ageet Forums

I couldn't get this to work..

What does under Authorisation in the User information section?

HI !

well i thought I was only who is suffering from this problem.... I also cannot get AGEphone work for me on HTC_TyTN ii and SIPPHONE account.

I am able to get sound from earpiece and can get connected to SIP server (not every time) and can check AUTOMATED service numbers! BUT.. When it comes time for real calling i can get connected to the second party too, at the same time when the calls start i HEAR a very strange sound (patpatpatpatpat) like some engine is running in my phone!!!!

Hope the team of AGEphone work on our problems related to use SIPPHONE. I would like to mention one thing that the TEAM is VERY SUPPORTING when ever you tell them your problem. I have experienced it personally.


Lets Hope for the Best !

Thanks in Advance to AGEphone team


Kipciaan

HI !

Please put your USERNAME/SIPPHONE ID in this !

In AGEphone Mobile 2, when your fill your USERNAME and you go to next entry it automatically copy your ID into Authorization and Display Name.

Its a Basic Feature of AGEphone Mobile 2


Best Regards

Kipciaan

Since you two had problems making Sipphone / Gizmo work here the complete settings as needed in the provider setup dialog:

Realm: proxy01.sipphone.com
SIP Domain: proxy01.sipphone.com
SIP Server: proxy01.sipphone.com (optional)
Registrar: proxy01.sipphone.com (optional)

User Name: your Gizmo username or number
Authorization: Gizmo username or number
Display Name: Gizmo username or number
Password: your Gizmo password

Be consistent about using your username or number. Mixing won't work.

I also tried calling from my mobile to the desktop or another phone and didn't experience any of the problems described. Raising the "Audio Buffer" under "3. Audio" might fix skipping in your conversations. You might also need to change a line in the "sipd.conf" under "My Documents \ My Phone Booth". The string is questions is

MediaTypes=3 0 8 101 which you should change to

Media Types=0 8 101

in order to have only G.711u / G.711a enabled. Uaually this step is not necessary as the codecs get negotiated automatically between client and server. Another possibility is to change the "Audio Packet Size" under "1. SIP Accounts" on page 3 of the provider options. Experiment with the setting and that strange "patpatpatpatpat" sound might go away.