Date: 2009-10-13 (tomorrow) Time: 19:00 UTC (3 PM EDT/12 PM PDT) Place: #fedora-fad on freenode
Hi Everyone,
We're meeting again tomorrow on #fedora-fad to keep talking about how we are doing in terms of getting ready for the Fedora Activity Day on 2009-10-23 & 24. Thanks to everyone who has been sending things to the list and helping to get the new version of Asterisk in place.
Before tomorrow's meeting I will also stage the text from all of the tickets on gobby so we can create a clearer plan of what we are going to work on. Be sure to get gobby set up and login before the meeting. Look for a document with a self descriptive title.
http://fedoraproject.org/wiki/Communicate/GobbyHowTo
Thanks, John
On Mon, Oct 12, 2009 at 16:12:03 -0700, John Poelstra poelstra@redhat.com wrote:
Date: 2009-10-13 (tomorrow) Time: 19:00 UTC (3 PM EDT/12 PM PDT) Place: #fedora-fad on freenode
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
Bruno Wolff III said the following on 10/12/2009 04:55 PM Pacific Time:
On Mon, Oct 12, 2009 at 16:12:03 -0700, John Poelstra poelstra@redhat.com wrote:
Date: 2009-10-13 (tomorrow) Time: 19:00 UTC (3 PM EDT/12 PM PDT) Place: #fedora-fad on freenode
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
From last week's meeting Jeff was creating a repo of the sanitized config files. Jeff any luck?
http://meetbot.fedoraproject.org/fedora-fad/2009-10-06/fedora-fad.2009-10-06...
On Mon, Oct 12, 2009 at 9:02 PM, John Poelstra poelstra@redhat.com wrote:
Bruno Wolff III said the following on 10/12/2009 04:55 PM Pacific Time:
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
From last week's meeting Jeff was creating a repo of the sanitized config files. Jeff any luck?
git://fedorapeople.org/~jcollie/ftalk-asterisk-configs.git
On Mon, Oct 12, 2009 at 10:58:23PM -0500, Jeffrey Ollie wrote:
On Mon, Oct 12, 2009 at 9:02 PM, John Poelstra poelstra@redhat.com wrote:
Bruno Wolff III said the following on 10/12/2009 04:55 PM Pacific Time:
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
From last week's meeting Jeff was creating a repo of the sanitized config files. Jeff any luck?
git://fedorapeople.org/~jcollie/ftalk-asterisk-configs.git
Also browseable here: http://fedorapeople.org/gitweb?p=jcollie/public_git/ftalk-asterisk-configs.g...
On Tue, Oct 13, 2009 at 00:46:35 -0400, "Paul W. Frields" stickster@gmail.com wrote:
On Mon, Oct 12, 2009 at 10:58:23PM -0500, Jeffrey Ollie wrote:
On Mon, Oct 12, 2009 at 9:02 PM, John Poelstra poelstra@redhat.com wrote:
Bruno Wolff III said the following on 10/12/2009 04:55 PM Pacific Time:
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
From last week's meeting Jeff was creating a repo of the sanitized config files. Jeff any luck?
git://fedorapeople.org/~jcollie/ftalk-asterisk-configs.git
Also browseable here: http://fedorapeople.org/gitweb?p=jcollie/public_git/ftalk-asterisk-configs.g...
Thanks. I'll be trying to get this setup on another machine by the end of the weekend. I got a couple of other things due by Thursday so I might not get to it until the weekend.
On Tuesday, October 13, 2009, Bruno Wolff III bruno@wolff.to wrote:
On Tue, Oct 13, 2009 at 00:46:35 -0400, "Paul W. Frields" stickster@gmail.com wrote:
On Mon, Oct 12, 2009 at 10:58:23PM -0500, Jeffrey Ollie wrote:
On Mon, Oct 12, 2009 at 9:02 PM, John Poelstra poelstra@redhat.com wrote:
Bruno Wolff III said the following on 10/12/2009 04:55 PM Pacific Time:
I'll probably be very late again, but will look through the logs later. One thing I'm still looking for is sample configs for Fedora Talk so that I can try to replicate some of Fedora Talk locally and write up what I find out before the FAD, for other people that want to set up test environments.
From last week's meeting Jeff was creating a repo of the sanitized config files. Jeff any luck?
git://fedorapeople.org/~jcollie/ftalk-asterisk-configs.git
Also browseable here: http://fedorapeople.org/gitweb?p=jcollie/public_git/ftalk-asterisk-configs.g...
Thanks. I'll be trying to get this setup on another machine by the end of the weekend. I got a couple of other things due by Thursday so I might not get to it until the weekend.
There is a good chance I'll not make today's meeting. I am in another meeting at that time (in person) and won't be able to be at a screen.
