Re: [Freeswitch-users] Should I choose FreeSwitch?
I know you excused the obvious bias but even then I typically as a policy do not recommend my own software to anyone. I am able to invite you to help work on the project and to welcome any input you have. Like you say, you will find that Asterisk is designed with PBX in mind where FreeSWITCH has a primary focus on being a switching platform and an application server. However, the way FreeSWITCH is designed it's more than possible to create a PBX environment just by way of configuration and possibly the addition of a few missing features that are mostly missing due to a non-urgency. I myself have done extensive work on the Asterisk project which does give me the right to comment on the quality of the code. I have many problems with Asterisk that have a great influence on why I started this project but I will not get into the details of them in this email. That said I still contribute to the Asterisk community to this day when I find time. If you do decide to use Asterisk I highly recommend you stick with the 1.2 release for your best chance of success. You also may want to consider Call Weaver, GNU Bayonne or YATE just to make sure you have considered all options. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Lucia Rotger <lrotger@aircomp.aero> To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, September 18, 2007 5:39:08 AM Subject: [Freeswitch-users] Should I choose FreeSwitch? I have to implement a production software PBX. After some googling I came out with two candidates: Asterisk and FreeSwitch. Asterisk looks like crap; granted, it's working in many installations and seems to be readier for prime time. But just looking at their bugs puts me away. Segfaults all over the place, crashes after handling a few hundred calls: in short, very poor, sloppy programming I think. Maybe I'm wrong here? I like the way things are handled here and I like the coding style a lot. I think it's a matter of time FS becomes a great piece of software. Also from looking at the bug tracker the issues don't seem to be like those plaguing Asterisk. What are your thoughts? apart from the obvious bias you have for FS since you are here? I'd also like to read documentation if I have to consider FreeSwitch. I see the wiki is incomplete in many places. There are variables/applications I can't even begin to guess what they do. Also I'm aware that FS is a platform and Asterisk if just a PBX so its ambitions are greater; in fact I only want a PBX not a communications framework but as I've said above I feel Asterisk is going to be a lot of pain. The PBX will have to handle no more than 24 calls through 12 BRI channels. The local phones will be SIP and the traffic will be mainly incoming calls although outgoing calls will be made also. I've never touched anything VoIP so I'm a total newbie. Thanks, Lucia Rotger _______________________________________________ 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 ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/
Anthony,
I know you excused the obvious bias but even then I typically as a policy do not recommend my own software to anyone. I am able to invite you to help work on the project and to welcome any input you have.
Thanks for the invitation; now I'm busy setting this up but I will install a test freeswitch sometime so I can get the hang of it. If you accept recommendations at this point I'd say accurate documentation is a plus; lack of it (or obsolete) is usually a showstopper. There is very good software out there but lack of documentation keeps it from being widely adopted (i.e. Zope).
Like you say, you will find that Asterisk is designed with PBX in mind where FreeSWITCH has a primary focus on being a switching platform and an application server. However, the way FreeSWITCH is designed it's more than possible to create a PBX environment just by way of configuration and possibly the addition of a few missing features that are mostly missing due to a non-urgency. I myself have done extensive
I like the generality of working with a framework, that way weird applications can be done.
work on the Asterisk project which does give me the right to comment on the quality of the code. I have many problems with Asterisk that have a great influence on why I started this project but I will not get into the details of them in this email. That said I still contribute to the
Oh my, don't take my comment personally :) In fact independently of the quality of the individual developers I know it's difficult to do good work (and very frustrating) if the problem hasn't been addressed correctly from the beginning, and this shows in Asterisk.
Asterisk community to this day when I find time. If you do decide to use Asterisk I highly recommend you stick with the 1.2 release for your best chance of success. You also may want to consider Call Weaver, GNU Bayonne or YATE just to make sure you have considered all options.
