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| 00:15.11 | WIMPy | +n |
| 00:15.16 | WIMPy | oops |
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| 00:29.18 | sawgood | Anyone know of a 'package' build already for Asterisk which would allow someone to 'dial' and extension (thus turning out a report via email in .PDF for the CDR)? |
| 00:29.34 | sawgood | So, if someone wanted a report of all their calls, they simply dial an extension? |
| 00:30.06 | WIMPy | System() |
| 00:30.43 | sawgood | so I could build it ... using the System() command ... |
| 00:30.49 | sawgood | I'll read on it now ... thanks! |
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| 00:31.22 | WIMPy | Shouldn't take more than a tail|mail. |
| 00:33.42 | sawgood | thank you! |
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| 00:38.41 | xpot-mobile | Question: if RDP packets are being sent to the IP of callee phone from caller phone... does that mean a re-invite is trying to take place? I see the callee phone responding to RDP packets from the server and ignoring packets coming directly from the caller phone. |
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| 00:41.09 | kaushal | Hi |
| 00:41.17 | pabelanger | xpot-mobile: RTP, RDP is for windows :p |
| 00:41.28 | kaushal | Anyone here comfortable with python ? |
| 00:41.31 | xpot-mobile | lol, RDP |
| 00:41.45 | xpot-mobile | i meant RTP of course ;) |
| 00:41.55 | pabelanger | kaushal: little |
| 00:42.02 | kaushal | pabelanger: ok |
| 00:42.14 | kaushal | pabelanger: let me pastebin the python code |
| 00:42.43 | kaushal | I have this python code http://paste.pocoo.org/show/476751/ |
| 00:43.13 | pabelanger | xpot-mobile: Usually, if RTP is directly between phones, a REINVITE would have happened. Unless you called the phone directly, via SIP URI |
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| 00:43.47 | kaushal | I have enabled 8 * 25 channels using call files. this code generates only 200 call files and not 240 numbers which is present in the mobileNo.txt file |
| 00:44.15 | kaushal | please suggest |
| 00:44.42 | kaushal | so what happens is it calls only 200 numbers and misses 40 numbers |
| 00:44.54 | kaushal | the reason being i have enabled 200 channels |
| 00:45.23 | WIMPy | And what is the question? |
| 00:45.39 | pabelanger | 8 * 25 is 200, so you are only looping 200 times |
| 00:45.52 | kaushal | so is there a way to check if the existing channel is free once first user hangs up and the rest of the 40 numbers are also accomodated |
| 00:45.55 | pabelanger | regardless of the number of lines in moblieNo.txt |
| 00:45.56 | kaushal | pabelanger: yes |
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| 00:46.03 | kaushal | thats the issue |
| 00:46.37 | WIMPy | Haven't we been over that two weeks ago? Use AMI. |
| 00:47.01 | pabelanger | You'll need to monitor asterisk and track the state of the channels, when it hangs up, create another call file |
| 00:47.31 | kaushal | WIMPy: i have the dialer which is written in java and use AMI |
| 00:47.41 | kaushal | so i get Error 101 |
| 00:47.41 | pabelanger | kaldemar: StarPy: https://github.com/asterisk-org/starpy |
| 00:47.52 | WIMPy | I guess you could generate calls from the h extension. Not sure how good that idea is. |
| 00:48.15 | kaushal | pabelanger: so how would i check for the state of the channels? |
| 00:48.23 | WIMPy | kaushal: And now with 200 calls, it's all fine? |
| 00:48.28 | kaushal | I mean what exactly i need to look |
| 00:48.40 | kaushal | WIMPy: yes |
| 00:48.58 | *** join/#asterisk User_CL (~UserRegis@pc-131-119-74-200.cm.vtr.net) |
| 00:49.11 | pabelanger | kaushal: you can originate a call with AMI, when it returns success or failure, move to the next number to dial |
| 00:49.13 | WIMPy | I need a decoder with state machine. Definitely. |
| 00:49.17 | User_CL | hi !!! |
| 00:49.32 | kaushal | pabelanger: Pleae help me understand about "monitor asterisk and track the state of the channels, when it hangs up, create another call file" |
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| 00:49.39 | kaushal | I am using call files |
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| 00:51.09 | xpot-mobile | pabelanger: ok, I notice that VPN_Side(10.1.10.147:2236 --> 192.168.1.180:2230), LOCAL_Side(192.168.1.16:2236 --> 192.168.1.180:2230)... 192.168.1.16 is the LOCAL_Side IP assigned to the VPN. Then LOCAL_Side(192.168.180:2230 --> 192.168.1.1:2236) with no response from 192.168.1.1 of course. Any idea why it would not reply back to 192.168.1.16? The * server is 192.168.1.10. I assume a REINVITE since the phones are sending RTP pack |
| 00:51.09 | xpot-mobile | ets to each other... or at least trying to. |
| 00:51.51 | pabelanger | kaushal: Asterisk generates events for the AMI, so you need to connect to it and monitor them. EG: Event: NewChannel, Event:Blah, Event:Hangup |
| 00:52.36 | kaushal | pabelanger: so with that python script it wont be able to do it ? |
| 00:52.52 | kaushal | pabelanger: still not understood |
| 00:53.01 | pabelanger | kaldemar: You can use StarPy, linked above |
| 00:53.29 | WIMPy | kaushal: Get a programmer, preferrably one who is comfortable with sockets. |
| 00:53.38 | kaushal | WIMPy: ok |
| 00:53.49 | pabelanger | xpot-mobile: without looking at the SIP trace, we cannot see what is happening |
| 00:54.03 | xpot-mobile | ok, I will get a pastebin |
| 00:57.41 | User_CL | hi... which is the better adapter sip ? |
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| 01:01.15 | modexi | hey room, i have a sip trunk with 4 DIDs bound to it. is there a way to set the outbound DID for queues? |
| 01:01.46 | xpot-mobile | pabelanger: http://pastebin.com/8k5pgXGw |
| 01:06.07 | xpot-mobile | modexi: yes |
| 01:06.34 | xpot-mobile | or rather an outbound did that if called will bring them into your queue |
| 01:06.39 | leifmadsen | User_CL: between which devices? |
| 01:07.10 | xpot-mobile | modexi: and the proper term is probably CallerID |
| 01:07.21 | leifmadsen | modexi: what do you mean outbound DID for the queues? are your queues calling people remotely? |
| 01:07.41 | modexi | xpot-mobile, i already have that. my provider is voip.ms and they have "value" and "premium" outbound routing which cost different amounts |
| 01:08.07 | leifmadsen | you'll have to be more clear in your questoin then |
| 01:08.15 | modexi | my incoming calls are routed to queues, which call a couple sip clients and also a couple cell phones as backups |
| 01:08.41 | leifmadsen | ok, so use a Local channel and Set(CALLERID(num)=4165551212) before you call the SIP channel |
| 01:08.49 | modexi | i want those backups to originate from XXX#, which costs less than YYY# |
| 01:08.57 | modexi | thank you very much. |
| 01:11.34 | modexi | @leifmadsen, sorry for my lack of understanding here, but is there a cleaner way to do that than creating an extension in extensions.conf that if called, does that before calling the cell phone? |
| 01:12.40 | leifmadsen | modexi: use separate peers for standard vs premium calls? |
| 01:13.00 | leifmadsen | then you can use callerid= I think on the peer |
| 01:13.11 | modexi | i see. thanks once again. |
| 01:15.06 | xpot-mobile | modexi: from what I understand of voip.ms standard vs premium route is dependant on which option you select in their portal. So if you dial XXX on standard route will cost one value, while XXX on premium route will cost another value. Am I way off here? |
| 01:16.07 | xpot-mobile | i don't think setting your CallerID is going to change the cost of your dial. |
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| 01:18.50 | modexi | xpot-mobile: i have 2 local numbers on premium and 2 8XX numbers on value |
| 01:19.29 | modexi | xpot-mobile: i was just trying to get the outgoing calls from queues.conf to go out through the local numbers for premium routing + lower costs |
