Hi List,
I am trying to get my budget sip phone to work with asterisk, which in
turn is configured to work with NuFone. I can get the phone to ring my
home PSTN'ed phone but as soon as I pick up my home phone it hangs.
Here's what I get in the log:
Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to
transmit frame type 1, while native formats is 4 (read/write = 1/1)
Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to
transmit frame type 1, while native formats is 4 (read/write = 1/1)
Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to
transmit frame type 1, while native formats is 4 (read/write = 1/1)
-- IAX2/NuFone/2 is ringing
Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to
transmit frame type 1, while native formats is 4 (read/write = 1/1)
Nov 4 18:37:44 NOTICE[1191013296]: channel.c:1696 ast_set_write_format:
Unable to find a path from G723 to ULAW
-- IAX2/NuFone/2 stopped sounds
-- IAX2/NuFone/2 answered SIP/2000-7e6e
Nov 4 18:37:49 WARNING[1191013296]: channel.c:2135
ast_channel_make_compatible: No path to translate from SIP/2000-7e6e(4)
to IAX2/NuFone/2(1)
Nov 4 18:37:49 WARNING[1191013296]: app_dial.c:1023 dial_exec: Had to
drop call because I couldn't make SIP/2000-7e6e compatible with
IAX2/NuFone/2
-- Hungup 'IAX2/NuFone/2'
== Spawn extension (from-sip, 3000, 1) exited non-zero on
'SIP/2000-7e6e'
-- Registered to '69.73.19.178', who sees us as 80.12.162.250:1026
Any ideas of what I am missing? It would seem that the key is "Unable to
find a path from G723 to ULAW" but I don't know what it means :-)
Cheers,
Jean-Michel.
Jean-Michel Hiver wrote:> Hi List, > > I am trying to get my budget sip phone to work with asterisk, which in > turn is configured to work with NuFone. I can get the phone to ring my > home PSTN'ed phone but as soon as I pick up my home phone it hangs. > > Here's what I get in the log: > > Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to > transmit frame type 1, while native formats is 4 (read/write = 1/1) > Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to > transmit frame type 1, while native formats is 4 (read/write = 1/1) > Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to > transmit frame type 1, while native formats is 4 (read/write = 1/1) > -- IAX2/NuFone/2 is ringing > Nov 4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to > transmit frame type 1, while native formats is 4 (read/write = 1/1) > Nov 4 18:37:44 NOTICE[1191013296]: channel.c:1696 ast_set_write_format: > Unable to find a path from G723 to ULAW > -- IAX2/NuFone/2 stopped sounds > -- IAX2/NuFone/2 answered SIP/2000-7e6e > Nov 4 18:37:49 WARNING[1191013296]: channel.c:2135 > ast_channel_make_compatible: No path to translate from SIP/2000-7e6e(4) > to IAX2/NuFone/2(1) > Nov 4 18:37:49 WARNING[1191013296]: app_dial.c:1023 dial_exec: Had to > drop call because I couldn't make SIP/2000-7e6e compatible with > IAX2/NuFone/2 > -- Hungup 'IAX2/NuFone/2' > == Spawn extension (from-sip, 3000, 1) exited non-zero on 'SIP/2000-7e6e' > -- Registered to '69.73.19.178', who sees us as 80.12.162.250:1026 > > Any ideas of what I am missing? It would seem that the key is "Unable to > find a path from G723 to ULAW" but I don't know what it means :-) > > Cheers, > Jean-Michel.Jean-Michel, Asterisk does not really have support for G723. It can do passthrough, but it will not transcode it. NuFone does not support it. That is where your error is coming from. Make sure that you do this: iax.conf: disallow=all allow=ilbc allow=gsm allow=ulaw sip.conf: disallow=all allow=ilbc allow=gsm allow=ulaw I think that should do what you need. Please read up on codecs at http://www.voip-info.org/wiki-Asterisk+codecs -- Kristian Kielhofner
Jean-Michel Hiver wrote:> Any ideas of what I am missing? It would seem that the key is "Unable to > find a path from G723 to ULAW" but I don't know what it means :-)add to sip.conf: disallow=all allow=ulaw and use support@nufone.net next time. Jeremy McNamara
> iax.conf: > > disallow=all > allow=ilbc > allow=gsm > allow=ulaw > > sip.conf: > > disallow=all > allow=ilbc > allow=gsm > allow=ulaw > > I think that should do what you need. Please read up on codecs at > http://www.voip-info.org/wiki-Asterisk+codecsWow, it works! And it works great too. I am quite impressed really. I have a ping of 400+ ms to switch-1.nufone.net yet when I call myself the quality of the call is actually pretty good. Thanks for the pointers to the codecs page. Cheers, Jean-Michel.