Nokia E65 & Nokia E51 vs Asterisk

StormRage · November 27, 2008, 10:41am

Hi,

I’m running Asterisk 1.4.22 with libpri 1.4.7 and dahdi 2.0.0. Connected to the PSTN via a TDM400P.

Softphones = no problem.

Only hardphones I have is Nokia based, especially the E65 and E51.

When an E65 or E51 user registers with Asterisk, I receive the following in the CLI:-

-- Saved useragent "Nokia RM-208 3.0633.69.00" for peer Smee
-- Got SIP response 400 "Bad Request" back from 192.168.0.21

Most interesting, is that calls from the Nokias to Softphones work quite well.

Yet… the moment one tries to make a call from a softphone to these Nokias, the Nokia phones goes … half dead …

With half dead, I mean:-

1.) You can’t cycle your call listings i.e. missed calls, received calls etc
2.) You can’t open your phone book via the shorcut key
3.) When going to the MENU, once you select any feature, the phone takes *** nearly *** forever to “activate” that chosen feature.

Only solution, presently is to bounce the phone, though not very pleasant

Would it have been Microsoft Software, one would live with it, BUT this is Linux with Asterisk connecting to Symbian. The BETTER / BEST option

Am I missing something very trivial… or does the developers need to check this out ?

Thanks ppl.

david55 · November 27, 2008, 3:36pm

What was sent to provoke the response? (sip debug or sip history output)

StormRage · November 28, 2008, 5:35am

Oops

StormRage · November 28, 2008, 5:57am

Hi,

Thanks for enquiring.

I do get the following in the CLI:

chan_sip.c:12949 handle_response: Remote host can’t match request CANCEL to call ‘13fc410e1685741c5227f5ff3c5ba51d@192.168.0.17’. Giving up.

May I ask which commands (more specifically) you require ?

This happens the moment the Nokia phones connect (register) with the Asterisk Server.

StormRage · November 28, 2008, 5:59am

Hi,

Thanks for enquiring.

I do get the following in the CLI, when phoning from a soft client to the Nokia phones.

chan_sip.c:12949 handle_response: Remote host can’t match request CANCEL to call ‘13fc410e1685741c5227f5ff3c5ba51d@192.168.0.17’. Giving up.

May I ask which commands (more specifically) you require on the debug side of things?

The “BAD REQUEST” happens the moment the Nokia phones connect (register) with the Asterisk Server.

david55 · November 28, 2008, 11:10am

The output that you get from

in particular the message that Asterisk sent to which the status 400 reply was returned and the contents of the that reply.

Although you may need to obfuscate sensitive information, note that the problem might be the use of specific non-alphanumeric characters.

[quote]centos*CLI> help sip set debug ip
Usage: sip set debug {off|on|ip addr[:port]|peer peername}
Globally disables dumping of SIP packets,
or enables it either globally or for a (single)
IP address or registered peer.[/quote]

StormRage · December 1, 2008, 5:58am

Hi David,

And so we learn as we go along - Thanks

Herewith the debug output:-

<— SIP read from 192.168.0.20:5060 —>
REGISTER sip:company.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK0n9cjig89thc7ujh0ge7gu2;rport
From: sip:StormRage@company.com;tag=a8ncjigfmthc7ism0ge7
To: sip:StormRage@company.com
Contact: sip:StormRage@192.168.0.20;expires=3600
CSeq: 1265 REGISTER
Call-ID: w8H9OHMVoIcZ10gsu2fo4klDHVZSDs
User-Agent: Nokia RM-208 3.0633.69.00
Max-Forwards: 70
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.20 : 5060 (NAT)
werewolf*CLI>
<— Transmitting (no NAT) to 192.168.0.20:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK0n9cjig89thc7ujh0ge7gu2;received=192.168.0.20;rport=5060
From: sip:StormRage@company.com;tag=a8ncjigfmthc7ism0ge7
To: sip:StormRage@company.com
Call-ID: w8H9OHMVoIcZ10gsu2fo4klDHVZSDs
CSeq: 1265 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:StormRage@192.168.0.17
Content-Length: 0

<------------>
– Registered SIP ‘StormRage’ at 192.168.0.20 port 5060 expires 3600
– Saved useragent “Nokia RM-208 3.0633.69.00” for peer StormRage
werewolf*CLI>
<— Transmitting (no NAT) to 192.168.0.20:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK0n9cjig89thc7ujh0ge7gu2;received=192.168.0.20;rport=5060
From: sip:StormRage@company.com;tag=a8ncjigfmthc7ism0ge7
To: sip:StormRage@company.com;tag=as2bfd6aec
Call-ID: w8H9OHMVoIcZ10gsu2fo4klDHVZSDs
CSeq: 1265 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: sip:StormRage@192.168.0.20;expires=3600
Date: Mon, 01 Dec 2008 05:41:01 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘w8H9OHMVoIcZ10gsu2fo4klDHVZSDs’ in 32000 ms (Method: REGISTER)
Scheduling destruction of SIP dialog ‘560726301da679590b073c5b53c67ba7@192.168.0.17’ in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.0.20:5060:
NOTIFY sip:StormRage@192.168.0.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK36961875;rport
From: “asterisk” sip:asterisk@192.168.0.17;tag=as1f2caffe
To: sip:StormRage@192.168.0.20
Contact: sip:asterisk@192.168.0.17
Call-ID: 560726301da679590b073c5b53c67ba7@192.168.0.17
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93

Messages-Waiting: yes
Message-Account: sip:asterisk@192.168.0.17
Voice-Message: 9/0 (0/0)

werewolf*CLI>
<— SIP read from 192.168.0.20:5060 —>
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.0.17:5060;branch=z9hG4bK36961875;rport=5060;received=192.168.0.17
To: sip:StormRage@192.168.0.20;tag=lljsjilro5hc6i6n0hk6
From: “asterisk” sip:asterisk@192.168.0.17;tag=as1f2caffe
Call-ID: 560726301da679590b073c5b53c67ba7@192.168.0.17
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (7 headers 0 lines) —
– Got SIP response 400 “Bad Request” back from 192.168.0.20
Really destroying SIP dialog ‘560726301da679590b073c5b53c67ba7@192.168.0.17’ Method: NOTIFY
Really destroying SIP dialog ‘6cT9OF5EoId6JghzyEHuTNfriNNnCx’ Method: REGISTER
– Registered SIP ‘Liz’ at 192.168.0.29 port 42714 expires 3600
– Saved useragent “eyeBeam release 1011s stamp 41121” for peer Liz
Really destroying SIP dialog ‘w8H9OHMVoIcZ10gsu2fo4klDHVZSDs’ Method: REGISTER

Unfortunately I do not see “enough” yet to be able to depict the problem…

In addition, may I ask how one can change the <Contact: > references from USER@DOMAIN.COM ? This being as the moment one user phones another, it does not reflect the domain name, but rather the IP. (Just to keep things clean)

Thanks

StormRage · December 1, 2008, 6:00am