Callback to two different number then bridge

Asterisk Asterisk Support
1

I don’t know how to describe this but I’ll try.

This is what I want accomplish by using Asterisk 1.4:

Lets say I am traveling out of country and want call my wife’s cellphone. So I called my home Asterisk box then hangup. Asterisk then call me back, meanwhile Asterisk calls my wife’s cellphone. So when both of us pick up then we can start talking.

I know how to make Asterisk call back by putting a call file on the /outgoing

I DON’T want use DISA. Because sometime Asterisk just couldn’t recognize the number I keyed in. So I am trying to avoid punching numbers.

My question is how to make these two call bridge(not sure this is the current term) together?

Please help. Thx.

[edit] Just found this : forums.whirlpool.net.au/forum-re … 26501.html
But they don’t have a solution.

2

I developed a complete callback prepaid system that is doing what you discribe with everything included (credits, detailed log files, procedure to upgrade and add accounts automaticly etc. etc.) and I’m offering callback prepaid as a service to the market.

If it is just for one or limited number of phonenumbers you could do some clever stuff with asterisk and call files.

make a callfile /var/spool/asterisk/test/test (or any other name)

Channel: /
MaxRetries: 2
RetryTime: 30
WaitTime: 25
Context: callback_test
Extension: s
Priority: 1

Add this to your /etc/asterisk/extensions.conf

[test]
exten => s,1,Answer()

exten => s,n,Hangup()
exten => h,1,System(cp /var/spool/asterisk/test/test /var/spool/asterisk/outgoing/test) ; copying the callfile, adjust the name of the callfile

[callback_test]
exten => s,1,Dial(${TRUNK}/,20,rt) ; Define the value of ${TRUNK} under [global] TRUNK=
exten => s,n,Hangup()

And you are up and running. I have tested this and it is working if you use a working trunk and proper phonenumbers. Some Asterisk magic in 14 lines. Good luck!

3

lesouvage, thanks for the reply.

Your solution is about callback but not to bridge another call channel. I want Asterisk callback to the initial caller as well as the fixed destination number at same time.

4

What happens with my example is:

[ul]person 1 calls asteriskbox (you can use a dedicated number, a (hidden) menu option or whatever to jump to the rest of the routine)

asterisk box calls person 1 back

as soon as person 1 picks up the phone asterisk starts calling person 2

as soon as person 2 picks up the phone asterisk will bridge the two channels so person 1 can talk to person 2.[/ul]

When this has happens both legs of the calls are setup from the Asterisk box

From what you have written I understand this is what you are looking for.

5

Thank you for the help, lesouvage. It works! :smile:

Another question: I tried to only use voipstunt as the trunk to make both the callback and the call destination(my wife’s cell phone) and CLI show the call is connected(answered) but no sound! If I use different trunks then it works fine.

Voipstunt does allow two(or more?) simultaneous calls. I can make two outgoing calls from the internal extensions by using Voipstunt at same time.

I aslo noticed that if I only use a local VoIP provider which allow me to make 3-way calling then the callback works without any problem.

So why can I just use Voipstunt ???

6

CLI log (start from Callback):
– Attempting call on SIP/1778xxxxxxx@VoipStunt for s@custom-out-sunming:1 (Retry 1)
> Channel SIP/VoipStunt-001a60d8 was answered.
– Executing [s@custom-out-sunming:1] Dial(“SIP/VoipStunt-001a60d8”, “SIP/18887258880@VoipStunt|40|rt”) in new stack
– Called 18887258880@VoipStunt
– SIP/VoipStunt-00137a40 is making progress passing it to SIP/VoipStunt-001a60d8
– SIP/VoipStunt-00137a40 answered SIP/VoipStunt-001a60d8 [color=red]<–seems normal, but dead silence![/color]
== Spawn extension (custom-out-sunming, s, 1) exited non-zero on ‘SIP/VoipStunt-001a60d8’
[Aug 8 11:52:32] NOTICE[685]: pbx_spool.c:351 attempt_thread: Call completed to SIP/1778xxxxxxx@VoipStunt

7

I’m not that an expert on this kind of sip issues. Are you sure you can make more then one call with the VoipStunt trunk, have you tested this with two seperated outgoing calls. There might be an issue with VoipStunt because this kind of bridging might interfere with some of the VOIP magic they have implemented on there node. I can’t think of anything concrete but maybe the real sip experts can give a pointer.

