[Asterisk-Users] Cisco SIP Weirdness (1750, not ATA)

[Eric Wieling]

I have the following in extentions.conf:

exten => 2111,1,Dial(SIP/2111 at gw1.langley)
exten => 2111,2,Voicemail(u2111)
exten => 2111,3,Hangup
exten => 2111,100,Voicemail(b2111)
exten => 2111,101,Hangup

I have the following in sip.conf:

; Cisco 1750
[gw1.langley]
type=friend
host=172.16.17.1
context=default
canreinvite=no

Like the ATA, lots of stuff doesn't work on the 1750 if I don't
have canreinvite=no

Extention 2111 is connected to an FXS port on the Cisco 1750.  I
can make and receive calls just fine.  If extention 2111 is busy
and a call comes into Asterisk (via Zap), Asterisk tries to send
the call to 2111 on the Cisco, the Cisco tells Asterisk that the
extention is busy.  Now comes the weird part. Asterisk sends the
call to voicemail and the voicemail tells the caller that the
person at extention 2111 is not available.  Shouldn't it tell
the user that extention 2111 is BUSY?

Here are the debug logs:

*CLI> 
*CLI> sip debug
SIP Debugging Enabled
*CLI>     -- Starting simple switch on 'Zap/1-1'
DEBUG[20491]: File chan_zap.c, Line 895 (zt_enable_ec): Enabled echo cancellation on channel 1
  == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>)
    -- Executing Goto("Zap/1-1", "2111|1") in new stack
    -- Goto (default,2111,1)
    -- Executing Dial("Zap/1-1", "SIP/2111 at gw1.langley") in new stack
Interface is eth0
IP Address is 172.16.17.7
We're at 172.16.17.7 port 24070
Answering with capability 2
Answering with capability 4
Answering with capability 8
10 headers, 11 lines
XXX Need to handle Retransmitting XXX:
INVITE sip:2111 at 172.16.17.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=207d7e2a
From: "PENSACOLA, FL" <sip:8503846785 at 172.16.17.7>;tag=611ff5f7
Contact: <sip:8503846785 at 172.16.17.7>
To: <sip:2111 at 172.16.17.1>
Call-ID: 3cd417467d369746768099b936dc3c3b at 172.16.17.7
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 5095 5095 IN IP4 172.16.17.7
s=session
c=IN IP4 172.16.17.7
t=0 0
m=audio 24070 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
 (no NAT) to 172.16.17.1:5060
    -- Called 2111 at gw1.langley
Sip read: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=207d7e2a
From: "PENSACOLA, FL" <sip:8503846785 at 172.16.17.7>;tag=611ff5f7
To: <sip:2111 at 172.16.17.1>;tag=CCC544-136
Date: Thu, 06 Mar 2003 23:05:39 GMT
Call-ID: 3cd417467d369746768099b936dc3c3b at 172.16.17.7
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


10 headers, 0 lines
Sip read: 
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=207d7e2a
From: "PENSACOLA, FL" <sip:8503846785 at 172.16.17.7>;tag=611ff5f7
To: <sip:2111 at 172.16.17.1>;tag=CCC544-136
Date: Thu, 06 Mar 2003 23:05:39 GMT
Call-ID: 3cd417467d369746768099b936dc3c3b at 172.16.17.7
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


10 headers, 0 lines
    -- Got SIP response 486 "Busy here" back from 172.16.17.1
XXX Need to handle Retransmitting XXX:
ACK sip:2111 at 172.16.17.1 SIP/2.0
Via: SIP/2.0/UDP 172.16.17.7:5060;branch=207d7e2a
From: "PENSACOLA, FL" <sip:8503846785 at 172.16.17.7>;tag=611ff5f7
To: <sip:2111 at 172.16.17.1>;tag=CCC544-136
Call-ID: 3cd417467d369746768099b936dc3c3b at 172.16.17.7
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 172.16.17.1:5060
DEBUG[2051]: File chan_sip.c, Line 541 (__sip_destroy): Detaching from SIP/gw1.langley-c320
  == No one is available to answer at this time
DEBUG[20491]: File chan_sip.c, Line 677 (sip_hangup): Asked to hangup channel not connected
    -- Executing VoiceMail("Zap/1-1", "u2111") in new stack
DEBUG[20491]: File chan_zap.c, Line 1655 (zt_answer): Took Zap/1-1 off hook
  == Parsing '/etc/asterisk/voicemail.conf': Found
DEBUG[20491]: File app_voicemail.c, Line 502 (leave_voicemail): vm/2111/unavail doesn't exist, doing what we can
    -- Playing 'vm-theperson'
    -- Playing 'digits/2'
    -- Playing 'digits/1'
    -- Playing 'digits/1'
    -- Playing 'digits/1'
    -- Playing 'vm-isunavail'
    -- Playing 'vm-intro'
    -- Playing 'beep'
    -- Recording to /var/spool/asterisk/vm/2111/INBOX/msg0006

*CLI> sip no debug
SIP Debugging Disabled
*CLI>