I have a recently installed FreePBX 2.7 running on Centos with kernel 2.6.18-164.11.1.el5 . The setup has a Digium TE220b interface card connect to a T1 line (PRI trunk). All my extensions are going to be softphones, so all extensions are SIP devices. I have set up a Zaptel trunk and a SIP trunk. I want to set up a network with all calls to extensions to be routed through the SIP trunk and calls to external network through the T1 line. The T1 line has 24 bearer channels and 24 DID allotted to me. I am able to call between two extensions/softphones. But whenever I try to call an external no it saya all circuits are busy, try to call again letter. I got a reply it might me due to SIP trunk not being configure properly. I am not sure what to do. Please help me. Here are the trunk configuration and SIP debug output with verbose level of 0. I have only one extension created as of now with extension 1000.
For internal routing of calls, i.e. within extensions. do I need to create a seperate SIP trunk all will the zap trunk do the work for me.
/etc/asterisk/sip_general_additional.conf
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
disallow=all
allow=g729
allow=ulaw
allow=g722
allow=gsm
allow=g723
allow=g726
allow=alaw
jbenable=yes
jbforce=yes
jbimpl=adaptive
jbresyncthreshold=1000
jblog=yes
jbmaxsize=200
defaultexpiry=120
maxexpiry=3600
srvlookup=no
minexpiry=60
allowguest=yes
registerattempts=0
registertimeout=20
notifyhold=yes
g726nonstandard=no
t38pt_udptl=no
videosupport=no
maxcallbitrate=1024
canreinvite=yes
rtptimeout=30
rtpholdtimeout=300
checkmwi=10
notifyringing=yes
rtpkeepalive=0
nat=no
/etc/asterisk/chan_dahdi.conf
[channels]
language=en
; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf
; XTDM20B Port #1,2 plugged into PSTN
;AMPLABEL:Channel %c - Button %n
context=from-pstn
signalling=pri_net
faxdetect=incoming
usecallerid=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
group=0
channel=1-2
Output from SIP Debug in asterisk CLI enclosed
<— SIP read from 129.107.212.84:9576 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-e6d8125be1fa9e15-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:9576
To: sip:[email protected]
From: "Mayank"sip:[email protected];tag=0c95b51d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite Beta release 4.0 v3 stamp 55153
Content-Length: 243
v=0
o=- 1 2 IN IP4 129.107.212.84
s=CounterPath X-Lite 4.0
c=IN IP4 129.107.212.84
t=0 0
m=audio 42658 RTP/AVP 0 8 3 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : hQY52WWh ZTmWDDe/ 129.107.212.84 42658
<------------->
— (12 headers 10 lines) —
Sending to 129.107.212.84 : 9576 (no NAT)
Using INVITE request as basis request - N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
<— Reliably Transmitting (NAT) to 129.107.212.84:9576 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-e6d8125be1fa9e15-1—d8754z-;received=129.107.212.84;rport=9576
From: "Mayank"sip:[email protected];tag=0c95b51d
To: sip:[email protected];tag=as7ce13328
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="77d86bbd"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.’ in 32000 ms (Method: INVITE)
Found user '1000’
dhcp-0-4-23-b4-f5-55*CLI>
<— SIP read from 129.107.212.84:9576 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-e6d8125be1fa9e15-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as7ce13328
From: "Mayank"sip:[email protected];tag=0c95b51d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 1 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
dhcp-0-4-23-b4-f5-55*CLI>
<— SIP read from 129.107.212.84:9576 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-c1b90a64516aa915-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:9576
To: sip:[email protected]
From: “Mayank"sip:[email protected];tag=0c95b51d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“77d86bbd”,uri="sip:[email protected]”,response=“dd5ec60015db3af2b8a18966b9498396”,algorithm=MD5
User-Agent: X-Lite Beta release 4.0 v3 stamp 55153
Content-Length: 243
v=0
o=- 1 2 IN IP4 129.107.212.84
s=CounterPath X-Lite 4.0
c=IN IP4 129.107.212.84
t=0 0
m=audio 42658 RTP/AVP 0 8 3 101
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 1 : hQY52WWh ZTmWDDe/ 129.107.212.84 42658
<------------->
— (13 headers 10 lines) —
Sending to 129.107.212.84 : 9576 (NAT)
Using INVITE request as basis request - N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
Found user '1000’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x190f (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 129.107.212.84:42658
Looking for 8172727409 in from-internal (domain 129.107.98.126)
list_route: hop: sip:[email protected]:9576
<— Transmitting (NAT) to 129.107.212.84:9576 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-c1b90a64516aa915-1—d8754z-;received=129.107.212.84;rport=9576
From: "Mayank"sip:[email protected];tag=0c95b51d
To: sip:[email protected]
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0
<------------>
Really destroying SIP dialog ‘[email protected]’ Method: INVITE
Audio is at 129.107.98.126 port 10702
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (NAT) to 129.107.212.84:9576 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-c1b90a64516aa915-1—d8754z-;received=129.107.212.84;rport=9576
From: "Mayank"sip:[email protected];tag=0c95b51d
To: sip:[email protected];tag=as7f75dc1d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Type: application/sdp
Content-Length: 291
v=0
o=root 13014 13014 IN IP4 129.107.98.126
s=session
c=IN IP4 129.107.98.126
t=0 0
m=audio 10702 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
dhcp-0-4-23-b4-f5-55*CLI>
<— Reliably Transmitting (NAT) to 129.107.212.84:9576 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-c1b90a64516aa915-1—d8754z-;received=129.107.212.84;rport=9576
From: "Mayank"sip:[email protected];tag=0c95b51d
To: sip:[email protected];tag=as7f75dc1d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 20
<------------>
<— SIP read from 129.107.212.84:9576 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 129.107.212.84:9576;branch=z9hG4bK-d8754z-c1b90a64516aa915-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as7f75dc1d
From: "Mayank"sip:[email protected];tag=0c95b51d
Call-ID: N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘N2MzZTg3NjEzNzQ2NWEyYjUyOTdiZTRlZGRiMjhiOGE.’ Method: ACK
dhcp-0-4-23-b4-f5-55*CLI>
<— SIP read from 129.107.212.84:9576 —>
<------------->
Thanks
Mayank