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/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] -- 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/[EMAIL PROTECTED]") in new stack -- Called [EMAIL PROTECTED] -- 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: > [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