Cant make calls to remote extensions

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 :slight_smile:

  • Executing [[email protected]:4] Set(“SIP/311-b75812c0”, “CALLERID(number)=311”) in new stack
    – Executing [[email protected]:6] Macro(“SIP/311-b75812c0”, “outisbusy|”) in new stack
    – Executing [[email protected]: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 [[email protected]: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

It doesn’t look like your connection is getting made. Try checking the /var/log/asterisk/full log around the time of the call and see what the error (if any) is in there.

1 Like

yup it was that , what was happening was that i have a vpn between the two sites so i had to put to another internal ip in each other trunk set as destination ip to make calls, it was a lil bit simple , but sometimes you dont catch it hahaha :v thanks