Re: [Asterisk-Users] CNAM lookup: new method for Caller ID Name delivery
John Todd wrote:
[cross-posted to -biz and -users since it could fall into either
category]
Interesting new product that has been introduced that I think some
would be interested in here (at least, those users in the United
States and perhaps Canada): CNAM delivery via IP lookup.
The problem: inbound calls on many PRI connections, and also over many
VoIP providers, do not include caller name. This means that all you
see is the caller ID number, but no name. Most PSTN lines these days
(if they are enabled with Caller ID) will also include a caller ID
name. So, you'd think that a well-configured Asterisk server should
somehow be able to deliver the same data, right?
A company called Accudata has come up with an IP-based CNAM lookup
tool. It's an HTTPS delivery method, with what I assume is XML as the
specification language. The nice part is that it really doesn't
matter what the backend looks like - Accudata has built app_getcnam
that automatically takes the 10 digit NANP number and spits back a 15
character caller name from within the Asterisk dialplan. You get the
caller ID from an inbound call (IP or PRI or any channel type as long
as it has an e.164 number associated with it) and then hand off the
${CALLERIDNUM} to this application, and get back a string with the
name. I don't have exact details on the system (see "disadvantages"
below) but it seems to be an interesting product.
Pricing:
At the "low volume" end of the scale (probably under 2000 queries
per month, but I didn't ask), the price is $0.0156 per lookup, which
is reasonable enough. I'm sure better price breaks come with volume.
Upsides:
1) They have direct Asterisk integration, using app_getcname.c as a
data method.
2) They at least are willing to talk to smaller customers who aren't
pushing millions of calls a month.
3) It's all IP - no unwarranted complexity of SS7 or other signalling.
Downsides:
1) They want you to sign an NDA before they'll discuss the methods
with you. I was not willing to sign an NDA to have an XML schema
example transmitted to me, so that was a non-starter. This really
angers me, actually - does anyone actually have a clue how many
lawyers need to get involved in an NDA, and what is it exactly that
the NDA is trying to do? NDAs are used in the USA for the most
frivolous and inane reasons. As if your competition didn't know what
you were doing? Please, let's be realistic here.
2) They have a $100 monthly minimum charge. If you only have a
billing volume of under $100, then you'll pay $100. So, if you have
under 6400 queries per month, you're paying for the honor of being
billed. This isn't that big a deal if you're an ITSP, but makes this
almost impossible for a smaller user to afford. (good opportunity for
a small reseller, especially if you are smart with caching.) I can't
say I disagree with them on this model to start, but I spent some time
doing the math for "small-time" usage, and at a $2 minimum and 50
included queries a month (and $.02 afterwards) this would make a very
nice market for a few thousand iPBX systems. Payment via Credit Card
or Paypal would be perfect; set it up once, forget about it. However,
that's not the model they chose, since they're not shooting for the
lower end of the market.
3) There may be hidden problems with the application; I haven't run
it, so I can't vouch for it.
Other notes:
The clever integrator of this application will save themselves some
lookup $ by caching the responses from the database into their own
database, along with a datestamp. Perhaps if an entry is >90 days
old, the system will re-lookup the entry in the Accudata database but
otherwise will present the memorized answer. (Hint: the caller ID's
of your inbound call pool is probably >80% redundant)
Contact information:
http://www.accudatatech.com
"Tracy Glick" <[EMAIL PROTECTED]> [sales contact]
"Kevin Nguyen" <[EMAIL PROTECTED]> [tech contact]
If anyone else has heard of an easy-to-use method for obtaining this
data via free or commercial methods, please follow-up to this post for
the archives. I don't speak for Accudata, nor am I a user of their
services, but it seems interesting so I'll pass it along to the group.
JT
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If it isn't agiast there agement, I would happy setup a "resale" server
for this just as you said, and probly at the prces you listed, I will
look into this abit more later today.
Only thing I use my asterisk server for, for the most part is a few
select hpones (very low usages), but my exist customers (who have a
higher volune), or people just wanting todo CNAM look ups could befit
from this.