Paul
John Poelstra said the following on 10/12/2009 04:12 PM Pacific Time:
Date: 2009-10-13 (tomorrow) Time: 19:00 UTC (3 PM EDT/12 PM PDT) Place: #fedora-fad on freenode
Hi Everyone,
We're meeting again tomorrow on #fedora-fad to keep talking about how we are doing in terms of getting ready for the Fedora Activity Day on 2009-10-23 & 24. Thanks to everyone who has been sending things to the list and helping to get the new version of Asterisk in place.
Before tomorrow's meeting I will also stage the text from all of the tickets on gobby so we can create a clearer plan of what we are going to work on. Be sure to get gobby set up and login before the meeting. Look for a document with a self descriptive title.
http://fedoraproject.org/wiki/Communicate/GobbyHowTo
Thanks, John
We had a brief meeting limited by the attendees availability and a poor use of MeetBot by me so here is a manual summary:
1) John will summarize the FedoraTalk Tickets 2) John will create some use cases for user and admin interaction with the system 3) <jcollie> one thing that i though of today is the ability to log in using the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
Log: http://meetbot.fedoraproject.org/fedora-fad/2009-10-13/fedora-fad.2009-10-13...
On Wed, Oct 14, 2009 at 12:38 PM, John Poelstra poelstra@redhat.com wrote:
- <jcollie> one thing that i though of today is the ability to log in using
the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
I think that the use case for this is that I'm away, but I need to call $CONTRIBUTOR and I want them to know that it's me instead of some random cell phone number, so we should be able to login and have it come from our extension.
Jeff, correct me if I'm wrong. And I thought the meeting was at 3PM EST, have I failed yet again? I'm incredibly busy with $DAYJOB as well :(
On Wed, Oct 14, 2009 at 12:58:20 -0400, Jon Stanley jonstanley@gmail.com wrote:
On Wed, Oct 14, 2009 at 12:38 PM, John Poelstra poelstra@redhat.com wrote:
- <jcollie> one thing that i though of today is the ability to log in using
the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
I think that the use case for this is that I'm away, but I need to call $CONTRIBUTOR and I want them to know that it's me instead of some random cell phone number, so we should be able to login and have it come from our extension.
Would caller id with the person's cell phone number solve this problem? If so, that might be easier to handle.
Jeff, correct me if I'm wrong. And I thought the meeting was at 3PM EST, have I failed yet again? I'm incredibly busy with $DAYJOB as well :(
It was pretty short. It was over by 2000 UTC when I got back. If you look at the log, you'll see it went pretty quick.
If you want to add to the wish list, I would like to eventually be able to forward calls to another sip server. Right now I sort of handle this (but something is broken) by having my asterisk box register with the fedora talk server so it can get calls. But I get so few calls (none except for my test calls) that I think it would be significantly less overhead to have a forwarding ability rather than have to essentially poll from my end.
On Wed, Oct 14, 2009 at 1:11 PM, Bruno Wolff III bruno@wolff.to wrote:
Would caller id with the person's cell phone number solve this problem? If so, that might be easier to handle.
That already happens - if I call in to my phone from a DID, my cell phone (and whatever "name" the telco provides - which seems to be more the CLLI of the switch) is displayed as the calling party.
Jon Stanley said the following on 10/14/2009 09:58 AM Pacific Time:
On Wed, Oct 14, 2009 at 12:38 PM, John Poelstra poelstra@redhat.com wrote:
- <jcollie> one thing that i though of today is the ability to log in using
the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
I think that the use case for this is that I'm away, but I need to call $CONTRIBUTOR and I want them to know that it's me instead of some random cell phone number, so we should be able to login and have it come from our extension.
When we get it figured it, please add to these pages: https://fedoraproject.org/wiki/Fedora_Talk_User_Cases https://fedoraproject.org/wiki/Fedora_Talk_Admin_Cases
Thanks, John
On Wed, Oct 14, 2009 at 13:14:47 -0400, Jon Stanley jonstanley@gmail.com wrote:
On Wed, Oct 14, 2009 at 1:11 PM, Bruno Wolff III bruno@wolff.to wrote:
Would caller id with the person's cell phone number solve this problem? If so, that might be easier to handle.
That already happens - if I call in to my phone from a DID, my cell phone (and whatever "name" the telco provides - which seems to be more the CLLI of the switch) is displayed as the calling party.
So is the problem then that you wouldn't recognize the cell phone number, but you would recognize the the developer extension. Or is it more you just want to know it's some developer? Is getting called by a developer you don't have a cell phone number for a common case?
