On Mon, 2011-09-26 at 21:54 -0400, John A. Sullivan III wrote:
> 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
I see the difference in the uri= field of Authorization, however I can
not reproduce the problem here: my sflphone completes the field
correctly with sip:extension@ip
The SIP RFC says the extension@ is optional, though in this case it
should probably be here.
Also the authorization step succeeds anyway:
-> sflphone sends an invite
<- asterix refuses it with code 401 Unauthorized
-> sflphone sends an invite with Authorization field set
<- asterix accepts it, send a code 100 Trying
<- asterix establishes the connection by sending code 200 OK, with a
complete SDP file.
Can you send the complete log of sflphone daemon?
kill all sflphone processes and run sflphone daemon in a terminal with:
/usr/lib/sflphone/sflphoned -c -d
then start the client
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