[Asterisk-Users] "Unable to find path from G729A to ULAW" on Sipphone.com

[Steven Sokol]

Barton/Tilghman,

Do I need g729 licenses for Asterisk?  I don't really need Asterisk to
be a party to the call (i.e. monitor the data stream) and the BT101
apparently includes a license for G.729.

Does anybody know if Sipphone/FWD recently changed their system to allow
only G.729 calls to traverse the 800 interface?

Thanks,

Steve

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Barton
Hodges
Sent: Tuesday, November 18, 2003 3:58 PM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] "Unable to find path from G729A to ULAW"
on Sipphone.com

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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