Hi Alberto,
Here's the output that you asked for:
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 11.3.0
SDP Session Name: Asterisk PBX 11.3.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|h263|testlaw)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Record on feature: automon
Record off feature: automon
Force rport: Auto (No)
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language:
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----
On Mon, Apr 29, 2013 at 7:12 PM, Alberto Llamas <albertollam...@gmail.com>wrote:
> Cual es el resultao de:
>
> *sip show settings*
>
>
> El 29/04/13 9:10, Moosa Khalid escribió:
>
> I'm trying to make a video call between two SIP peers registered on a
> single asterisk box. Following is the config of my sip peers
>
> [107]
> defaultuser=107
> secret=107
> type=friend
> host=dynamic
> context=default
> canreinvite=yes
> videosupport=yes
> dtmfmode=rfc2833
> qualify=yes
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=h263
>
> [701]
> defaultuser=701
> secret=701
> type=friend
> host=dynamic
> context=default
> videosupport=yes
> qualify=yes
> canreinvite=yes
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> allow=h263
>
> I've also enabled videosupport in the global sip settings.
> I'm using the softphone Xlite 4.5 which supports h263 codec on for both
> clients. *Asterisk version is 11.3.0 LTS*. Clients registered are on same
> network. Asterisk shows the following output on console on a call b/w
> peers. Audio is fine but of course no video thanks to the following
> warning.
>
> * == Using SIP VIDEO CoS mark 6*
> * == Using SIP RTP CoS mark 5*
> * -- Executing [96107@default:1] Dial("SIP/701-00000014",
> "SIP/107,20,rt") in new stack*
> * == Using SIP VIDEO CoS mark 6*
> * == Using SIP RTP CoS mark 5*
> * -- Called SIP/107*
> * -- SIP/107-00000015 is ringing*
> * -- SIP/107-00000015 is ringing*
> *[Apr 29 18:35:12] WARNING[11363][C-0000000d]: chan_sip.c:10141
> process_sdp: Ignoring video stream offer because port number is zero*
> *
> *
> This happens regardless of which sip peer originates the call.
>
>
>
> --
> _____________________________________________________________________
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>
> --
> Alberto Llamas
> Ingeniero de Telecomunicaciones
> Digium Certified Asterisk Administrator
> Digium Certified Asterisk Professional
> Linux Administrator
>
>
> --
> _____________________________________________________________________
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> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
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