I have a need to connect to an asterisk box running iax -- what happened to mod_iax I can't find it anywhere? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore. /b On Sep 6, 2010, at 9:18 PM, covici@ccs.covici.com wrote:
I have a need to connect to an asterisk box running iax -- what happened to mod_iax I can't find it anywhere?
Thanks.
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
On Sep 6, 2010, at 9:18 PM, covici@ccs.covici.com wrote:
I have a need to connect to an asterisk box running iax -- what happened to mod_iax I can't find it anywhere?
Thanks.
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
If you had attended ClueCon you would have know this! ;) The Doc and his Mr. Fusion couldn't save it. /b On Sep 6, 2010, at 9:32 PM, covici@ccs.covici.com wrote:
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch. On Mon, Sep 6, 2010 at 6:37 PM, Brian West <brian@freeswitch.org> wrote:
If you had attended ClueCon you would have know this! ;) The Doc and his Mr. Fusion couldn't save it. /b On Sep 6, 2010, at 9:32 PM, covici@ccs.covici.com wrote:
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
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Hi, There is another solution. As discussed at Cluecon. There is most of the iax code available in the opal/ptlib libraries. Sure - opal&ptlib don't build in freeswitch - some issue about this was mentioned at cluecon this year. However, the build script in freeswitch (from when I last looked) uses a branch version of ptlib and trunk version of opal.. Not a good mix. Better to use the trunk revision of ptlib&opal -- where ptlib&opal come from the same SVN revision number. Ok - there needs to be an iax registrar put into the opal code - bounties anyone? and you never know - it would be quite doable.. The voip protocol side of iax is useful - and great for people to use when connecting from an unknown position to some server on the public internet. Sufficiently useful to make it worth using. Who is game to work with me in getting the answer? Derek. On Mon, 6 Sep 2010, Jeffrey Leung wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
On Mon, Sep 6, 2010 at 6:37 PM, Brian West <brian@freeswitch.org> wrote:
If you had attended ClueCon you would have know this! ;) The Doc and his Mr. Fusion couldn't save it. /b On Sep 6, 2010, at 9:32 PM, covici@ccs.covici.com wrote:
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
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-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
While IAX is good for penetrating NAT issues. There is no good solution to load balance IAX registration or calls. So if you want to build something scalable, it is better to use SIP. Henry Huang Unified Communication System R&D, Founder of UniC Solution US: +1 (626) 606-3306 台灣(Taiwan): +886 933847619 Chat Google Talk: red_rain_seven@gmail.com Skype: unicsolution MSN: b_ball_henry@hotmail.com Contact Me [image: Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image: Facebook] <http://www.facebook.com/profile.php?id=595148342>[image: Twitter]<http://twitter.com/unicsolution> On Tue, Sep 7, 2010 at 12:06 PM, Derek Smithies <derek@indranet.co.nz>wrote:
Hi, There is another solution.
As discussed at Cluecon. There is most of the iax code available in the opal/ptlib libraries.
Sure - opal&ptlib don't build in freeswitch - some issue about this was mentioned at cluecon this year. However, the build script in freeswitch (from when I last looked) uses a branch version of ptlib and trunk version of opal.. Not a good mix. Better to use the trunk revision of ptlib&opal -- where ptlib&opal come from the same SVN revision number.
Ok - there needs to be an iax registrar put into the opal code - bounties anyone? and you never know - it would be quite doable..
The voip protocol side of iax is useful - and great for people to use when connecting from an unknown position to some server on the public internet. Sufficiently useful to make it worth using.
Who is game to work with me in getting the answer?
Derek.
On Mon, 6 Sep 2010, Jeffrey Leung wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as
an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
On Mon, Sep 6, 2010 at 6:37 PM, Brian West <brian@freeswitch.org> wrote:
If you had attended ClueCon you would have know this! ;) The Doc and his Mr. Fusion couldn't save it. /b On Sep 6, 2010, at 9:32 PM, covici@ccs.covici.com wrote:
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
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-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/
"How did you make it work??" "Oh, the usual, get everything right". _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, Surely not. On Tue, 7 Sep 2010, Henry Huang wrote:
While IAX is good for penetrating NAT issues. There is no good solution to load balance IAX registration or calls. So if you want to build something scalable, it is better to use SIP.
The iax server issues a call transfer command, to transfer the incoming call to a different server. you do have to write the logic to decide when/where to transfer the calls. Derek. -- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
IAX is also great for saving bandwidth by using the "trunk" mode. Regards HASSAN On Tue, Sep 7, 2010 at 16:26, Derek Smithies <derek@indranet.co.nz> wrote:
Hi, Surely not.
