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Is there a section to talk about Java ME? Apps, etc. (7)
to General by teroyk - 15 hrs, 58 mins ago - more...
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2007-12-02
, 11:29
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Posts: 15 |
Thanked: 6 times |
Joined on Nov 2007
@ england
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#22
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in general im finding then nat traversal using default setting in sofia are working fine for me with two accounts (voipuser.org and sipgate.co.uk) but i've also found that sip domain diailing fails (sip-id@domain.com) and hope that this is resolved in the future as it's one of the ways in which i call friends on different sip providers even when they don't have reciprocal arrangements.
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2007-12-02
, 12:37
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Posts: 1,648 |
Thanked: 2,122 times |
Joined on Mar 2007
@ UNKLE's Never Never Land
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#23
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Originally Posted by rheve
It seems that this bug affects other providers too (www.voipdiscount.com and www.voipbuster.com have been reported).
I am able to "register" with my SIP provider (Free, french ISP), but no way to get outgoing/incoming call working. As far as I know it is a known issue, but not solved yet (https://bugs.maemo.org/show_bug.cgi?id=1699)
please vote in bugzilla if this concerns you.
Chris
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2007-12-02
, 18:47
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Posts: 465 |
Thanked: 149 times |
Joined on Oct 2007
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#24
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Originally Posted by badgerbalti
SIP URIS work fine here, here's two that work for me:
but i've also found that sip domain diailing fails (sip-id@domain.com) and hope that this is resolved in the future as it's one of the ways in which i call friends on different sip providers even when they don't have reciprocal arrangements.
1234@loligo.com
613@fwd.pulver.com
Do you have any luck with those?
I did have a problem initially, but that was due to my Asterisk server not being configured to dial anything other than regular phone numbers, or local extensions.
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2007-12-05
, 09:31
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Posts: 35 |
Thanked: 18 times |
Joined on Feb 2007
@ France
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#25
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I had a similar problem with using sofia and gizmoproject SIP service.
To fix it I forced the Transport (Accounts / Edit / Advanced) to use just UDP instead of Auto.
Hope that helps.
p.s. Has anyone got a way to find / fix the RTP port ?
Last edited by cybergypsy; 2007-12-05 at 10:05.
To fix it I forced the Transport (Accounts / Edit / Advanced) to use just UDP instead of Auto.
Hope that helps.
p.s. Has anyone got a way to find / fix the RTP port ?
Last edited by cybergypsy; 2007-12-05 at 10:05.
I am setting up N800 instead one of two computers. Computers are running soft phone sip clients (X-LITE). Accounts serviced by outside VOIP providers. I can call from one account to another (2 computers setup scenario) by using PSTN dialing 1(XXX)XXX-XXXX and VOIP dialing XXXXXXXXXX@PROVIDER.COM/NET. After setting up N800 replacing one of computers I am receiving calls to N800 and can dial PSTN number to my second account. However attempting to dial VOIP number format failing. Calling log of my account indicated that failing calls was decoded and directed as calls to New Zealand instead of US calls. I suspect that N800 2008 SIP client doesn't send VOIP domain info and system can't properly decode destination.
Any thoughts?
Another problem with dialer - it is disabled after sending phone number until hung up. In siltation when caller needs to call phone extension no dialer available.
Can you guys confirm problems or show errors on my side?