_
Re: [Asterisk-Users] SIP native bridge problem
So it's a problem with my NAT, that's what I thought.. Ok, I got another question, with SIP nantive brideing happening, do the CDRs asterisk keeps still good enough for billing, or are they only good for the short time asterisk is in the media stream? Kevin P. Fleming wrote: Nathan Goodwin wrote: Does anyone have any ideas how I could fix this, this is sort of important, if it's just me because of my NAT causing it, would doing so part forwarding and disable NAT support on asterisk and the Sipura fix this problem? It's almost impossible to fix this problem. Here's the scenario: Your SPA-2000 initiates a call to the * server, and then * initiates a call to your provider. When the provider answers, * tells the SPA-2000, and it starts sending RTP to *. By doing so, your NAT/firewall expects to receive packets back from the _same IP address and port they were sent to_. While * is still in the media path, this is how it works, and things are fine. However, when * tries to re-invite the SIP provider to send audio directly to your SPA-2000, the packets now arrive from a different IP (and probably a different port number). Any decent NAT/firewall will drop them on the floor. Thus, no audio from the provider. It is _possible_ for this to work if * happens to reinvite your SPA-2000 _first_, and it starts sending audio directly to the SIP provider, thus opening a different IP/port combination through the firewall. However, this is not reliable, and there is no way to force the reinvites to happen in a particular order (or to complete in a particular amount of time). You can disallow reinvite for your SPA-2000 but leave it turned on for your provider, in which case one direction of the media stream can bypass *. Otherwise, you need a NAT/firewall that understands SIP so it can be aware of the changes as they occur (or you need to use a SIP proxy on the NAT/firewall, like siproxd). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP native bridge problem
I'm having a problem, I'm not sure if it has todo with the fact that my phone is behind a NAT or not, but here it is.. My problem is when I call out, my asterisk system routes the call to my SIP provider, whoever, as soon as the other party answers, asterisk tries to make a native bridge for the call, and then the call drops instantly. However, if I keep asterisk in the middle (by anyable transfers), no bridge is made and the call stays just fine. My setup is so: Sipura-2000 -> NAT (Netgear router) -> cable/internet -> colocated asterisk server -> SIP provider The native bride I assume is asterisk trying to connect the RTP stream directly from the Sipura to my SIP provider (thus asterisk keeping it's self out of the media stream), and this is exactly what I would like to have. But I can't for the life of be figure ot why it's just hanging up once the bridge is made. Does anyone have any ideas how I could fix this, this is sort of important, if it's just me because of my NAT causing it, would doing so part forwarding and disable NAT support on asterisk and the Sipura fix this problem? I'll welcome any input, Nathan Goodwin Diamonleaf Communications LLC ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bellster - cool :-)
I like that idea, and if it worked that way I could gladly add a few international routes. Gustavo Russo wrote: A couple of proposals for this excelent idea : - Credits accounting based in minutes and not in number of calls. - The possibility for the owner of the route to add a specific multiplier factor for the route, for instance between 1 and 10. This way cellular or expensive routes can give more credits than a cheap one. G. - Original Message - From: "Steven P. Donegan" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, January 22, 2005 10:38 PM Subject: Re: [Asterisk-Users] Bellster - cool :-) Agreed - the announcement is not needed - although it's kind of neat - perhaps it could be something that was optional/configurable from the bellster web page? Nathan Goodwin wrote: I love belster, I added a route for the 518 area code, (that covers most of upstate NY), only thing I wish I could do is get rid of the message that says how many credits I have left. I would rather it just report congested is the call can't go though (doto lack of credits), that way I could make Bellster my default route, then use another if it doesn't work as a backup. I made a few test calls to different places using Bellster, surprizingly the quility was very good. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bellster - cool :-)
I love belster, I added a route for the 518 area code, (that covers most of upstate NY), only thing I wish I could do is get rid of the message that says how many credits I have left. I would rather it just report congested is the call can't go though (doto lack of credits), that way I could make Bellster my default route, then use another if it doesn't work as a backup. I made a few test calls to different places using Bellster, surprizingly the quility was very good. Steven P. Donegan wrote: OK, I have done all the stuff at my end and at Bellsters end to add 21 new area codes (all of california) to the Bellster dial plan. Pretty cool deal! I hope others go for this quickly - as it could be a really nice co-op. I do suggest to Jeff - do some sort of calling trunk -vs- routed trunk match to make sure that someone can't run their credits sky-high by making calls through themselves. I did all my test calls through my own trunks and voila I have credits available. Jeff - you rock :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
Wouldn't that make routing free calls illegal as well, your still bypassing? Miguel Cavazos wrote: Thanx but that is consider in Mexico bypass and its illegal, second we are just doing a test with real traffic to get feedback of any weird thing going on. Testing Chan_unicall stability is our goal. If you can send alot of traffic while we are doing test i would thank you for that. Till now we have only got 4 channels at most busy and we need to see if it will handle a full E1 to test then with 2,3 and 4 E1's On 13/01/2005, at 12:49 PM, Nathan Goodwin wrote: I tried to contact you off list, but your system rejected my e-mail, I was wondering ig you planed on selling minutes for routes into Mexico once you where done testing, if so, could you please contact me off list with your rates for Mexico City, or anyplace else in Mexico you service, thank you. Nathan Goodwin Diamondleaf LLC Miguel Cavazos wrote: Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten => _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall
I tried to contact you off list, but your system rejected my e-mail, I was wondering ig you planed on selling minutes for routes into Mexico once you where done testing, if so, could you please contact me off list with your rates for Mexico City, or anyplace else in Mexico you service, thank you. Nathan Goodwin Diamondleaf LLC Miguel Cavazos wrote: Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can be with REAL traffic. Add this to your extensions.conf only gsm as a codec is going to be permitted. exten => _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt) -- Saludos, Miguel Cavazos ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual Modems
After searching the archives, I came acrross a few people mentioning this, but I never saw anything about what became of it. Has anyone tried to make a virtual modem that could be directly handled by astrisk, I saw a while ago that someone was going to try and make one using the same DSP libaries that the WinModems use, but then nothing. Would do this even be possible, and if so, what kind of connection speeds could one hope to achive, with the compression and such? I don't need a 56k connect (though it would be nice), but be able todo 28.8 would be fine. :) Also, would using the new T.38 (I think it is), that was designed for faxes help modem calls any? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
