Hi im trying to call from my extensions to another city , where i have another server with free pbx , i call from xxx extension to xxx extension and i cant connect the call , i was just making a debug to my extension and i enabled Nat on the config of my extension and this is the output of the debug i put
- Executing [continue@macro-dialout-trunk:4] Set(“SIP/311-b75812c0”, “CALLERID(number)=311”) in new stack
– Executing [1173@from-internal:6] Macro(“SIP/311-b75812c0”, “outisbusy|”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/311-b75812c0”, “”) in new stack
Audio is at 172.16.1.5 port 15618
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
SURENVIOS-NEIVA*CLI>
<— Transmitting (NAT) to 172.16.1.111:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.1.111:5060;branch=z9hG4bKbe3949a09376695a;received=172.16.1.111
From: “Aux Fact y Gestion2” sip:[email protected];tag=688dba7f557f9678
To: sip:[email protected];tag=as421553a0
Call-ID: [email protected]
CSeq: 58952 INVITE
User-Agent: FPBX-2.9.0(1.4.26)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 2952 2952 IN IP4 172.16.1.5
s=session
c=IN IP4 172.16.1.5
t=0 0
m=audio 15618 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
– Executing [s@macro-outisbusy:2] Playback(“SIP/311-b75812c0”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/311-b75812c0> Playing ‘all-circuits-busy-now’ (language ‘es’)
SURENVIOS-NEIVA*CLI>
<— SIP read from 172.16.1.111:5060 —>
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 172.16.1.111:5060;branch=z9hG4bKbe3949a09376695a
From: “Aux Fact y Gestion2” sip:[email protected];tag=688dba7f557f9678
To: sip:[email protected]
Supported: path
Call-ID: [email protected]
CSeq: 58952 CANCEL
User-Agent: Grandstream GXP280 1.2.5.3
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
can some one please help me ? the trunk its fine but i cant make any call to remote extensions, those extensions are sip