| 01:20.10 | modexi | xpot-mobile: outbound calls seem to be going out one of the 8XX numbers, if i read my billing correctly |
| 01:22.13 | xpot-mobile | modexi: I imagine your using SIP termination with voip.ms... so the outbound call is not really going out over 8XX DID... rather you are likely sending that CallerID with your call. The cost you are accrued is dependant on who you are calling and not where you are calling from. |
| 01:25.11 | modexi | ah, that makes sense. i think i misinterpreted something on their website. |
| 01:26.21 | modexi | "This defines the route the system will use when you place a toll free call to Canada or the USA" -- i interpreted that at first as any call coming from the 8XX number, but now that i re-read that, i interpret that as an outgoing call to any 8XX number |
| 01:28.41 | xpot-mobile | :) |
| 01:29.31 | modexi | i guess the calls are billed as they are and it's really not enough money to worry about. my main concern was the premium routing of outbound calls without the $1.05/m charge for premium 8XX routing |
| 01:31.58 | modexi | and further inspection indicates it is using the premium routing... so sorry for wasting your time. i appreciate your help, though. |
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| 01:50.14 | p3nguin | As far as I know, you can't choose premium routing for some calls and value route for others from the same account. |
| 01:51.38 | p3nguin | xpot-mobile: You were right, that's exactly how it works. Premium costs more than value. |
| 01:52.28 | p3nguin | You'll pay about 1.2 cents per minute on premium and 1.05 cents per minute on value, if I remember the figures correctly. |
| 01:53.19 | p3nguin | modexi: Also, there is no such thing as an outbound DID. DID is direct INWARD dialing. |
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| 02:37.16 | BugKhaM | How to limit sip session from a user? |
| 02:37.38 | BugKhaM | I am using call-limit=xx but it doesn't work in all cases |
| 02:38.11 | BugKhaM | iptables? |
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| 04:28.43 | PhoenixMage | Hi guys, anyone know how I can configure a cisco 7975G's switch? I want the PC on one vlan and the phone on the voice vlan |
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| 04:43.21 | Nugget | PhoenixMage: your question doesn't make sense |
| 04:44.20 | PhoenixMage | Nugget: The 7975G has the usual switch in it, one port hardwired to the phone, one pc port and the uplink port |
| 04:44.38 | Nugget | I am aware of tht (although I'm not sure it's a "switch" |
| 04:44.42 | Nugget | but your question doesn't make sense |
| 04:44.48 | PhoenixMage | I want the PC port configured for one port the phone port configured to the vlan and the uplink as a trunk |
| 04:44.59 | Nugget | oh, you mean it has three ethernet ports? |
| 04:45.08 | PhoenixMage | yeah sorry |
| 04:45.13 | PhoenixMage | should be more clear |
| 04:45.22 | Nugget | nutty, sorry. I didn't realize that |
| 04:45.33 | PhoenixMage | I am assuming you configure the vlans in the cnf.xml file? |
| 04:45.33 | Nugget | I thought it just had the two like the 45/65 phones |
| 04:45.59 | PhoenixMage | Nugget: 2 physical ports, one hardwired |
| 04:46.15 | Nugget | then I'm confused again :) |
| 04:46.19 | PhoenixMage | lol |
| 04:46.23 | Nugget | what do you mean by "hardwired"? |
| 04:46.25 | PhoenixMage | sorry |
| 04:46.31 | Nugget | there's an in port and an out port. |
| 04:46.37 | PhoenixMage | it exists but its tied to the phone |
| 04:46.37 | Nugget | and I'm pretty sure it's just a hub |
| 04:46.39 | WIMPy | internal |
| 04:46.46 | PhoenixMage | no its not a hub |
| 04:46.46 | WIMPy | What's confusing about that? |
| 04:47.03 | PhoenixMage | definately a switch otherwise you could easily vlan hop |
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| 04:47.59 | PhoenixMage | Googling it seems that you have to configure it from the upstream (cisco) switch and it cant be done in the cnf.xml |
| 04:48.33 | WIMPy | How is that supposed to work? |
| 04:48.59 | Nugget | ah, I see what you mean now |
| 04:49.24 | PhoenixMage | WIMPy: CDP |
| 04:49.58 | WIMPy | Does that onclude vlan autoconfiguration? |
| 04:50.45 | PhoenixMage | WIMPy: what do you mean? |
| 04:51.14 | PhoenixMage | Seems there is a switchport voice vlan <vlan #> command |
| 04:51.29 | WIMPy | The only thing I know about cdp is how to switch it off :-) |
| 04:51.49 | PhoenixMage | heh, it can be noisy |
| 04:52.03 | WIMPy | yes |
| 04:56.18 | PhoenixMage | not sure if you can set it as an access port and have both a voice vlan and a data vlan |
| 04:56.21 | PhoenixMage | will find out when I get home |
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| 05:24.14 | irroot | morning folks |
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| 05:27.39 | justdave | so, I've been experimenting a little with using regcontext and regexten in sip.conf with the intention of dropping identical configs on multiple asterisk servers, and using DUNDI to route the calls to the one the user is actually connected to at the time... |
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| 05:28.13 | justdave | and adding a user to the context when they register seems to work fine, but I haven't figured out how to make that dialplan entry go away when the user is no longer connected |
| 05:29.22 | justdave | if the client actually unregisters it goes away |
| 05:29.43 | justdave | but if they don't unregister, it seems to be there forever, even after their registration times out |
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| 05:31.19 | irroot | justdave that seems odd you using realtime ?? if the peer expires from cache it might work then |
| 05:31.32 | justdave | nope, no realtime, just sip.conf |
| 05:32.06 | kaldemar | justdave: see regextenonqualify |
| 05:32.35 | justdave | where's that documented? |
| 05:32.52 | kaldemar | justdave: in the sample sip.conf, at least in 1.8. |
| 05:33.17 | justdave | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf doesn't list it, and my current sip.conf doesn't have it in comments :) |
| 05:33.25 | justdave | gotta go find a distribution copy of it then |
| 05:33.38 | kaldemar | in 1.6.2 too. |
| 05:33.52 | kaldemar | quit using voip-info as a configuration reference. |
| 05:34.05 | kaldemar | http://svn.digium.com/svn/asterisk/tags/1.8.6.0/configs/sip.conf.sample |
| 05:35.20 | justdave | that files doesn't have it |
| 05:35.47 | justdave | oh, it's regextenonqualify |
| 05:35.54 | justdave | missed the "on" in the middle |
| 05:36.00 | justdave | got it |
| 05:36.01 | justdave | thanks |
| 05:36.09 | irroot | its not inside it ON top :P |
| 05:36.22 | irroot | thx kaldemar |
| 05:36.44 | kaldemar | no problemo. |
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| 08:51.53 | x1user | Hi, I have few phones connected with chan mobile. How can i access the asterisk system not via sip client, but via the number of some phone??? |
| 08:52.19 | x1user | Instead from calling trough a sip client, i want to call from a standart gsm phone? |
| 08:53.48 | kaldemar | call the number of an asterisk connected phone. |
| 08:55.43 | x1user | Yes but then the phone just rings. |
| 08:55.53 | x1user | I doesnot goes to the context in the dialplan |
| 08:56.29 | kaldemar | then the phones are not connected to asterisk properly. |
| 08:58.12 | x1user | It is connected, with state free. |
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| 09:18.47 | tuxx- | hiya |
| 09:21.56 | tuxx- | is there an option for the queue that when ever agent is in use, the caller exits the queue? I tried it with leavewhenempty=strict, but that only works if the agent is either unavailable or paused i think. |
| 09:23.33 | irroot | tuxx ringinuse / ignorebusy |
