[asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer

2007-06-25 Thread falz 
.96.5
t=0 0
m=audio 16816 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
-- Started music on hold, class 'default', on SIP/302-0824f618
-- Stopped music on hold on SIP/302-0824f618
ivan*CLI>
<--- SIP read from 192.168.96.18:50422 --->
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.96.18:5060;branch=z9hG4bK77d2c2d8
From: ;tag=001200348d021a566e942586-7ba72702
To: "User Name 1" ;tag=as2a70a5c3
Call-ID: [EMAIL PROTECTED]
Max-Forwards: 70
Date: Mon, 25 Jun 2007 17:09:59 GMT
CSeq: 101 ACK
User-Agent: Cisco-CP7960G/8.0
Content-Length: 0


<->
--- (10 headers 0 lines) ---
ivan*CLI>
<--- SIP read from 192.168.96.16:50074 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK7f41a807;rport
From: ;tag=as2c9302cc
To: "User Name 1" ;tag=001200347d27001a7e20b127-2129053d
Call-ID: [EMAIL PROTECTED]
Date: Mon, 25 Jun 2007 17:06:25 GMT
CSeq: 103 INVITE
Warning: 399 SDP Not Acceptable
Server: Cisco-CP7960G/8.0
Contact: 
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0


<->
--- (12 headers 0 lines) ---
set_destination: Parsing
 for address/port
to send to
set_destination: set destination to 192.168.96.16, port 5060
Transmitting (no NAT) to 192.168.96.16:5060:
ACK sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK7f41a807;rport
From: ;tag=as2c9302cc
To: "User Name 1" ;tag=001200347d27001a7e20b127-2129053d
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms (Method:
ACK)
set_destination: Parsing
 for address/port
to send to
set_destination: set destination to 192.168.96.18, port 5060
Reliably Transmitting (no NAT) to 192.168.96.18:5060:
BYE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK27d10c95;rport
From: "User Name 1" ;tag=as2a70a5c3
To: ;tag=001200348d021a566e942586-7ba72702
Call-ID: [EMAIL PROTECTED]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (companyname-default, 304, 1) exited non-zero on
'SIP/302-0824f618'
Scheduling destruction of SIP dialog
'[EMAIL PROTECTED]' in 32000 ms
(Method: ACK)
set_destination: Parsing
 for address/port
to send to
set_destination: set destination to 192.168.96.16, port 5060
Reliably Transmitting (no NAT) to 192.168.96.16:5060:
BYE sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK06cc5909;rport
From: ;tag=as2c9302cc
To: "User Name 1" ;tag=001200347d27001a7e20b127-2129053d
Call-ID: [EMAIL PROTECTED]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
ivan*CLI>
<--- SIP read from 192.168.96.18:50422 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK27d10c95;rport
From: "User Name 1" ;tag=as2a70a5c3
To: ;tag=001200348d021a566e942586-7ba72702
Call-ID: [EMAIL PROTECTED]
Date: Mon, 25 Jun 2007 17:09:59 GMT
CSeq: 104 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0


<->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: ACK
ivan*CLI>
<--- SIP read from 192.168.96.16:50075 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK06cc5909;rport
From: ;tag=as2c9302cc
To: "User Name 1" ;tag=001200347d27001a7e20b127-2129053d
Call-ID: [EMAIL PROTECTED]
Date: Mon, 25 Jun 2007 17:06:25 GMT
CSeq: 104 BYE
Server: Cisco-CP7960G/8.0
Content-Length: 0


<->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: ACK


=
=

Any help or thoughts would be appreciated!

--falz

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Re: [asterisk-users] Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer

2007-06-25 Thread falz 
On 6/25/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> On Mon, 2007-06-25 at 12:51 -0500, falz wrote:
> >
> > However, on the first business day, we realized that when transferring
> > calls (not using call parking, using the built in transfer buttons on
> > a Cisco 7960) would not work. This error would occur:
> >
> I had this problem when I first upgraded from 1.2 to 1.4 on all my IP
> Phones.  What I did to fix it was add "canreinvite=no" to all phones and
> this solved the problem.

Interesting. Just after posting to this list, I downgraded back to
1.2. I'll re-upgrade again later and try this out in the next few
days. I didn't have this defined anywhere, so it must default to yes
in the each device's area in sip.conf.

Thanks for the tip, I was way off track on what I was debugging
(codecs and such)

--falz

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