Unable to register phone on Zultys MX250 r8.0.3: SIP 400 Error

[IsaacSchneider]

Hello,

I just received a T46G running firmware version 28.71.0.70 and found it is sending malformed registration packets. I am trying to register to a Zultys MX250 running firmware r8.0.3. The Zultys sends back SIP/2.0 400 Bad Request. I know the account is good as I can register several different phones to the account without issue.

This is the SIP trace of the registration. One thing I noticed is the phone does not retry the registration the three time as setup in the phones config.

=================================================
REGISTER sip:192.168.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK666461973
From: ""199"" <sip:isaacschneider@192.168.1.250>;tag=1872100210
To: ""199"" <sip:isaacschneider@192.168.1.250>
Call-ID: 455264881@192.168.1.113
CSeq: 1 REGISTER
Contact: <sip:isaacschneider@192.168.1.113:5062>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink-T46G 28.71.0.70 28.1.0.128.0.0.0 001565457bb1 00:15:65:45:7b:b1
Expires: 3600
Mac: 001565457bb1
Content-Length: 0

SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK666461973
From: ""199"" <sip:isaacschneider@192.168.1.250>;tag=1872100210
To: ""199"" <sip:isaacschneider@192.168.1.250>
Call-ID: 455264881@192.168.1.113
CSeq: 1 REGISTER
Contact: <sip:isaacschneider@192.168.1.113:5062>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink-T46G 28.71.0.70 28.1.0.128.0.0.0 001565457bb1 00:15:65:45:7b:b1
Expires: 3600
Mac: 001565457bb1
Content-Length: 0
=================================================

Any help would be appreciated.

Thanks,
Isaac

[Yealink Support]

Hi Isaac,
1. Did you fill the Server Host in correct field? Our T46G can support two server host, you fill the SIP Server 1?
2. Could you provide PCAP trace, syslog(level 6) and config.bin file to us, so we can analyze?
Before you export the syslog, please set log level as 6, and reboot the phone, then click Start,and reproduce the issue, then click Stop,and export the trace, syslog, config,bin to us.
(About where to export these files, please refer to attached screenshot.)
These three files are very important for us, hope you can kindly understand.
Thanks.

BTW, if you think it's not suitable to upload your three files here, you can send the email to our email support@yealink.com

[IsaacSchneider]

(06-11-2013 11:06 AM)Yealink Support Wrote:  Hi Isaac,
1. Did you fill the Server Host in correct field? Our T46G can support two server host, you fill the SIP Server 1?

Yes I have the correct host name (ip address/domain)

(06-11-2013 11:06 AM)Yealink Support Wrote:  2. Could you provide PCAP trace, syslog(level 6) and config.bin file to us, so we can analyze?

Sure, I have attached my test configs, logs and pcap.

Thanks,
Isaac

[Yealink Support]

Thanks Isaac, let me submit to our engineer to check, any update will let you know ASAP.

[IsaacSchneider]

There are two things that stand out in the registration message. First the Max Forwards parameter on all of the phones I have seen appear just after the Via parameter. Second is the Contact line not giving all of the parameters most other phones are presenting.

I looked up the RFCs and they do not specifically state in which order most of the parameters must appear. However the RFC examples clearly show the Max Forwards parameter always after the Via parameter. This follows the behavior of other vendors sip ua.

As for the Contact line the RFC states it is optional for the extended parameters but is suggested/recommended to provide additional parameters. And again on other phones they all have additional parameters like Registration Timeout and additional unique login session ids, rport,.. etc.

What I did not check but is something the RFC requires is the unique session ids have to be unsigned 32bit numbers.

-Isaac

[Yealink Support]

Hi Isaac,
Our R&D are modifying this issue now, we will send you the new firmware once it's ready.
Thanks.

[Yealink Support]

Hi Isaac,
Please download the new firmware which fixed register issue on Zultys:
ftp://Cathy:6br6pcyl@ftp.yealink.com/fir...1.0.85.rom
After upgrade to this firmware, please reset to factory default.
Any test result please let us know.
Thanks.

[IsaacSchneider]

Hello,

The phone has successfully registered on a MX250. I have yet to test out the functionality as I am not physically near the phone. But initial programming looks good.

Thank you for the quick turn around, I know this will make a alot of people happy.

-Isaac

[IsaacSchneider]

I am curious, At first glance it looks like only the User-Agent field had changed between the versions. Can you give some release notes or is this a test build?

After a second to test the phone I found a problem. When dialing out the call would get to 100 Trying back from the pbx but then it would timeout. If this helps I have included a pcap of the call flow below.

