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Hi, help please! we try to connect 7911G to asterisk 1.6 skinny or sip dosen't matter (we have both phone) internal call softphone to sofphone - is OK but when call to Cisco7911 - it's ringing but it's not answer, I don't know why?

-- Executing [333@sip-dial:1] Dial("SIP/vadim-0000000a", "SIP/cis@192.168.1.39") in new stack

== Using SIP RTP CoS mark 5 -- Called cis@192.168.1.39 -- SIP/192.168.1.39-0000000b is ringing == Spawn extension (sip-dial, 333, 1) exited non-zero on 'SIP/vadim-0000000a'

sip.conf [cis] type=friend
host=192.168.1.39
nat=yes
username=cis dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=sip-dial callerid=333

[vadim] type=friend secret=12345
host=dynamic
nat=yes
username=vadim dtmfmode=rfc2833 disallow=all allow=ulaw context=sip-dial callerid=888

Extension.conf [sip-dial]

exten => s,1,Answer() exten => s,n,Playback(msg) exten => s,n,Wait(5)

exten => 111,1,Dial(SIP/andrey) exten => 888,1,Dial(SIP/vadim) exten => 222,1,Dial(SIP/vital)

exten => 333,1,Dial(SIP/cis,20,tr) exten => 333,n,Hangup

Hi, help please! we try to connect 7911G to asterisk 1.6 skinny or sip dosen't matter (we have both phone) internal call softphone to sofphone - is OK but when call to Cisco7911 - it's ringing but it's not answer, I don't know why?

-- Executing [333@sip-dial:1] Dial("SIP/vadim-0000000a", "SIP/cis@192.168.1.39") in new stack

== Using SIP RTP CoS mark 5 -- Called cis@192.168.1.39 -- SIP/192.168.1.39-0000000b is ringing == Spawn extension (sip-dial, 333, 1) exited non-zero on 'SIP/vadim-0000000a'

sip.conf sip.conf

[cis] type=friend
host=192.168.1.39
nat=yes
username=cis dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=sip-dial callerid=333

[vadim] type=friend secret=12345
host=dynamic
nat=yes
username=vadim dtmfmode=rfc2833 disallow=all allow=ulaw context=sip-dial callerid=888

Extension.conf [sip-dial]

exten => s,1,Answer() exten => s,n,Playback(msg) exten => s,n,Wait(5)

exten => 111,1,Dial(SIP/andrey) exten => 888,1,Dial(SIP/vadim) exten => 222,1,Dial(SIP/vital)

exten => 333,1,Dial(SIP/cis,20,tr) exten => 333,n,Hangup

Hi, help please! we try to connect ТЕЛ 7911G to asterisk 1.6 skinny or sip dosen't matter (we have both phone) internal call softphone to sofphone Прошивка Sip 11.8-4-2S набрать на тел можно, звонок проходит но при поднятии трубки - is OK but when call to Cisco7911 - it's ringing but it's not answer, I don't know why?тишина, отбой нормально срабатывает, между софтовыми все работает.

-- Executing [333@sip-dial:1] Dial("SIP/vadim-0000000a", "SIP/cis@192.168.1.39") in new stack

== Using SIP RTP CoS mark 5 -- Called cis@192.168.1.39 -- SIP/192.168.1.39-0000000b is ringing == Spawn extension (sip-dial, 333, 1) exited non-zero on 'SIP/vadim-0000000a'

sip.conf

[cis] type=friend
host=192.168.1.39
nat=yes
username=cis dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw context=sip-dial callerid=333

[vadim] type=friend secret=12345
host=dynamic
nat=yes
username=vadim dtmfmode=rfc2833 disallow=all allow=ulaw context=sip-dial callerid=888

Extension.conf [sip-dial]

exten => s,1,Answer() exten => s,n,Playback(msg) exten => s,n,Wait(5)

exten => 111,1,Dial(SIP/andrey) exten => 888,1,Dial(SIP/vadim) exten => 222,1,Dial(SIP/vital)

exten => 333,1,Dial(SIP/cis,20,tr) exten => 333,n,Hangup333,1,Dial(SIP/cis,20,tr)