| 09:24.20 | tuxx- | mkay, but will that make the caller exit the queue when all agents are in use? |
| 09:24.46 | tuxx- | i see now that i already have the option 'ringinuse=no' |
| 09:25.27 | irroot | ok it wont exit it will wait for them to be unbusy |
| 09:30.09 | irroot | tuxx- can see where to do this |
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| 09:36.55 | tuxx- | right :) |
| 09:41.42 | irroot | tuxx- on further look if the device state is handled properly it should work |
| 09:44.50 | tuxx- | hmkay, well i say all my queue agents have the status "(In Use)". But my caller stayed in the queue. Having the options 'leavewhenempty=strict' and 'ringinuse=no' |
| 09:44.56 | tuxx- | s/say/saw |
| 09:46.05 | tuxx- | meh, gonna test it out in my development enviornment... they want me to test it live here, but debugging is awfull with this much calls going through asterisk ;p |
| 09:46.11 | wdoekes2 | in use is not empty, afaik |
| 09:46.27 | wdoekes2 | you could set a queue timeout, I think |
| 09:46.48 | wdoekes2 | although that would not be immediate, of course |
| 09:48.00 | wdoekes2 | *empty = (QUEUE_EMPTY_PENALTY | QUEUE_EMPTY_INVALID | QUEUE_EMPTY_PAUSED | QUEUE_EMPTY_UNAVAILABLE); |
| 09:48.11 | wdoekes2 | does not include QUEUE_EMPTY_INUSE |
| 09:48.28 | wdoekes2 | tuxx-: leavewhenempty=strict,inuse ? |
| 09:48.44 | *** join/#asterisk ocx (c27e0e65@gateway/web/freenode/ip.194.126.14.101) |
| 09:49.04 | tuxx- | gonna test it out |
| 09:50.42 | *** part/#asterisk lister_ (~lister@scarda1.lnk.telstra.net) |
| 09:50.45 | *** join/#asterisk lister_ (~lister@scarda1.lnk.telstra.net) |
| 09:51.15 | irroot | tuxx- cool make a new option perhaps for this i still dont see a usage case |
| 09:52.05 | *** join/#asterisk TheNoOne (~thenoone@91.114.16.34) |
| 09:52.31 | TheNoOne | hi ⦠where do i configure the EC taps when using mg2 with dahdi in asterisk 1.6? |
| 09:52.38 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 09:53.33 | TheNoOne | dahdi show channel 1 .. says "Echo Cancillation: 1taps" |
| 09:53.38 | TheNoOne | i thought 32 is min.? |
| 09:54.50 | tuxx- | irroot: doesnt work :-( |
| 09:55.01 | tuxx- | eh |
| 09:55.04 | tuxx- | wdoekes2: doesnt work ;\ |
| 09:55.15 | tuxx- | irroot: do you mean i should make a feautre request out of this? |
| 09:56.59 | irroot | yeah |
| 10:01.18 | wdoekes2 | tuxx-: which asterisk version? |
| 10:01.54 | StaRetji | p3nguin: ping |
| 10:02.26 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 10:03.22 | *** join/#asterisk iulhk (~carryoff@119.152.220.11) |
| 10:29.09 | *** join/#asterisk sunfone (~jeff@c-67-184-36-158.hsd1.il.comcast.net) |
| 10:30.21 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 10:34.40 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 10:37.38 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 10:54.01 | stix | Why are my channels sometimes called SIP/CME22403-0000084? I don't know where this CME is coming from⦠normally the channels are something like SIP/0700-00000652 |
| 10:54.02 | *** join/#asterisk etc (~root@124-168-89-27.dyn.iinet.net.au) |
| 10:54.14 | etc | hello asterisk overlords. |
| 10:57.37 | etc | my sip.conf is http://paste2.org/p/1653540 - i can't register against my sip server - missing username in From: digest response, eg From: "123" <sip:p123@host.com>;tag=456 - any ideas? |
| 10:59.35 | kaldemar | etc: try register => user:secret:authuser@host/extension |
| 11:00.19 | kaldemar | stix: do you know where 22403 is coming from? |
| 11:04.01 | etc | kaldemar: no luck |
| 11:04.45 | kaldemar | etc: show what happens |
| 11:06.09 | *** join/#asterisk garymc (~chatzilla@host81-139-71-64.in-addr.btopenworld.com) |
| 11:06.14 | etc | the response to the challenge doesn't have the "fromuser" before <sip:> |
| 11:06.37 | etc | (and registration fails) |
| 11:07.13 | etc | I've got a working sample from a polycom registering and am comparing - the major difference appears to be the challenge response |
| 11:08.28 | kaldemar | you were talking about digest earlier |
| 11:10.12 | etc | oh, you're right. my mistake. |
| 11:10.18 | etc | I do mean the From: line |
| 11:10.46 | etc | the [authentication] does the digest as I need it. |
| 11:11.04 | etc | i figured the fromuser would do the From: |
| 11:11.59 | kaldemar | fromuser is a device option, not for [general]. |
| 11:12.33 | etc | ah, this is where my asterisk'isms fail. |
| 11:13.18 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 11:13.25 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 11:14.05 | etc | is a "friend" type for outgoing and incoming? |
| 11:14.36 | kaldemar | friend creates both peer and user. a peer is really enough. |
| 11:14.44 | etc | ; * The type=friend is a device type that accepts both incoming and outbound calls, where Asterisk match on the From: username on incoming calls. (A synonym for friend is "user"). This is a type you use for your local SIP phones. |
| 11:14.46 | etc | ah. |
| 11:15.07 | kaldemar | where did you copy that? |
| 11:16.46 | etc | sip.conf |
| 11:16.53 | kaldemar | from sample config, i see. that is an odd remark that synonym for friend is "user", since it is not. oh well, the peer section is bullshit also. |
| 11:17.22 | etc | http://paste2.org/p/1653566 has my current config. it's not using the xyzzy device - how do I tell it to use it? |
| 11:17.58 | kaldemar | register statements belong to [general], not under a peer definition. |
| 11:19.48 | etc | how does the register which peer definition to use? |
| 11:20.44 | etc | err, how does the register know which peer definition to use? |
| 11:20.48 | kaldemar | afaik register statements don't use peer definitions, expect with callbackextension. |
| 11:21.02 | kaldemar | s/with/for/ |
| 11:21.17 | kaldemar | grrr.. except.. |
| 11:22.31 | etc | now i'm confused. |
| 11:23.31 | kaldemar | register statements have little to do with peer definitions. |
| 11:23.43 | etc | the [general] defines a register, but can't have a fromuser. during the register, there is a challenge. |
| 11:24.02 | etc | i need the fromuser to be in the challenge response From: |
| 11:24.33 | kaldemar | looks like you can't. what server is requiring it? |
| 11:25.15 | etc | BroadWorks |
| 11:26.17 | etc | i don't quite understand why the peer definition doesn't use register statements |
| 11:28.05 | kaldemar | registrations are really separate from calls in SIP. registrations are a way to let the other end know where you are, nothing more. |
| 11:30.53 | etc | perhaps i'm not using asterisk as it was intended. |
| 11:31.01 | etc | i just to register as a SIP client. |
| 11:31.51 | kaldemar | or broadworks is requiring a display name even though such a requirement does not exist in the SIP protocol. |
| 11:32.23 | etc | ah. |
| 11:32.44 | kaldemar | i have never heard of rejecting a registration because of a missing display name. |
| 11:33.22 | kaldemar | on the contrary, people make asterisk register to servers all the time, successfully. |
| 11:35.34 | etc | i've done this succesfully once before - but i can't recall if i ended up changing the source. |
| 11:35.38 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 11:36.42 | etc | while i'm asking - what's the 'right' way to shutdown asterisk? |
| 11:37.42 | kaldemar | there are many. the usual is via init script. |
| 11:41.10 | *** join/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 11:41.19 | etc | i meant from console (asterisk -r) |
| 11:41.49 | etc | i'm recompling code to test - pkill -9 asterisk works for me, but wondering if right. |
| 11:43.10 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust646.15-3.cable.virginmedia.com) |
| 11:44.12 | kaldemar | etc: "core show help core stop" will tell you more. |
| 11:44.59 | etc | excellent. thank you |
| 11:45.01 | kaldemar | gracefully means that asterisk stops accepting new calls and shuts down when the last channel is down. |