From my past experience I tinkered a bit and changed two settings. First I changed the Local SIP port on Account1 to use 5060 (I have seen this with other SIP phones cause problems). Second I removed the G722 from the codec list on Account1. The Zultys doesn't support g722 natively. After that I was able to successfully place a call.

Even though the use of a port other than 5060 on the phone is legal according to RFCs. I have seen this cause problems on many other systems and routers with SIP-ALG. Can you set the default to port 5060 for all accounts (the current is 5062).

Also can you remove g722 from the default codec list. As most of the PBXs I have seen do not support that codec natively. I think if you are limiting the default codec list. The list should only contain g711 a/u only. I know this should work if you offer a list the to ends should just negotiate the list. I blame Zultys for this one but have seen the same behavior on other pbx systems.

Thanks,
Isaac

==========================================
Warning! Gibberish below
==========================================
1 SIP 192.168.1.113:5062 192.168.1.250:5060 Rx 6/19/2013 5:34:56 PM
INVITE sip:500@192.168.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1048703307
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>
Call-ID: 1478442744@192.168.1.113
CSeq: 1 INVITE
Contact: <sip:isaacschneider@192.168.1.113:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.71.0.85
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 308

v=0
o=- 20002 20002 IN IP4 192.168.1.113
s=SDP data
c=IN IP4 192.168.1.113
t=0 0
m=audio 11784 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

2 SIP 192.168.1.250:5060 192.168.1.113:5062 Tx 6/19/2013 5:34:56 PM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1048703307
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>;tag=1962-zultys--10359999761341_2581221965-1341
Call-ID: 1478442744@192.168.1.113
CSeq: 1 INVITE
Server: Zultys MX250 v8.0.3 build 1
Content-Length: 0" 11DHW-00-00001-2A2

3 SIP 192.168.1.250:5060 192.168.1.113:5062 Tx 6/19/2013 5:34:56 PM
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1048703307
WWW-Authenticate: Digest realm=""192.168.1.250"",nonce=""MX250:2581227489"",algorithm=MD5,qop=""auth"",opaque=""MX250:2581227489""
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>;tag=1962-zultys--10359999761341_2581221965-1341
Call-ID: 1478442744@192.168.1.113
CSeq: 1 INVITE
Contact: sip:500@192.168.1.250:5060
Server: Zultys MX250 v8.0.3 build 1
Content-Length: 0

4 SIP 192.168.1.113:5062 192.168.1.250:5060 Rx 6/19/2013 5:34:56 PM
ACK sip:500@192.168.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1048703307
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>;tag=1962-zultys--10359999761341_2581221965-1341
Call-ID: 1478442744@192.168.1.113
CSeq: 1 ACK
Content-Length: 0

5 SIP 192.168.1.113:5062 192.168.1.250:5060 Rx 6/19/2013 5:34:56 PM
INVITE sip:500@192.168.1.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1219870465
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>
Call-ID: 1478442744@192.168.1.113
CSeq: 2 INVITE
Contact: <sip:isaacschneider@192.168.1.113:5062>
Authorization: Digest username=""isaacschneider"", realm=""192.168.1.250"", nonce=""MX250:2581227489"", uri=""sip:500@192.168.1.250:5060"", response=""2e77b011ff72746e00e734c15b27470c"", algorithm=MD5, cnonce=""0a4f113b"", opaque=""MX250:2581227489"", qop=auth, nc=00000001
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.71.0.85
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 308

v=0
o=- 20002 20002 IN IP4 192.168.1.113
s=SDP data
c=IN IP4 192.168.1.113
t=0 0
m=audio 11784 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv

6 SIP 192.168.1.250:5060 192.168.1.113:5062 Tx 6/19/2013 5:34:56 PM
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.113:5062;branch=z9hG4bK1219870465
From: <sip:isaacschneider@192.168.1.250>;tag=1657902058
To: <sip:500@192.168.1.250>;tag=1963-zultys--10359999761341_2581348541-1341
Call-ID: 1478442744@192.168.1.113
CSeq: 2 INVITE
Server: Zultys MX250 v8.0.3 build 1
Content-Length: 0

[Yealink Support]

Hi Isaac,
1. In this new firmware, we fixed these issues:
1)Can't register to Zultys PBX, because T46G user agent is blank
2)T46G can't be recognized by Asterisk PBX
3)The SIP Server do not support some domain name
4)Don't have MWI

2. About 5060 port, sorry because this port is for IP direct call, so now we use port 5062, our product department will evaluate this again.

3. You can remove G722 codec by auto provisioning if BPX do not support this.
-------------------------------------
account.X.codec.6.enable = 0
account.X.codec.6.payload_type = G722
account.X.codec.6.priority = 4
account.X.codec.6.rtpmap = 9
--------------------------------------
Anyway thanks for your suggestion.