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Jim Gottlieb

2009-Jun-13 23:31 UTC

[asterisk-users] Polycom registration errors

I'm evaluating using Polycom phones for our call center and I've set  
up my first phone (a SoundPoint 560) to give it a try.

The phone is working and can successfully place and receive calls.   
But every minute, there's an error in the log file:

chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed
for
'192.168.200.99' - Username/auth name mismatch

Turning on SIP debug, it appears it's asterisk trying to register with  
the phone:

Using latest REGISTER request as basis request
Sending to 192.168.200.99 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.200.99:5060:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP  
192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99
From: "6193644850" <sip:6193644850 at
jtsd05>;tag=A1BB38FF-7161AAEA
To: <sip:6193644850 at jtsd05>;tag=as3d68239c
Call-ID: 20f907fe-db323389-f4569534 at 192.168.200.99
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

But then, the From: and To: lines seem to both show it from hostname  
jtsd05, though there's also the line saying it's going to  
192.168.200.99 (the phone).

I've played with all sorts of settings in sip.conf, but the messages  
persist.  Here's what I've got:

[hft0]
type=friend
username=hft0
secret=mysecret
context=outtrunk-office
host=192.168.200.99
disallow=all
allow=ulaw
dtmfmode=rfc2833
progressinband=no ;Polycom phones have trouble with the  
progressinband=never
callerid="HFT Booth 0 <(619) 364-4850>"
allowsubscribe=yes

And some of the Polycom phone config:
    reg reg.1.displayName="HFT0"
    reg.1.address="6193644850"
    reg.1.label="4850"
    reg.1.type="private"
    reg.1.lcs=""
    reg.1.csta=""
    reg.1.thirdPartyName=""
    reg.1.auth.userId="hft0"
    reg.1.auth.password="mysecret"
    reg.1.auth.optimizedInFailover=""
    reg.1.musicOnHold.uri=""
    reg.1.server.1.address="jtsd05"
    reg.1.server.1.port=""
    reg.1.server.1.transport="DNSnaptr"
    reg.1.server.2.transport="DNSnaptr"
    reg.1.server.1.expires=""
    reg.1.server.1.expires.overlap=""
    reg.1.server.1.register=""
    reg.1.server.1.retryTimeOut=""
    reg.1.server.1.retryMaxCount=""
    reg.1.server.1.expires.lineSeize=""
    reg.1.server.1.lcs=""
    reg.1.outboundProxy.address=""

Any ideas would be welcomed.  Thanks...

...Jim Gottlieb, San Diego, California

Jeff LaCoursiere

2009-Jun-14 00:19 UTC

head link

[asterisk-users] Polycom registration errors

On Sat, 13 Jun 2009, Jim Gottlieb wrote:
> I'm evaluating using Polycom phones for our call center and I've
set
> up my first phone (a SoundPoint 560) to give it a try.
>
> The phone is working and can successfully place and receive calls.
> But every minute, there's an error in the log file:
>
> chan_sip.c: Registration from '<sip:6193644850 at jtsd05>'
failed for
> '192.168.200.99' - Username/auth name mismatch
>
> Turning on SIP debug, it appears it's asterisk trying to register with
> the phone:
>
> Using latest REGISTER request as basis request
> Sending to 192.168.200.99 : 5060 (non-NAT)
> Transmitting (no NAT) to 192.168.200.99:5060:
> SIP/2.0 404 Not found
> Via: SIP/2.0/UDP
> 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99
> From: "6193644850" <sip:6193644850 at
jtsd05>;tag=A1BB38FF-7161AAEA
> To: <sip:6193644850 at jtsd05>;tag=as3d68239c
> Call-ID: 20f907fe-db323389-f4569534 at 192.168.200.99
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
This isn't asterisk registering with the phone - it is asterisk sending 
back the "404 Not Found".
>
> But then, the From: and To: lines seem to both show it from hostname
> jtsd05, though there's also the line saying it's going to
> 192.168.200.99 (the phone).
>
> I've played with all sorts of settings in sip.conf, but the messages
> persist.  Here's what I've got:
>
> [hft0]
> type=friend
> username=hft0
> secret=mysecret
> context=outtrunk-office
> host=192.168.200.99
Change the above to host=dynamic
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> progressinband=no ;Polycom phones have trouble with the
> progressinband=never
> callerid="HFT Booth 0 <(619) 364-4850>"
> allowsubscribe=yes
>
> And some of the Polycom phone config:
>    reg reg.1.displayName="HFT0"
>    reg.1.address="6193644850"
>    reg.1.label="4850"
>    reg.1.type="private"
>    reg.1.lcs=""
>    reg.1.csta=""
>    reg.1.thirdPartyName=""
>    reg.1.auth.userId="hft0"
>    reg.1.auth.password="mysecret"
>    reg.1.auth.optimizedInFailover=""
>    reg.1.musicOnHold.uri=""
>    reg.1.server.1.address="jtsd05"
Can the phone resolve this unqualified name?
>    reg.1.server.1.port=""
>    reg.1.server.1.transport="DNSnaptr"
>    reg.1.server.2.transport="DNSnaptr"
>    reg.1.server.1.expires=""
>    reg.1.server.1.expires.overlap=""
>    reg.1.server.1.register=""
>    reg.1.server.1.retryTimeOut=""
>    reg.1.server.1.retryMaxCount=""
>    reg.1.server.1.expires.lineSeize=""
>    reg.1.server.1.lcs=""
>    reg.1.outboundProxy.address=""
>
> Any ideas would be welcomed.  Thanks...
>
> ...Jim Gottlieb, San Diego, California
>
I think host=dynamic will fix you up.