If you type “sip set debug” and make a phonecall the output might give a lead to the answer to this problem. You can turn the sip debug off by typing “sip set debug off” on the cli. Please post the sip debug info.

Most of the time this kind of silence has a relation with rtp ports that aren’t available and/or don’t pass the firewall properly. The signaling channel (port 5060) is up and running but the dynamic assigned rtp ports (10.000 to 20.000) used for the voice/sound/data do not pass the firewall.

The cli output looks, as you wrote, normal for a succesfull setup of a call.

I’m happy to hear that the routine is working

8

I just checked that I can make more then one call with the VoipStunt trunk at the same time. I called my wife and asked her to hold, then I use another extension to call my cellphone. Everything seems work fine. Last, I login to my Voipstunt website account and I saw the two recent calls records.

So I did the debug(callback and dial out) but I feel dizzy when I read it. :blush:

Here is the log:

[NOTE]:
17783229858, my cellphone number, the number to callback
6042885566, Asterisk DID
18887258880, Destination number

NSLU2*CLI> sip set debug
SIP Debugging enabled
Reliably Transmitting (NAT) to 192.168.1.28:5060:
OPTIONS sip:1005@192.168.1.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK5418efcf;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as4b77f4ae
To: sip:1005@192.168.1.28:5060
Contact: sip:asterisk@192.168.1.77
Call-ID: 6f36d59e132697df616673cd6b729b43@192.168.1.77
CSeq: 102 OPTIONS
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

NSLU2*CLI>
<— SIP read from 192.168.1.28:5060 —>
SIP/2.0 200 OK
To: sip:1005@192.168.1.28:5060;tag=6bedd76af22633d0i0
From: “asterisk” sip:asterisk@192.168.1.77;tag=as4b77f4ae
Call-ID: 6f36d59e132697df616673cd6b729b43@192.168.1.77
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK5418efcf
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘285e78c535b2945c1824d98912e9875b@192.168.1.77’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.107:5060:
NOTIFY sip:1001@192.168.1.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK30f86819;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as2e869264
To: sip:1001@192.168.1.107:5060
Contact: sip:asterisk@192.168.1.77
Call-ID: 285e78c535b2945c1824d98912e9875b@192.168.1.77
CSeq: 102 NOTIFY
User-Agent: Grandstream 486
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.77
Voice-Message: 0/0 (0/0)

Really destroying SIP dialog ‘6f36d59e132697df616673cd6b729b43@192.168.1.77’ Method: OPTIONS
NSLU2*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK30f86819;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as2e869264
To: sip:1001@192.168.1.107:5060
Call-ID: 285e78c535b2945c1824d98912e9875b@192.168.1.77
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘285e78c535b2945c1824d98912e9875b@192.168.1.77’ Method: NOTIFY
Reliably Transmitting (NAT) to 192.168.1.114:5060:
OPTIONS sip:1002@192.168.1.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK08af2e44;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as41e3e2b9
To: sip:1002@192.168.1.114:5060
Contact: sip:asterisk@192.168.1.77
Call-ID: 5d24d0565c97133a0ab039b633e0edfa@192.168.1.77
CSeq: 102 OPTIONS
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

NSLU2*CLI>
<— SIP read from 192.168.1.114:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK08af2e44;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as41e3e2b9
To: sip:1002@192.168.1.114:5060
Call-ID: 5d24d0565c97133a0ab039b633e0edfa@192.168.1.77
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘5d24d0565c97133a0ab039b633e0edfa@192.168.1.77’ Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.1.107:5060:
OPTIONS sip:1001@192.168.1.107:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK52effe55;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as30db3bfe
To: sip:1001@192.168.1.107:5060
Contact: sip:asterisk@192.168.1.77
Call-ID: 0652467566dd397b37305e950f35bd6c@192.168.1.77
CSeq: 102 OPTIONS
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Reliably Transmitting (NAT) to 192.168.1.28:5061:
OPTIONS sip:dynaguy@192.168.1.28:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK1619553e;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as12eabb95
To: sip:dynaguy@192.168.1.28:5061
Contact: sip:asterisk@192.168.1.77
Call-ID: 3dc10bfc6d71880a71508cdd556e318d@192.168.1.77
CSeq: 102 OPTIONS
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

