hello! i have a problem with my asterisk configuration.I am using AsterisNow with Freepbx 2.11 and would like to configure IAX2 trunking between two servers in the same LAN.
I followed an article found on the net but i am still getting this message: “All lines are busy”.If someone could help me i will be happy 
What article did you follow. Post your config. Post some logs.
it is writen by Sylvain Thérien but i am unable to paste it here!
thanks but even after using this link i have the same message! “all lines are busy”
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here are some logs with the asterisk console when i try to make a call :
– Executing [[email protected]:1] Macro(“SIP/611-00000005”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [[email protected]:1]
Set(“SIP/611-00000005”, “TOUCH_MONITOR=1409812564.5”) in new stack
– Executing [[email protected]:2] Set(“SIP/611-00000005”, “AMPUSER=611”) in new stack
– Executing [[email protected]:3]
GotoIf(“SIP/611-00000005”, “0?report”) in new stack
– Executing [[email protected]:4] ExecIf(“SIP/611-00000005”, “1?Set(REALCALLERIDNUM=611)”) in new stack
– Executing [[email protected]:5]
Set(“SIP/611-00000005”, “AMPUSER=611”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/611-00000005”, “0?limit”) in new stack
– Executing [[email protected]:7] Set(“SIP/611-00000005”, “AMPUSERCIDNAME=karim”) in new stack
– Executing [[email protected]:8]
GotoIf(“SIP/611-00000005”, “0?report”) in new stack
– Executing [[email protected]:9] Set(“SIP/611-00000005”, “AMPUSERCID=611”) in new stack
– Executing [[email protected]:10] Set(“SIP/611-00000005”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [[email protected]:11] Set(“SIP/611-00000005”, “CALLERID(all)=“karim” <611>”) in new stack
– Executing [[email protected]:12]
GotoIf(“SIP/611-00000005”, “0?limit”) in new stack
– Executing [[email protected]r-callerid:13] ExecIf(“SIP/611-00000005”, “1?Set(GROUP(concurrency_limit)=611)”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/611-00000005”, “1?Set(CHANNEL(language)=fr)”) in new stack
– Executing [[email protected]:15] GosubIf(“SIP/611-00000005”, “7?sub-ccss,s,1(from-internal,8710)”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/611-00000005”, “0?Return()”) in new stack
– Executing [[email protected]:2] Set(“SIP/611-00000005”, “CCSS_SETUP=TRUE”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/611-00000005”, “0?monitor_config,1(from-internal,8710):monitor_default,1(from-internal,8710)”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/611-00000005”, “0?is_exten”) in new stack
– Executing [[email protected]:2] StackPop(“SIP/611-00000005”, “”) in new stack
– Executing [[email protected]:3] Return(“SIP/611-00000005”, “FALSE”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/611-00000005”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,30)
– Executing [[email protected]:30] Set(“SIP/611-00000005”, “CALLERID(number)=611”) in new stack
– Executing [[email protected]:31] Set(“SIP/611-00000005”, “CALLERID(name)=karim”) in new stack
– Executing [[email protected]:32] Set(“SIP/611-00000005”, “CDR(cnum)=611”) in new stack
– Executing [[email protected]:33] Set(“SIP/611-00000005”, “CDR(cnam)=karim”) in new stack
– Executing [[email protected]:34] Set(“SIP/611-00000005”, “CHANNEL(language)=fr”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/611-00000005”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [[email protected]:3] GosubIf(“SIP/611-00000005”, “0?sub-diversion-header,s,1()”) in new stack
– Executing [[email protected]:4] Set(“SIP/611-00000005”, “INTRACOMPANYROUTE=YES”) in new stack
– Executing [[email protected]:5] Set(“SIP/611-00000005”, “MOHCLASS=default”) in new stack
– Executing [[email protected]:6] ExecIf(“SIP/611-00000005”, “1?Set(TRUNKCIDOVERRIDE=611)”) in new stack
– Executing [[email protected]:7] Set(“SIP/611-00000005”, “_NODEST=”) in new stack
– Executing [[email protected]:8] Gosub(“SIP/611-00000005”, “sub-record-check,s,1(out,8710,)”) in new stack
– Executing [[email protected]:1] Set(“SIP/611-00000005”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/611-00000005”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [[email protected]:7] Set(“SIP/611-00000005”, “__MON_FMT=wav”) in new stack
– Executing [[email protected]:8] GotoIf(“SIP/611-00000005”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [[email protected]:11] ExecIf(“SIP/611-00000005”, “0?Return()”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/611-00000005”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [[email protected]:13] GotoIf(“SIP/611-00000005”, “0?out,1”) in new stack
– Executing [[email protected]:14] Set(“SIP/611-00000005”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [[email protected]:15] Set(“SIP/611-00000005”, “NOW=1409812564”) in new stack
– Executing [[email protected]:16] Set(“SIP/611-00000005”, “__DAY=04”) in new stack
– Executing [[email protected]:17] Set(“SIP/611-00000005”, “__MONTH=09”) in new stack
and my config:
TRUNK
trunk name: iax2
outbound CallerID:710
Dial pattern: 8|7XX
PEER Details:
host=198.168.x.x
username=710
secret=kadi710
type=friend
trunk=yes
qualify=yes
qualifyfreqok=25000
transfer=no
forceencryption=yes
encryption=yes
auth=md5
Incoming Settings
USER context:611
USER Details:
context=from-internal
secret=karim611
type=user
Register String: 710:[email protected]
OUTBOUND ROUTES
Route Name: local_8
Route CID: 611
Route Type: intra-company
Dial Patterns: 8|7XX
Trunk Sequence for Matched Routes: iax2
i didn’t send all the logs because they are many and i am unable to post all in one time i don’t know why…
I have a voice quality problem (voice very choppy) when I use IAX2 trunk between a “Freepbx 12 with asterisk version 12” and another system running Asterisk 1.8.
The error message WARNING[25042][C-000000ca]: chan_iax2.c:1793 compress_subclass: Can’t compress subclass 2097217
I am using Alaw codec, even if I do not specify any codecs the same error.
I hope someone can assist fixing this issue.