yes, according to the following post.

http://lists.digium.com/pipermai ... 04-June/000547.html

also note "The first thing you have to remember is that Asterisk is *not* a SIP proxy, it's a SIP PBX." Any SIP proxy such as iptel or whatever does not stay in the middle of the media stream

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vsp also includes a registar server,  and parameters such "canreinvite=yes" must be enabled as default. No point for them to stay in the middle.

if your call goes out to pstn, the vsp will redirect you to their pstn gateways.

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回復 6851# hersvim


   
Thanks for elaborating the merit and demerit of using CANREIVITE to us.  I have read the passage in 2004 that you advised.  In the last paragraph of the post, I also found that there is still a limitation of using "canreinvite=yes" if our devices are set behind NAT which will cause one audio stream when they communicate to the one on the outside.  In fact, many of the members here had experienced this before.  Can we expect a better suggestion to resolve the problem when we deploy our Asterisk Server and other related ATA or IP phones behind NAT?  

BTW, I've just learnt from http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite that the parameter of "canreinvite=yes" has a change in certain extent in the migration from Asterisk 1.2 to 1.4.

QUOTE:

The "canreinvite" option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: "yes", "no", "nonat", "update".

UNQUOTE


Thanks for sharing.

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回復 2083# yhfung

yhfung兄:
Siemens S685 IP
有6-SIP VOIP  可否同时和2-sip voip 进行三方通话?

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回復 6854# keernest

角色C兄和眾高手已經轉移陣地, 我已經PM給你!

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回復 6855# 雯雯


    谢谢

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回復  keernest

角色C兄和眾高手已經轉移陣地, 我已經PM給你!
雯雯 發表於 2010-5-3 11:46


麻烦也告诉我

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回復 6857# helenmak

已經PM你了! 請查收!

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PM埋我, thanks

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回復 6859# tamtsin038

已經PM你了! 請查收!

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