NSLU2*CLI>
<— SIP read from 192.168.1.28:5061 —>
SIP/2.0 200 OK
To: sip:dynaguy@192.168.1.28:5061;tag=c918aa9a456dfa60i1
From: “asterisk” sip:asterisk@192.168.1.77;tag=as12eabb95
Call-ID: 3dc10bfc6d71880a71508cdd556e318d@192.168.1.77
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK1619553e
Server: Linksys/PAP2-3.1.22(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
— (10 headers 0 lines) —
NSLU2*CLI>
<— SIP read from 192.168.1.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK52effe55;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as30db3bfe
To: sip:1001@192.168.1.107:5060
Call-ID: 0652467566dd397b37305e950f35bd6c@192.168.1.77
CSeq: 102 OPTIONS
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘3dc10bfc6d71880a71508cdd556e318d@192.168.1.77’ Method: OPTIONS
Really destroying SIP dialog ‘0652467566dd397b37305e950f35bd6c@192.168.1.77’ Method: OPTIONS
Scheduling destruction of SIP dialog ‘2792f4085771d25326debe1f2a77364c@192.168.1.77’ in 6400 ms (Method: NOTIFY)
Reliably Transmitting (NAT) to 192.168.1.114:5060:
NOTIFY sip:1002@192.168.1.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK3da2f5e3;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as355d8ab7
To: sip:1002@192.168.1.114:5060
Contact: sip:asterisk@192.168.1.77
Call-ID: 2792f4085771d25326debe1f2a77364c@192.168.1.77
CSeq: 102 NOTIFY
User-Agent: Grandstream 486
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.77
Voice-Message: 0/0 (0/0)

NSLU2*CLI>
<— SIP read from 192.168.1.114:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:5060;branch=z9hG4bK3da2f5e3;rport
From: “asterisk” sip:asterisk@192.168.1.77;tag=as355d8ab7
To: sip:1002@192.168.1.114:5060
Call-ID: 2792f4085771d25326debe1f2a77364c@192.168.1.77
CSeq: 102 NOTIFY
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘2792f4085771d25326debe1f2a77364c@192.168.1.77’ Method: NOTIFY
NSLU2*CLI>
<— SIP read from 64.69.89.228:5060 —>
NOTIFY sip:16042739158@24.87.16.226 SIP/2.0
Via: SIP/2.0/UDP 64.69.89.228;branch=z9hG4bK506b.24785d68fffd128b70a5f4a7375066d1.0
Via: SIP/2.0/UDP 64.69.89.229:5064;branch=z9hG4bK0446969288709210ff765bbfa430f93e;rport=5064
Max-Forwards: 16
From: PortaUM sip:PortaUM@64.69.89.229:5064;tag=cfdf016cbce592860fbd5b803e07cb01
To: sip:16042739158@64.69.89.228
Call-ID: 7dc75f3c7a9b40ae8b4b6bcf0587e250@64.69.89.229
CSeq: 1 NOTIFY
Contact: PortaUM sip:PortaUM@64.69.89.229:5064
Expires: 300
User-Agent: Sippy
Event: message-summary
Content-Length: 22
Content-Type: application/simple-message-summary

