Summary:ASTERISK-00733: Grandstream SIP phones drop digits, set to SIP DTMF Mode INFO
Reporter:ltropiano (ltropiano)Labels:
Date Opened:2004-01-04 13:02:00.000-0600Date Closed:2011-06-07 14:05:05
Versions:Frequency of
Description:As of this build, GS phones set to INFO DTMF drop digits when dialing out...

It's random which digits too... our Cisco phones work just fine.  The Dialplan hasn't changed.  

I dialed "96988647" ...

   -- Executing SetVar("SIP/50-13c0", "MYDIGIT=50") in new stack
   -- Executing GotoIf("SIP/50-13c0", "1?4:3") in new stack
   -- Goto (default,969887,4)
   -- Executing GotoIf("SIP/50-13c0", "0?5:6") in new stack
   -- Goto (default,969887,6)
   -- Executing SetCallerID("SIP/50-13c0", "Rocksteady <512-427-1350>") in new stack
   -- Executing Macro("SIP/50-13c0", "sbc-outdial|Zap/g1|69887") in new stack
   -- Executing AGI("SIP/50-13c0", "call_log.agi|s") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
   -- AGI Script call_log.agi completed, returning 0
   -- Executing Dial("SIP/50-13c0", "Zap/g1/69887") in new stack
   -- Called g1/69887
   -- Hungup 'Zap/1-1'
Comments:By: ltropiano (ltropiano) 2004-01-04 13:06:36.000-0600

Sorry this should have been selected as "bug" not feature.  My oops.

By: Brian West (bkw918) 2004-01-04 14:37:02.000-0600

Show us your macro and extensions.conf entry.

By: ltropiano (ltropiano) 2004-01-04 14:46:41.000-0600

exten => s,1,AGI(call_log.agi,${EXTEN})
exten => s,2,Dial(${ARG1}/${ARG2})
exten => s,3,Congestion
exten => s,103,Playback(ss-noservice)
exten => s,104,Congestion

exten => _91900X.,1,Congestion
exten => _991900X.,1,Congestion
exten => _9976XXXX,1,Congestion
exten => _991NXX976X.,1,Congestion
exten => _9.,1,SetVar(MYDIGIT=${CALLERIDNUM})
exten => _9.,2,GotoIf($[${MYDIGIT} < 99]?4:3)
exten => _9.,3,SetVar(MYDIGIT=00)
exten => _9.,4,GotoIf($[${MYDIGIT} = 0000000000]?5:6)
exten => _9.,5,SetVar(MYDIGIT=00)
exten => _9.,6,SetCallerID(Rocksteady <512-427-13${MYDIGIT}>)
exten => _9.,7,Macro(sbc-outdial,${PRITRUNK},${EXTEN:1})

(This was working a previous release... fyi, with the GS phones).  The Cisco phones continue to work just fine.

From sip.conf:

callerid="Lenny Tropiano @ Home" <50>

By: Brian West (bkw918) 2004-01-04 14:49:25.000-0600

This isn't a bug in asterisk but those crappy grandstream phones.

By: ltropiano (ltropiano) 2004-01-04 14:51:34.000-0600

Very well could be.  Although since it was working and stop, and I hadn't changed firmware revs, I could only assume it was what I did change (CVS revs in Asterisk).   I bought 5 GS phones and 25 Cisco 7940/60s... and stopped buying GS phones.  Quality is just not there.

By: jrollyson (jrollyson) 2004-01-09 01:03:50.000-0600

Grandstream phones will drop digits if dialed too quickly. There's nothing Asterisk can do about digits it never got.

By: Brian West (bkw918) 2004-01-09 11:21:15.000-0600

This is a Grandstream Issue.