[asterisk-users] Users Conference - [EMAIL PROTECTED]:30 PM EDT: Founders of Voicepulse
For this week's conference, the two founders of Voicepulse, Ravi Sakaria and Ketan Patel, will be joining us. For those of you who are not aware, Voicepulse is an asterisk friendly VOIP provider that has won awards for service and innovation. We will also have Trixbox news, updates, as well as discount codes. Lastly, we are working feverishly to bring you more information regarding legal issues surrounding VOIP in the coming weeks. So please join us for this week's conference: http://www.AsteriskUsersConference.org You can find out more about Voicepulse at: http://www.voicepulse.com Voicepulse's asterisk section located at: http://connect.voicepulse.com Info about the founders of Voicepulse: http://www.voicepulse.com/corporate/Management.aspx ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. I'd say that duni.conf is a reference, and you expect it to be an introductory document. A reference should be comprehensive. It is best used after you've grasped the basic concepts, and together with a text search. Asterisk's sample configuration files actually serve a role of a reference. The config files can be both a reference and an introduction. Look at sip.conf. Most of the examples in that file are relatively simple, what you would expect for a beginner to set up most of the time. There are also some more complex examples in that file. Lastly, the sip.conf file has a good section that explains pretty much any option that could be used in sip.conf. We should strive to make all of the conf files similar to sip.conf and iax.conf. I don't disagree with you that a separate intro document is needed but there is no reason that the conf files could not serve a broader purpose. Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? I don't see this as the case. Most people who use * are comfortable with the level of complexity that is present in DUNDi, they just don't know where to start. 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? Absolutely. If you need any help in putting this together or if you simply need people to review a tutorial, I would be glad to assist. I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Thank you for being a part of the conference last Friday. Your participation is greatly appreciated. Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 330 Speakerphone
Anyone who has experience with the Polycom 330 know if the speakerphone is loud enough to be heard in a 20 foot x 20 foot room? The context is a classroom where announcements will need to be made. The phone will be wall mounted at the front of classroom. Thanks, Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Users Conference Friday at 12:30 PM, EDT
You can listen or join the Asterisk Users Conference Fridays at 12:30 PM
EDT
Today's subject suggestions:
FAX capabilities, what's your solution?
Multiple asterisk server implimentation: ENUM, DUNDI or even two servers
connected
Your subjects?
Share your ideas, ask your questions!
See http://x2z.eu for instructions on how to join or listen
irc://irc.freenode.net/asterisk-users-conference
Note that the SIP channel will only be open from about 12:20PM EDT.
Testing before then will give you the message your PIN is not valid but if
it answers, you're good.
; SIP call
; exten = AUC,1,Dial(SIP/[EMAIL PROTECTED],60,D(22622#${YOUR_PIN}#))
If you would like to talk about services or products your company provides
and answer users' questions, contact me off list. Anyone is welcome to be a
guest and answer users' questions.
Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox,
Adhearsion
Listen to the archived recordings here:
http://x2z.eu/astusers.htm
I would certainly recommend that people participate in this
conference. It is more than a call or podcast, it is a forum
where we can come together to discuss issues, ask questions, and be
part of the community. If you want to discuss any topic related to
Asterisk or if you simply want to find out what others are doing,
this is a great resource.
Matthew
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[asterisk-users] IAX Encryption
I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it looks like the encryption is being setup normally when trunking is enabled. I just can't tell if the voice frame within the trunked packet is encrypted. Any assistance would be appreciated. Thanks, Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modification of Caller ID based on context
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250).
This is set in sip.conf for each user. Each user has a local DID
as well (hosted through Vitelity, for example (555)111-). The
problem is that this extension was being passed to the outside
world. I currently have a SetCallerID command changing the
CallerID to our main office number, but some users want their DID
sent, not the general number.
The problem is that if their caller ID is set to their DID, when
users hit redial on their phones internally they dial out and
back in. I corrected this by putting each DID in extensions.conf
under their three digit extension, but that seems a bit like a
kludge obviously.
I'm looking for a method of sending the internal three digit
extension only when a user is dialing another user internally,
otherwise it will send their DID. Is their a method to do this in
the dial plan? Anyone have an example of how to accomplish this?
Thanks in advance.
Mike,
I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 = 555111 ... extdid/250 =
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:
exten = NXXNXX,n,Set(outgoingCID=MAINNUMBER)
exten = NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)
exten = NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})
exten = NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})
...
You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 = 200 ...
didext/5551112272 = 250.
exten = NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)
OR you could do something similar with LOCAL channels or with a Dial
command.
-Matthew
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