Messages-Waiting: no

<------------->
— (14 headers 1 lines) —
NSLU2*CLI>
<— Transmitting (NAT) to 64.69.89.228:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP 64.69.89.228;branch=z9hG4bK506b.24785d68fffd128b70a5f4a7375066d1.0;received=64.69.89.228
Via: SIP/2.0/UDP 64.69.89.229:5064;branch=z9hG4bK0446969288709210ff765bbfa430f93e;rport=5064
From: PortaUM sip:PortaUM@64.69.89.229:5064;tag=cfdf016cbce592860fbd5b803e07cb01
To: sip:16042739158@64.69.89.228;tag=as00ee1de2
Call-ID: 7dc75f3c7a9b40ae8b4b6bcf0587e250@64.69.89.229
CSeq: 1 NOTIFY
User-Agent: Grandstream 486
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<------------>
NSLU2*CLI>
<— SIP read from 209.17.160.129:5060 —>
OPTIONS sip:6042885566@24.87.16.226 SIP/2.0
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK6ee38e4e
From: “asterisk” sip:asterisk@209.17.160.129;tag=as5c29bc09
To: sip:6042885566@24.87.16.226
Contact: sip:asterisk@209.17.160.129
Call-ID: 57ce2b9d5f0988457c64ac792d3e6d09@209.17.160.129
CSeq: 102 OPTIONS
User-Agent: DV VOIP
Date: Wed, 08 Aug 2007 22:59:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Looking for 6042885566 in mydomain-incoming (domain 24.87.16.226)

<— Transmitting (NAT) to 209.17.160.129:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.17.160.129:5060;branch=z9hG4bK6ee38e4e;received=209.17.160.129
From: “asterisk” sip:asterisk@209.17.160.129;tag=as5c29bc09
To: sip:6042885566@24.87.16.226;tag=as6768b065
Call-ID: 57ce2b9d5f0988457c64ac792d3e6d09@209.17.160.129
CSeq: 102 OPTIONS
User-Agent: Grandstream 486
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:24.87.16.226
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘57ce2b9d5f0988457c64ac792d3e6d09@209.17.160.129’ in 32000 ms (Method: OPTIONS)
– Executing [h@custom-CallBack-sunli:3] Hangup(“SIP/6042885566-0019ac00”, “”) in new stack
== Spawn extension (custom-CallBack-sunli, h, 3) exited non-zero on ‘SIP/6042885566-0019ac00’
– Attempting call on SIP/17783229858@VoipStunt for s@custom-DISA-out-sunming:1 (Retry 1)
Audio is at 24.87.16.226 port 15866
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 194.120.0.198:5060:
INVITE sip:17783229858@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK6a7fb944;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 102 INVITE
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1712 1712 IN IP4 24.87.16.226
s=session
c=IN IP4 24.87.16.226
t=0 0
m=audio 15866 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Really destroying SIP dialog ‘7d8e080b13b884372a218c7d2f9238d5@209.17.160.129’ Method: ACK
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK6a7fb944;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com
Contact: sip:17783229858@194.120.0.198:5060
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
WWW-Authenticate: Digest realm=“sipdiscount.com”,nonce=“3157290015”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 194.120.0.198:5060:
ACK sip:17783229858@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK6a7fb944;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 102 ACK
User-Agent: Grandstream 486
Max-Forwards: 70
ontent-Length: 0

Audio is at 24.87.16.226 port 15866
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 194.120.0.198:5060:
INVITE sip:17783229858@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK7577c3c8;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 103 INVITE
User-Agent: Grandstream 486
Max-Forwards: 70
Authorization: Digest username=“dynaguy”, realm=“sipdiscount.com”, algorithm=MD5, uri="sip:17783229858@sip.voipstunt.com", nonce=“3157290015”, response=“15b47fcd67b89e82bc7e54c3a667f197”, opaque=""
Date: Wed, 08 Aug 2007 22:58:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1712 1713 IN IP4 24.87.16.226
s=session>
c=IN IP4 24.87.16.226
t=0 0
m=audio 15866 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK7577c3c8;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com
Contact: sip:17783229858@194.120.0.198:5060
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Length: 0

