Below are my details:
-
The IP addresses of SIP provider - proxy1.bandtel.com and
proxy2.bandtel.com / DNS server details - 65.175.129.149 / 66.237.65.90 -
The Asterisk system - 216.146.88.205 (public IP)/ 10.1.1.68 (internal
IP) subnet mask - 255.255.255.0 -
The SIP phone? - We dont use SIP phone to establish the calls, We use
the RUBY AGI for making the automated calls. -
The bindaddr in my mgcp.conf file is 0.0.0.0
I could connect the call but once i pick the call the dialstatus still gives me as “NOANSWER”.
The message i get in Asterisk prompt is
– No one is available to answer at this time (1:0/0/0)
My sip provider is clear that he is sending the 200 ok, but its some configuration which has to be done in sip.conf i guess to receive this signal.
Thanks in advance and appreciate your response.
sip.conf
;
; SIP Configuration example for Asterisk
;
; Syntax for specifying a SIP device in extensions.conf is
; SIP/devicename where devicename is defined in a section below.
;
; You may also use
; SIP/username@domain to call any SIP user on the Internet
; (Don't forget to enable DNS SRV records if you want to use this)
;
; If you define a SIP proxy as a peer below, you may call
; SIP/proxyhostname/user or SIP/user@proxyhostname
; where the proxyhostname is defined in a section below
;
; Useful CLI commands to check peers/users:
; sip show peers Show all SIP peers (including friends)
; sip show users Show all SIP users (including friends)
; sip show registry Show status of hosts we register with
;
; sip debug Show all SIP messages
;
; reload chan_sip.so Reload configuration file
; Active SIP peers will not be reconfigured
;
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
autocreatepeer=yes
domain=proxy1.bandtel.com
callprogress=yes
disallow=all
allow=ulaw
trustrpid = yes
sendrpid = yes
progressinband=yes
promiscredir = yes
dtmfmode = rfc2833
qualify=yes
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[]]@host[:port][/extension]
register => 2068200001:*******@registrar.bandtel.com:5060
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
externip = 216.146.88.205
localnet=10.1.1.68/255.255.255.0
nat=yes
[proxy2.bandtel.com]
type=peer
secret=******
username=2068200001
host=proxy2.bandtel.com
fromuser=2068200001
fromdomain=proxy2.bandtel.com
dtmfmode=rfc2833
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
progressinband=yes
promiscredir=yes
nat=yes
context=dialout
authuser=2068200001
outboundproxy=proxy2.bandtel.com
outboundproxyport=5060
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:topsecret@digium.com
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
;------------------------------------------------------------------------------
; Users and peers have different settings available. Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; callingpres callingpres
; permit permit
; deny deny
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; allowoverlap allowoverlap
; allowsubscribe allowsubscribe
; allowtransfer allowtransfer
; subscribecontext subscribecontext
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
; We match on IP address of the proxy for incoming calls
; since we can not match on username (caller id)
;type=peer
;context=from-fwd
;host=fwd.pulver.com
;[sip_proxy-out]
;type=peer ; we only want to call out, not be called
;secret=guessit
;username=yourusername ; Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider requires ";user=phone" on URI
;call-limit=5 ; permit only 5 simultaneous outgoing calls to this peer
;outboundproxy=proxy.provider.domain ; send outbound signaling to this proxy, not directly to the peer
; Call-limits will not be enforced on real-time peers,
; since they are not stored in-memory
;port=80 ; The port number we want to connect to on the remote side
; Also used as "defaultport" in combination with "defaultip" settings
;------------------------------------------------------------------------------
; Definitions of locally connected SIP devices
;
; type = user a device that authenticates to us by "from" field to place calls
; type = peer a device we place calls to or that calls us and we match by host
; type = friend two configurations (peer+user) in one
;
; For device names, we recommend using only a-z, numerics (0-9) and underscore
;
; For local phones, type=friend works most of the time
;
; If you have one-way audio, you probably have NAT problems.
