Alcatel Unleashed

To all,

I made a couple of tests this week with different sip softPhones. The one that answered all my expectations is eyeBeam from CounterPath ( http://www.counterpath.com ).

I had a couple of issues with it, and the technical support (thank to Andre) answered my emails right away to give me a hand to have it working.

The SDP RFC is ambiguous at best about whether this is legal, in the main body of the RFC is says something along the lines of "attribute names that aren't registered (I forget the wording they use) should start with "X-". But the grammar in one of the appendix seems to explicitly forbid the "-" character in an a= line. So depending on how the Alcatel and other phone interpreted the RFC, it may be rejecting that attribute, even though the RFC also says unrecognized attributes should be ignored.

Since getting the manufacturers to follow that part of the RFC could be difficult, the most recent version (1.5.7) should allow you to omit the x-rtp-session-id by doing the following:
Dial ***7469, filter for "session", and set system:sdp:include_x_rtp_session_id to 0.

Once this is done, you will be able to get eyeBeam work with your SIP trunk perfectly. The features are excellent, and the GUI interface is great.

On the top of that, eyeBeam is compatible with MacOS X !

HIGHLY RECOMMENDED !

it worked, nice. Can you post the configuration of the extensions, the trunk group and the eyeBeam? Will be great

This is an example of SIP TRUNK CREATION without authentication and no external GW:
Type: T2
Remote Network: 4 (choose one that you do not use, 4 in my case)
Q931: ABC-F
T2 Specification: SIP

When you go down hierarchie one level, the IP compression type should be G711

Then when you go down hierarchie again, you should have at least 2 for the number of SIP access for your trunk.

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Go under Translator / Network Routing, and modify the REMOTE NETWORK that you used before (4 in our case):
Protocol: QSIG

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After that, go back to the main menu, then go down to the SIP menu ...

SIP PROXY:
Minimal Authentication Method:: SIP None

and that's it..

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User creation:
Directory number: 1111
Set Type:: External
URL Username:1111
SIP Password: 1111

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eyeBeam configuration
Display Name: Your name
User Name: Extension you created
Password: Password that you set in the extension
Authorization username: extension@password
Domain: IP address of the MAIN CPU

Check Register with domain and receive incoming calls[b/]
Then select Proxy and put the IP address of the MAIN CPU.

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That's it !

it worked. however how can you make an outgoing call?

I tried it too. Works fine also with DDI and outgoing calls

Yes, you can make outgoing call if you leave the default configuration for the Trunk COS.

X-Lite is definitely the best SIP client (IMHO).
5 points from 5!
User interface, configuration etc.
. Works fine with OXE. Did not test it with OXO yet.
Compared it with SIPPS from NERO, sipXphone and ExpressTalk.

We check X-Lite in 6.2 and it works but now test in release 7 and call failed with message "Bad Request".

Vad, this might be related to my first post.

do a

motortrace 2
killall motorsip
traced


and see if you have the error message that I talked about in the 1st post..[/b]

I am using Rel 7 in my lab system,Using X-lite I am able to resgister on network. Problem I face

1) I try call other extn on pbx the message is CALL FAILED:BAD REQUEST
2) If I call SIP Phone from other extn it rings I pick up the call message is CALL ESTABLISHED but on calling extn I still hear ring back tone.No communication.