SCCP and Direct RTP streams

milesje

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#1
I am using Elastix 2.0.2 with SCCP-b V3 RC3. Phones are Cisco 7971 & 7970. I need to have the phones maintain their call if the PBX is shut off. I thought the directrtp option in the sccp.conf file would allow this to happen but so far no luck. Does anyone know if this is possible with asterisk and SCCP?
 

fmvillares

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#2
if the call is mantained into 2 sccp only phones using direct rtp or the same using reinvite into 2 sip phones the call is mantained...if asterisk gets cut off all transcoding and mixed protocol calls would be dropped....
 

milesje

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#3
All phones on the system are SCCP and the directrtp flag is set in the sccp.conf file on the Asterisk box. Is there a setting on the actuall phone that needs be set as well?

With two SCCP phones connected if the asterisk box is lost both phones do drop the call.

Edit: well let me rephrase that when the asterisk box is lost the sccp phones will still show as connected but there is no audio on either phone.
 

fmvillares

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#4
Re: Re:SCCP and Direct RTP streams

well as you may know sccp is not fully supported in asterisk and i dont have any of them to try to reproduce here the issue but in sip works just fine in reinvite mode...maybe in new versions or some misconfig in your phones..but in my case i will try to change all phones to sip in asterisk
 

milesje

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#5
Unfortunatly we are unable to use SIP; SCCP is our only option, and yes I know that SCCP is a third party package and not supported by Asterisk. I was hoping to find someone though who does also uses skinny firmware phones that has been able to get this to work.
 

fmvillares

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#6
Re: Re:SCCP and Direct RTP streams

ok sorry cant help you more than this...you re a using a quite like frankenstein solution for asterisk and this type of things are the prize to pay there...and remember asterisk was not created to be a p2p like pbx like call manager or sphere etc...asterisk controls everything from the pbx rtp etc by default to monitor record etc...but with reinvite options you could revert this in sip only...
 

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