On Tue, 7 Sep 2010, Henry Huang wrote:
While IAX is good for penetrating NAT issues. There is no good solution to load balance IAX registration or calls. So if you want to build something scalable, it is better to use SIP.
The iax server issues a call transfer command, to transfer the incoming call to a different server.
you do have to write the logic to decide when/where to transfer the calls.
Derek. -- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/
"How did you make it work??" "Oh, the usual, get everything right".
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True. But, over TDM interconnects (like a IPLC), this saves you from buying expensive compression equipment. Regards HASSAN On Tue, Sep 7, 2010 at 18:12, Brian West <brian@freeswitch.org> wrote:
All bollocks! The savings are negated when you start to lose packets and the whole trunk gets hozed.
/b
On Sep 7, 2010, at 6:43 AM, Nyamul Hassan wrote:
IAX is also great for saving bandwidth by using the "trunk" mode.
Regards HASSAN
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If its a TDM interconnect then how is IAX going to save you? I think you might mean IP interconnect over an IPL? /b On Sep 7, 2010, at 8:20 AM, Nyamul Hassan wrote:
True. But, over TDM interconnects (like a IPLC), this saves you from buying expensive compression equipment.
Regards HASSAN
Yes. IP over IPLC. Regards HASSAN On Tue, Sep 7, 2010 at 18:26, Brian West <brian@freeswitch.org> wrote:
If its a TDM interconnect then how is IAX going to save you? I think you might mean IP interconnect over an IPL?
/b
On Sep 7, 2010, at 8:20 AM, Nyamul Hassan wrote:
True. But, over TDM interconnects (like a IPLC), this saves you from buying expensive compression equipment.
Regards HASSAN
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Brian, is that a protocol - or implementation issue? My understanding is that the protocol is going to save you bytes on the network... However, sip&h323 also have trunking type systems, which will also save you bytes on the network.. Derek. Further - when the packet loss starts to happen, you have other issues.. Derek. On Tue, 7 Sep 2010, Brian West wrote:
All bollocks! The savings are negated when you start to lose packets and the whole trunk gets hozed.
/b
On Sep 7, 2010, at 6:43 AM, Nyamul Hassan wrote:
IAX is also great for saving bandwidth by using the "trunk" mode.
Regards HASSAN
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-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
On 09/07/2010 12:06 PM, Derek Smithies wrote:
Hi, There is another solution.
As discussed at Cluecon. There is most of the iax code available in the opal/ptlib libraries.
Sure - opal&ptlib don't build in freeswitch - some issue about this was mentioned at cluecon this year. However, the build script in freeswitch (from when I last looked) uses a branch version of ptlib and trunk version of opal.. Not a good mix. Better to use the trunk revision of ptlib&opal -- where ptlib&opal come from the same SVN revision number.
Ok - there needs to be an iax registrar put into the opal code - bounties anyone? and you never know - it would be quite doable..
The voip protocol side of iax is useful - and great for people to use when connecting from an unknown position to some server on the public internet. Sufficiently useful to make it worth using. From an unknown position on the internet STUN usually works just fine. The * people loving saying STUN is all very bad, but a recent exchange on their mailing list seems to indicate none of them understand it. :-)
If you like this media through the signalling server scheme, I have a new car I think you'll love. Every trip it makes goes though the forecourt of the dealer who supplied it. :-) Steve
our current position on iax is as follows; mod_iax is in the unsupported repo and already is probably not compiling due to recent changes to the core. The protocol exposed backwards incompatibilities that led to crashes so we no longer trust it in our codebase. We feel that in order to support IAX we either need to go with derek and opal or build a completely standalone robust IAX stack which is a much more complicated, expensive and almost academic of a solution at this juncture. On Tue, Sep 7, 2010 at 9:07 AM, Steve Underwood <steveu@coppice.org> wrote:
On 09/07/2010 12:06 PM, Derek Smithies wrote:
Hi, There is another solution.
As discussed at Cluecon. There is most of the iax code available in the opal/ptlib libraries.
Sure - opal&ptlib don't build in freeswitch - some issue about this was mentioned at cluecon this year. However, the build script in freeswitch (from when I last looked) uses a branch version of ptlib and trunk version of opal.. Not a good mix. Better to use the trunk revision of ptlib&opal -- where ptlib&opal come from the same SVN revision number.
Ok - there needs to be an iax registrar put into the opal code - bounties anyone? and you never know - it would be quite doable..
