Mitel Forums - The Unofficial Source
Mitel Forums => MiVoice Office 250/Mitel 5000 => Topic started by: jmorgan121 on June 14, 2018, 09:04:03 PM
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Does anyone know of a sip softphone app that would function as a sip extension on the 250 without phone manager?
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Jmorgan121,
Any SIP softphone or hardware is capable of working on the MiVO-250 without the MiVoice Office Application Suite (phone manager).
The issue is that SIP devices on the system are limited to a small subset of features that both the SIP device and the MiVO-250 supports.
Also the MCS only allows the new 6900 series SIP devices access to the MiVO-250’s OAI feature set, and then not all of the features that traditional phones, digital and IP, have access to.
Hope that answers your question.
Thanks,
TE
Sent from my iPhone using Tapatalk
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Does anyone know of a sip softphone app that would function as a sip extension on the 250 without phone manager?
Any "generic" SIP softphone app usually works... Bria, X-Lite, MicroSIP, Zoiper, Linphone, or whatever else you want. Just remember that generic SIP phones are mostly functionally equivalent to single line phones and have limited functionality compared to Mitel specific devices or the Phone Manager softphone.
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We have a Phone Manager demo account. I have set up the Phone Manager mobile client as well as the 6940. I just can't get a generic sip client to register. I've tried SessionTalk and Zoiper. I enter my credentials and keep getting 404 not found.
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This will register directly to the MiVO250, phone manager has no part in this, make sure to set your credentials there. In general, I leave everything default and just set outbound credentials.
X-Lite is the defacto standard for this for testing if your having issues. That is literally what support will will have you try.
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I am not familiar with the 250 and am just curious. Is there any setting where you have to say the terminal is 3rd party?
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jmorgan121,
If you are getting a 404 error then the user doesn't exist or the domain is not correct. The 404 error typically comes from the server so in this case I would say that the client you setup is not sending the correct credentials to the MiVO-250.
Wait, you are not trying to get the SIP Device to register with the Phone Manager MCS are you?
bojo387,
No, there is nothing like that in the MiVO-250, but if you want the Phone Manager client to see the device properly then you have to tell it that the SIP Device is 3rd-party when you setup the credentials in it or it will register to the MiVO-250 and disconnect the SIP Device instead of monitor it.
Thanks,
TE
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bojo387,
No, there is nothing like that in the MiVO-250, but if you want the Phone Manager client to see the device properly then you have to tell it that the SIP Device is 3rd-party when you setup the credentials in it or it will register to the MiVO-250 and disconnect the SIP Device instead of monitor it.
Thanks,
TE
I appreciate the explanation, TE! Thanks! :)
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I'm not trying to register with the Phone Manager. I was just stating that I have successfully configured SIP extensions and authentication on the 250 for Phone Manager. The register request is making it to the ICP, I just keep getting an error. I've tried 2 different sip clients.
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.250:42155;received=192.168.2.250;branch=z9hG4bK-524287-1---ac8b894dc4931028;rport=42155
From: "Testing" <sip:Testing@192.168.2.200>;tag=bf6cf613
To: "Testing" <sip:Testing@192.168.2.200>;tag=Mitel-5000_769429219-4881
Call-ID: pXL6LLEXSMRBYD0YCzpDPA..
CSeq: 22 REGISTER
Contact: <sip:Testing@192.168.2.200:5060>
Retry-After: 4
Allow: NOTIFY,REGISTER,REFER,SUBSCRIBE,INFO,INVITE,ACK,OPTIONS,CANCEL,BYE
User-Agent: Mitel-5000-ICP-6.3.6.21
Content-Length: 0
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I just figured out if I set up authentication to use the extension number of the phone as the username and password, it registers. However, I get the A149 insecure password alarm.
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Only the username has to be the extension. It's working. Thanks everybody
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Just a note, if you are only using SIP inside your LAN (no port forward from public internet on port 5060) you can just disable the SIP Password alarm in system flags... If you are actually using it off-premise, then I would change the password to something more secure.
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Yeah. I changed the password. I'm using it through a VPN
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I have a different issue with the sip softphone (eyebeam).
In MiVioce 250, I created a sip phone and set in-bound authentication username / password (all are extension number for simple)
When the softphone tried to register, it got 403 - Forbidden.
Anything wrong I did?
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WarTiger,
Do you have static binding turned on? If you do then is the SIP soft-phones’ IP address correct?
Is the SIP soft-phone using the correct credentials?
Thanks,
TE
Sent from my iPhone using Tapatalk
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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?
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WaTiger,
If you turn off Authentication and just use the Passcode under the extension can you get it to work?
Thanks,
TE
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If I turn off Authentication, the softphone will show 401 unauthorized.
Any other settings I need to change?
Or licence limitation?
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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
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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
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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
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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?
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In fact, I don’t see where I can transfer or Park from the Phone Manager mobile client either. Surely I’m missing something
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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.
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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
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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)
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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
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It should be license limitation.
So we have no choice.
Stick with Mitel phones :(
Thank you for your help!