Nortel 1535 all incoming calls fail

[bjefferys]

I can make outgoing calls. I am on 2.76 firmware. Incoming calls fail locally or externally. I get a sip 501 Method Not Implemented error. Behind NAT or internal. Same issue. See log. Looks like PING not implemented.

<--- SIP read from xxx.xxx.xxx.xxx:1025 --->
PING sip:pbx.xxxxxx.com:5060 SIP/2.0
From: <sip:[email protected]>;tag=3023a0-9001a8c0-13c4-45026-a05-32fbc7cf-a05
To: <sip:[email protected]>
Call-ID: [EMAIL="[email protected]"][email protected][/EMAIL]
CSeq: 18 PING
Via: SIP/2.0/UDP 192.168.1.144:5060;rport;branch=z9hG4bK-a05-27242a-26126475
Proxy-Require: com.nortelnetworks.firewall
Contact: sip:[email protected]
Max-Forwards: 70
Supported: replaces
User-Agent: Nortel IP Phone 1535 (0.2.76.0305)
Content-Length: 0
 
<------------->
--- (12 headers 0 lines) ---
Sending to 76.188.103.25 : 1025 (NAT)
<--- Transmitting (NAT) to xxx.xxx.xxx.xxx:1025 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 192.168.1.144:5060;branch=z9hG4bK-a05-27242a-26126475;received=76.xxx.xxx.xxx;rport=1025
From: <sip:[email protected]>;tag=3023a0-9001a8c0-13c4-45026-a05-32fbc7cf-a05
To: <sip:[email protected]>;tag=as41b1982a
Call-ID: [EMAIL="[email protected]"][email protected][/EMAIL]
CSeq: 18 PING
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
 
<------------>
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:1025:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK2eff288c;rport
From: "Unknown" <sip:[email protected]>;tag=as4491fb23
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [EMAIL="[email protected]"][email protected][/EMAIL]CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 13 Dec 2010 17:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
 
---
<--- SIP read from xxx.xxx.xxx.xxx:1025 --->
SIP/2.0 200 OK
From: "Unknown"<sip:[email protected]>;tag=as4491fb23
To: <sip:[email protected]:5060>;tag=302540-9001a8c0-13c4-45026-a19-2105c3eb-a19
Call-ID: [EMAIL="[email protected]"][email protected][/EMAIL]CSeq: 102 OPTIONS
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport=5060;branch=z9hG4bK2eff288c
Supported: replaces
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
User-Agent: Nortel IP Phone 1535 (0.2.76.0305)
Content-Type: application/sdp
Content-Length: 626
v=0
o=LGEIPP 0 0 IN IP4 xxx.xxx.xxx.xxxs=SIP Call
c=IN IP4 xxx.xxx.xxx.xxxt=0 0
m=audio 19000 RTP/AVP 0 8 18 4 95 99
c=IN IP4 76.188.103.25
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:95 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
m=video 19001 RTP/AVP 34 98 102
c=IN IP4 76.188.103.25
b=TIAS:186000
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1
a=rtpmap:98 H264/90000
a=rtpmap:102 H264/90000
a=fmtp:98 packetization-mode=0
a=fmtp:102 packetization-mode=1
a=framerate:15.0
a=maxprate:15.0
a=sendrecv
<------------->
--- (11 headers 29 lines) ---
Really destroying SIP dialog '[email protected] Method: OPTIONS

 

[tshif]

Have you enabled Allow Anonymous Inbound SIP Calls?

 

[bjefferys]

Yeah. Anon is enabled. I have to test again. This might only be happening on wifi. Will return with test results.

 

[Astrosmurfer]

It is because you have Allow Anonymous SIP Calls enabled that you can make "outgoing calls". More accurately calls from the 1535 to Asterisk extensions. In this configuration, Asterisk does not require registration/authentication for calls from the 1535 to reach any Asterisk destination.

This is definitely not my configuration of choice but, I do understand that some people may wish to allow Anonymous SIP so as to accept inbound SIP URI calls.

However, without registration, Asterisk has no way of finding the 1535 so calls meant to terminate at the 1535 will fail.

It sounds like you are not properly registering the 1535 to Asterisk. It is also possible for bogus NAT configuration to cause asymmetric call or audio problems. I'd recommend starting with an internal 1535 extension until you are successful.

What do you get for extension 704 with the sip show peers command?

In any case, the ping not implemented stuff will not impede call progress.

Edit: Wait. Am I reading this right? You are calling from 704 to 704?

 

[blanchae]

I usually use *65 (speak extension), *43 (echo test) or *60 (speak time) for testing from one phone. If they work then you are registered properly and don't have a NAT issue. Which by the way, do you have NAT=yes or no for the phone extension configuration in FreePBX?

 

[bjefferys]

Factory reset

I factory reset the phone and set it back up with the web provisioning. All is good now.

BTW, I do have NAT set as yes and could call all feature codes, extensions, and trunks. sip show peer 704 showed ok before I reset the phone. Just could not receive incoming calls. Something got fargled in the phone. Thanks for all your responses. I love this forum.

EDIT: I was calling ext 704 from my softphone (ext 706).

 

[blanchae]

I've seen some strange things happen if NAT=yes on a local phone (should be no) and strange things if NAT=no on a remote phone (should be yes). Usually one way audio or can call out but not in or the phone will register but can't dial out.