The voip protocol side of iax is useful - and great for people to use when connecting from an unknown position to some server on the public internet. Sufficiently useful to make it worth using. From an unknown position on the internet STUN usually works just fine. The * people loving saying STUN is all very bad, but a recent exchange on their mailing list seems to indicate none of them understand it. :-)
If you like this media through the signalling server scheme, I have a new car I think you'll love. Every trip it makes goes though the forecourt of the dealer who supplied it. :-)
Steve
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Hi, On Tue, 7 Sep 2010, Steve Underwood wrote:
If you like this media through the signalling server scheme, I have a new car I think you'll love. Every trip it makes goes though the forecourt of the dealer who supplied it. :-)
yeah - we could invent snap - shit not another protocol with a TCP port for signalling. Media is in rtp packets for ease of interchange with h.323 and sip.. 42424 would be the port I would use - for tcp and udp. 31415 would also do, I suppose.. But that is hard to get enough people to use it, to get the number of users to make it worthwhile. Derek. -- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
Yuck. On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
On Mon, Sep 6, 2010 at 6:37 PM, Brian West <brian@freeswitch.org> wrote:
If you had attended ClueCon you would have know this! ;) The Doc and his Mr. Fusion couldn't save it. /b On Sep 6, 2010, at 9:32 PM, covici@ccs.covici.com wrote:
OK, thanks. I guess I am screwed trying to get to that box. Brian West <brian@freeswitch.org> wrote:
A tear in time and space has caused the flux capacitor to not function properly so we had to retire the module since the lib didn't work right anymore.
/b
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Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
On Tue, Sep 7, 2010 at 8:23 AM, Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
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Bad habits are hard to kick.
Another scenario is, in countries where regular SIP / H323 is blocked, IAX is mostly not so, and therefore helps. I personally prefer to work with H323 / SIP. But, ground realities are a bit different at times. Regards HASSAN On Tue, Sep 7, 2010 at 18:28, Tony Graziano <tgraziano@myitdepartment.net>wrote:
On Tue, Sep 7, 2010 at 8:23 AM, Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
Bad habits are hard to kick.
Derek: I agree that your method would work, but I was talking about more native load balancer without rewriting the logics. Henry Huang Unified Communication System R&D, Founder of UniC Solution US: +1 (626) 606-3306 台灣(Taiwan): +886 933847619 Chat Google Talk: red_rain_seven@gmail.com Skype: unicsolution MSN: b_ball_henry@hotmail.com Contact Me [image: Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image: Facebook] <http://www.facebook.com/profile.php?id=595148342>[image: Twitter]<http://twitter.com/unicsolution> On Tue, Sep 7, 2010 at 8:33 PM, Nyamul Hassan <mnhassan@usa.net> wrote:
Another scenario is, in countries where regular SIP / H323 is blocked, IAX is mostly not so, and therefore helps.
I personally prefer to work with H323 / SIP. But, ground realities are a bit different at times.
Regards HASSAN
On Tue, Sep 7, 2010 at 18:28, Tony Graziano <tgraziano@myitdepartment.net>wrote:
On Tue, Sep 7, 2010 at 8:23 AM, Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
Bad habits are hard to kick.
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On 09/07/2010 02:33 PM, Nyamul Hassan wrote:
Another scenario is, in countries where regular SIP / H323 is blocked, IAX is mostly not so, and therefore helps.
I personally prefer to work with H323 / SIP. But, ground realities are a bit different at times.
In that case why not use an openvpn link? Regards, Patrick
We are using iax trunk feature to save bandwidth, it makes lot of sense when your branches/offices are connected through a dedicated radio/fiber, we end up saving millions per year using IAX, @Brian, for me it always worked like a charm, never faced quality/packet loss issues with ~20 concurrent calls over iax trunk in business hours. I hope someone will take this challenge someday and rewrite more robust iax stack for FreeSWITCH :) Thanks and best regards, -m On Tue, Sep 7, 2010 at 8:49 PM, Patrick Lists < freeswitch-list@puzzled.xs4all.nl> wrote:
On 09/07/2010 02:33 PM, Nyamul Hassan wrote:
Another scenario is, in countries where regular SIP / H323 is blocked, IAX is mostly not so, and therefore helps.
I personally prefer to work with H323 / SIP. But, ground realities are a bit different at times.
In that case why not use an openvpn link?
