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tjardickOffline
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Joined: Dec 12, 2003
Posts: 949

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Posted: Feb 09, 2005 - 02:59 PM Reply with quote Back to top
Update:-

Page about this on the sipX Wiki here:-

http://sipx-wiki.calivia.com/index.php/ ... h_Asterisk

I'm really interested in the advantages that SipX might have over Asterisk (if any), so don't mind people showing the advantages of SipX compared to Asterisk and maybe SER.

Allthough i think it is not in anyone's interrest to start a SipX vs. Asterisk battle Smile

What I would like to know, as both Asterisk and SipX can hand off calls to external gateway's, is if SipX PBX has better support for PBX behind NAT.

I think the main issue with asterisk on this point is that it doesn't support STUN, so it is always hard to setup Asterisk behind NAT on a dynamic IP connection (and I know there is IAX but that is not the point of my question - sometimes you just have/want to use SIP).

What else is there about features like: Voicemail, IVR, Call/voicestream proxying (e.g. asterisk on the router to support multple phones behind NAT without portmaps for each phone), SIMPLE (Sip Instant Messaging) etc. etc.

Any active SipX users/developers that might want to react on these things ?

Kind regards,

Tjardick
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dhublerOffline



Joined: Feb 09, 2005
Posts: 1

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Posted: Feb 09, 2005 - 05:36 PM Reply with quote Back to top
I work on the configuration server. Saw a comparison for sipX 2.4 v.s. Asteriks 1.0.3 here:

http://www.linuxworld.com.au/index.php/ ... p;4;fpid;4

Note: I don't claim to be a SIP expert here

Gateways: yes
STUN : No (although better phones should really handle this, i see point about voicestream proxying and that would be an interesting way to support phones that do not support STUN)
SIMPLE: No .. although I'm not an expert on subtle SIP flavors (...mmm SIP flavors Wink )
Voicemail:yes
IVR: yes
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ScaredyCatOffline



Joined: Jul 18, 2004
Posts: 3

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Posted: Mar 01, 2005 - 08:56 AM Reply with quote Back to top
Fundamentally,

Asterisk is NOT a sip proxy whereas SipX and SER are. You can get around some of the problems with Asterisk not being a proxy using dialplan tricks etc but you still don;t get some of the very cool features of proxys (eg forking)

Obviously SipX is for SIP only, asterisk can be used with sip, mgcp, h323, IAX (duh) etc.

ScaredyCat
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deanOffline
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Joined: Dec 13, 2003
Posts: 7172
Location: London
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Posted: May 26, 2005 - 02:17 PM Reply with quote Back to top
This is what LinuxWorld had to say about it:-

Quote:
SIPxchangeV.2.4

Unlike Asterisk's purely open source approach, Pingtel's SIPxchange has a more commercial flavor. It's still an open source product, the base PBX, sipX, can be downloaded from the SIPfoundry Web site for free, but if you pay Pingtel's modest price, you get additional support, plus plug-ins and tools such as media gateway services.

Most companies will opt for a dual-processor platform with power and disk redundancy for their IP PBX, but even here, the cost of moving to SIPxchange is very low compared to other commercial competitors we reviewed.

Pingtel's distribution of SIPxchange runs on Red Hat Linux and is compatible with most commercially available SIP-based IP phones. The 2.4 distribution includes Web-based graphical management interfaces, an automated attendant, and voice mail, among other features.

I tested the Pingtel software on a 1U server provided by the company. Most of the phones we used were Cisco 7912, 7940, and 7960 SIP, but we also included Pingtel's sipX softphone, which could be useful for traveling employees. All phones were connected to an Ethernet switch on a dedicated, closed network.

Initial installation of the sipX and Pingtel software required a considerable amount of manual intervention. Luckily, after the software was installed, the graphical interfaces eased the configuration burden, and there are scripts for many tasks.

However, adding phones to the network isn't always easy. You have to enter each phone's MAC (media access control) address individually before you can use the phone. This means either typing that address in or using a bar-code scanner, because the Pingtel software has no way to query the phones for this information. Plus, you'll have to repeat this process each time you add a new phone to the network. Unless you have a very small phone network, this could rapidly become a burden on the IT staff.

On the other hand, once you add the phones, managing them is made easier by a well-designed interface that allows you to control the phones' features and functions. You can, for example, assign specific functions to specific buttons on the phones centrally, and Pingtel includes graphical interfaces for user inboxes and for control of voice mail.

After the initial setup, I found SIPxchange to be very effective and easy to use. The user interfaces are convenient, and the graphical management interfaces are well-designed and intuitive, so you won't have to hire multiple sipX experts to run your phone system (although you will still need at least one). This is a solution that should fit in well with your enterprise, especially if you have experience with commercial distributions of open source software.


Do we have any existing sipX PBX users, or anyone else here using sipX based software of any other kind?

We're interested in hearing from you. We're currently considering porting our existing SER based SIP Server over to repro (SER has not been maintained in quite a while, which is beginning to concern us a little).

Dean
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clivecarter332Offline



Joined: Nov 18, 2004
Posts: 78
Location: Blaina, Abertillery, Gwent
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Posted: May 26, 2005 - 08:07 PM Reply with quote Back to top
Quote:

Do we have any existing sipX PBX users, or anyone else here using sipX based software of any other kind?

We're interested in hearing from you. We're currently considering porting our existing SER based SIP Server over to repro (SER has not been maintained in quite a while, which is beginning to concern us a little).
Dean

'er indoors has agrred to an increase in my pocket money ! New Dell server on order, BUT could only afford a Celeron Processor.
If this will run SipX I will be giving it a go.
Watch this space !
Clive
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