---------- Forwarded message ----------
From: Salvatore Frandina <[email protected]>
Date: 2010/1/21
Subject: App_Conference: SIPp SDP and DTMF mode
To: Neil Stratford <[email protected]>, Mihai Balea <[email protected]>,
[email protected]
Hi,
I'm using SIPp application http://sipp.sourceforge.net/ to generate a SIP
call to open source PBX Asterisk.
Command to call the extension (extension) where the IP is IP address of
Asterisk
[code]sipp -m 1 -d 36000000 -s extension -sf uac_modified.xml IP [/code]
In the configuration file uac_modified.xml there are the following lines
[code]
INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
User-Agent: sipp
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 0 97 8 18 3 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-11,16
a=rtpmap:0 PCMU/8000
a=rtpmap:97 SPEEX/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
m=video [media_port] RTP/AVP 115
a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
a=rtpmap:115 H263-1998/90000
a=sendrecv
[/code]
The call work well between SIPp and softphone (Eyebeam, X-lite), i can see
all the messages in the Asterisk CLI.
If i try to use App_conference application the SIPp user work only without
video support.
Scenarios: there is a conference (DTMF mode enabled) where there are two or
more users when i press a digit to see a generic user, the SIPp user returns
the following error
[code]
sipp: The following events occured:
2010-01-21 16:07:10:392 1264086430.392045: Aborting call on unexpected
message for Call-Id '[email protected]': while pausing (index 5), received
'INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.4:5060;branch=z9hG4bK11e48a8e;rport
Max-Forwards: 70
From: sut <sip:[email protected]:5060>;tag=as47ad7951
To: sipp <sip:[email protected]:5061>;tag=1
Contact: <sip:[email protected] <sip%[email protected]>>
Call-ID: [email protected]
CSeq: 102 INFO
User-Agent: Asterisk PBX 1.6.2.0
Content-Type: application/media_control+xml
Content-Length: 205
<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>
'.
[/code]
If i disable the video support without the following lines
[code] m=video [media_port] RTP/AVP 115
a=fmtp:115 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
a=rtpmap:115 H263-1998/90000
a=sendrecv [/code]
the DTMF mode works well and there is no error.
The problem it's difficult can you help me?
Thank you very much
--
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: [email protected]
_______________________________________
--
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: [email protected]
_______________________________________
------------------------------------------------------------------------------
Throughout its 18-year history, RSA Conference consistently attracts the
world's best and brightest in the field, creating opportunities for Conference
attendees to learn about information security's most important issues through
interactions with peers, luminaries and emerging and established companies.
http://p.sf.net/sfu/rsaconf-dev2dev
_______________________________________________
Sipp-users mailing list
[email protected]
https://lists.sourceforge.net/lists/listinfo/sipp-users