On Wed, Oct 14, 2009 at 12:11 PM, Bruno Wolff III bruno@wolff.to wrote:
On Wed, Oct 14, 2009 at 12:58:20 -0400, Jon Stanley jonstanley@gmail.com wrote:
On Wed, Oct 14, 2009 at 12:38 PM, John Poelstra poelstra@redhat.com wrote:
- <jcollie> one thing that i though of today is the ability to log in using
the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
I think that the use case for this is that I'm away, but I need to call $CONTRIBUTOR and I want them to know that it's me instead of some random cell phone number, so we should be able to login and have it come from our extension.
Would caller id with the person's cell phone number solve this problem? If so, that might be easier to handle.
Yes/no. Yes, in that the system should pass through your cell phone number to the callee. No, in that a significant number of people wouldn't want their personal cell phone number being passed on to the callee. Also, you might not recognize most people's cell phone numbers, plus there may be many cases where caller id information isn't sent or is incorrect etc.
Jeff, correct me if I'm wrong. And I thought the meeting was at 3PM EST, have I failed yet again? I'm incredibly busy with $DAYJOB as well :(
It was pretty short. It was over by 2000 UTC when I got back. If you look at the log, you'll see it went pretty quick.
If you want to add to the wish list, I would like to eventually be able to forward calls to another sip server. Right now I sort of handle this (but something is broken) by having my asterisk box register with the fedora talk server so it can get calls. But I get so few calls (none except for my test calls) that I think it would be significantly less overhead to have a forwarding ability rather than have to essentially poll from my end.
We're handling this as one-off items that need someone from F-I to set up - for example both Dennis Gilmore and I have IAX trunks set up.
On Wed, 14 Oct 2009, Jeffrey Ollie wrote:
On Wed, Oct 14, 2009 at 12:11 PM, Bruno Wolff III bruno@wolff.to wrote:
On Wed, Oct 14, 2009 at 12:58:20 -0400, Jon Stanley jonstanley@gmail.com wrote:
On Wed, Oct 14, 2009 at 12:38 PM, John Poelstra poelstra@redhat.com wrote:
- <jcollie> one thing that i though of today is the ability to log in using
the fas voip credentials from a cell phone when using the did numbers --after meeting question from John, "What exactly does this mean and how would it be used? What would it improve that we can't do now?"
I think that the use case for this is that I'm away, but I need to call $CONTRIBUTOR and I want them to know that it's me instead of some random cell phone number, so we should be able to login and have it come from our extension.
Would caller id with the person's cell phone number solve this problem? If so, that might be easier to handle.
Yes/no. Yes, in that the system should pass through your cell phone number to the callee. No, in that a significant number of people wouldn't want their personal cell phone number being passed on to the callee. Also, you might not recognize most people's cell phone numbers, plus there may be many cases where caller id information isn't sent or is incorrect etc.
Would it be possible for me to tell asterisk, I come from this number and this number, please give me a proper "Mike McGrath" id from those numbers?
Although that probably doesn't work with businesses since everyone in the westford office might look like the same person.
-Mike
On Wed, Oct 14, 2009 at 2:38 PM, Bruno Wolff III bruno@wolff.to wrote:
On Wed, Oct 14, 2009 at 12:48:59 -0500, Mike McGrath mmcgrath@redhat.com wrote:
Would it be possible for me to tell asterisk, I come from this number and this number, please give me a proper "Mike McGrath" id from those numbers?
Although that probably doesn't work with businesses since everyone in the westford office might look like the same person.
You would also want to make sure people didn't treat this as absolute truth as with some kinds of phone service you can supply any number you want as the caller id number. So it is possible to spoof these numbers.
Yes, very easily spoofable.
On Wed, Oct 14, 2009 at 12:48:59 -0500, Mike McGrath mmcgrath@redhat.com wrote:
Would it be possible for me to tell asterisk, I come from this number and this number, please give me a proper "Mike McGrath" id from those numbers?
Although that probably doesn't work with businesses since everyone in the westford office might look like the same person.
You would also want to make sure people didn't treat this as absolute truth as with some kinds of phone service you can supply any number you want as the caller id number. So it is possible to spoof these numbers.
Most likely I'll miss or be very late to the next meeting. I have made some good prgress with the asterisk setup instructions. The server is up, running and accepting sip packets. I haven't figured out how to get registrations accepted. (This is probably related to the domain name not matching the server name, but I'm still playing with it.) It would help if people tried out the instructions before the Tuesday meeting so that they could ask Jeff questions if some stuff there is unclear. I'll be spending some more time on this today as I would like to get to the point where twinkle to twinkle calls (through the server) work. But I am not sure I'll have that working before Tuesday. It should be pretty close configwise, but I need to figure out what needs to change and that may take a while.