I've also read somewhere that 1.2 is more stable than 1.4. I appreciate you mentioning the other options. I already knew of them but as I said the candidates are down to Asterisk and Freeswitch (well only Asterisk now but still). Thanks again for your kind reply :) Lucia
Gateways are usually the better option because you can configure your phones to use it as a failover for WHEN your Asterisk server deadlocks and crashes. A gateway would also provide better echo cancellation and jitter buffer support then a card based solution. There is a list of BRI gateways on the voip-info wiki here: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways On 9/18/07, Lucia Rotger <lrotger@aircomp.aero> wrote:
Anthony,
I know you excused the obvious bias but even then I typically as a policy do not recommend my own software to anyone. I am able to invite you to help work on the project and to welcome any input you have.
Thanks for the invitation; now I'm busy setting this up but I will install a test freeswitch sometime so I can get the hang of it. If you accept recommendations at this point I'd say accurate documentation is a plus; lack of it (or obsolete) is usually a showstopper. There is very good software out there but lack of documentation keeps it from being widely adopted (i.e. Zope).
Like you say, you will find that Asterisk is designed with PBX in mind where FreeSWITCH has a primary focus on being a switching platform and an application server. However, the way FreeSWITCH is designed it's more than possible to create a PBX environment just by way of configuration and possibly the addition of a few missing features that are mostly missing due to a non-urgency. I myself have done extensive
I like the generality of working with a framework, that way weird applications can be done.
work on the Asterisk project which does give me the right to comment on the quality of the code. I have many problems with Asterisk that have a great influence on why I started this project but I will not get into the details of them in this email. That said I still contribute to the
Oh my, don't take my comment personally :) In fact independently of the quality of the individual developers I know it's difficult to do good work (and very frustrating) if the problem hasn't been addressed correctly from the beginning, and this shows in Asterisk.
Asterisk community to this day when I find time. If you do decide to use Asterisk I highly recommend you stick with the 1.2 release for your best chance of success. You also may want to consider Call Weaver, GNU Bayonne or YATE just to make sure you have considered all options.
I've also read somewhere that 1.2 is more stable than 1.4.
I appreciate you mentioning the other options. I already knew of them but as I said the candidates are down to Asterisk and Freeswitch (well only Asterisk now but still).
Thanks again for your kind reply :) Lucia
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Hi, Does that mean the phones must be configured to try ASterisk first, and if they don't succeed then try to reach a SIP VoIP provider like those in that list? If I understood you correctly, how do I know if a particular SIP phone can do that? or do I insert a SIP proxy between the phones and Asterisk and when Asterisk becomes unreachable the proxy connects the phones elsewhere? I'm assuming a SIP proxy works like a web proxy? Regards, Lucia On Wed, 19 Sep 2007 04:25:41 +0200, Jason Garland <jgarland@gmail.com> wrote:
Gateways are usually the better option because you can configure your phones to use it as a failover for WHEN your Asterisk server deadlocks and crashes. A gateway would also provide better echo cancellation and jitter buffer support then a card based solution.
There is a list of BRI gateways on the voip-info wiki here: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways
If your phone can use DNS SRV records it can do it. Or if the phone supports multiple proxy servers. Sent from my iPhone On Sep 19, 2007, at 7:15 AM, "Lucia Rotger" <lrotger@aircomp.aero> wrote:
Hi,
Does that mean the phones must be configured to try ASterisk first, and if they don't succeed then try to reach a SIP VoIP provider like those in that list?
If I understood you correctly, how do I know if a particular SIP phone can do that? or do I insert a SIP proxy between the phones and Asterisk and when Asterisk becomes unreachable the proxy connects the phones elsewhere? I'm assuming a SIP proxy works like a web proxy?
Regards, Lucia
On Wed, 19 Sep 2007 04:25:41 +0200, Jason Garland <jgarland@gmail.com> wrote:
Gateways are usually the better option because you can configure your phones to use it as a failover for WHEN your Asterisk server deadlocks and crashes. A gateway would also provide better echo cancellation and jitter buffer support then a card based solution.
There is a list of BRI gateways on the voip-info wiki here: http://www.voip-info.org/wiki/index.php?page=VOIP+Gateways
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participants (3)
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Anthony Minessale
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Jason Garland
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Lucia Rotger