I am having some issues using Asterisk as a PRI gateway with Sipxecs. For the
most part it works for inbound and outbound calling however
when a call is received on a PRI channel and then send to a SipXecs extension
which has a forwarding rule to ring the extension and a mobile device at the
same time
asterisk quickly cancels the call to the extension while allowing the mobile to
ring.
I have 2 media gateways and 2 sipxecs proxies this behavior is not happening
when the call comes from GW2 then gets forwarded out GW1 (or vice versa)
Call --> PRI ---> Asterisk PRI GW 1 ---> Sipxecs (Forward Rule "simultaneous
ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + Mobile) SipX Exten Rings One
Time while mobile rings as expected.
Some of my calls come in another gateway and when this happens the call is
handled properly:
Call --> PRI ---> Asterisk PRI GW 2 ---> Sipxecs (Forward Rule "simultaneous
ring") ---> Asterisk PRI GW 1 ----> (SipX Exten + Mobile) Expected result both
extensions ring
Both Asterisk PRI GWs are set up as unmanaged gateways in sipxecs.
Peer Def in asterisk look like this:
[general]
trustrpid = yes
sendrpid = yes
progressinband=never
srvlookup=yes
[GW01]
type=friend
port=5060
insecure=invite,port
host=GW01.domain.com
context=default
dtmfmode=rfc2833
[GW02]
type=friend
port=5060
insecure=invite,port
host=GW02.domain.com
context=default
dtmfmode=rfc2833
Dialplan is basically
[inbound]
exten => _XXXX,1,AGI(route.php)
exten => _XXXX,2, Dial(${ext...@domain.com)
[outbound]
exten => _NXXNXXXXXX,1,Dial(DAHDI/g1/${EXTEN}
[default]
include => inbound
include => outbound
here is the sip debug from server --- calling my did which routes to exten 2945
on sipxecs
Content-Length: 316
Expires: 60
X-Sipx-Authidentity:
<sip:2...@domain.com;signature=4AFC3F27%3A433dc76eea085f80717687d8084654a2>
X-Sipx-Handled: XSIPX02-IP-ADDRESS-67.107.93.2
v=0
o=root 1085943255 1085943255 IN IP4 GW01-IP-ADDRESS
s=Asterisk PBX 1.6.2.0-rc4
c=IN IP4 GW01-IP-ADDRESS
t=0 0
m=audio 15766 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
<------------->
--- (21 headers 14 lines) ---
<--- Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0bd6f69;received=SIPX02-IP-ADDRESS
Via: SIP/2.0/TCP
SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac70fb6983d8aeacefa051a75c83ce64f8c8~0a78ca617d5b460168faf046fcaf2f1b;id=22276-565
Via: SIP/2.0/UDP
SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac6651daee59948275599d7b41f51a249b4d~1bfa448fba164a3d273549fca4a8a79d
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Record-Route:
<sip:SIPX02-IP-ADDRESS:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EYXMyMzQ0ZjE5MA%60%60.900_ntap%2Aid%7EMjIyNzYtNTY1%214bc4cb52b2d1e947feccb17805166d0b>
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:6185591...@gw01-ip-address>
Content-Length: 0
<------------>
-- Now forwarding DAHDI/11-1 to 'Local/6932...@default' (thanks to
SIP/DOMAIN.com-00001844)
Scheduling destruction of SIP dialog
'44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060:
CANCEL sip:2...@domain.com SIP/2.0
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport
Max-Forwards: 70
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.2.0-rc4
Content-Length: 0
---
Scheduling destruction of SIP dialog
'44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms (Method: INVITE)
-- Executing [6932...@default:1] Dial("Local/6932...@default-e585;2",
"DAHDI/g2/6932833") in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/6932833
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 200 OK
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 CANCEL
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
CANCEL sip:6932...@gw01-ip-address;sipx-noroute=Voicemail;transport=udp SIP/2.0
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 CANCEL
Max-Forwards: 20
Via: SIP/2.0/UDP
SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0bd6f69
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Sending to SIPX02-IP-ADDRESS : 5060 (no NAT)
Scheduling destruction of SIP dialog
'44c969f1000b2c574b3245e779126...@gw01-ip-address' in 32000 ms (Method: CANCEL)
plastmg01*CLI>
<--- Reliably Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;rport=5060
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
plastmg01*CLI>
<--- Transmitting (no NAT) to SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
SIPX02-IP-ADDRESS;branch=z9hG4bK-sipXecs-ac739a93395457cf85554d6d8810b0bd6f69;received=SIPX02-IP-ADDRESS
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 CANCEL
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 408 Request timeout
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=023e4750
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 INVITE
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 408 Request timeout
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=023e4750
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 INVITE
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- DAHDI/30-1 is proceeding passing it to Local/6932...@default-e585;2
-- Local/6932...@default-e585;1 is proceeding passing it to DAHDI/11-1
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 408 Request timeout
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=023e4750
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 INVITE
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
plastmg01*CLI>
<--- SIP read from UDP:SIPX02-IP-ADDRESS:5060 --->
SIP/2.0 408 Request timeout
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=023e4750
Call-Id: 44c969f1000b2c574b3245e779126...@gw01-ip-address
Cseq: 102 INVITE
Via: SIP/2.0/UDP GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;rport=5060
Server: sipXecs/4.0.2 sipXecs/sipXproxy (Linux)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (no NAT) to SIPX02-IP-ADDRESS:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;rport=5060
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
-- DAHDI/30-1 is making progress passing it to Local/6932...@default-e585;2
-- DAHDI/30-1 is making progress passing it to Local/6932...@default-e585;2
-- Local/6932...@default-e585;1 is making progress passing it to DAHDI/11-1
-- Local/6932...@default-e585;1 is making progress passing it to DAHDI/11-1
Retransmitting #2 (no NAT) to SIPX02-IP-ADDRESS:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;rport=5060
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
-- DAHDI/30-1 answered Local/6932...@default-e585;2
-- Local/6932...@default-e585;1 answered DAHDI/11-1
-- Native bridging DAHDI/11-1 and DAHDI/30-1
== Spawn extension (default, 6932833, 1) exited non-zero on
'Local/6932...@default-e585;2'
Retransmitting #3 (no NAT) to SIPX02-IP-ADDRESS:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;rport=5060
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
Retransmitting #4 (no NAT) to SIPX02-IP-ADDRESS:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
GW01-IP-ADDRESS:5060;branch=z9hG4bK5d22be3e;received=SIPX02-IP-ADDRESS;rport=5060
From: "6185591324" <sip:6185591...@gw01-ip-address>;tag=as2344f190
To: <sip:2...@domain.com>;tag=a70a3f79
Call-ID: 44c969f1000b2c574b3245e779126...@gw01-ip-address
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.0-rc4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
finally asterisk will report something like the following .... (note this is
not from the above call so the call-id is different)
[Nov 12 11:02:57] WARNING[6378]: chan_sip.c:3782 retrans_pkt: Maximum retries
exceeded on transmission 1e88918e19216bcb4a4d43fb7793c...@gw01-ip-address for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
Really destroying SIP dialog '1e88918e19216bcb4a4d43fb7793c...@gw01-ip-address'
Method: CANCEL
It seem that asterisk just wants to forward the call to the mobile device and
cancel the extens call
Can anyone advise me on a working config for this ?
_______________________________________________
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
sipXecs IP PBX -- http://www.sipfoundry.org/
