[SR-Users] tel2sip is not working correctly

[Jyoti Bansal]

Hello all,
I am using kamailio code to establish calls with the real ue's. Our calls
are getting established with an issue , whenever invite is send from A
party to B party with teluri
that tel uri is converted to sip uri. While converting to sip uri some
extra bytes are getting added after user=phone and the call ends.
Can someone help us on this issue of getting some extra bytes added in the
sip uri?  for example :-
sip:405874112224411 at ims.mnc123.mcc456.3gppnetwork.org;user=phone��

Thanks,

*Jyoti Bansal*

Software Engineer .

Great Software Laboratory | www.gslab.com

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