Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not familiar with network configuration really, although I do understand most of the basics. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/30/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040812/64838868/attachment.htm
--- Kurtz <kurtz@lightspeed.ca> wrote:> Why is it that the wiki indirectly recommends SER (or another proxy) out in front of Asterisk. > If Asterisk can use radius, and provide the rest of AAA they why ? Incidentall\y, I'm not > familiar with network configuration really, although I do understand most of the basics. >Asterisk is not a SIP proxy, it is a UAS, and also a SIP Registrar. Many use SER and Asterisk together, SER as SIP proxy and Asterisk as PSTN gateway. The advantage is that this combination is highly scalable. Not sure whether Asterisk supports RADIUS authentication. AFAIK, it is not supported, but i beleive some works is in progress in this direction. Do a search on the archives, and you'll get many links on this. Regards, Girish __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail
Hi All,
Can someone help me clear up some stuff? I am about to implement asterisk
for a office of about 20 people. I plan on running SIP phones for everyone.
(a mix of Cisco Sets and Xlite soft phones)
We will place the Asterisk server at a collocation provider and have in
connected to the PSTN via 2 PRIs. (digium card)
When customers call our 800 number they will be sent to asterisk. When they
enter an extension I want asterisk to check if that SIP users is logged in
and if not transfer the call back out over PSTN (to a cell phone)
Now, here is where things are a little foggy... I want put a local Asterisk
server here in the office so that the SIP users connect to it thereby
reducing the chatter across the WAN. I would like to have the two Asterisk
servers communicate via IAX.
Questions:
1. Does this scenario pass muster? Is my thinking logical or does anyone
have a better suggestion?
2. Is this possible? Can the remote Asterisk server check to see if the SIP
user is logged in to the local Asterisk server before sending the call
across the WAN?
3. Should I be using SER vs. another Asterisk server? The problem I see with
this is that it doesn't support IAX. I believe that is the preferred method?
Am I right?
Thanks for all the help from the OSS community. Great software!!!
~chris
____________________________________________
Christopher Jacob Eye Street Software
Program Manager, 14151 Newbrook Drive
Partner Products Suite 250
301.305.0991 Chantilly, VA 20151
www.eyestreet.com
Hi Everyone, Just a curious question. Has anyone heard of any service provider who is using Asterisk and SER to provide their VOIP services? Thanks Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050226/2b0fe47d/attachment.htm
I'm a little confused between the pros/cons or benefits of one Asterisk or SER. I've been using Asterisk for a little bit and I know it's a very powerful and scalable platform (in terms of capacity and functionality). However, I've read in some posts that some people are using SER + * and I read no SER's page that it can even do what * does. So, if I have an application where I wish to offer telephony services (call origination/termination) with "applications" such as prepaid services, conferencing, IVRs, "ACD", dialers, etc, would a single installation of * be sufficient? By single I mean * alone (it could be a cluster) and not with SER. I guess to some degree it may depend on the number of clients, but imagine trying to offer a service similar to Vonage. I buy a "simple" SIP phone, bring it home, sign up for service at www.beyourownprovider.com, register my phone on the site and voila - ready to make and/or receive calls. I can call other members, similar to FWD or I can terminate to PSTN. Now, multiply this scenario to hundreds or thousands of SIP phones spread all over the place, in-front and behind NATs. What would be your approach? Would you still use SER for anything? Is this the right list to post this question? Thanks, Daniel
SER is a SIP proxy, Asterisk is a PBX, and application server. SER passes calls from place to place and does not get in the audio path. SER uses SIP, * is able to transcode, and convert Protocols. You can build an IVR, VM, and PBX with Asterisk. SER is like a traffic cop, where * is the car wash, garage, gas station, etc.... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Daniel Salama Sent: Thursday, April 21, 2005 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and SER I'm a little confused between the pros/cons or benefits of one Asterisk or SER. I've been using Asterisk for a little bit and I know it's a very powerful and scalable platform (in terms of capacity and functionality). However, I've read in some posts that some people are using SER + * and I read no SER's page that it can even do what * does. So, if I have an application where I wish to offer telephony services (call origination/termination) with "applications" such as prepaid services, conferencing, IVRs, "ACD", dialers, etc, would a single installation of * be sufficient? By single I mean * alone (it could be a cluster) and not with SER. I guess to some degree it may depend on the number of clients, but imagine trying to offer a service similar to Vonage. I buy a "simple" SIP phone, bring it home, sign up for service at www.beyourownprovider.com, register my phone on the site and voila - ready to make and/or receive calls. I can call other members, similar to FWD or I can terminate to PSTN. Now, multiply this scenario to hundreds or thousands of SIP phones spread all over the place, in-front and behind NATs. What would be your approach? Would you still use SER for anything? Is this the right list to post this question? Thanks, Daniel _______________________________________________ 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
Hello, when SER redirects calls to asterisk can it be redirected to a realtime peer in asterisk.i cld redirect the call to a static peer but if someone can guide me through for realtime peer settings. Thank you, AA
Hello all, I have setup my Asterisk and SER boxes. implementing the ser.cfg and extensions.conf logic i am able to make calls from asterisk to ser and vice versa. is it possible to make simultaneous calls to a ser client from different asterisk clients without getting a 486 busy from SER. Thanks, AA
Hi all I have SER installed and running But ser send all voice message to email But i would like to integrate with Asterisk IVR in the like , did some one integrated this kind of setup if so kindly guide me how can i do that ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060514/37468182/attachment.htm
I have been reading about integrating Asterisk with SER to help Asterisk deal with large volume of registrations (mainly). I was planning on fronting Asterisk with SER for that purpose. Not that I have the traffic at this moment, but because I wanted to get the infrastructure in place. However, my providers are using G711 codec and I offer G711 and G729 to my clients because they don't have the best broadband service available. So, if my clients are talking G729, I suppose I will have to always keep Asterisk in the media path so as to do codec translation. Is that correct? I was also planning on using SER's nathelper, but if Asterisk _HAS_ to be in the media path, there may not be a need for SER's nathelper. Is this assumption correct? If my purpose of using SER is basically to alleviate registration load and help route (possibly load balance) traffic among multiple Asterisk servers as well as SIP providers, do I really need SER? Would you recommend it? Granted, I have been running both Asterisk and SER as separate systems for a while and they both seem very stable to me. Thanks, Daniel