# Useing Asterisk as a front end to SipX

**URL:** <https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323>\
**Category:** Asterisk Support\
**Created:** [April 13, 2008, 9:52pm UTC](https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323 "2008-04-13T21:52:14Z")\
**Posts on this page:** 4\
**Page:** 1

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**Author:** ![rspalding](https://avatars.discourse-cdn.com/v4/letter/r/97f17d/32.png) [@rspalding](https://community.asterisk.org/u/rspalding)\
**Post date:** [April 13, 2008, 9:52pm UTC](https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323/1 "2008-04-13T21:52:14Z")

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We are currently trying to setup a VOIP system using Polycom Phones…have a SipX server setup but have been having trouble getting connected to Vitelity with two way communication.

My partner is suggesting this …

Quote.“Idea is to use Asterisk to punch through NAT, sipX to do everything else. It appears that Asterisk is already registered to Vitelity and will handle a call from outside, but does not yet send it on to sipX and thus does not get to a phone.”

Any comments?

I am wondering why we have to do that?

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**Author:** ![leon-tcs](https://avatars.discourse-cdn.com/v4/letter/l/3ab097/32.png) [@leon-tcs](https://community.asterisk.org/u/leon-tcs)\
**Post date:** [April 15, 2008, 3:01pm UTC](https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323/2 "2008-04-15T15:01:51Z")

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what features do you want in the sipX. I think one sip server will be stable than 2.

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**Author:** ![yesmat](https://avatars.discourse-cdn.com/v4/letter/y/b487fb/32.png) [@yesmat](https://community.asterisk.org/u/yesmat)\
**Post date:** [May 1, 2008, 3:15am UTC](https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323/3 "2008-05-01T03:15:05Z")

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sipX is a great product and much more scalable than Asterisk, it can support few thousands of extensions. HA between 2 servers could be setup in a breaze.

It laso has a great interafce and endpoint support second to none.

The real big problem with sipX is NAT. It doesn’t suuport NAT traversal and having Asterisk in front, may solve this issue.

Good luck

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**Author:** ![ziro\_axis](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/ziro_axis/32/212_2.png) [@ziro\_axis](https://community.asterisk.org/u/ziro_axis)\
**Post date:** [May 3, 2008, 10:14pm UTC](https://community.asterisk.org/t/useing-asterisk-as-a-front-end-to-sipx/20323/4 "2008-05-03T22:14:13Z")

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Dear yesmat;  
what you mean by SipXesc is stable than Asterisk;  
actually i want to install one VoIP PBX so what is my choices, SipXesc or asteisk, or both of them!  
i have a miniProject about building up on VoIP PBX i searche the free ones i found Asterisk and now you are saying SipXesc is better.  
thus i hope you can help me in chosing what is suitable for my application!  
and thanks for your support