All pastes #287114 Raw Edit

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#287114 ·published 2006-12-21 04:27 UTC
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zapata.conf
------------------
[trunkgroups]

[channels]
language                  =  en
switchtype                =  national
signalling                =  pri_cpe
pridialplan               =  national
prilocaldialplan          =  national
faxdetect                 =  both
relaxdtmf                 =  yes
echocancel                =  yes
echocancelwhenbridged     =  yes
overlapdial               =  no
echotraining              =  yes
usecallingpres            =  yes
amaflags                  =  default
context                   =  default
group                     =  1
channel                   => 1-23



zaptel.conf
-----------
loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24



debug & verbose to 100 -- sip debug peer 2627309
----------------------------------------------------

*CLI>
< Protocol Discriminator: Q.931 (8)  len=39
< Call Ref: len= 2 (reference 11/0xB) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a2]
< Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
<                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
<                              Ext: 1  User information layer 1: u-Law (34)
< [18 03 a1 83 81]
< Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0
<                        ChanSel: Reserved
<                       Ext: 1  Coding: 0   Number Specified   Channel Type: 3
<                       Ext: 1  Channel: 1 ]
< [6c 0c 21 83 32 31 38 39 36 39 34 36 35 31]
< Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
<                           Presentation: Presentation allowed of network provided number (3) '2189694651' ]
< [70 08 a1 32 36 32 37 33 30 39]
< Called Number (len=10) [ Ext: 1  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '2627309' ]
-- Making new call for cr 11
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
> Protocol Discriminator: Q.931 (8)  len=11
> Call Ref: len= 2 (reference 11/0xB) (Terminator)
> Message type: CALL PROCEEDING (2)
> [18 04 e9 81 83 81]
> Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0
>                        ChanSel: Reserved
>                       Ext: 1  DS1 Identifier: 1
>                       Ext: 1  Coding: 0   Number Specified   Channel Type: 3
>                       Ext: 1  Channel: 1 ]
    -- Accepting call from '2189694651' to '2627309' on channel 1/1, span 1
    -- Executing Macro("Zap/1-1", "standard|7309|SIP/2627309") in new stack
    -- Executing Dial("Zap/1-1", "SIP/2627309|25") in new stack
We're at 10.0.5.10 port 14492
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 10.3.237.50:5060:
INVITE sip:2627309@10.3.237.50 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>
Contact: <sip:2189694651@10.0.5.10>
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 Dec 2006 04:22:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 210

v=0
o=root 58037 58037 IN IP4 10.0.5.10
s=session
c=IN IP4 10.0.5.10
t=0 0
m=audio 14492 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 2627309
< Protocol Discriminator: Q.931 (8)  len=9
< Call Ref: len= 2 (reference 11/0xB) (Originator)
< Message type: RELEASE (77)
< [08 02 82 86]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Public network serving the local user (2)
<                  Ext: 1  Cause: Channel unacceptable (6), class = Normal Event (0) ]
-- Processing IE 8 (cs0, Cause)
    -- Channel 1/1, span 1 got hangup
Scheduling destruction of call '55679cf10570be052ccb3d0d53153774@10.0.5.10' in 32000 ms
  == Spawn extension (macro-standard, s, 1) exited non-zero on 'Zap/1-1' in macro 'standard'
  == Spawn extension (macro-standard, s, 1) exited non-zero on 'Zap/1-1'
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 11/0xB) (Terminator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
    -- Hungup 'Zap/1-1'

<-- SIP read from 10.3.237.50:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>;tag=70EBCA4E-65750045
CSeq: 102 INVITE
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
Contact: <sip:2627309@10.3.237.50>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291
Content-Length: 0


--- (9 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.3.237.50:5060:
CANCEL sip:2627309@10.3.237.50 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '55679cf10570be052ccb3d0d53153774@10.0.5.10' in 32000 ms

<-- SIP read from 10.3.237.50:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>
CSeq: 102 CANCEL
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
Contact: <sip:2627309@10.3.237.50>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291
Content-Length: 0


--- (9 headers 0 lines) ---

<-- SIP read from 10.3.237.50:5060:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>;tag=70EBCA4E-65750045
CSeq: 102 INVITE
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
Contact: <sip:2627309@10.3.237.50>
User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291
Content-Length: 0


--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.3.237.50:5060:
ACK sip:2627309@10.3.237.50 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.10:5060;branch=z9hG4bK6ed6de62;rport
From: "2189694651" <sip:2189694651@10.0.5.10>;tag=as6738f942
To: <sip:2627309@10.3.237.50>;tag=70EBCA4E-65750045
Contact: <sip:2189694651@10.0.5.10>
Call-ID: 55679cf10570be052ccb3d0d53153774@10.0.5.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---