[Asterisk-Users] "Unable to find path from G729A to ULAW" on
Sipphone.com
Miguel Cavazos
miguel at cavazos.com.mx
Tue Nov 18 09:20:26 MST 2003
i followed what you said didint work heres what console says i cant do
the 1800 call anyway
-- Executing Macro("SIP/101-8376", "callerid-pstn") in new stack
-- Executing SetVar("SIP/101-8376", "SIP_CODEC=g729") in new stack
-- Executing Dial("SIP/101-8376", "SIP/*18006927753 at fwd") in new
stack
-- Called *18006927753 at fwd
-- SIP/fwd-2e46 is making progress passing it to SIP/101-8376
-- SIP/fwd-2e46 answered SIP/101-8376
== Spawn extension (asterisk, 18006927753, 3) exited non-zero on
'SIP/101-8376'
-- Executing Macro("SIP/101-c43c", "callerid-pstn") in new stack
-- Executing SetVar("SIP/101-c43c", "SIP_CODEC=g729") in new stack
-- Executing Dial("SIP/101-c43c", "SIP/*18006927753 at fwd") in new
stack
-- Called *18006927753 at fwd
-- SIP/fwd-bc38 is making progress passing it to SIP/101-c43c
-- SIP/fwd-bc38 answered SIP/101-c43c
== Spawn extension (asterisk, 18006927753, 3) exited non-zero on
'SIP/101-c43c'
On Tue, 2003-11-18 at 21:58, Barton Hodges wrote:
> asterisk-users-admin at lists.digium.com wrote:
> > I seem to be having a problem with transcoding and/or agreeing on a
> > valid codec. I am running a new image pulled from CVS at 1:30 PM
> CST.
> > The issue occurs when I try to make a call to a toll-free number
> over
> > sipphone.com.
> >
> > Here's what I see in the console:
> >
> > NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
> > Unable to find a path from G729A to ULAW
> > NOTICE[1259545280]: File channel.c, Line 1448
> (ast_set_write_format):
> > Unable to find a path from ULAW to G729A
> >
> > Before somebody tells me "UTFG", I ALREADY HAVE. Somebody else had
> a
> > similar issue last week and there was no real resolution posted. So
> > here it is again. I have all of the codecs that I support
> > enabled in my
> > sip.conf. Here is the relevant section:
> >
> > ;
> > ; SIP Configuration for Asterisk
> > ;
> > [general]
> > port = 5060 ; Port to bind to
> > bindaddr = 0.0.0.0 ; Address to bind to
> > context = default ; Default for incoming calls
> > srvlookup = yes ; Enable SRV lookups on outbound calls
> > pedantic = yes ; Enable slow, pedantic checking for
> > Pingtel ;tos=lowdelay
> > ;tos=184
> > maxexpirey=3600 ; Max length of incoming registration we
> allow
> > defaultexpirey=120 ; Default length of incoming/outoing
> > registration ;notifymimetype=text/plain ; Allow overriding of
> > mime type in NOTIFY ;videosupport=yes ; Turn on
> support
> > for SIP video disallow=all ; Disallow all codecs
> > allow=ulaw ; Allow codecs in order of
> preference
> > allow=alaw ; Allow codecs in order of
> preference
> > allow=gsm allow=ilbc
> >
> > register => 17476692375:[MYSECRET]@sipphone.com/1101
> >
> > [sipphone]
> > type=peer
> > username=17476692375
> > secret=[MYSECRET]
> > host=proxy01.sipphone.com
> > fromuser=SteveSokol
> > fromdomain=sipphone.com
> > canreinvite=no
> >
> > ; ==END OF SIP.CONF FILE===
> >
> > The issue occurs whenever any calls that route over the sipphone
> peer
> > are made to a toll-free number. The calling phone (either my GS100
> or
> > my X-LITE softphone) rings two or three times then gives me
> > busy. Here
> > is the entire debug output:
> >
> > -- Executing Dial("SIP/1101-1f83",
> > "SIP/18884510851 at sipphone.com|20|tr") in new stack
> > -- Called 18884510851 at sipphone.com
> > NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format):
> > Unable to find a path from G729A to ULAW
> > NOTICE[1234379840]: File channel.c, Line 1448
> (ast_set_write_format):
> > Unable to find a path from ULAW to G729A
> > -- SIP/sipphone.com-e7b3 is making progress passing it to
> > SIP/1101-1f83
> > -- SIP/sipphone.com-e7b3 answered SIP/1101-1f83
> > -- Attempting native bridge of SIP/1101-1f83 and
> > SIP/sipphone.com-e7b3 NOTICE[1242768320]: File channel.c, Line 1478
> > (ast_set_read_format): Unable to find a path from G729A to ULAW
> > NOTICE[1242768320]: File channel.c, Line 1448
> (ast_set_write_format):
> > Unable to find a path from ULAW to G729A
> > WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked
> to
> > transmit frame type 4, while native formats is 256 (read/write =
> 4/4)
> > == Spawn extension (default, 918884510851, 1) exited non-zero on
> > 'SIP/1101-1f83'
> >
> > The problem does NOT occur when I call another sipphone.com user
> (i.e.
> > GS100 -> Asterisk -> Sipphone -> GS100). Those calls go through
> just
> > fine. The toll free calls were working last week. Is it me, or is
> > it Sipphone.com?
> >
> > Any suggestions would be greatly appreciated.
> >
> > Steve
>
> I've been having the same types of problems (I'm probably the guy
> you're referring to who had the same problems last week). This is the
> solution I have found to work reliably so far.
>
> Configure the Grandstream BT101 with the following codecs, in the
> following order:
> choice 1: G.729A/B (g729)
> choice 2: PCMU (ulaw)
> choice 3: PCMA (alaw)
> choice 4: G.729A/B (g729)
> choice 5: PCMU (ulaw)
> choice 6: PCMA (alaw)
>
> Configure the codecs in sip.conf like this:
> disallow=all
> allow=all
> allow=ulaw
> allow=alaw
> allow=g729
>
> Configure the entry in extensions.conf to use a certain codec when
> necessary (I've found it necessary only when calling through the 800
> gateway provided to both FWD and SIPphone):
> ; FWD
> exten => _1800NXXXXXX,1,Macro(callerid-pstn)
> exten => _1800NXXXXXX,2,SetVar(SIP_CODEC=g729)
> exten => _1800NXXXXXX,3,Dial(SIP/*${EXTEN}@fwd)
> ; SIPphone
> ;exten => _1800NXXXXXX,1,Macro(callerid-pstn)
> ;exten => _1800NXXXXXX,2,SetVar(SIP_CODEC=g729)
> ;exten => _1800NXXXXXX,3,Dial(SIP/*${EXTEN}@sipphone)
>
> I hope this helps,
>
> Barton
>
>
>
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