[Asterisk-Users] Zyxel 2000W (WI-FI) Problems

[Alexander Scheerschmidt]

Oh yeah, i forgot, do you hav installed the latest firmware ? If not,
download it and install.
My config (Zyxel phone):
 

SIP PROXY

  _____  

SIP URI	 sip:  @ 10.0.0.10 : 5060 	
SIP Server Address	   	
SIP Server Port	   	
Registrar Server Address	   	
Registrar Server Port	   	
Register Expiry Time(sec.)	   	
OPTIONS Interval Timer	   	
Session Expiry Time(sec.)	   	
Display Name	     	

  _____  

Authentication	 	

  _____  

Registrar Username	     	
Registrar Password	     	

  _____  

Registration Status	 Registered	
  _____  

 

PHONE SETTINGS

  _____  

Default Voice Codec	  G.729, 8k G.711u, 64k G.711a, 64k	
Speaking Volume(-14~14)	  -14 -13 -12 -11 -10 -9 -8 -7 -6 -5 -4 -3 -2 -1 0 1
2 3 4 5 6 7 8 9 10 11 12 13 14  	
Listening Volume(-14~14)	  -14 -13 -12 -11 -10 -9 -8 -7 -6 -5 -4 -3
-2 -1 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14  	
RTP Port	 	
Jitter Buffer	 Small  Medium  Large  	
Voice Frames per Packet	 Small  Medium  Large  	
DTMF Relay	  disable inband(RFC2833) outband	
DTMF Payload(0~127)	   	

	

	    	
 
Regards,
Alexander
 
 From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Thore
Sent: Monday, 09 May 2005 10:20
To: ASTERIKS
Subject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems


Hi!

 

Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I
answer the phone I am ringing.

It works fine if I call the 2000W from other phones.

 

I have tried many sip settings. I use this now:

[205]

type=friend

username=205

secret=passwd205

callerid="Zyxel" <205>

host=dynamic

context=local

nat=yes

canreinvite=no

disallow=all

allow=g729

dtmfmode=rfc2833

 

 

 

 

Sip debug:

 headers, 0 lines

Retransmitting #4 (NAT):

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rpo
rt=5060

From: <sip:205 at 60.64.250.253;user=phone>;tag=C8355813679C716AFCA

To: <sip:202 at 60.64.250.253>;tag=as3bcc72b4

Call-ID: 24472-D1B9-1FA6-8959-E629AA6722FB at 192.168.253.149

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:202 at 60.64.250.253>

Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1"

Content-Length: 0

 

 

 to 60.64.250.254:5060

Retransmitting #5 (NAT):

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP
192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rpo
rt=5060

From: <sip:205 at 60.64.250.253;user=phone>;tag=C8355813679C716AFCA

To: <sip:202 at 60.64.250.253>;tag=as3bcc72b4

Call-ID: 24472-D1B9-1FA6-8959-E629AA6722FB at 192.168.253.149

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: <sip:202 at 60.64.250.253>

Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1"

Content-Length: 0 

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