| 11:47.16 | etc | no luck, even with the "fromuser" in the From: |
| 11:47.33 | etc | i'm guessing it's the :5060 in various lines that are breaking it. |
| 11:48.25 | kaldemar | you could enable sip debug and pastebin a failing registration |
| 11:49.11 | etc | i'm watching the sip debug - it's behaving as it should - i'm comparing with something that works and something that doesn't - will give a pastebin example in a second |
| 11:49.30 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 11:50.27 | ocx | does Skype allow extension creation on their voip service? for example would that work? Dial(technology/user[:password]@remote_host[:port][/remote_extension]) |
| 11:54.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 11:54.55 | batfastad | Hi everyone. Is there a way to query an asterisk box to get the SIP presence of all connected extensions (eg: offline, available, busy)? Or do I have to manually create a presence subscription to each extension? |
| 11:55.30 | etc | ocx: no |
| 11:56.05 | ocx | no way to integrate asterisk with skype ? |
| 11:56.25 | etc | ocx: correct |
| 11:56.35 | ocx | shame |
| 11:56.36 | etc | batfastad: sip show channels ? |
| 11:56.59 | etc | kaldemar: http://paste2.org/p/1653610 |
| 11:58.36 | leifmadsen | ocx: ya you missed the boat on buying chan_skype |
| 11:59.01 | ocx | you can buy that one? |
| 11:59.10 | leifmadsen | no, you can't (like I just said) |
| 11:59.29 | ocx | that would be a revolution |
| 11:59.38 | leifmadsen | no, it wasn't :) |
| 11:59.50 | ocx | but would heavily load their servers too for free :) |
| 12:00.02 | leifmadsen | http://www.digium.com/en/products/software/skypeforasterisk.php |
| 12:00.35 | ocx | btw are you the author of the definitive guide? |
| 12:00.37 | batfastad | etc: Interesting, that could work. But if I create presence subscriptions to the extensions, do the phones send it direct to the subscriber or does it go through the asterisk box first? |
| 12:01.02 | leifmadsen | ocx: I am the co-author, yes |
| 12:01.09 | ocx | great work man |
| 12:01.26 | ocx | easy to ready and very fruitful |
| 12:01.31 | leifmadsen | batfastad: sounds like something you might want to use the AMI for |
| 12:01.38 | leifmadsen | ocx: awesome! glad you're enjoying it |
| 12:01.50 | leifmadsen | is reading Pro Puppet right now for managing Asterisk deployments... |
| 12:02.53 | kaldemar | etc: the difference between those two is most likely the correct secret. |
| 12:02.56 | batfastad | leifmadsen: AMI being Asterisk Management Interface? |
| 12:03.03 | leifmadsen | yes |
| 12:03.20 | etc | kaldemar: that's all i can think of now - with the exception of the :5060 port |
| 12:03.28 | batfastad | leifmadsen: AMI being the asterisk CLI? |
| 12:04.23 | leifmadsen | no |
| 12:04.29 | leifmadsen | ~ami |
| 12:04.30 | infobot | extra, extra, read all about it, ami is the Asterisk Manager Interface, a way to control an Asterisk server via a TCP/IP socket. See http://voip-info.org/wiki/view/Asterisk+manager+API |
| 12:05.58 | ocx | how can i resume a dialplan after a call is bridged, for example DIAL() priotity1 and some other applications to be done after the call is completed, |
| 12:06.36 | ocx | i am only able to execute applications after the dial in case of a timeout on the dial command |
| 12:06.42 | leifmadsen | infobot: no, ami is <reply> AMI is the Asterisk Manager Interface, a way to control an Asterisk server (and retrieve information) via a TCP/IP socket. More information is available at http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html and http://voip-info.org/wiki/view/Asterisk+manager+API |
| 12:06.42 | infobot | okay, leifmadsen |
| 12:10.10 | *** join/#asterisk coppice (~chatzilla@116.92.20.245) |
| 12:10.25 | mirko_brankovic | ocx: use Dial(....,,g) with g option or maybe some sub routine |
| 12:10.54 | ocx | that would continue with the next priority once the dial appl exits? |
| 12:11.31 | kaldemar | etc: put the secret in the registration line and try it, in case the realm doesn't match. |
| 12:12.09 | mirko_brankovic | ocx: yes i think so |
| 12:16.27 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 12:16.54 | aberrios | offtopic but does anyone know anywhere in the UK that repairs polycom phones that are out of warranty? |
| 12:18.00 | *** join/#asterisk voiper (442c9108@gateway/web/freenode/ip.68.44.145.8) |
| 12:18.40 | etc | kaldemar: no luck |
| 12:20.21 | voiper | Hi Guys, anyone know how to get pass this error ? Unable to create channel of type 'DAHDI' (cause 0 - Unknown) - Dahdi version 2.5.01 asterisk version 1.6.2.20. Dahdi show channels shows all the 31 channels and Dahdi status shows OK (no alarms) |
| 12:22.03 | voiper | this is the output from dahdi show status : Wildcard TE121 Card 0 OK 2 0 0 CCS HDB3 0 db (CSU)/0-133 feet (DSX-1) |
| 12:22.20 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 12:23.36 | voiper | this is the error i get while dialing out : [Sep 16 16:13:56] WARNING[4196]: app_dial.c:1780 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
| 12:23.50 | *** part/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 12:26.59 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 12:31.37 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 12:32.20 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:32.28 | *** join/#asterisk irroot (~irroot@41.204.216.10) |
| 12:33.44 | *** join/#asterisk jetlag (~jetlag@pool-71-188-8-68.cmdnnj.east.verizon.net) |
| 12:40.16 | ocx | getting this error when trying to use the goto application |
| 12:40.16 | ocx | <PROTECTED> |
| 12:40.22 | ocx | how can i debug that? |
| 12:43.47 | mirko_brankovic | ocx: what does 'sip show peers' tells you |
| 12:44.46 | ocx | its 2 lines |
| 12:45.07 | ocx | i will paste in #pastebin |
| 12:46.02 | ocx | 1001/1001 148.135.38.89 D 58675 Unmonitored 1000/1000 148.135.38.89 D 27737 Unmonitored |
| 12:46.50 | ocx | these 2 sip can call each other with no problem |
| 12:47.03 | *** join/#asterisk stix (~stix@109.56.180.174) |
| 12:48.14 | mirko_brankovic | hmm thay should be monitored |
| 12:48.57 | mirko_brankovic | like : 4 sip peers [Monitored: 4 online, 0 offline Unmonitored: 0 online, 0 offline] |
| 12:50.11 | mirko_brankovic | maybe your goto app has nothing left to do and ends call, try to use Hangup() at the end |
| 12:52.15 | ocx | this ism y extensions.conf |
| 12:52.16 | ocx | http://pastebin.com/3Mt7dXU2 |
| 12:52.22 | ocx | trying to get the goto working |
| 12:52.32 | ocx | line 17 |
| 12:52.55 | *** join/#asterisk scubes13 (~scubes13@cpe-024-088-120-157.sc.res.rr.com) |
| 12:53.13 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-dztmnzqylgixfzyq) |
| 12:53.39 | *** join/#asterisk dms (~dms@nat/digium/x-zfowakdwakxdknrq) |
| 12:55.39 | mirko_brankovic | ocx: try with exten => 600,1,Goto(TestMenu,start,1); |
| 12:55.42 | ocx | when executuin line 21 it is exiting |
| 12:55.47 | eduzimrs | Asterisk 1.6 works fine with CentOS6.0 64bits ? |
| 12:55.48 | ocx | yeai tried that |
| 12:55.53 | ocx | replacing the start with 1 |
| 12:55.57 | ocx | and vice versa :/ |
| 12:57.06 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:58.39 | mirko_brankovic | ocx: than try to answer on 600 first exten => 600,1,Answer(); and than Goto(exten => 600,1,Goto(TestMenu,start,2); |
| 12:59.17 | ocx | ok |
| 12:59.32 | *** join/#asterisk coppice (~chatzilla@116.92.20.245) |
| 13:00.56 | ocx | [Sep 16 11:59:57] WARNING[6032]: pbx.c:4818 add_pri: Unable to register extension '600', priority 1 in 'internal', already in use |
| 13:04.11 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 13:04.56 | ocx | it is working |
| 13:04.59 | ocx | thanks mirko |
| 13:05.06 | ocx | you should answer first as you said |
| 13:05.09 | ocx | before the goto |
| 13:05.17 | *** join/#asterisk serafie (~erin@nat/digium/x-zzqsvqhkpygbvtub) |
