how to change the call envolope

[john_usc]

I am getting call transfer failed. my voip provider told me that I need to send this From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as0efda1e7
with my public ip like From: "asterisk" <sip:asterisk@192.34.xx.xx>;tag=as0efda1e7

how would I change that? Please let me know. My production is down

[Feb 3 23:19:28] Retransmitting #4 (NAT) to 64.2.142.93:5060:
OPTIONS sip:64.2.142.xxx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK02965c80;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as0efda1e7
To: <sip:64.2.142.xxx;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 408fe0bc6e6f062472fede28421d41ef@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Feb 2011 04:19:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Feb 3 23:19:28] Really destroying SIP dialog '408fe0bc6e6f062472fede28421d41ef@192.168.1.2' Method: OPTIONS
[Feb 3 23:19:28] Retransmitting #4 (NAT) to 64.2.142.93:5060:
OPTIONS sip:64.2.142.xxx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK605c54fb;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as325f3529
To: <sip:64.2.142.xxx;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 131dbde53fd611d845eac1253a045f2d@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Feb 2011 04:19:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

[williamconley]

ordinarily your provider should be able to give you the sip.conf entries, which you would put in "Admin->Carriers->Account Entry" for this provider.

a possible entry is "externip=xxx.xxx.xxx.xxx". This can actually be put directly in the sip.conf so it will apply to all contexts or directly in specific carriers who require it.

but seriously, if they are a voip provider with special requirements ... they should be able to give you the asterisk methods to achieve those results. NAT is a standard hurdle that must be crossed by virtually every provider, and there are many solutions available.

[john_usc]

these are vitelity guys. I am getting frustrated with them though

[williamconley]

You'll need to substitute your own host entries if they do not have you on the same servers as this client, and this is a "dial 6" dialplan with a 10 digit plus 1 for direct dialing from agent workstations:

account entry:

[vitel-inbound]
type=friend
dtmfmode=auto
host=64.2.142.90
context=trunkinbound
disallow=all
allow=all
insecure=port,invite
canreinvite=no
nat=yes

[vitel-outbound]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=g729
canreinvite=no
nat=yes

globals:

DIAL6TRUNK = SIP/vitel-outbound

dialplan:

exten=>_61NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=>_61NXXNXXXXXX,n,NoOp(Dial(SIP/${EXTEN:1}@vitel-outbound,,tTor))
exten=>_61NXXNXXXXXX,n,Dial(${DIAL6TRUNK}/${EXTEN:1},,To)
exten=>_61NXXNXXXXXX,n,Hangup

exten=>_1NXXNXXXXXX,1,Dial(${DIAL6TRUNK}/${EXTEN:1},,To)
exten=>_1NXXNXXXXXX,n,Hangup

sip.conf:

externip=xxx.xx.xx.xx

substitute your firewall's external ip address