On Mon, 2011-09-26 at 17:57 -0400, Rafaël Carré wrote:
> On Mon, 2011-09-26 at 16:54 -0400, John A. Sullivan III wrote:
> > On Tue, 2011-09-13 at 01:42 -0400, John A. Sullivan III wrote:
> > > On Tue, 2011-09-13 at 00:56 -0400, John A. Sullivan III wrote:
> > > > Hmm . . . now that I finally have my Debian Squeeze SFLPhone 0.9.13
> > > > installation and have successfully registered to Asterisk, I am unable
> > > > to place a call. All the key strokes are misinterpreted and every call
> > > > says that is not a valid extension. I would not think this is a DTMF
> > > > issue but I did set it to both SIP and RTP.
> > > >
> > > > When I dial from a different phone and observe the Asterisk console, I
> > > > see Execute and the extension I've dialed. When SFLPhone dials, I see
> > > > Execut and just an "s". I see no information on stdout or stderr for
> > > > SFLPhone and there does not seem to be much logging. Any idea what I've
> > > > done wrong? Thanks - John
> > > <snip>
> > > I backleveled to 0.9.9 and have the same problem. I did not have this
> > > problem in 0.9.8. Thanks - John
> > <snip>
> > With Rafael's help, I was able to build debs for 0.9.14 for Squeeze.
> > However, I still have the same problem. The digits are not being sent
> > to our Asterisk PBX. For example, we have an echo test at extension
> > 213. Here is what happens when I dial 213 (either by number or by
> > clicking on the phone pad in the application):
> >
> > == Extension Changed 312[a100] new state InUse for Notify User othersip
> > -- Executing [s@a100:1] Answer("SIP/mysip-0000021c", "0.5") in new stack
> > -- Executing [s@a100:2] Playback("SIP/mysip-0000021c", "im-sorry") in
> > new stack
> > -- <SIP/mysip-0000021c> Playing 'im-sorry.slin' (language 'en')
> > [Sep 26 15:40:57] NOTICE[9967]: channel.c:3137 __ast_read: Dropping
> > incompatible voice frame on SIP/mysip-0000021c of format g722 since our
> > native format has changed to 0x200 (speex)
> > -- Executing [s@a100:3] Wait("SIP/mysip-0000021c", "0.0.5") in new stack
> > -- Executing [s@a100:4] Playback("SIP/mysip-0000021c",
> > "you-dialed-wrong-number") in new stack
> > -- <SIP/mysip-0000021c> Playing 'you-dialed-wrong-number.slin'
> > (language 'en')
> > -- Executing [s@a100:5] Wait("SIP/mysip-0000021c", "0.4") in new stack
> > -- Executing [s@a100:6] Playback("SIP/mysip-0000021c", "vm-goodbye") in
> > new stack
> >
> > Notice that it has dialed "s", i.e., it has fallen through the dial
> > plan. I have tried DTMF via RTP (as we normally do) and via SIP. I did
> > notice the g722 error which I thought was strange because the account is
> > set to use speex first. So, I thought there might be a bug where it was
> > using g722 anyway and disabled g722. Then I got:
> >
> > == Extension Changed 312[a100] new state InUse for Notify User othersip
> > -- Executing [s@a100:1] Answer("SIP/mysip-00000222", "0.5") in new stack
> > -- Executing [s@a100:2] Playback("SIP/mysip-00000222", "im-sorry") in
> > new stack
> > -- <SIP/mysip-00000222> Playing 'im-sorry.slin' (language 'en')
> > [Sep 26 15:44:24] NOTICE[9976]: channel.c:3137 __ast_read: Dropping
> > incompatible voice frame on SIP/mysip-00000222 of format ulaw since our
> > native format has changed to 0x200 (speex)
> > -- Executing [s@a100:3] Wait("SIP/mysip-00000222", "0.0.5") in new stack
> > -- Executing [s@a100:4] Playback("SIP/mysip-00000222",
> > "you-dialed-wrong-number") in new stack
> >
> > So I disabled everything but speex but still have the same problem of
> > not passing the number dialed and falling through the dial plan:
> >
> > == Extension Changed 312[a100] new state InUse for Notify User othersip
> > -- Executing [s@a100:1] Answer("SIP/mysip-00000223", "0.5") in new stack
> > -- Executing [s@a100:2] Playback("SIP/mysip-00000223", "im-sorry") in
> > new stack
> > -- <SIP/mysip-00000223> Playing 'im-sorry.slin' (language 'en')
> > -- Executing [s@a100:3] Wait("SIP/mysip-00000223", "0.0.5") in new stack
> > -- Executing [s@a100:4] Playback("SIP/mysip-00000223",
> > "you-dialed-wrong-number") in new stack
> > -- <SIP/mysip-00000223> Playing 'you-dialed-wrong-number.slin'
> > (language 'en')
> > == Spawn extension (a100, s, 4) exited non-zero on 'SIP/mysip-00000223'
> > -- Executing [h@a100:1] Hangup("SIP/mysip-00000223", "") in new stack
> > == Spawn extension (a100, h, 1) exited non-zero on 'SIP/mysip-00000223'
> >
> > This is using the very latest git - pulled today just before building.
> > What do we do next? Thanks - John
>
> Try a network capture on the computer running sflphone, with wireshark
>
Hello, Rafael. I think I see it. I will send you the traces privately
and securely via our PassIT service. However, in comparing a successful
Snom phone with SFLPhone, when the Snom sends the authorization in the
final INVITE, it sets uri=sip:213@<ip address> however SFLPhone sets it
to uri=sip:<ip address>. In the Snom trace I will send, it is packet
11. In the two SFLPhone traces I will send, they are packet 4. Thanks
- John
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