On Sun, Oct 18, 2009 at 9:25 AM, Bruno Wolff III bruno@wolff.to wrote:
Most likely I'll miss or be very late to the next meeting. I have made some good prgress with the asterisk setup instructions. The server is up, running and accepting sip packets. I haven't figured out how to get registrations accepted. (This is probably related to the domain name not matching the server name, but I'm still playing with it.) It would help if people tried out the instructions before the Tuesday meeting so that they could ask Jeff questions if some stuff there is unclear. I'll be spending some more time on this today as I would like to get to the point where twinkle to twinkle calls (through the server) work. But I am not sure I'll have that working before Tuesday. It should be pretty close configwise, but I need to figure out what needs to change and that may take a while.
This sounds like a good place to give updates. I'm waiting on ixs (Andreas Thienemann) to build the icecast rpm for EPEL. I might just grab the source today and put it up on publictest15 anyway. The only real questions I have are what mount points we want to have and authentication. I'm going to look into each method of authentication and see what I can come up with.
Cheers,
Clint
Auth via FAS would be swell. :)
On 10/18/09, Clint Savage herlo1@gmail.com wrote:
On Sun, Oct 18, 2009 at 9:25 AM, Bruno Wolff III bruno@wolff.to wrote:
Most likely I'll miss or be very late to the next meeting. I have made some good prgress with the asterisk setup instructions. The server is up, running and accepting sip packets. I haven't figured out how to get registrations accepted. (This is probably related to the domain name not matching the server name, but I'm still playing with it.) It would help if people tried out the instructions before the Tuesday meeting so that they could ask Jeff questions if some stuff there is unclear. I'll be spending some more time on this today as I would like to get to the point where twinkle to twinkle calls (through the server) work. But I am not sure I'll have that working before Tuesday. It should be pretty close configwise, but I need to figure out what needs to change and that may take a while.
This sounds like a good place to give updates. I'm waiting on ixs (Andreas Thienemann) to build the icecast rpm for EPEL. I might just grab the source today and put it up on publictest15 anyway. The only real questions I have are what mount points we want to have and authentication. I'm going to look into each method of authentication and see what I can come up with.
Cheers,
Clint
Fedora-infrastructure-list mailing list Fedora-infrastructure-list@redhat.com https://www.redhat.com/mailman/listinfo/fedora-infrastructure-list
On Sun, Oct 18, 2009 at 10:25:59 -0500, Bruno Wolff III bruno@wolff.to wrote:
I'll be spending some more time on this today as I would like to get to the point where twinkle to twinkle calls (through the server) work. But I am not sure I'll have that working before Tuesday.
I got the twinkle to twinkle connection through the server to work. This is about as far as I can take things and I think the instructions are ready for review by the gurus.
I didn't test any conferencing type functions.
The changes I needed to make seem to be related to changes in Asterisk since 1.4.
On Sun, Oct 18, 2009 at 17:18:33 -0400, Jon Stanley jonstanley@gmail.com wrote:
Auth via FAS would be swell. :)
It looks like currently md5 hashes for your voip password are stored in the asterisk config files. I suspect using FAS passwords for sip is going to be a bad idea as clients save them without proper protection. I don't know if the protocol properly protects them either. (Though it does use challange response, so there is some hope of that.)
On Sun, Oct 18, 2009 at 3:18 PM, Jon Stanley jonstanley@gmail.com wrote:
Auth via FAS would be swell. :)
Agreed. However, this is just for connecting the asterisk client to an audio streaming server. I don't know that fas would be ultimately how we authenticate. Maybe the person who starts the recording would need a fas account and when the recording was done, asterisk-ices could send it on to the user who authenticated. I'm not sure yet, will keep thinking on this line.
Clint
On Sun, Oct 18, 2009 at 4:30 PM, Bruno Wolff III bruno@wolff.to wrote:
On Sun, Oct 18, 2009 at 17:18:33 -0400, Jon Stanley jonstanley@gmail.com wrote:
Auth via FAS would be swell. :)
It looks like currently md5 hashes for your voip password are stored in the asterisk config files. I suspect using FAS passwords for sip is going to be a bad idea as clients save them without proper protection. I don't know if the protocol properly protects them either. (Though it does use challange response, so there is some hope of that.)
Those MD5 hashes aren't your FAS password, they are your VoIP password, which is a separate setting.
On Sun, Oct 18, 2009 at 7:49 PM, Jeffrey Ollie jeff@ocjtech.us wrote:
Those MD5 hashes aren't your FAS password, they are your VoIP password, which is a separate setting.
Right, I looked into the FAS Asterisk plugin, and the password (VoIP password, NOT the FAS password) looks to be stored in the db cleartext - see http://git.fedorahosted.org/git/?p=fas.git;a=blob;f=plugins/fas-plugin-aster... line 40 (the md5 is constructed at runtime)
So if we need the information, it should be *really* easily available.
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