; If Asterisk is on a public IP, and the phone is inside of a NAT device
; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
;[grandstream1]
;type=friend
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
;dtmfmode=info ; either RFC2833 or INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call and 1 incoming call at a time
; from the phone to asterisk
; 1 for the explicit peer, 1 for the explicit user,
; remember that a friend equals 1 peer and 1 user in
; memory
; This will affect your subscriptions as well.
; There is no combined call counter for a "friend"
; so there's currently no way in sip.conf to limit
; to one inbound or outbound call per phone. Use
; the group counters in the dial plan for that.
;
;mailbox=1234@default ; mailbox 1234 in voicemail context "default"
;disallow=all ; need to disallow=all before we can use allow=
;allow=ulaw ; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
;allow=g729 ; Pass-thru only unless g729 license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for more information
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
;type=friend
;regexten=1234 ; When they register, create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234@default,1233@default ; Subscribe to status of multiple mailboxes
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234@context,2345 ; Mailbox(-es) for message waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in the MWI notify message
; defaults to global vmexten which defaults to "asterisk"
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;[polycom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls from this user
;secret=blahpoly
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;username=polly ; Username to use in INVITE until peer registers
; Normally you do NOT need to set this parameter
;disallow=all
;allow=ulaw ; dtmfmode=inband only works with ulaw or alaw!
;progressinband=no ; Polycom phones don't work properly with "never"
;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;insecure=port ; Allow matching of peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it's 1 second to reply
; Helps with NAT session
; qualify=yes uses default value
;
; Call group and Pickup group should be in the range from 0 to 63
;
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5
;defaultip=192.168.0.60 ; IP address to use if peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this account based on IP address
;permit=192.168.0.60/255.255.255.0
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than 200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP address that packet is
; received from instead of trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is
; behind a NAT).
;defaultip=192.168.0.4 ; IP address to use until registration
;username=goran ; Username to use when calling this device before registration
; Normally you do NOT need to set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set for all calls from this device
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes ; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on or DTMF reception will work improperly.sip_debug
*CLI> Reliably Transmitting (NAT) to 65.175.129.133:5060:
OPTIONS sip:proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK22cfd3c4;rport
From: "asterisk" <sip:asterisk@10.1.1.68>;tag=as7d7f766d
To: <sip:proxy2.bandtel.com>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 111b5688643541676b70b1552f812a5b@10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Dec 2007 09:56:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK22cfd3c4;rport=5060
From: asterisk <sip:asterisk@10.1.1.68>;tag=as7d7f766d
To: <sip:proxy2.bandtel.com>
Call-ID: 111b5688643541676b70b1552f812a5b@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '111b5688643541676b70b1552f812a5b@10.1.1.68' Method: OPTIONS
-- Attempting call on Local/outbound@dialout for outbound-handler@dialout:1 (Retry 1)
-- Executing [outbound@dialout:1] Answer("Local/outbound@dialout-78ba,2", "") in new stack
-- Executing [outbound@dialout:2] Wait("Local/outbound@dialout-78ba,2", "50") in new stack
-- Executing [outbound-handler@dialout:1] Dial("Local/outbound@dialout-78ba,1", "SIP/011919960466622@proxy2.bandtel.com|70|gM(outbound-connect^agi://10.1.1.68/ivr/speak^---+%0Aname%3A+sanchu%0Aid%3A+1%0A^)") in new stack
