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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barton Hodges Sent: Tuesday, November 18, 2003 3:58 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] "Unable to find path from G729A to ULAW" on Sipphone.com [EMAIL PROTECTED] 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:[EMAIL PROTECTED]/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/[EMAIL PROTECTED]|20|tr") in new stack > -- Called [EMAIL PROTECTED] > 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/[EMAIL PROTECTED]) ; SIPphone ;exten => _1800NXXXXXX,1,Macro(callerid-pstn) ;exten => _1800NXXXXXX,2,SetVar(SIP_CODEC=g729) ;exten => _1800NXXXXXX,3,Dial(SIP/[EMAIL PROTECTED]) I hope this helps, Barton _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
