Asterisk 1.2beta2 and Flash Operator Panel

Asterisk Asterisk Support
1

Hi all.

I installed Asterisk 1.2beta 2 days ago without any problem. But today, I’ve tried to use Flash Operator Panel.
When I try to do a transfer with IP Phones (Pingtel and Cisco 7960), Asterisk says :

(when i drag’n drop from cisco to pingtel)
– Got SIP response 400 “Bad Request” back from 172.20.42.29
> Channel SIP/THECISCO-b582 was never answered.

or
(when i drag n drop from pingtel to cisco)
> Channel SIP/PINGTEL-c253 was answered.
– Executing Dial(“SIP/PINGTEL-c253”, “SIP/thecisco|20|rtT”) in new stack
– Called thecisco
– Got SIP response 400 “Bad Request” back from 172.20.42.29
– SIP/thecisco-12ca is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Nov 3 18:31:43 WARNING[23555]: pbx.c:2420 __ast_pbx_run: Timeout, but no rule ‘t’ in context ‘interne’

As you can see, in the second case, my pingtel rings but not my cisco. In first case, no one rings.

With softphones, transfers work very well. Otherwise, with pingtel (or cisco) and a softphone, i have the same problem.

I don’t understand because Flash Operator worked with Asterisk 1.0.9.

Thanks for your help.

Aurelien

2

Run “sip debug thecisco” in Asterisk, re-test, and post back the debug text.

3

drag n drop form cisco to pingtel :

[quote]We’re at 172.20.42.40 port 17434
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 172.20.42.29:5060:
INVITE sip:thecisco@172.20.42.29:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.42.40:5060;branch=z9hG4bK61c881ef;rport
From: “900 Cisco” sip:900@interne@172.20.42.40;tag=as2a663336
To: sip:thecisco@172.20.42.29:5060
Contact: sip:900@interne@172.20.42.40
Call-ID: 57db28db4e51acf31f40b7dc3b7e0e09@172.20.42.40
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Nov 2005 10:13:43 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 23744 23744 IN IP4 172.20.42.40
s=session
c=IN IP4 172.20.42.40
t=0 0
m=audio 17434 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Got SIP response 400 "Bad Request" back from 172.20.42.29

Transmitting (no NAT) to 172.20.42.29:5060:
ACK sip:thecisco@172.20.42.29:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.42.40:5060;branch=z9hG4bK61c881ef;rport
From: “900 Cisco” sip:900@interne@172.20.42.40;tag=as2a663336
To: sip:thecisco@172.20.42.29:5060
Contact: sip:900@interne@172.20.42.40
Call-ID: 57db28db4e51acf31f40b7dc3b7e0e09@172.20.42.40
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Destroying call '57db28db4e51acf31f40b7dc3b7e0e09@172.20.42.40’
c-boromir*CLI>[/quote]

drag n drop from pingtel to cisco :

[quote]SIP Debugging Enabled for IP: 172.20.42.29:5060
– Executing Dial(“SIP/PINGTEL-668a”, “SIP/thecisco|20|rtT”) in new stack
We’re at 172.20.42.40 port 10524
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (no NAT) to 172.20.42.29:5060:
INVITE sip:thecisco@172.20.42.29:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.42.40:5060;branch=z9hG4bK73143703;rport
From: “100 Pingtel” sip:100@externe@172.20.42.40;tag=as08c66653
To: sip:thecisco@172.20.42.29:5060
Contact: sip:100@externe@172.20.42.40
Call-ID: 080c26df33837b0316b6ce5106ccade4@172.20.42.40
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 04 Nov 2005 10:17:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 23744 23744 IN IP4 172.20.42.40
s=session
c=IN IP4 172.20.42.40
t=0 0
m=audio 10524 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Called thecisco
-- Got SIP response 400 "Bad Request" back from 172.20.42.29

Transmitting (no NAT) to 172.20.42.29:5060:
ACK sip:thecisco@172.20.42.29:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.42.40:5060;branch=z9hG4bK73143703;rport
From: “100 Pingtel” sip:100@externe@172.20.42.40;tag=as08c66653
To: sip:thecisco@172.20.42.29:5060
Contact: sip:100@externe@172.20.42.40
Call-ID: 080c26df33837b0316b6ce5106ccade4@172.20.42.40
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/thecisco-c3f2 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
Destroying call '080c26df33837b0316b6ce5106ccade4@172.20.42.40’
Nov 4 11:17:46 WARNING[31147]: pbx.c:2420 __ast_pbx_run: Timeout, but no rule ‘t’ in context 'interne’
Nov 4 11:18:06 WARNING[23771]: chan_sip.c:1189 retrans_pkt: Maximum retries exceeded on transmission 065bd4ba42cfa23750e538d661300dd9@172.20.42.40 for seqno 103 (Non-critical Request)
c-boromir*CLI>[/quote]