Hi
I have a Te411 PRI card connected to parlay voxtream i60. Every call
that comes on asterisk over zap channel and goes on to SIP Voice Blue
gsm gateway disconects after this timeout.
This is complete sip debug log. I also described how sip communication
is done in this matter. My configuration for sip is very simple i have
a trunk number 5 called gsm_gw_1_1-peer with following settings. Voice
Blue is ip gsm gateway and it is working ok on several instalations
but never with PRI card.
This disconnect happens because calling equipment doesn't get any
response from Asterisk on zap channel that call is in progress.
Why aren't message from sip forwarded to zap channel?
Would it be better if Ringing message would be sent from voice blue
instead of session progress?
[general]
canreinvite=no
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=alaw
[gsm_gw_1_1-peer]
type=peer
host=192.168.0.100
dtmfmode=inband
context=from-mux
canreinvite=no
Asterisk PBX VoiceBlue
INVITE
-------------------------------------------->
TRYING
<--------------------------------------------
SESSION PROGRESS
<--------------------------------------------
CANCEL
-------------------------------------------->
OK
<--------------------------------------------
REQUEST TERMINATED
<--------------------------------------------
ACK
--------------------------------------------->
Starting simple switch on 'Zap/3-1'
-- Accepting overlap call from '38626540259' to '041656699'
on
channel 0/3, span 1
-- Executing Goto("Zap/3-1",
"outrt-005-IpGsmGateway13|0038641656699|1") in new stack
-- Goto (outrt-005-IpGsmGateway13,0038641656699,1)
-- Executing Macro("Zap/3-1",
"dialout-trunk|5|0038641656699|") in new stack
-- Executing GotoIf("Zap/3-1", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("Zap/3-1", "user-callerid") in new
stack
-- Executing DBget("Zap/3-1",
"AMPUSER=DEVICE/38626540259/user")
in new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=38626540259/user
-- DBget: Value not found in database.
-- Executing DBget("Zap/3-1",
"AMPUSERCIDNAME=AMPUSER//cidname")
in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname
-- DBget: Value not found in database.
-- Executing GotoIf("Zap/3-1", "1?5") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("Zap/3-1", "Using CallerID
38626540259") in new stack
-- Executing Macro("Zap/3-1",
"record-enable|38626540259|OUT") in new stack
-- Executing GotoIf("Zap/3-1", "0 > 0?2:4") in new
stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("Zap/3-1",
"recordingcheck|20051129-095434|1133254470.611") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20051129-095434|1133254470.611: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("Zap/3-1", "No recording needed") in
new stack
-- Executing Macro("Zap/3-1", "outbound-callerid|5") in
new stack
-- Executing GotoIf("Zap/3-1", "1?3") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing DBget("Zap/3-1",
"USEROUTCID=AMPUSER/38626540259/outboundcid") in new stack
-- DBget: varname=USEROUTCID, family=AMPUSER, key=38626540259/outboundcid
-- DBget: Value not found in database.
-- Executing GotoIf("Zap/3-1", "1?6") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing NoOp("Zap/3-1", "CallerID set to
38626540259") in new stack
-- Executing SetGroup("Zap/3-1", "OUT_5") in new stack
-- Executing CheckGroup("Zap/3-1", "") in new stack
-- Executing SetVar("Zap/3-1",
"DIAL_NUMBER=0038641656699") in new stack
-- Executing SetVar("Zap/3-1", "DIAL_TRUNK=5") in new
stack
-- Executing AGI("Zap/3-1", "fixlocalprefix") in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("Zap/3-1", "OUTNUM=10038641656699")
in new stack
-- Executing Cut("Zap/3-1", "custom=OUT_5|:|1") in new
stack
-- Executing GotoIf("Zap/3-1", "0?16") in new stack
-- Executing Dial("Zap/3-1",
"SIP/gsm_gw_1_1-peer/10038641656699")
in new stack
We're at 192.168.0.99 port 13554
Adding codec 0x8 (alaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
INVITE sip:10038641656699@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>
Contact: <sip:38626540259@192.168.0.99>
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 Nov 2005 08:54:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 158
v=0
o=root 6100 6100 IN IP4 192.168.0.99
s=session
c=IN IP4 192.168.0.99
t=0 0
m=audio 13554 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
---
-- Called gsm_gw_1_1-peer/10038641656699
asterisk014*CLI>
<-- SIP read from 192.168.0.100:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>;tag=0050C229C70B-204011259
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 INVITE
Contact: <sip:192.168.0.100:5060>
User-Agent: VoiceBlue V-02.07.14
Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, NOTIFY
Content-Length: 0
--- (10 headers 0 lines)---
asterisk014*CLI>
<-- SIP read from 192.168.0.100:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>;tag=0050C229C70B-204011259
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 INVITE
Contact: <sip:192.168.0.100:5060>
User-Agent: VoiceBlue V-02.07.14
Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 140
v=0
o=VoiceBlue 36016 9541 IN IP4 192.168.0.100
s=GSM call
c=IN IP4 192.168.0.100
t=0 0
m=audio 10166 RTP/AVP 8
a=rtpmap:8 PCMA/8000
--- (11 headers 7 lines)---
Found RTP audio format 8
Peer audio RTP is at port 192.168.0.100:10166
Found description format PCMA
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0
(nothing), combined - 0x0 (nothing)
-- SIP/gsm_gw_1_1-peer-5e45 is making progress passing it to Zap/3-1
-- Channel 0/3, span 1 got hangup request
Reliably Transmitting (no NAT) to 192.168.0.100:5060:
CANCEL sip:10038641656699@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>
Contact: <sip:38626540259@192.168.0.99>
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of call
'719681054033b15e5d384f845dd5953f@192.168.0.99' in 15000 ms
== Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on
'Zap/3-1' in macro 'dialout-trunk'
== Spawn extension (outrt-005-IpGsmGateway13, 0038641656699, 1)
exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
<-- SIP read from 192.168.0.100:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>;tag=0050C229C70B-204011259
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 CANCEL
User-Agent: VoiceBlue V-02.07.14
Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, NOTIFY
Content-Length: 0
--- (9 headers 0 lines)---
asterisk014*CLI>
<-- SIP read from 192.168.0.100:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>;tag=0050C229C70B-204011259
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 INVITE
User-Agent: VoiceBlue V-02.07.14
Allow: INVITE, BYE, ACK, CANCEL, OPTIONS, REFER, NOTIFY
Content-Length: 0
--- (9 headers 0 lines)---
Transmitting (no NAT) to 192.168.0.100:5060:
ACK sip:10038641656699@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.99:5060;branch=z9hG4bK5b6a23e6;rport
From: "38626540259"
<sip:38626540259@192.168.0.99>;tag=as6809c997
To: <sip:10038641656699@192.168.0.100>;tag=0050C229C70B-204011259
Contact: <sip:38626540259@192.168.0.99>
Call-ID: 719681054033b15e5d384f845dd5953f@192.168.0.99
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0