[Asterisk-Users] Cannot dial out with Polycom 501 after upgrade

Jim Rice jim at bydesignpublishing.com
Mon Apr 17 09:53:58 MST 2006


On Mon, 2006-04-17 at 10:45 -0400, Andrew Kohlsmith wrote:
> a sip debug on the asterisk console will give you a ton of data if it's 
> getting to the asterisk box...

*CLI> sip debug

... stuff scrolled off screen ...

m=audio 2230 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

--- (15 headers 11 lines)---
Using INVITE request as basis request -
f7b4c5b6-dad1e9f8-c6bca7a7 at 10.0.0.201
Sending to 10.0.0.201 : 5060 (non-NAT)
Found user '201jim'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.201:2230
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|
alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 6600546 in office (domain 10.0.0.1)
Reliably Transmitting (no NAT) to 10.0.0.201:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP
10.0.0.201;branch=z9hG4bKfaa55b2b1A2706FC;received=10.0.0.201
From: "201 JIM" <sip:201jim at 10.0.0.1>;tag=5B7B9FBA-441AF0A9
To: <sip:6600546 at 10.0.0.1;user=phone>;tag=as7bb06f68
Call-ID: f7b4c5b6-dad1e9f8-c6bca7a7 at 10.0.0.201
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6600546 at 10.0.0.1>
Content-Length: 0


---

<-- SIP read from 10.0.0.201:5060:
ACK sip:6600546 at 10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.201;branch=z9hG4bKfaa55b2b1A2706FC
From: "201 JIM" <sip:201jim at 10.0.0.1>;tag=5B7B9FBA-441AF0A9
To: <sip:6600546 at 10.0.0.1;user=phone>;tag=as7bb06f68
CSeq: 2 ACK
Call-ID: f7b4c5b6-dad1e9f8-c6bca7a7 at 10.0.0.201
Contact: <sip:201jim at 10.0.0.201>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.5.0043
Max-Forwards: 70
Content-Length: 0


--- (11 headers 0 lines)---
Destroying call 'f7b4c5b6-dad1e9f8-c6bca7a7 at 10.0.0.201'




-- 
Jim Rice
by Design Publishing
11626 N. Tracey Road
Hayden, Idaho  83835




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