| 13:06.27 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:06.31 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.65.185) |
| 13:09.53 | *** part/#asterisk stix (~stix@193.89.191.209) |
| 13:10.21 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 13:20.26 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 13:32.44 | *** join/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 13:33.58 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 13:37.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:38.18 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:38.56 | etc | kaldemar: would Error loading module 'res_crypto': /usr/lib/asterisk/modules/res_crypto.so: cannot open shared object file: No such file or directory affect registration ? |
| 13:43.37 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 13:53.30 | *** join/#asterisk thecardsmith (~doug@pool-71-161-220-199.burl.east.myfairpoint.net) |
| 13:54.14 | thecardsmith | If I want to take a lot of SIP connections in/out of an asterisk IVR -- is there any type of PCI device I can use to offload some of the work from the processor? |
| 13:54.21 | thecardsmith | I'm kinda coming from an old school T1 mentality |
| 13:55.13 | thecardsmith | and I'm used to say... a processor on a PCI card that uses hardware to help record a call, instead of doing in software on the machine |
| 13:55.43 | thecardsmith | am I coming from outerspace? or is there an option? or do most people just use straight-up-into-a-nic? |
| 13:57.27 | coppice | why would you want to offload work from the main processor? |
| 13:58.06 | eduzimrs | anyone knows this erros: res_odbc.c:1459 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified |
| 13:58.53 | anonymouz666 | eduzimrs: you need to setup your ODBC correctly, the book explains that. please follow. |
| 14:00.01 | thecardsmith | fair question! ...i'm trying to figure out how I would build an asterisk box to scale up. Right now I have windows IVRs (which I am dying to replace), and they can take ~48 calls. But let's say... I want to take an order of magnitude -- 480 calls, and record all of those phone calls (or at least, a percentage) |
| 14:00.24 | thecardsmith | so I was thinking, SIP is probably the answer. those boxes take 2 T1's in, 2 out |
| 14:00.30 | eduzimrs | anonymouz666: ok, wich book? |
| 14:02.42 | mirko_brankovic | eduzimrs: http://www.asteriskdocs.org/ |
| 14:03.48 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:09.06 | *** join/#asterisk floh79 (~quassel@178-26-49-226-dynip.superkabel.de) |
| 14:09.33 | floh79 | Hi, could someone of you point me to a doc where I can read which port to forward for sip (voip) please? |
| 14:09.42 | floh79 | I need to setup our firewall for voip. |
| 14:10.07 | eduzimrs | ok, but im running compiling * from source and this func odbc is not available, odbc libs are already installed, im using cenOS6 64bits with * 1.6 |
| 14:10.16 | floh79 | I did forward UDP for 5060. So connection is estabilished but no voice. :( |
| 14:10.31 | eduzimrs | im running make menuselect and its not available there too |
| 14:15.13 | mirko_brankovic | eduzimrs: Asterisk 1.6 comes with no addons, so you have to install addons too |
| 14:16.24 | Qwell | func_odbc is not in addons |
| 14:16.42 | Qwell | You need to run ./configure after installing the development libs. |
| 14:25.12 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-npggnboploekstcr) |
| 14:25.30 | Katty | ATTENTION ASTERISKABUHLS |
| 14:25.36 | Katty | IT IS FRIDAY |
| 14:25.42 | Katty | DO YOU KNOW WHERE YOUR DANCING PANTS ARE!? |
| 14:26.56 | eduzimrs | @Qwell firts i install the d.libs, after ./configure ok? |
| 14:29.02 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:29.41 | eduzimrs | addons must be installed after * is completly installed right? |
| 14:29.44 | *** part/#asterisk mirko_brankovic (~mirko_bra@212.200.146.253) |
| 14:32.35 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:34.15 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-189.dsl.stlsmo.sbcglobal.net) |
| 14:34.35 | KavanS | porting config from asterisk 1.4 to 1.8, dialed system, menu sounds are not playing via the Background command...any suggestions? |
| 14:34.47 | KavanS | logs don't seem to give anything solid... |
| 14:36.11 | LemensTS | I shouldnt run into problems running a 2.6Ghz celeron on debian 6 with 2 nics (one local one net), as firewall/dhcp server and asterisk 1.8 with 6 fxo modules on pci card. 7 sip phones, probably 20 devices on the dhcp server.... |
| 14:36.29 | KavanS | LemensTS, not at all... |
| 14:36.40 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:37.07 | KavanS | I think even a p3 could get away running that :) but I'm not a top tier #asterisk expert :) |
| 14:37.08 | LemensTS | KavanS: cool. it seems alot easier to run the asterisk boxes as firewall/dhcp servers companies than trying to bandaid them up under cheap firewall routers.... |
| 14:37.18 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:37.36 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:39.18 | LemensTS | I never use makemenu select when compiling asterisk, how much difference in memory/power is their by deselecting things I dont need like moh, iax, speex, etc? |
| 14:39.32 | KavanS | now that I do not know... |
| 14:39.57 | LemensTS | I basically just use g711, g729, sip, voicemail, and standard stuff like that |
| 14:40.13 | LemensTS | Cool. Been wanting to find out for a while.... |
| 14:40.18 | KavanS | g729 will cost your cpu a little bit, but nothing serious in my experience |
| 14:40.53 | LemensTS | If i do g729 to g729 it seems to cost alot more than g711 to g711 even by not transcoding, from what i have read |
| 14:40.57 | LemensTS | that sound right to you? |
| 14:41.24 | LemensTS | like over half cpu power |
| 14:41.34 | KavanS | hrm, interesting |
| 14:41.44 | KavanS | not in my experience... |
| 14:41.58 | LemensTS | that was just on the voip-info server load page that someone had said. |
| 14:42.32 | LemensTS | Yea i didnt see how it could be that great of a difference if not transcoding. I think they may have meant to say they were transcoding |
| 14:42.53 | *** join/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 14:43.33 | mort_gib | <LemensTS> g729 is much ahrder on the CPU, no surprise ther |
| 14:43.49 | mort_gib | <LemensTS> It's more controlling the version you need and the options you need |
| 14:44.06 | mort_gib | And then of course controlling your environment more |
| 14:44.45 | batfastad | Hi everyone. Is there a way to list the useragents of all connected SIP clients? |
| 14:51.33 | LemensTS | mort_gib: even when just doing g729 passthru? |
| 14:51.53 | mort_gib | Do you provide conference rooms?? |
| 14:52.03 | LemensTS | mort_gib: nope |
| 14:52.23 | mort_gib | Well, I always compile, not to save CPU or mem |
| 14:52.43 | mort_gib | But because I want to control version and stuff I need... |
| 14:52.52 | Qwell | The CPU usage does not differ between codecs if you're using passthrough... |
| 14:53.20 | *** join/#asterisk mindCrime_ (~chatzilla@rrcs-70-62-112-146.midsouth.biz.rr.com) |
| 14:53.35 | LemensTS | Qwell: thanks |
| 14:54.05 | Qwell | That's like asking if a phone weighs more when it's using one codec over another. |
| 14:54.44 | LemensTS | Can you make a fully g729 system (voicemail, moh, etc) where it never has to transcode? |
| 14:54.50 | Qwell | sure |
| 14:55.40 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 14:56.03 | LemensTS | I would need to re-encode the voicemail and moh sound files as g729 manually, as there is not a way to compile asterisk to do that correct? |
| 14:56.28 | Qwell | All of the prompts are available in g.729. |
| 14:59.03 | LemensTS | Qwell: I assume those will show up in /var/lib/asterisk/sounds/en when i recompile with g729 support? |
