[asterisk-users] Call forwarding

Sat Sep 9 02:49:55 MST 2006

Hello to all asterisk users, 

I have a problem with call forwarding.

My extensions.conf:

[outbound]
exten =>  _*22*XXX,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4})
exten =>  _*22*,1,DBdel(CFIM/${CALLERID(num)})

Have three stations, 301, 302 and 303. When dial on 301 following
number:

*22*302

it should redirect all calls targeted to 301 to number 302. But it
doesn`t work.

If anyone of you has experience with call forwarding, your help will be
appreciated. Thank you very much.

Vladimir 

Here is SIP output:

---

<-- SIP read from 192.168.0.10:5060:
ACK sip:*22*302 at 192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-4b96baec
From: 301 <sip:301 at 192.168.0.1>;tag=c3d74f8b3bb05e94o0
To: <sip:*22*302 at 192.168.0.1>;tag=as3e772365
Call-ID: 704a9017-a9ad3778 at 192.168.0.10
CSeq: 101 ACK
Max-Forwards: 70
Contact: 301 <sip:301 at 192.168.0.10:5060>
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 0

--- (10 headers 0 lines)---

<-- SIP read from 192.168.0.10:5060:
INVITE sip:*22*302 at 192.168.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610
From: 301 <sip:301 at 192.168.0.1>;tag=c3d74f8b3bb05e94o0
To: <sip:*22*302 at 192.168.0.1>
Call-ID: 704a9017-a9ad3778 at 192.168.0.10
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="301",realm="asterisk",nonce="46339822",uri="sip:*22*302 at 192.168.0.1",algorithm=MD5,response="3284e4149a3abe9e0c4c454af19aa7b5"
Contact: 301 <sip:301 at 192.168.0.10:5060>
Expires: 240
User-Agent: Sipura/SPA2002-3.1.2(a)
Content-Length: 422
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 437796 437796 IN IP4 192.168.0.10
s=-
c=IN IP4 192.168.0.10
t=0 0
m=audio 16406 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (15 headers 19 lines)---
Using INVITE request as basis request - 704a9017-a9ad3778 at 192.168.0.10
Sending to 192.168.0.10 : 5060 (NAT)
Found user '301'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.10:16406
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x51d (g723|ulaw|alaw|
g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for *22*302 in outbound (domain 192.168.0.1)
list_route: hop: <sip:301 at 192.168.0.10:5060>
Transmitting (NAT) to 192.168.0.10:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 <sip:301 at 192.168.0.1>;tag=c3d74f8b3bb05e94o0
To: <sip:*22*302 at 192.168.0.1>
Call-ID: 704a9017-a9ad3778 at 192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*22*302 at 192.168.0.1>
Content-Length: 0

---
    -- Executing Set("SIP/301-503d", "DB(CFIM/301)=302") in new stack
    -- Executing Hangup("SIP/301-503d", "") in new stack
Reliably Transmitting (NAT) to 192.168.0.10:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 <sip:301 at 192.168.0.1>;tag=c3d74f8b3bb05e94o0
To: <sip:*22*302 at 192.168.0.1>;tag=as0c8d34d4
Call-ID: 704a9017-a9ad3778 at 192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*22*302 at 192.168.0.1>
Content-Length: 0

---
Retransmitting #1 (NAT) to 192.168.0.10:5060:
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
192.168.0.10:5060;branch=z9hG4bK-dd489610;received=192.168.0.10
From: 301 <sip:301 at 192.168.0.1>;tag=c3d74f8b3bb05e94o0
To: <sip:*22*302 at 192.168.0.1>;tag=as0c8d34d4
Call-ID: 704a9017-a9ad3778 at 192.168.0.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:*22*302 at 192.168.0.1>
Content-Length: 0