RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
--- Uriel Carrasquilla [EMAIL PROTECTED] wrote: John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility I think the cost is about the same as for putting a web server at a hosting facility. But I don't think you need high bandwidth. SER simply sets up the call. I don't think the audio data actually goes through SER. It goes directly between the two end points. This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris Albertson wrote: This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. http://www.voip-info.org/wiki-Asterisk+sip+reinvite As I understand this Asterisk sets up the call with itself as endpoints, then moves the stream tobypass the PBX and go directly with a SIP reinvite. Some clients does not support this, and with those you have to configure asterisk to stay in the media path for this client with canreinvite=no in SIP.conf. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the shortcuts that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT John: Thank you for responding. I am in the process of installing SER and hope to have it ready by this weekend. I am in the process of installing some equipment at a local colo. I have to tell you, at the expense of offending you, that I use MS-Outlook and the responses go to the tope of the messages. At work I use Lotus Notes and the same thing happens. Before, I used PROFS (on mainframes) and the same principle applied. All in all, 20+ years of using this principle for e-mails at both work and home. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. However, this is not about me but the * group and the well being of this list. Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. Regards, Uriel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Andre:
This makes a lot of sense. I had used Asterisk in the past to play the role
of Gatekeeper for directing traffic to the appropriate Asterisk acting as a
PSTN gateway. IAX does a heck of a good job in that configuration.
However, with SIP, I have run into nothing but trouble with registrations
falling off.
I have read the SER manual I am going to jump into it, now that I know that
in practice it works and it is not only theory in a manual.
Thank you,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andres
Sent: Tuesday, October 14, 2003 12:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a
front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER
Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so
that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
log(1, This is a Long Distance Call\n);
route(6);
break;
};
};
___
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Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using . As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html -- Al Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
OK OK OK, I got it. See my response inside the body of your E-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw Sent: Tuesday, October 14, 2003 8:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? [URIEL] - I have to learn how to quote with Outlook. Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using . As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. [URIEL] you are absolutely right and I do apologize. Ignorance is not an excuse. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html [URIEL] Thank you. -- Al Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 13-10 17:11, John Todd wrote: [...] SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. SER can can become very helpful when it is run in the public internet and clients are behind NATs. For this case SER contains many NAT helping functions that can rewrite header fields, test if a client comes from behind a NAT, ping clients behind NATs (to keep the NAT binding open) and force RTP proxy usage when necesary. Along with RTP proxy SER can help any *symmetric* SIP user agent to get through NAT. (A symmetric SIP user agent is a user agent that uses the same source port for receiving signalling and media as for sending them. Vast majority of SIP user agents as of today is symmetric, including Windows Messenger, Cisco phones, Grandstream phone a.s.o.). There is also support for proxy behind NAT, but it is mostly untested yet. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris: I am glad to see someone else asking the same question I have been asking myself. As soon as I get my public IP address, I will install SER on the public side and Asterisk behind a NAT (with dynamic IP) to see if I can get around problems I have when my SIP (UA) behind their own NAT on the other side of my Internet connection. If you make any progress, please share. I will do the same. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: Monday, October 13, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT At 11:26 PM -0400 10/13/03, Uriel Carrasquilla wrote: From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) Reply-To: [EMAIL PROTECTED] Date: Mon, 13 Oct 2003 23:26:59 -0400 John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the shortcuts that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
log(1, This is a Long Distance Call\n);
route(6);
break;
};
};
.
.
.
route[6] {
rewritehostport(your_asterisk_box:5050);
if (!t_relay()) {
sl_reply_error();
};
}
Andres
http://www.telesip.net
I suspect it is very inexpensive to put a SER server in a hosting facility
to forward traffic to multiple Asterisks based on Least Cost Routing.
My problem is that my experience is with Asterisk and not with SER.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Monday, October 13, 2003 8:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.
SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.
SER is GPL'd and has very good user documentation. Don't know how well
the above will work. The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive
One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing to edit *.conf files.
See here for details http://www.iptel.org/ser/
=
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
KG6OMK
SER is an excellent option as a front end to Asterisk. It is a
true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been
the primary focus of Asterisk development. In fact, Asterisk's SIP
implementation is very limited (though it is extremely pragmatic.)
However, moving to SER does not solve any of the issues about the
proxy being behind a NAT, and I believe that SER will have the same
problems (though I could be wrong on this; I haven't experimented
with SER's ability to work from behind a NAT.) SIP clients work
well enough behind NAT (most of them, anyway) but the servers are a
different story.
I really like SER's third-party addons for account administration;
Asterisk is significantly more complex, and probably would not be as
easily converted to such a front end. In fact, SER has a very
complex routing/scripting language that is not easily administered
with a web front end, so I think that SER and Asterisk suffer from
the same problems. If someone were to come up with a simple way to
administer voicemail.conf and sip.conf from a web tool, that would go
far to making Asterisk a bit more user-accessible...
JT
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