| 14:59.52 | Qwell | There is no "g.729 support" compile option. |
| 15:00.00 | Qwell | You select the formats of sounds you want, and you install them. |
| 15:00.29 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 15:00.47 | LemensTS | Qwell: Ok I understand, appreciate the help |
| 15:02.59 | *** join/#asterisk irroot (~irroot@41.49.185.231) |
| 15:07.05 | *** join/#asterisk lcat (~lcat@187.45.254.107) |
| 15:10.45 | Katty | PARTY PANTS |
| 15:11.31 | irroot | Katty next sip of this beer is for you |
| 15:11.52 | irroot | ~beer Katty |
| 15:11.52 | infobot | ACTION pours a quart of Yuengling Lager for Katty |
| 15:14.25 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:15.05 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 15:22.45 | tzanger | Ooh yuengling |
| 15:22.55 | tzanger | bellies up to the bar |
| 15:24.17 | beek | asks for one. |
| 15:27.23 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 15:27.47 | *** join/#asterisk iamaham (~iamaham1@web2.supergreenhosting.com) |
| 15:27.54 | iamaham | greetings, I believe one of my fxo chips are bad. |
| 15:28.06 | iamaham | How do you initiate a call from the cli to go out via a specific channel? |
| 15:29.39 | *** join/#asterisk dms (~dms@nat/digium/x-fpcighnfvrnvmpjv) |
| 15:29.47 | irroot | iamaham originate |
| 15:30.33 | SuPrSluG | http://www.voip-info.org/wiki/view/Asterisk+CLI+dial |
| 15:31.42 | batfastad | Hi everyone. Anyone know a way to list the useragents of connected SIP devices through the asterisk CLI? |
| 15:32.53 | Qwell | You've gotta love the people that wait three whole minutes for an answer before leaving. |
| 15:33.42 | irroot | Qwell indeed |
| 15:34.29 | *** join/#asterisk iamaham (~iamaham1@web2.supergreenhosting.com) |
| 15:34.42 | iamaham | How do you use originate or have an example given source SIP # and output channel # |
| 15:34.46 | Qwell | <irroot> iamaham originate |
| 15:35.11 | Kobaz | core show application Originate |
| 15:35.27 | iamaham | no such command 'core' |
| 15:35.41 | WIMPy | Time to upgrade |
| 15:36.26 | iamaham | hrm I ended up just making a bunch of concurrent calls till I got to the channel |
| 15:36.28 | iamaham | quit |
| 15:38.23 | Katty | ewe beer |
| 15:38.33 | Katty | i don't want no beer! |
| 15:38.41 | VoipForces | Hi all, have a question.To make it short I issue an outbound call via AMI and when it is answered I transfer it to a SIP extension using a simple Dial(SIP/XYZ). But right now while the Dial is doing it's thing, the person that got the call hears ringing. Is there a way to supress that ringing ? |
| 15:38.46 | beek | hands Katty a glass of wine. |
| 15:39.23 | Katty | :< |
| 15:39.28 | Katty | don't like wine either! |
| 15:39.30 | beek | VoipForces: Look at the parameters to the Dial() application. |
| 15:39.36 | Katty | how about a bottle of malibu or ammareto? |
| 15:39.42 | Katty | and a shot glass |
| 15:39.43 | beek | Done! |
| 15:39.45 | Katty | shot glass optional |
| 15:39.54 | beek | takes the wine back and exchanges it for ammareto. |
| 15:39.59 | Katty | :>>> |
| 15:40.02 | Katty | YAY |
| 15:40.04 | beek | drinks the wine |
| 15:40.05 | VoipForces | beek: I did but did not see one that supress ringing... |
| 15:40.13 | Katty | can't drink yet |
| 15:40.16 | Katty | it's not 2pm yet |
| 15:40.33 | Katty | ...not that i... start drinking at 2... |
| 15:40.36 | beek | VoipForces: But you can play MOH, which could be silence. |
| 15:40.50 | VoipForces | beek: yeah that makes sense... |
| 15:41.49 | Katty | i miss zeeek |
| 15:41.54 | azv4 | Any Panasonic Digital Hybrid pros know off the top of their head if there is a mode a port can be in that will prevent programming of features for the phone on that port? I have a port where FWD/DND wont work, and user cant get voicemail |
| 15:41.59 | beek | He's on #vuc right now |
| 15:42.19 | Katty | :> |
| 15:42.31 | Katty | i foundsed him ^_^ |
| 15:43.01 | Katty | if anyone would like to join the asterisk christmas card exchange, please let me know! |
| 15:43.12 | beek | Katty: I signed up. |
| 15:43.23 | Katty | you sure did |
| 15:43.58 | voiper | anyone has openr2 settings for india ? |
| 15:47.38 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 15:48.38 | irroot | voiper used it recently the docs have good info im sure you will find it therew |
| 15:50.44 | voiper | hi Irroot, the docs don't have anything for india (the one from google code) |
| 15:51.15 | voiper | i goto a point the calls are hitting channel bank but somehow it is not dialing out |
| 15:51.35 | VoipForces | beek: m(none) works great thanks. |
| 15:52.51 | irroot | voiper make sure the R2 is digital and not ana type i have had a channel bank that did not work as it was not digital |
| 15:53.02 | beek | VoipForces: No problem. |
| 15:54.26 | voiper | was it valiant ? |
| 15:54.31 | Katty | irroot: you should join the christmas card exchange! |
| 15:55.04 | irroot | hehe i dont even bother with a tree |
| 15:55.44 | irroot | i could get the girlfriend to she does the christmass thing |
| 16:03.16 | *** part/#asterisk batfastad (~benbradle@93-97-161-95.zone5.bethere.co.uk) |
| 16:07.31 | eduzimrs | im compiling asterisk add-ons 1.6 so this error occurs: chan_mobile.c:1035: error: incompatible types when assigning to type âunion ast_frame_subclassâ from type âlong long unsigned intâ |
| 16:07.56 | eduzimrs | many erros on this chan_mobile.c |
| 16:08.52 | eduzimrs | any idea? |
| 16:13.41 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 16:14.45 | DrDamnit | When telnetting to the AMI, and attempting to log in, sending the action: login with credentials fails the first time. Repeating the request a second time authenticates. Is this a bug or user error on my end? |
| 16:14.45 | Nugget | telnet is eeeeeeevil! |
| 16:14.55 | Qwell | DrDamnit: user error |
| 16:15.01 | DrDamnit | I thinik so too. |
| 16:15.30 | DrDamnit | I am sending: action: login \n\r username: user \r\n secret: password \r\n \r\n, is this correct? |
| 16:16.34 | DrDamnit | What's interesting, is if I send \r\n \r\n, THEN the same request, it goes through. It's like I have to initialize something... |
| 16:16.58 | WIMPy | without the spaces around the newlines |
| 16:17.20 | DrDamnit | there are no spaces on the newlines (I added the spaces for clarity in here). |
| 16:19.10 | DrDamnit | The error is "Missing action in request", but I have action:login in there. |
| 16:20.45 | DrDamnit | http://pastebin.com/6P630uUL |
| 16:20.49 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.36.250.adsl.dynamic.totbb.net) |
| 16:21.34 | BugKhaM | I saw cmd ReadFile but can anyone tell me if there's a cmd WriteFile available? |
| 16:21.53 | Qwell | DrDamnit: \r\n or \n\r? You're using them interchangably.. |
| 16:22.13 | DrDamnit | crlf |
| 16:24.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 16:24.43 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 16:24.47 | DrDamnit | any thoughts on that pastebin I just put up? This has to be a simple solution that I am over looking.... |
| 16:26.23 | p3nguin | Dr. Dammit: If you're trying to automate something, consider using expect. |
| 16:27.06 | Qwell | DrDamnit: According to your message above, you used \n\r for action. |
| 16:27.16 | DrDamnit | I know. That's a type-o. |
| 16:27.35 | DrDamnit | It should be \r\b (CRLF). |
| 16:27.47 | DrDamnit | aaaahhhh! \r\n |
| 16:28.06 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:29.19 | DrDamnit | p3nguin: I am writing a .NET client to do some simple management tasks, however, when I kept getting this error, I went to manually type it into the telnet interface, and was able to reproduce the issue. So, I must not be doing something write when I do it manuall (and therefore, the .NET client is also going to do it wrong). Of course, this is a simple interface, it shouldn't be this complicated. |
