Inbound calles

General Help
1

when i route my did number to extention directly i can’t hear the caller on my extension .
the call is disconnecting after 32 sec
when i enable debug for the extension i get

== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 5060
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
Reliably Transmitting (no NAT) to 172.16.10.1:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.3:5060;branch=z9hG4bK22b55511
Max-Forwards: 70
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.0
Date: Sat, 04 Dec 2010 09:48:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 600374747 600374747 IN IP4 192.168.150.3
s=Asterisk PBX 1.8.0
c=IN IP4 192.168.150.3
t=0 0
m=audio 13344 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=ptime:20
a=sendrecv

-- Called 101

<— SIP read from UDP:172.16.10.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.150.3:5060;received=192.168.150.3;branch=z9hG4bK22b55511
Call-ID: [email protected]:5060
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected]
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:172.16.10.1:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.150.3:5060;received=192.168.150.3;branch=z9hG4bK22b55511
Call-ID: [email protected]:5060
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected];tag=tjKhwck…0gCb4i7sh-2bIgydq-v78ON
CSeq: 102 INVITE
Contact: sip:[email protected]:5060
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0

<------------->
— (9 headers 0 lines) —
– SIP/101-00000080 is ringing

<— SIP read from UDP:172.16.10.1:5060 —>

<------------->

<— SIP read from UDP:172.16.10.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.3:5060;received=192.168.150.3;branch=z9hG4bK22b55511
Call-ID: [email protected]:5060
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected];tag=tjKhwck…0gCb4i7sh-2bIgydq-v78ON
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: sip:[email protected]:5060
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 3500444448 3500444449 IN IP4 172.16.10.1
s=pjmedia
c=IN IP4 172.16.10.1
t=0 0
a=X-nat:0
m=audio 10005 RTP/AVP 0 96
a=rtcp:10006 IN IP4 172.16.10.1
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
— (11 headers 12 lines) —
Found RTP audio format 0
Found RTP audio format 96
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 96
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.10.1:10005
list_route: hop: sip:[email protected]:5060
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 172.16.10.1:5060
Transmitting (no NAT) to 172.16.10.1:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.3:5060;branch=z9hG4bK75d6e7b7
Max-Forwards: 70
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected]:5060;tag=tjKhwck…0gCb4i7sh-2bIgydq-v78ON
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.0
Content-Length: 0

--

Scheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 172.16.10.1:5060
Reliably Transmitting (no NAT) to 172.16.10.1:5060:
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.3:5060;branch=z9hG4bK60745816
Max-Forwards: 70
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected]:5060;tag=tjKhwck…0gCb4i7sh-2bIgydq-v78ON
Call-ID: [email protected]:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

== Spawn extension (macro-dial-one, s, 37) exited non-zero on ‘SIP/To Lab-0000007f’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/To Lab-0000007f’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 101, 1) exited non-zero on ‘SIP/To Lab-0000007f’

<— SIP read from UDP:172.16.10.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.3:5060;received=192.168.150.3;branch=z9hG4bK60745816
Call-ID: [email protected]:5060
From: “737000073” sip:[email protected];tag=as48a18b4f
To: sip:[email protected];tag=tjKhwck…0gCb4i7sh-2bIgydq-v78ON
CSeq: 103 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: INVITE
Reliably Transmitting (no NAT) to 172.16.10.1:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.150.3:5060;branch=z9hG4bK69a1a50f
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as55e4e9b1
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.0
Date: Sat, 04 Dec 2010 09:48:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

-- Remote UNIX connection disconnected

<— SIP read from UDP:172.16.10.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.150.3:5060;received=192.168.150.3;branch=z9hG4bK69a1a50f
Call-ID: [email protected]:5060
From: “Unknown” sip:[email protected];tag=as55e4e9b1
To: sip:[email protected];tag=z9hG4bK69a1a50f
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: CSipSimple
Content-Type: application/sdp
Content-Length: 386

v=0
o=- 3500444466 3500444466 IN IP4 172.16.10.1
s=pjmedia
c=IN IP4 172.16.10.1
t=0 0
m=audio 10001 RTP/AVP 0 8 97 104 98 3 96
a=rtcp:10002 IN IP4 172.16.10.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 speex/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:98 speex/16000
a=rtpmap:3 GSM/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
<------------->
— (13 headers 17 lines) —

when i enable

2

Hi
i have cisco router with ngx fw

3

You could try setting canreinvite=no on the trunk for your DID.

Also, what firewall/router are you using? It is likely to be NAT or an ALG (http://www.voip-info.org/wiki/view/Routers+SIP+ALG) on the firewall causing the issue.

4

I have had similar issues in the past and it turned out to be SIP issues with the firewall. Do you have a router firewall? Have you opened port 5060 for SIP traffic? or better still have you added your trunks ip to have access to your voip switch on the firewall?

Thanks

5

i have open all the udp port to my pbx

6

v=0
o=- 3500454438 3500454438 IN IP4 192.168.10.102
s=pjmedia
c=IN IP4 192.168.10.102
t=0 0
m=audio 10001 RTP/AVP 0 8 97 104 98 3 96
a=rtcp:10002 IN IP4 192.168.10.102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 speex/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:98 speex/16000
a=rtpmap:3 GSM/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15

I DON"T HAVE 192.168.10 NETWORK AT ALL