[Asterisk-Users] Re: asterisk-grandstream call
Andres
andres at telesip.net
Tue Feb 10 15:42:56 MST 2004
Bill Michaelson wrote:
>
>
>>>I am trying to muddle my way tthrough getting something - actually
>>>anything to work - with Asterisk. I've acquired a Grandstream phone and
>>>I've got * on a Red Hat 9 box. I've gotten to a point where I can see
>>>(via ethereal) that the phone REGISTER's successfully with asterisk, and
>>>then I try to dial into voicemail. This is what I observe in the packet
>>>trace...
>>>
>>>GS: INVITE -> *
>>>*: Status 100 (Trying) -> GS
>>>*: Status 200 (OK with session description) -> GS
>>
>>
>
>Does the GS then send an ACK? It should. If it doesn't then this
>probably means that it hasn't received the 200 response. (firewall
>issue?)
>
>If it is sending the ACK, then it is probably a codec issue, as has
>been already mentioned. GS doesn't always seem to do very well in
>codec selection.
>
>Doug
>
>
> -------------------------
> Thanks for that hint. I see what you mean. When configured for FWD,
> the GS does indeed send an ACK at an equivalent point in the protocol.
>
> But no, the GS does not send an ACK when configured for my * box. I
> suppose the * box is expecting it, because about one second later, the
> * box resends the 200 message - this in spite of the fact that has
> started spewing RTP furiously. Both devices are on the same LAN, with
> no intervening firewall, and the OK ought to be visible to the GS (the
> packet even contains the expected destination MAC ID, derived earlier
> via ARP).
>
> That makes two mysteries: 1) why doesn't the GS seem to see the OK?
> and 2) why does * send the RTP stream in spite of the fact that it has
> not received the ACK from the GS? Shouldn't it wait?
>
> Regarding codec selection, I see a minor difference between the FWD
> and the local * box test cases, but I know nothing about the
> negotiation protocol...
>
> With FWD, the OK message lists 3 Media Formats:
>
> Media Description, name and address (m): audio 10496 RTP/AVP 0 8 101
> Media Type: audio
> Media Port: 10496
> Media Proto: RTP/AVP
> Media Format: 0
> Media Format: 8
> Media Format: 101
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
>
> But with the local box, it lists one other - note the addition of GSM...
>
> Media Description, name and address (m): audio 16708 RTP/AVP 3 0 8 101
> Media Type: audio
> Media Port: 16708
> Media Proto: RTP/AVP
> Media Format: 3
> Media Format: 0
> Media Format: 8
> Media Format: 101
> Media Attribute (a): rtpmap:3 GSM/8000
Bingo...GS chokes with GSM...just disallow it in your sip.conf:
disallow=all
allow=alaw
allow=ulaw
> Media Attribute (a): rtpmap:0 PCMU/8000
> Media Attribute (a): rtpmap:8 PCMA/8000
> Media Attribute (a): rtpmap:101 telephone-event/8000
> Media Attribute (a): fmtp:101 0-16
>
> Don't see much else different in the packets.
>
> It might also be relevant that the FWD connection, which works
> successfully, is through a firewall with NAT.
>
> Still fishing... thanks for your attention - much appreciate not being
> alone here!
>
>
--
Andres
Network Admin
http://www.telesip.net
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