<------------->
— (10 headers 0 lines) —
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK7577c3c8;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com;tag=cb1710accb2b10ac46aa48a783196
Contact: sip:17783229858@194.120.0.198:5060
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Type: application/sdp
Content-Length: 202

v=0
o=dynaguy 1186613945 1186613945 IN IP4 80.239.235.163
s=SIP Call
c=IN IP4 80.239.235.163
t=0 0
m=audio 25482 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 80.239.235.163:25482
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x7 (g723|gsm|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 80.239.235.163:25482
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK7577c3c8;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com;tag=cb1710accb2b10ac46aa48a783196
Contact: sip:17783229858@194.120.0.198:5060
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Type: application/sdp
Content-Length: 202

v=0
o=dynaguy 1186613953 1186613953 IN IP4 80.239.235.163
s=SIP Call
c=IN IP4 80.239.235.163
t=0 0
m=audio 25482 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 80.239.235.163:25482
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x7 (g723|gsm|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 80.239.235.163:25482
list_route: hop: sip:17783229858@194.120.0.198:5060
set_destination: Parsing sip:17783229858@194.120.0.198:5060 for address/port to send to
set_destination: set destination to 194.120.0.198, port 5060
Transmitting (NAT) to 194.120.0.198:5060:
ACK sip:17783229858@194.120.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK66c48217;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
To: sip:17783229858@sip.voipstunt.com;tag=cb1710accb2b10ac46aa48a783196
Contact: sip:dynaguy@24.87.16.226
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 103 ACK
User-Agent: Grandstream 486
Max-Forwards: 70
Content-Length: 0

   > Channel SIP/VoipStunt-0019fa10 was answered.
-- Executing [s@custom-DISA-out-sunming:1] Dial("SIP/VoipStunt-0019fa10", "SIP/18887258880@VoipStunt|40|rt") in new stack

Audio is at 24.87.16.226 port 19512
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 194.120.0.198:5060:
INVITE sip:18887258880@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK42dad317;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 102 INVITE
User-Agent: Grandstream 486
Max-Forwards: 70
Date: Wed, 08 Aug 2007 22:58:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1712 1712 IN IP4 24.87.16.226
s=session
c=IN IP4 24.87.16.226
t=0 0
m=audio 19512 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called 18887258880@VoipStunt

NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK42dad317;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com
Contact: sip:18887258880@194.120.0.198:5060
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
WWW-Authenticate: Digest realm=“sipdiscount.com”,nonce=“3227747312”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 194.120.0.198:5060:
ACK sip:18887258880@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK42dad317;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 102 ACK
User-Agent: Grandstream 486
Max-Forwards: 70
ontent-Length: 0

Audio is at 24.87.16.226 port 19512
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 194.120.0.198:5060:
INVITE sip:18887258880@sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK3c8fafd1;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com
Contact: sip:dynaguy@24.87.16.226
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 103 INVITE
User-Agent: Grandstream 486
Max-Forwards: 70
Authorization: Digest username=“dynaguy”, realm=“sipdiscount.com”, algorithm=MD5, uri="sip:18887258880@sip.voipstunt.com", nonce=“3227747312”, response=“484365091a6e42d0bca7578172ab3915”, opaque=""
Date: Wed, 08 Aug 2007 22:58:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1712 1713 IN IP4 24.87.16.226
s=session>
c=IN IP4 24.87.16.226
t=0 0
m=audio 19512 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK3c8fafd1;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com
Contact: sip:18887258880@194.120.0.198:5060
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Length: 0

<------------->
— (10 headers 0 lines) —
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK3c8fafd1;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com;tag=ce1710acce2b10ac46aa186f10ac57
Contact: sip:18887258880@194.120.0.198:5060
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Type: application/sdp
Content-Length: 202

v=0
o=dynaguy 1186613953 1186613953 IN IP4 195.219.64.186
s=SIP Call
c=IN IP4 195.219.64.186
t=0 0
m=audio 41696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 195.219.64.186:41696
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x7 (g723|gsm|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 195.219.64.186:41696
– SIP/VoipStunt-0019ac00 is making progress passing it to SIP/VoipStunt-0019fa10
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK3c8fafd1;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com;tag=ce1710acce2b10ac46aa186f10ac57
Contact: sip:18887258880@194.120.0.198:5060
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Type: application/sdp
Content-Length: 202

v=0
o=dynaguy 1186613957 1186613957 IN IP4 195.219.64.186
s=SIP Call
c=IN IP4 195.219.64.186
t=0 0
m=audio 41696 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