j

Dave Fullerton

2009-Jun-15 21:04 UTC

head link

[asterisk-users] Polycom registration errors

Jim Gottlieb wrote:> I'm evaluating using Polycom phones for our call center and I've
set
> up my first phone (a SoundPoint 560) to give it a try.
> 
> The phone is working and can successfully place and receive calls.   
> But every minute, there's an error in the log file:
> 
> chan_sip.c: Registration from '<sip:6193644850 at jtsd05>'
failed for
> '192.168.200.99' - Username/auth name mismatch
> 
> Turning on SIP debug, it appears it's asterisk trying to register with
> the phone:
> 
> Using latest REGISTER request as basis request
> Sending to 192.168.200.99 : 5060 (non-NAT)
> Transmitting (no NAT) to 192.168.200.99:5060:
> SIP/2.0 404 Not found
> Via: SIP/2.0/UDP  
> 192.168.200.99;branch=z9hG4bKd732a3486F528693;received=192.168.200.99
> From: "6193644850" <sip:6193644850 at
jtsd05>;tag=A1BB38FF-7161AAEA
> To: <sip:6193644850 at jtsd05>;tag=as3d68239c
> Call-ID: 20f907fe-db323389-f4569534 at 192.168.200.99
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
> 
> But then, the From: and To: lines seem to both show it from hostname  
> jtsd05, though there's also the line saying it's going to  
> 192.168.200.99 (the phone).
> 
> I've played with all sorts of settings in sip.conf, but the messages  
> persist.  Here's what I've got:
> 
> [hft0]
> type=friend
> username=hft0
> secret=mysecret
> context=outtrunk-office
> host=192.168.200.99
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> progressinband=no ;Polycom phones have trouble with the  
> progressinband=never
> callerid="HFT Booth 0 <(619) 364-4850>"
> allowsubscribe=yes
> 
> And some of the Polycom phone config:
>     reg reg.1.displayName="HFT0"
>     reg.1.address="6193644850"
>     reg.1.label="4850"
>     reg.1.type="private"
>     reg.1.lcs=""
>     reg.1.csta=""
>     reg.1.thirdPartyName=""
>     reg.1.auth.userId="hft0"
>     reg.1.auth.password="mysecret"
>     reg.1.auth.optimizedInFailover=""
>     reg.1.musicOnHold.uri=""
>     reg.1.server.1.address="jtsd05"
>     reg.1.server.1.port=""
>     reg.1.server.1.transport="DNSnaptr"
>     reg.1.server.2.transport="DNSnaptr"
>     reg.1.server.1.expires=""
>     reg.1.server.1.expires.overlap=""
>     reg.1.server.1.register=""
>     reg.1.server.1.retryTimeOut=""
>     reg.1.server.1.retryMaxCount=""
>     reg.1.server.1.expires.lineSeize=""
>     reg.1.server.1.lcs=""
>     reg.1.outboundProxy.address=""
> 

Try changing reg.1.address to "hft0". My hunch is asterisk is looking
at
the from of "6193644850 at jtsd05" and going "huh? I don't
know a
6193644850 at jtsd05".

-Dave

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