| 16:29.33 | DrDamnit | I am probably making it complicated, so need a second set of eyes to tell me what I did wrong.... |
| 16:30.07 | p3nguin | Dr. Dammit: Maybe expect isn't the right tool for you, but it is still a very good tool! |
| 16:30.37 | moy | voiper: your best bet is try itu first |
| 16:31.16 | moy | and then take it from there, you can use the r2proto.conf to modify advanced settings to match a new variant ... |
| 16:31.28 | p3nguin | beek: Still no luck on the Vyatta port redirection... any other advice? |
| 16:33.31 | DrDamnit | p3nguin: Link? |
| 16:33.46 | p3nguin | Dr. Dammit: Link of what? |
| 16:34.23 | beek | p3nguin: Did you open the appropriate ports throught the firewall? |
| 16:35.07 | DrDamnit | beek: there is no firewall. I am connecting directly. Did you see the pastebin? http://pastebin.com/6P630uUL |
| 16:35.18 | eduzimrs | many erros on this chan_mobile.c installing * addons .... anyone knows? |
| 16:35.24 | p3nguin | beek: Yes, I did. I have a feeling, based on the lines you pasted for me the other night, that you didn't fully understand what I was trying to accomplish. The lines of yours look to me like they are regular old port forwarding through nat, and that's not what I'm trying to do. |
| 16:35.36 | DrDamnit | beek: sorry. Missed that was directed at p3nguin. |
| 16:35.47 | beek | Then I misunderstood what you were asking. |
| 16:36.14 | beek | p3nguin: If you'd like, join the ##vyatta channel and we can carry on there. There are a number of Vyatta employees hanging out there as well. |
| 16:36.28 | p3nguin | beek: I'll be there! |
| 16:39.58 | *** join/#asterisk Rufus (Rufus@unaffiliated/rufus) |
| 16:40.03 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:41.32 | Katty | someone needs to adopt me. |
| 16:41.40 | DrDamnit | Qwell: Turns out it was the telnet client (Putty). Using the regular telnet client, it works. So, the problem is in how my code is creating the packets. Thanks though! |
| 16:42.51 | Defraz | So I want to forward my phone to a 1800 number and then have it choose the extension after waiting for teh greeting and the ivr that answers. It always answers and plays the menu after 5 seconds then expects and extension to be entered. |
| 16:43.42 | Defraz | the database put CF 1010 18005551212wwwww2020 doesn't seem to work |
| 16:43.46 | Defraz | I thougth w was a pause. |
| 16:46.18 | Defraz | Is there a better way to do a call forward like that? |
| 16:46.30 | Defraz | Or a way that works? |
| 16:47.36 | VoipForces | Defraz: problem is probably that your wwww2020 is send as part of the dial string and not dialed once the call is bridged. |
| 16:47.54 | Defraz | oh true |
| 16:48.08 | VoipForces | Defraz: Better way would be to have a unique DID on the other side of heve them put an inbound callerid rule. |
| 16:48.17 | Defraz | yea they won't do that. |
| 16:48.34 | Defraz | it is some avaya system that they have a main number. |
| 16:48.41 | Defraz | they won't change crap for me. |
| 16:48.50 | VoipForces | Try the D([called][:calling]) options of Dial |
| 16:49.23 | Defraz | Thanks I will take a look |
| 16:51.36 | Defraz | can I put www in the calling side |
| 16:53.09 | Defraz | I guess I will try it. |
| 17:00.55 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:07.20 | VoipForces | Have a customer with like 40 agent desks using Twinkle softphone with USB headsets and about 20 Aastra deskphones. No prob with the deskphone, but the softphone, the agent side, the voice during a call sometimes fades out and back in. Any idea ? |
| 17:08.19 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
| 17:15.53 | Merlin | haven't heard of that issue before voipforces |
| 17:16.25 | VoipForces | Merlin: thanks, puzzles me, having the customer check their workstations. |
| 17:16.39 | Merlin | what is the ILEC term for when you have an inbound DID hit one PRI, and if that PRI is full or down, it moves to another PRI? |
| 17:16.51 | Merlin | i am having a mental block right now |
| 17:18.59 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:19.15 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 17:22.26 | Katty | is it 5 yet :< |
| 17:22.54 | acidfoo | somewhere it is |
| 17:23.21 | *** join/#asterisk bchia (~Adium@nat/digium/x-xqcrwlihumozzboz) |
| 17:25.39 | _Corey_ | Merlin: DTO aka Direct Trunk Overflow |
| 17:27.08 | Merlin | is it just for overflow? or can it be used for failover too? |
| 17:27.13 | _Corey_ | yep |
| 17:27.17 | Merlin | awesome thank you |
| 17:27.49 | _Corey_ | You can typically pull the original number called from the secondary PRI by looking at the RDNIS field |
| 17:27.58 | _Corey_ | (in case that matters to you) |
| 17:28.00 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 17:29.10 | Hive | Does anyone use fax for asterisk? I'm having a problem sending a fax to what I'm guessing is another virtual fax system with the error; "Unsupported SDP media type in offer: image 50106 udptl t38" |
| 17:33.30 | eduzimrs | many erros on this chan_mobile.c installing * addons .... anyone knows? |
| 17:34.40 | Qwell | eduzimrs: What version of Asterisk-Addons? What version of Asterisk? |
| 17:38.56 | eduzimrs | addons 1.6.2.4 asterisk 1.8.6.0 |
| 17:39.50 | eduzimrs | @Qwell: addons 1.6.2.4 asterisk 1.8.6.0 |
| 17:41.05 | Qwell | You don't need addons with 1.8.. |
| 17:41.57 | eduzimrs | ok, can u help me with a problem with odbc connect? |
| 17:41.57 | *** join/#asterisk lcat (~lcat@187.45.254.107) |
| 17:42.08 | eduzimrs | @Qwell: ok, can u help me with a problem with odbc connect? |
| 17:42.11 | Katty | tosses an icecube down Qwell's shirt. |
| 17:42.32 | Qwell | watches the steam roll out of his shirt. |
| 17:42.51 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 17:43.10 | eduzimrs | @Qwell: res_odbc.c:1507 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Driver's SQLAllocHandle on SQL_HANDLE_HENV failed |
| 17:44.01 | Katty | someone help me think |
| 17:44.10 | Katty | trent reznor put out a new album, i believe with his wife |
| 17:44.17 | Qwell | How to Destroy Angels |
| 17:44.22 | Katty | ty |
| 17:45.31 | Katty | Qwell: http://music.atbigelow.org/group/How%20to%20Destroy%20Angels/How%20to%20Destroy%20Angels/ <- does that look like the full album? |
| 17:45.59 | Qwell | umm |
| 17:46.06 | Qwell | http://store.destroyangels.com/ |
| 17:46.10 | Qwell | free |
| 17:46.45 | Katty | see, you guys know everything |
| 17:46.49 | Qwell | pretty sure everything he does is free now |
| 17:51.47 | Katty | :> |
| 17:55.54 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 18:10.21 | eduzimrs | [Sep 16 15:10:05] WARNING[1643]: res_odbc.c:1507 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Driver's SQLAllocHandle on SQL_HANDLE_HENV failed |
| 18:10.28 | eduzimrs | anyone knows? |
| 18:16.36 | luckyaba | anyone know of any SIP providers that do not chage a per trunk fee? |
| 18:19.56 | *** join/#asterisk m_tadeu (~quassel@89.181.27.241) |
| 18:29.57 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 18:30.24 | eduzimrs | WARNING[1966]: res_config_mysql.c:442 realtime_multi_mysql: MySQL RealTime: Invalid database specified: 'snep' (check res_mysql.conf) ... so there is no such file "res_mysql.conf" in * 1.8 |
| 18:30.50 | eduzimrs | instead is res_config_mysql |
| 18:31.41 | anonymouz666 | eduzimrs: cara, pelo amor de deus. leia a documentação. você está a horas pedindo algo que é muito fácil de resolver. |
| 18:49.10 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:51.33 | m_tadeu | is there a way to share a contact list for a softphone? |
| 18:51.49 | Qwell | m_tadeu: that's something the phone would have to support. ie; ask them |
| 18:52.19 | m_tadeu | Qwell: thanx |
| 18:52.25 | WIMPy | Don't you use a browser for contacts? |
| 18:59.37 | *** join/#asterisk furia (~herman.jo@d51528B76.access.telenet.be) |
