[Asterisk-Users] Asterisk and Cisco phones chan_sccp vschan_skinny vs native SIP and one-way audio

[Michael J. Tubby B.Sc.]

----- Original Message ----- 
From: "Julien Goodwin" <asterisk-lists at studio442.com.au>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Tuesday, February 01, 2005 3:42 AM
Subject: Re: [Asterisk-Users] Asterisk and Cisco phones chan_sccp
vschan_skinny vs native SIP and one-way audio

On Mon, Jan 31, 2005 at 04:05:23PM -0000, Michael J. Tubby B.Sc (Hons) G8TIC
arranged a set of bits into the following:

>>
>> I've recently built a couple of Asterisk boxes and want to migrate
>> away from CallManager to Asterisk.
>>
>> On my Asterisk box I have about 8 Grandstream BT101s and a
>> Cisco 7905G in SIP mode, on my CallManager I have about 10
>> x 30VIP, 2 x 7940 and a 7960.
>>
>> I've built Asterisk version 1.0.5 along with Zozo's chan_sccp
>> (CVS latest from last night) and got it partially working.  All devices
>> are on the inside of a private network at the moment (192.168.144.0/24)
>> and I'm having some issues with devices on chan_sccp.
>
> That's chan_sccp from chan-sccp.sourceforge.net?
>
Can't remember... might have been source-forge, might have been from
Zozo... I'll have to check

>> The 30VIPs can place and receive calls but I have a one-way
>> audio problem.  The 7960 can receive calls but when I place calls
>> from it I end up directly in the voicemail "unavailable" and the SIP
>> phone doesn't ring.
>
> I know about the problems with the 30vip and am slowly fixing them. but
> the issue with the sound is seemingly a bug in asterisk that says to
> tell the other device that 127.0.0.1 is the best route for RTP.

I've seen this mentioned in discussions elsewhere already implying that
Asterisk does a dns lookup on its own name as the recommended solution
is to check/change the /etc/hosts file to ensure that the server's own
name does not point to 127.0.0.1 but points to the IP address on which
it is reachable... eg. in my case gate.tubby.org needs /etc/hosts to look
like:

    127.0.0.1    localhost.localdomain localhost
    192.168.144.1    gate.tubby.org gate

Could you add something like:

    bind_address=a.b.c.d

or

    rtp_proxy=a.b.c.d

to force/ensure that the SCCP phone sent its RTP payload to the Asterisk
server which could, in turn, re-send the RTP stream to the ultimate
destination?


>> Looking at the network the SIP device opens an RTP stream to the
>> Cisco (30VIP or 7960) but the Cisco device doesn't send RTP
>> back to the SIP phone...  can anyone point me in the right direction
>> with this?

>> A more general question: with Cisco phones being removed from
>> a CallManager environment, is it best to keep them in Skinny/SCCP
>> mode or change out to SIP?  The 30VIPs can only do SCCP/Skinny
>> so which of the two channel drivers in Asterisk should I use for
>> best effect?

> chan_sccp has more features (and will have several more once I can get a
> current generation phone to test them on), but is less stable.

:o)

> As for the phones that support SIP, my view is that unfortunatly for now
> SIP is the better choice for stability and feature support.
>

Okay... I'll load my 7960 and 7940 with SIP then (for now) no probs...
but I would still like to migrate my 10-20 users with 30VIPs spread
around the world off CallManager and on to Asterisk...

> Thanks,
> Julien
> chan_sccp developer


Thanks!

Mike


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