Sip softphone mobile client

[WaTiger]

Thank you for your help.
The static binding is off.
Like I mentioned, I just set everything (username/password/passcode) to extension number.
So I am sure inputting correct information to softphone.
I read several posts, it looks quite simple.
But I just can't make it work.
Any suggestions?

[Tech Electronics]

WaTiger,

If you turn off Authentication and just use the Passcode under the extension can you get it to work?

Thanks,

TE

[WaTiger]

If I turn off Authentication, the softphone will show 401 unauthorized.
Any other settings I need to change?
Or licence limitation?

[Tech Electronics]

WaTiger,

You will always get a 401 error initially as that is how the SIP Protocol stack works. The initial 401 error is to notify the device to use the nonce provided to encrypt its username and password.

If you could attach your SIP logs then we may be able to point to a more definitive answer.

Thanks,

TE

[WaTiger]

Hi, TE
Thank you for your time and patience.
Here is the log.
In MiVoice 250 turn off in-bound authentication.
In eyebeam input user name (extension number)& password (passcode)
Authorization user name keep blank
Domain is pbx ip
ebybeam debug log:
[18-06-27]00:31:59.459 | Info (more) RESIP | "Root System Cert Store opened" |
[18-06-27]00:31:59.460 | Info (more) RESIP | "22 certs loaded of type 1" |
[18-06-27]00:31:59.460 | Info (more) RESIP | "My System Cert Store opened" |
[18-06-27]00:31:59.460 | Info (more) CCM | "Creating transport... [URI:1000@192.168.2.9]" | sua::CSIPAccess::CTransportCreator::Start
[18-06-27]00:31:59.460 | Info (min) CCM | "Not using firewall SIP proxy. Reasons:   [URI:1000@192.168.2.9]" | sua::CSIPAccess::CTransportCreator::UseFirewallSIPProxy
[18-06-27]00:31:59.461 | Info (more) RESIP:TRANSPORT | "Creating UDP transport host= port=9296 ipv4=1" |
[18-06-27]00:31:59.461 | Info (more) CCM | "DSCP marking is already set correct for SIP or DNS signaling socket, where socket=1276, where transport type=3, where resip file is W:\HEAD\SUA\Library\sipfoundry\main\resip\stack\InternalTransport.cxx, where resip line # is 122, where DSCP value=0" | sua::CCMAfterSocketCreationFuncPtr
[18-06-27]00:31:59.461 | Info (more) RESIP:TRANSPORT | "Creating TCP transport host= port=9296 ipv4=1" |
[18-06-27]00:31:59.461 | Info (more) RESIP:TRANSPORT | "Creating TLS transport for domain  interface= port=9297" |
[18-06-27]00:31:59.462 | Warning (min) RESIP:DNS | "local hostname does not contain a domain part" |
[18-06-27]00:31:59.586 | Info (more) RESIP:DUM | "Got a DumFeatureMessage0418B4C8" |
[18-06-27]00:31:59.708 | Info (more) RESIP:DUM | "Got: SipResp: 403 tid=9c17c2201916d05f cseq=REGISTER contact=1000@192.168.2.9:5060 / 1 from(wire)" |
[18-06-27]00:31:59.708 | Info (min) CCM | "Response code to SIP request did not match any entry specified in retry-response-list. Response: 403[URI:1000@192.168.2.9]" | sua::CSIPRegistrationWatcher::OnSIPRegistrationDestroyed

[Tech Electronics]

WaTiger,

You are not getting a 401 response you are getting a 403 response.

[18-06-27]00:31:59.708 | Info (more) RESIP:DUM | "Got: SipResp: 403 tid=9c17c2201916d05f cseq=REGISTER contact=1000@192.168.2.9:5060 / 1 from(wire)" |
[18-06-27]00:31:59.708 | Info (min) CCM | "Response code to SIP request did not match any entry specified in retry-response-list. Response: 403[URI:1000@192.168.2.9]" | sua::CSIPRegistrationWatcher::OnSIPRegistrationDestroyed

So, with a 403 Error the server is getting your request properly, but is refusing to fulfill it because it doesn't like what it is getting; typically a bad username and/or password.

So, it looks as though you are trying to setup x1000 as a SIP extension on your MiVO-250 [192.168.2.9]. Did you change your SIP port to be something other than 5060 or maybe I am reading this wrong?

By the way Eyebeam is no longer a supported SIP device as it was discontinued by Counterpath. Have you tried the X-Lite or maybe another phone like the 3CX free SIP Phone; this is what I use?

Thanks,

TE

[jmorgan121]

Sessiontalk client worked for me. Haven’t figured out a way to transfer calls though without having to put the call on hold, intercom who I want to transfer to and tell them to wait a few seconds and reverse transfer, then hang up and retrieve the original call and put it back on hold so the reverse transfer can take place. Any thoughts TE?

[jmorgan121]

In fact, I don’t see where I can transfer or Park from the Phone Manager mobile client either. Surely I’m missing something

[WaTiger]

PBX sip port 5060 udp, local sip port random
I tried 3cx softphone and got same result
There is one Mitel sip phone in system and works fine
The whole system was configured by anther technician one year ago.
I am a recruit for Mitel devices, so I am not sure about the configuration.

[Tech Electronics]

WaTiger,

Alright since I don't know anything about the Eyebeam let's start with the MiVO-250 setup and make sure it isn't something setup wrong there.

1. Go to System > Devices and Feature Codes > Phones > 1000 > Passcode: Delete the current passcode if there are ##### there.
1a. Verify what your SIP Peer Group extension is going to be; it starts with P and can be found near the bottom.

2. Go to System > Devices and Feature Codes > SIP Peer Groups > SIP Phone Peer > P#### > Configuration

Under here we need to verify a couple of things before going on.

Do you have either the FQDN or the IP Address filled in. If so clear out the FQDN and put 255.255.255.255 in the IP Address.

3. Go to System > Devices and Feature Codes > SIP Peer Groups > SIP Phone Peer > P#### > Authentication

Set Inbound Authentication to No and then clear out the Inbound Authentication username and passcode. This will force it to use the Passcode under the phone for authentication; which you cleared out in step 1.

Also, if you have anything in the Outbound Authentication fields then clear those out as well.

Now, at this point your SIP Device requires no Password to be logged into. Within your SIP Device however you will still need to supply the Username to be 1000 so it will tell the MiVO-250 which device extension you want to be.

Let me know if that works for you.

Thanks,

TE

[WaTiger]

Hi, TE
I followed your instruction and create a new sip extension.
Actually same process as before.
Remove passcode and no authentication.
But still got same 403 error. Tried several software (xlite/3cx/eyebeam)

[Tech Electronics]

WaTiger,

Are you sure that you have licensing for this new SIP Device?

At this point there is no password required by the system so it is either you don't have a license for this device or the programming on the SIP Device is inaccurate.

Thanks,

TE

[WaTiger]

It should be license limitation.
So we have no choice.
Stick with Mitel phones
Thank you for your help!