster? |
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| 13:38.41 | atekinalp | hi when i try to transfer a call using dial application i get "SIP/2.0 403 No Rates Found for Account 2661" what does this mean? |
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| 14:32.29 | slav3_kitten | off to the zoo an what not |
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| 14:44.12 | ruben23 | hi guys |
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| 17:50.55 | ChannelZ | rjvvliet: do you have access? IE if asterisk is running as root but you're a normal user trying to connect, no worky. |
| 17:51.34 | rjvvliet | ChannelZ: Thank s for you response, i figured out the problem 7mins ago. |
| 17:52.07 | rjvvliet | ChannelZ: it was the wronmg kernel source with the rwrong kernel running, this made DAHDI not compile and complain about CRC_CCITT |
| 17:54.01 | rjvvliet | ChannelZ: i'am using CentOS to build from scratch, i was using Debian previously, and it has been a while sind i have build Asterisk from scratch. |
| 17:56.57 | ChannelZ | Good. I know it was from hours ago but I just sat down and it happened to be on my screen :) |
| 17:59.17 | rjvvliet | ChannelZ: Yea, thanks for replying, at the time i had no idea where to continue. i know CRC_CCITT is needed for DAHDI, but did not see the errors, DAHDI installed whithout complaining so i missed it untill i checked my screen buffer. |
| 18:00.08 | rjvvliet | ChannelZ: Have to say, this is one i'll not easy forget ;-) |
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| 18:08.07 | Thorondor | Hi |
| 18:08.57 | Thorondor | Has anyone seen a Cisco 7912 phone with SIP firmware working with asterisk? |
| 18:10.57 | Thorondor | I can see there is an issue with the phone arbitrarily suppressing sound |
| 18:11.42 | *** join/#asterisk greenwolf (aefeb1ea@gateway/web/freenode/ip.174.254.177.234) |
| 18:12.16 | greenwolf | does anyone know why my asterisk server keeps -- ast_get_srv: SRV lookup for '_sip._UDP.sip.flowroute.com' mapped to host sip-nv1.flowroute.com, port 5060 |
| 18:12.41 | greenwolf | its about every 15-20 seconds that this is being put out in the CLI |
| 18:13.30 | rjvvliet | greenwolf: i think it is because verbose or debug is set to the level where this message is reported. |
| 18:13.51 | greenwolf | so its not the asterisk server sending out this to my carrier? |
| 18:14.44 | rjvvliet | greenwolf: It's asterisk reporting to you that is resolved the SRV record. |
| 18:15.16 | greenwolf | ok so my question is...should this be reporting this often i guess? |
| 18:17.01 | *** join/#asterisk _-Jon-_ (~jon@air.xerodns.org) |
| 18:17.06 | _-Jon-_ | Hello all |
| 18:17.15 | rjvvliet | greenwolf: not sure but i would say yes, the SRV records desination could changed. |
| 18:17.38 | greenwolf | ok thanks alot rjvvliet |
| 18:17.40 | ChannelZ | I imagine actually that message comes up every time you call |
| 18:18.01 | greenwolf | just didnt want asterisk to use all its resources and slow it down |
| 18:18.01 | greenwolf | ya kno |
| 18:18.53 | _-Jon-_ | I had a question which I'm not sure if this is provider specific, or if there is something in the configs that can do what I need.. I have two DIDs from the same provider (under the same credentials), one is billed a per minute rate, and the other is free. Is there some way I can get Asterisk to specify which outgoing DID to use in this case? |
| 18:19.07 | rjvvliet | greenwolf: yea, i know, but DNS requests are not that resource intensive |
| 18:19.25 | greenwolf | ok lol i didnt think so :) |
| 18:19.41 | ChannelZ | _-Jon-_: I guess that depends on the provider and how they consider 'what DID you dial out of it' |
| 18:20.12 | ChannelZ | If they identify/bill it based on the CID you set then that's easy |
| 18:20.29 | _-Jon-_ | Hmmm damn, I was hoping there was a config option to try to get it to specify. Kind of like CID.. Sometimes the provider will let you set your own, and others force you to use what they want |
| 18:20.41 | ChannelZ | just Set(CALLERID(num)=12345) before you Dial() |
| 18:20.48 | _-Jon-_ | Let me give that a shot! |
| 18:20.50 | _-Jon-_ | Thanks |
| 18:21.04 | ChannelZ | The way to separate the 'accounts' are through different SIP peers but if they make you use the same one, you'd have to ask them. |
| 18:22.10 | _-Jon-_ | Hmm yeah I tried authenticating with the free DID, but no go. I'll try to CID trick and if not, I'll have to contact them |
| 18:22.17 | _-Jon-_ | Thank you very much for your help :) |
| 18:22.33 | ChannelZ | good luck |
| 18:22.38 | _-Jon-_ | Thanks! |
| 18:30.25 | rjvvliet | i'am curious, has anyone here has done the DCAP exam? |
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| 21:38.32 | billytwowilly | anybody have any experience getting v-tech phones to work with chan_mobile? |
| 21:45.39 | jpsharp | Are you talking about VTEC cordless phones? |
| 21:45.52 | billytwowilly | yes. I have one that has bluetooth support. |
| 21:46.02 | billytwowilly | you can hit "cell" on it and connect it to your cell phone to make calls. |
| 21:46.10 | billytwowilly | I'm trying to connect it to asterisk to make calls instead. |
| 21:47.07 | billytwowilly | branded v-tech in Canada |
| 21:48.32 | jpsharp | I haven't done it with the vtech phones, I just remember using chan_mobile with my Treo700p was a cast iron pain in the ass. |
| 21:48.49 | billytwowilly | so far this has been a pain in the ass... |
| 21:49.02 | jpsharp | Well, actually, chan_mobile was the easy part. Pairing it up with the linux BT stack was the hard part. |
| 21:50.00 | billytwowilly | I think I have that bit working... not sure. this vtech phone doesn't stay discoverable all the time, just when you put it into "add new device" mode |
| 21:50.15 | billytwowilly | I can see it in kde bluetooth taskbar thing. |
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| 21:57.06 | billytwowilly | all the docs I see on the web relating to bluetooth seem to be from 2008 and earlier. |
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