| 19:03.08 | m_tadeu | I'm distributing a free softphone so that clients can call me at no charge. But I'd like to make it easier for them and give them a contact list for the several services that I supply |
| 19:03.15 | furia | hello pabelanger - here furia (herman) - I have private channel open,but doesn't seem to work |
| 19:03.29 | m_tadeu | WIMPy: ^^^ |
| 19:03.30 | pabelanger | furia: hi |
| 19:03.36 | pabelanger | furia: ya, here is fine |
| 19:03.43 | pabelanger | so, where are you at now? |
| 19:04.19 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:04.34 | furia | ok - so my ast 1.8.6.0 is working fine - but only after rebuilding it |
| 19:05.23 | pabelanger | Right, I suspect you are not running any of the patches we do for building the package |
| 19:06.35 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:06.35 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:06.36 | pabelanger | furia: pb the output of: DEB_BUILD_OPTIONS="debug" apt-get -b source asterisk |
| 19:08.38 | furia | no idea - I just executed the instructions for the backtrace, but without rebuilding my ast config expected that modules are in /usr/lib/asterisk /modules and they weren't until Ididthebuild |
| 19:11.31 | furia | DEB_BUILD_OPTIONS="debug" apt-get -b source asterisk is what I did indeed - output was sent before with again some public key warnings |
| 19:12.39 | pabelanger | furia: pastebin the output if you don;t mind |
| 19:14.48 | furia | difficult now - need to proceed testing with my 1.8.6.0 Asterisk - don't want to take the risk now to screw it again, but Iwas thinli |
| 19:16.59 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:16.59 | furia | sorry - was away too soon - but I was thinking to retry the upgrade again when there is a newer package, and then come back if still not working ? |
| 19:17.17 | pabelanger | sure |
| 19:18.18 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 19:18.47 | furia | just one problem I still have now : these xmldoc warnings on startup - is there no way to disable them without rebuilding ? e.g. in modules.conf somehow ? |
| 19:21.05 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust646.15-3.cable.virginmedia.com) |
| 19:26.45 | timeshell | How does one kill rogue meetme sessions? |
| 19:27.20 | timeshell | Without restarting asterisk? |
| 19:27.30 | leifmadsen | channel request hangup ? |
| 19:27.43 | furia | paul - still there ? see my question above on xmldoc warnings |
| 19:30.27 | timeshell | Thank you leif |
| 19:30.37 | timeshell | Never had to do that before. |
| 19:30.38 | timeshell | :p |
| 19:31.33 | hardwire | Nugget: quit asking about hardwiring. |
| 19:31.39 | hardwire | srsly |
| 19:34.20 | pabelanger | furia: not sure, something you likely did when you compiled the source. Or remove the xmldoc file from your HDD |
| 19:42.27 | furia | pabalanger: don't find any on HD, so probably unmarked during make menuselect - and now resulting inthese warnings ? |
| 19:48.23 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
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| 20:44.25 | *** part/#asterisk VoipForces (~Adium@69.165.199.195) |
| 20:44.53 | Nugget | eyes hardwire |
| 20:45.16 | Qwell | hardwires telnet |
| 20:45.20 | Qwell | aww |
| 20:45.22 | Nugget | hah |
| 20:51.29 | *** join/#asterisk Gonz0 (~Gonz0@41.102.207.230) |
| 20:52.21 | Gonz0 | hi im looking for someone who speak french for some informations plz |
| 20:52.55 | dr0ck | je |
| 20:53.23 | Gonz0 | do you speak french dr0ck ? |
| 20:53.36 | dr0ck | meant to say "je suis muzzy" |
| 20:53.45 | Gonz0 | :| |
| 20:53.51 | dr0ck | and no, im just here for the joke :) |
| 20:53.56 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 20:54.09 | Gonz0 | hf |
| 20:55.02 | pabelanger | translate.google.com ? |
| 20:55.23 | Gonz0 | ill try |
| 20:56.41 | hardwire | eats Qwell |
| 20:57.00 | hardwire | Qwell: how did you know I was using telnetlib today? :) |
| 20:58.12 | Gonz0 | i have 10 analogic phones (i mean no ip ) and 1 external line how to connect my phones with the asterisk server :o |
| 20:59.09 | Gonz0 | i mean my phones have rj11 and lan is with rj45 so i can use ATA adapter and a switch or something like that ? |
| 21:02.15 | hardwire | analogic. |
| 21:02.16 | malcolmd | Gonz0: yup, that'd work |
| 21:02.17 | hardwire | is a neat word. |
| 21:05.47 | Gonz0 | i have another mistake, asterisk can assign a local number to all phones (i have 10 phones) ans how he know phone1 and other phones there are no IPs in phones |
| 21:06.57 | *** join/#asterisk evo (570e5a9c@debian/developer/evo) |
| 21:10.49 | Gonz0 | so in summary i need a server with asterisk, a 10 ports switch and 10 rj11 to rj45 ?! |
| 21:10.57 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 21:11.48 | evo | hello guys, my boss recently bought HiPath TAPI software by Siemens to manage our telephony system (for example workstations can start calls with Outlook); now I'd like to interact with the system with free software, I was wondering if Asterisk could be used to dialog with phones using TAPI .. thanks |
| 21:15.53 | *** join/#asterisk scatterp (~scatterp@02dfe731.bb.sky.com) |
| 21:16.24 | scatterp | hi can any one hel me i got one way audio im using sipgate ive tried fwding ports and using a stun server but nothing seems to fix it :/ |
| 21:17.11 | scatterp | my router suports milkfish what ever that is |
| 21:21.40 | pabelanger | Gonz0: Yes, you can either use a PCI card (EG: Digium TDM400P) for your Asterisk server or an external ATA (SIP->analog ports converter) |
| 21:25.00 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:25.53 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 21:26.50 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 21:39.00 | malcolmd | pabelanger: TDM400 is EOL...TDM410 is the current product in that segment |
| 21:43.21 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 21:48.44 | luke-jr | is there some kind of Asterisk exploit being used in the wild to root systems? |
| 21:48.56 | luke-jr | possibly on older Asterisk systems? |
| 21:54.31 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:00.22 | *** join/#asterisk JustinCampbell (~justinCam@c-71-224-204-129.hsd1.pa.comcast.net) |
| 22:06.59 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 22:14.50 | *** join/#asterisk Uberi (~Anthony@209.141.132.3) |
| 22:15.21 | *** part/#asterisk Uberi (~Anthony@209.141.132.3) |
| 22:15.42 | *** join/#asterisk Tuncay (~Tuncay@g231220246.adsl.alicedsl.de) |
| 22:15.51 | Tuncay | hi Uberi |
| 22:16.23 | *** part/#asterisk Tuncay (~Tuncay@g231220246.adsl.alicedsl.de) |
| 22:23.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-dztmnzqylgixfzyq) |
| 22:27.56 | *** join/#asterisk nighty^ (~nighty@69-165-220-105.dsl.teksavvy.com) |
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| 22:50.36 | pabelanger | luke-jr: Anything is possible, but without information about what is happening it is hard to say |
| 23:01.36 | scatterp | hi can any one help me i got one way audio im using sipgate/bria ive tried fwding ports and using a stun server but nothing seems to fix it :/ oh and my router suports milkfish what ever that is |
| 23:06.14 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 23:17.35 | luke-jr | pabelanger: two old Asterisk systems within a few weeks both wiped clean |
| 23:20.10 | luke-jr | *seemingly* unrelated, unless the attacker has inside info of business relationships |
| 23:23.22 | pabelanger | luke-jr: well, it is possible. Unless you have been actively patching, you could be open for attack. Here are all the known security releases: http://downloads.asterisk.org/pub/security/ |
| 23:24.29 | *** part/#asterisk etc (~root@124-168-89-27.dyn.iinet.net.au) |
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| 23:59.03 | rdegges | Sup guys. |
| 23:59.05 | rdegges | How goes it. |