Audio is at 10.1.1.68 port 10988
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 65.175.129.133:5060:
INVITE sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Date: Tue, 04 Dec 2007 09:56:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 30571 30571 IN IP4 10.1.1.68
s=session
c=IN IP4 10.1.1.68
t=0 0
m=audio 10988 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 011919960466622@proxy2.bandtel.com
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport=5060
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport=5060
Record-Route: <sip:65.175.129.133;ftag=as6de88472;lr>
From: 2068200001 <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="65.175.129.133",nonce="ec13ba9157d5c517c1efb05fc30fe17247552714"
<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 65.175.129.133:5060:
ACK sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK5c12b644;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Audio is at 10.1.1.68 port 10988
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 65.175.129.133:5060:
INVITE sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Contact: <sip:2068200001@10.1.1.68>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Authorization: Digest username="2068200001", realm="65.175.129.133", algorithm=MD5, uri="sip:011919960466622@proxy2.bandtel.com", nonce="ec13ba9157d5c517c1efb05fc30fe17247552714", response="2511411b6a656bedf21d7549795c2e1a", opaque=""
Date: Tue, 04 Dec 2007 09:56:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 30571 30572 IN IP4 10.1.1.68
s=session
c=IN IP4 10.1.1.68
t=0 0
m=audio 10988 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport=5060
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport=5060
Record-Route: <sip:65.175.129.133;ftag=as6de88472;lr>
From: 2068200001 <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>;tag=2a3ad5978d87b904137d93d2eef8307b
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 INVITE
Server: Sippy
Content-Length: 169
Content-Type: application/sdp
v=0
o=prxlax01 0 0 IN IP4 216.168.169.108
s=sip call
t=0 0
m=audio 19206 RTP/AVP 0 101
c=IN IP4 216.168.169.87
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 8 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 216.168.169.87:19206
Found description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=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 216.168.169.87:19206
-- SIP/proxy2.bandtel.com-08ffb4f0 is making progress passing it to Local/outbound@dialout-78ba,1
Reliably Transmitting (NAT) to 65.175.129.133:5060:
OPTIONS sip:proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0f2bd66a;rport
From: "asterisk" <sip:asterisk@10.1.1.68>;tag=as2183c12e
To: <sip:proxy2.bandtel.com>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 038483a862b18f72336099c06b716996@10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Dec 2007 09:57:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK0f2bd66a;rport=5060
From: asterisk <sip:asterisk@10.1.1.68>;tag=as2183c12e
To: <sip:proxy2.bandtel.com>
Call-ID: 038483a862b18f72336099c06b716996@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '038483a862b18f72336099c06b716996@10.1.1.68' Method: OPTIONS
-- Executing [outbound@dialout:3] NoOp("Local/outbound@dialout-78ba,2", "status=") in new stack
-- Executing [outbound@dialout:4] AGI("Local/outbound@dialout-78ba,2", "agi://10.1.1.68/ivr/unanswered") in new stack
-- AGI Script agi://10.1.1.68/ivr/unanswered completed, returning 0
-- Executing [outbound@dialout:5] Hangup("Local/outbound@dialout-78ba,2", "") in new stack
== Spawn extension (dialout, outbound, 5) exited non-zero on 'Local/outbound@dialout-78ba,2'
Scheduling destruction of SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Scheduling destruction of SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' in 6400 ms (Method: INVITE)
== Spawn extension (dialout, outbound-handler, 1) exited non-zero on 'Local/outbound@dialout-78ba,1'
[Dec 4 03:57:24] NOTICE[30721]: pbx_spool.c:351 attempt_thread: Call completed to Local/outbound@dialout
Retransmitting #1 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Retransmitting #2 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Retransmitting #3 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Retransmitting #4 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Retransmitting #5 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Retransmitting #6 (NAT) to 65.175.129.133:5060:
CANCEL sip:011919960466622@proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK658d6d72;rport
From: "2068200001" <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
To: <sip:011919960466622@proxy2.bandtel.com>
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "2068200001" <sip:2068200001@proxy2.bandtel.com>;privacy=off;screen=no
Content-Length: 0
---
Really destroying SIP dialog '3c0b902d3343200937839d954e419945@proxy2.bandtel.com' Method: INVITE
<--- SIP read from 65.175.129.133:5060 --->
BYE sip:2068200001@10.1.1.68 SIP/2.0
Via: SIP/2.0/UDP 65.175.129.133;branch=z9hG4bKf68c.1386723063d6adb5111ac6fbb4807938.0
Via: SIP/2.0/UDP 65.175.129.133:5061;branch=z9hG4bKbbdef7a200b9da53df8377e11dda6d13;rport=5061
Max-Forwards: 16
From: <sip:011919960466622@proxy2.bandtel.com>;tag=2a3ad5978d87b904137d93d2eef8307b
To: 2068200001 <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 100 BYE
Contact: Anonymous <sip:65.175.129.133:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 1139039190-878584165-3706522926-1638668998
h323-conf-id: 1139039190-878584165-3706522926-1638668998
<------------->
--- (13 headers 0 lines) ---
<--- Transmitting (no NAT) to 65.175.129.133:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 65.175.129.133;branch=z9hG4bKf68c.1386723063d6adb5111ac6fbb4807938.0;received=65.175.129.133
Via: SIP/2.0/UDP 65.175.129.133:5061;branch=z9hG4bKbbdef7a200b9da53df8377e11dda6d13;rport=5061
From: <sip:011919960466622@proxy2.bandtel.com>;tag=2a3ad5978d87b904137d93d2eef8307b
To: 2068200001 <sip:2068200001@proxy2.bandtel.com>;tag=as6de88472
Call-ID: 3c0b902d3343200937839d954e419945@proxy2.bandtel.com
CSeq: 100 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Dec 4 03:58:04] NOTICE[30597]: chan_sip.c:7292 sip_reregister: -- Re-registration for 2068200001@registrar.bandtel.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 65.175.129.133:5060:
REGISTER sip:registrar.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK2ab21b67;rport
From: <sip:2068200001@registrar.bandtel.com>;tag=as728b665d
To: <sip:2068200001@registrar.bandtel.com>
Call-ID: 7acf76d45513c15e437162be24accd15@10.1.1.68
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="2068200001", realm="registrar.bandtel.com", algorithm=MD5, uri="sip:registrar.bandtel.com", nonce="47551ebbe34379de594cebaae453f9a285c1227f", response="2df694733d16158b78608764bb35800d", opaque=""
Expires: 120
Contact: <sip:s@10.1.1.68>
Event: registration
Content-Length: 0
---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK2ab21b67;rport=5060
From: <sip:2068200001@registrar.bandtel.com>;tag=as728b665d
To: <sip:2068200001@registrar.bandtel.com>;tag=e266e8809bf60e12ed80013e395353e4-a437
Call-ID: 7acf76d45513c15e437162be24accd15@10.1.1.68
CSeq: 112 REGISTER
PortaBilling: currency:USD
Contact: <sip:s@10.1.1.68>;expires=295
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '7acf76d45513c15e437162be24accd15@10.1.1.68' in 32000 ms (Method: REGISTER)
[Dec 4 03:58:04] NOTICE[30597]: chan_sip.c:12289 handle_response_register: Outbound Registration: Expiry for registrar.bandtel.com is 295 sec (Scheduling reregistration in 280 s)
Reliably Transmitting (NAT) to 65.175.129.133:5060:
OPTIONS sip:proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport
From: "asterisk" <sip:asterisk@10.1.1.68>;tag=as55bc3120
To: <sip:proxy2.bandtel.com>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Dec 2007 09:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
Retransmitting #1 (NAT) to 65.175.129.133:5060:
OPTIONS sip:proxy2.bandtel.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport
From: "asterisk" <sip:asterisk@10.1.1.68>;tag=as55bc3120
To: <sip:proxy2.bandtel.com>
Contact: <sip:asterisk@10.1.1.68>
Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 Dec 2007 09:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
<--- SIP read from 65.175.129.133:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.1.1.68:5060;branch=z9hG4bK799eaac9;rport=5060
From: asterisk <sip:asterisk@10.1.1.68>;tag=as55bc3120
To: <sip:proxy2.bandtel.com>
Call-ID: 29d7b41406896bea38a4afa92ba02d26@10.1.1.68
CSeq: 102 OPTIONS
Server: Sippy
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '29d7b41406896bea38a4afa92