Regards, Patrick
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Perhaps that would be something you could spend some of the millions per year you are saving to finance. It could benefit any project that wants to support IAX if it were properly licensed to BSDish. On Tue, Sep 7, 2010 at 1:59 PM, Ghulam Mustafa <mustafa.pk@gmail.com> wrote:
We are using iax trunk feature to save bandwidth, it makes lot of sense when your branches/offices are connected through a dedicated radio/fiber, we end up saving millions per year using IAX,
@Brian, for me it always worked like a charm, never faced quality/packet loss issues with ~20 concurrent calls over iax trunk in business hours.
I hope someone will take this challenge someday and rewrite more robust iax stack for FreeSWITCH :)
Thanks and best regards,
-m
On Tue, Sep 7, 2010 at 8:49 PM, Patrick Lists <freeswitch-list@puzzled.xs4all.nl> wrote:
On 09/07/2010 02:33 PM, Nyamul Hassan wrote:
Another scenario is, in countries where regular SIP / H323 is blocked, IAX is mostly not so, and therefore helps.
I personally prefer to work with H323 / SIP. But, ground realities are a bit different at times.
In that case why not use an openvpn link?
Regards, Patrick
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Hi, Most of it is there in mod opal. Just need someone to put up a bounty, and people like me can finish the opal code. Derek. On Tue, 7 Sep 2010, Ghulam Mustafa wrote:
We are using iax trunk feature to save bandwidth, it makes lot of sense when your branches/offices are connected through a dedicated radio/fiber, we end up saving millions per year using IAX,
@Brian, for me it always worked like a charm, never faced quality/packet loss issues with ~20 concurrent calls over iax trunk in business hours.
I hope someone will take this challenge someday and rewrite more robust iax stack for FreeSWITCH :)
Thanks and best regards,
-m
On Tue, Sep 7, 2010 at 8:49 PM, Patrick Lists <freeswitch-list@puzzled.xs4all.nl> wrote: On 09/07/2010 02:33 PM, Nyamul Hassan wrote: > Another scenario is, in countries where regular SIP / H323 is blocked, > IAX is mostly not so, and therefore helps. > > I personally prefer to work with H323 / SIP. But, ground realities are > a bit different at times.
In that case why not use an openvpn link?
Regards, Patrick
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-- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity@ekiga.net mail: mustafa.pk@gmail.com web: cyrenity.wordpress.com
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
But what idiot had the ip address in the data -- bad idea or so it seems to me. Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
---------------------------------------------------- Alternatives:
---------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
The same idiot that put the IP in the UDP header? /b On Sep 7, 2010, at 11:35 AM, covici@ccs.covici.com wrote:
But what idiot had the ip address in the data -- bad idea or so it seems to me.
Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
Hi, the voip protocols (sip & h323) had their genisys two decades ago - long before there were agreed on "best practices". putting the ip address in the packets made things like transfer easier to do, or directing the media to a different box. Derek. On Tue, 7 Sep 2010, covici@ccs.covici.com wrote:
But what idiot had the ip address in the data -- bad idea or so it seems to me.
Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly.
/b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
---------------------------------------------------- Alternatives:
---------------------------------------------------- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
In reality, it is not easy! ICE, TURN, STUN, UPnP. wth so many RFCs. that is why Skype is more popular even it is not open standard. On Tue, Sep 7, 2010 at 5:23 AM, Brian West <brian@freeswitch.org> wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote:
If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, cause most sip phones I have tried suck. Typically, the UI is broken, and hard to follow. When a user gets an iax phone, all they have to do is set the locn of the remote server, and "it just works". Remember, your average user does not know what a file is, or how to use one. "if you know how to set it up properly". ok - this statement means that SIP is good for 0.1% of the population.. Derek. On Tue, 7 Sep 2010, Brian West wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
This thread seems to go into a discussion on pro and cons of IAX in particular cases... As anything, IAX can have users that find it useful. Problem is, the "open" (as in BSD compatible license) implementation of the IAX stack is broken, and to write a new stack is a *really* big work, that takes time from professional programmers and testers. So, at end, problem is: the peoples/companies that would like to use IAX with FreeSWITCH (or other software not GPLed) are willing/able to raise enough money to fund the development of a working and reliable IAX stack? Maybe the people/companies interested can start a fund raising via web, launching a sort of consortium for that... -giovanni On Tue, Sep 7, 2010 at 11:15 PM, Derek Smithies <derek@indranet.co.nz> wrote:
Hi, cause most sip phones I have tried suck.
Typically, the UI is broken, and hard to follow.
When a user gets an iax phone, all they have to do is set the locn of the remote server, and "it just works".
Remember, your average user does not know what a file is, or how to use one.