<------------->
— (11 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 195.219.64.186:41696
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x7 (g723|gsm|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 195.219.64.186:41696
list_route: hop: sip:18887258880@194.120.0.198:5060
set_destination: Parsing sip:18887258880@194.120.0.198:5060 for address/port to send to
set_destination: set destination to 194.120.0.198, port 5060
Transmitting (NAT) to 194.120.0.198:5060:
ACK sip:18887258880@194.120.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK042745c0;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com;tag=ce1710acce2b10ac46aa186f10ac57
Contact: sip:dynaguy@24.87.16.226
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 103 ACK
User-Agent: Grandstream 486
Max-Forwards: 70
Content-Length: 0

-- SIP/VoipStunt-0019ac00 answered SIP/VoipStunt-0019fa10

Really destroying SIP dialog ‘128875828-268651078@192.168.1.114’ Method: REGISTER
Really destroying SIP dialog ‘112315814-1515129281@192.168.1.107’ Method: REGISTER
Really destroying SIP dialog ‘57ce2b9d5f0988457c64ac792d3e6d09@209.17.160.129’ Method: OPTIONS
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
BYE sip:dynaguy@24.87.16.226 SIP/2.0
Via: SIP/2.0/UDP 194.120.0.198:5060;branch=z9hG4bK7577c3c8
From: sip:17783229858@sip.voipstunt.com;tag=cb1710accb2b10ac46aa48a783196
To: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
Contact: sip:17783229858@194.120.0.198:5060
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 0 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 194.120.0.198 : 5060 (NAT)

<— Transmitting (NAT) to 194.120.0.198:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.120.0.198:5060;branch=z9hG4bK7577c3c8;received=194.120.0.198
From: sip:17783229858@sip.voipstunt.com;tag=cb1710accb2b10ac46aa48a783196
To: “6042885566” sip:dynaguy@24.87.16.226;tag=as6267cbeb
Call-ID: 55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226
CSeq: 0 BYE
User-Agent: Grandstream 486
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:dynaguy@24.87.16.226
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘33d1edac4307ab180108675c5de5e272@24.87.16.226’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:18887258880@194.120.0.198:5060 for address/port to send to
set_destination: set destination to 194.120.0.198, port 5060
Reliably Transmitting (NAT) to 194.120.0.198:5060:
BYE sip:18887258880@194.120.0.198:5060 SIP/2.0
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK2ab190dc;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com;tag=ce1710acce2b10ac46aa186f10ac57
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 104 BYE
User-Agent: Grandstream 486
Max-Forwards: 70
Authorization: Digest username=“dynaguy”, realm=“sipdiscount.com”, algorithm=MD5, uri=“sip:18887258880@194.120.0.198:5060”, nonce=“3227747312”, response=“d6b843bc149158d230fd1d8a3540ff9e”, opaque=""
Content-Length: 0

== Spawn extension (custom-DISA-out-sunming, s, 1) exited non-zero on ‘SIP/VoipStunt-0019fa10’
[Aug 8 15:59:03] NOTICE[1712]: pbx_spool.c:351 attempt_thread: Call completed to SIP/17783229858@VoipStunt
Really destroying SIP dialog ‘55de90b75c19f5cd3b3e41e7639cbaac@24.87.16.226’ Method: BYE
NSLU2*CLI>
<— SIP read from 194.120.0.198:5060 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 24.87.16.226:5060;branch=z9hG4bK2ab190dc;rport
From: “6042885566” sip:dynaguy@24.87.16.226;tag=as1ef98f26
To: sip:18887258880@sip.voipstunt.com;tag=ce1710acce2b10ac46aa186f10ac57
Contact: sip:18887258880@194.120.0.198:5060
Call-ID: 33d1edac4307ab180108675c5de5e272@24.87.16.226
CSeq: 104 BYE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘33d1edac4307ab180108675c5de5e272@24.87.16.226’ Method: INVITE
NSLU2CLI> sip set debug off
SIP Debugging Disabled
NSLU2CLI>