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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-4b96baec
From: 301 <sip:[EMAIL PROTECTED]>;tag=c3d74f8b3bb05e94o0
To: <sip:[EMAIL PROTECTED]>;tag=as3e772365
Call-ID: [EMAIL PROTECTED]
CSeq: 101 ACK
Max-Forwards: 70
Contact: 301 <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK-dd489610
From: 301 <sip:[EMAIL PROTECTED]>;tag=c3d74f8b3bb05e94o0
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="301",realm="asterisk",nonce="46339822",uri="sip:[EMAIL
PROTECTED]",algorithm=MD5,response="3284e4149a3abe9e0c4c454af19aa7b5"
Contact: 301 <sip:[EMAIL PROTECTED]: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 - [EMAIL PROTECTED]
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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=c3d74f8b3bb05e94o0
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=c3d74f8b3bb05e94o0
To: <sip:[EMAIL PROTECTED]>;tag=as0c8d34d4
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]>;tag=c3d74f8b3bb05e94o0
To: <sip:[EMAIL PROTECTED]>;tag=as0c8d34d4
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
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