"if you know how to set it up properly". ok - this statement means that SIP is good for 0.1% of the population..
Derek.
On Tue, 7 Sep 2010, Brian West wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/
"How did you make it work??" "Oh, the usual, get everything right". _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618
Hi, the opal library contains an IAX implementation that did voice calls to digium. At one stage, it did call transfer. along came the iax management, who added this call token thing to the iax protocol. The opal code contains call token handling. Remaining? to write an iax registrar, and find a way to ensure that all this can talk with freeswitch. license? the opal implementation of iax and h323 and sip is all MPL. connectivity with freeswitch? There is code in freeswitch to talk to opal to do h323 calls. Since opal uses a normalised approach to voip protocols and consequently all protocols look the same to the management software, there is little to do to get connectivity with freeswitch. code quality? the opal code for iax is heavily documented, is readable and nice. it is heavily multi-threaded, so will cope with huge call volumes. So what is remaining?
Maybe the people/companies interested can start a fund raising via web, launching a sort of consortium for that... Exactly - but there is only one way to go that has a chance of working, which is to use the opal code.
Derek. On Tue, 7 Sep 2010, Giovanni Maruzzelli wrote:
This thread seems to go into a discussion on pro and cons of IAX in particular cases...
As anything, IAX can have users that find it useful.
Problem is, the "open" (as in BSD compatible license) implementation of the IAX stack is broken, and to write a new stack is a *really* big work, that takes time from professional programmers and testers.
So, at end, problem is: the peoples/companies that would like to use IAX with FreeSWITCH (or other software not GPLed) are willing/able to raise enough money to fund the development of a working and reliable IAX stack?
Maybe the people/companies interested can start a fund raising via web, launching a sort of consortium for that...
-giovanni
On Tue, Sep 7, 2010 at 11:15 PM, Derek Smithies <derek@indranet.co.nz> wrote:
Hi, cause most sip phones I have tried suck.
Typically, the UI is broken, and hard to follow.
When a user gets an iax phone, all they have to do is set the locn of the remote server, and "it just works".
Remember, your average user does not know what a file is, or how to use one.
"if you know how to set it up properly". ok - this statement means that SIP is good for 0.1% of the population..
Derek.
On Tue, 7 Sep 2010, Brian West wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/
"How did you make it work??" "Oh, the usual, get everything right". _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Derek Smithies Ph.D. IndraNet Technologies Ltd. ph +64 3 365 6485 Web: http://www.indranet-technologies.com/ "How did you make it work??" "Oh, the usual, get everything right".
On Sep 7, 2010, at 7:27 PM, Derek Smithies wrote:
Snip... So what is remaining?
Maybe the people/companies interested can start a fund raising via web, launching a sort of consortium for that... Exactly - but there is only one way to go that has a chance of working, which is to use the opal code.
Has anyone figured out how much money we're talking about?
--FC
Hi, SIP phones have been made easy for their target market to use, and end users are not their target market. Service providers are the target market, and SIP phones are mostly made easy for those people to provision and control in volume. Unless a strong end user market for SIP phones develops, this won't change. Most IAX phones sit outside this, as they are mostly made by people who simply can't get into the high volume service provider business, where they want to be. Usually they are second division phones, as well as second division suppliers. The good hardware runs SIP. Steve On 09/08/2010 05:15 AM, Derek Smithies wrote:
Hi, cause most sip phones I have tried suck.
Typically, the UI is broken, and hard to follow.
When a user gets an iax phone, all they have to do is set the locn of the remote server, and "it just works".
Remember, your average user does not know what a file is, or how to use one.
"if you know how to set it up properly". ok - this statement means that SIP is good for 0.1% of the population..
Derek.
On Tue, 7 Sep 2010, Brian West wrote:
Why is everyone dead set on using IAX? you know SIP works fine and can traverse NAT if you know how to set it up properly. /b
On Sep 7, 2010, at 8:16 AM, Steven Ayre wrote:
Yuck.
On 7 September 2010 02:56, Jeffrey Leung <curriegrad2004@gmail.com> wrote: If you do really want to use IAX, there is a solution: Use Asterisk as an IAX protocol translator and have it to forward all the calls via SIP to Freeswitch.
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (15)
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Anthony Minessale
-
Brian West
-
covici＠ccs.covici.com
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Derek Smithies
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Frank Carmickle
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Ghulam Mustafa
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Giovanni Maruzzelli
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Henry Huang
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Jeffrey Leung
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jesse
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Nyamul Hassan
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Patrick Lists
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Steve Underwood
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Steven Ayre
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Tony Graziano