Video Calls mit Asterisk?

Hi,

ich bin heut in der sip.conf über die Option "supportvideo=no" gestossen. :lol:

Also schnell mal auf yes gesetzt, auf voip-info.org gelesen, dass ich bei den codecs noch allow=261 und allow263 und allow=263p setzen muss und dann fröhlich mit eyebeam probiert.

Ich bekomm aber immer nur Audiogespräche hin, drücke ich auf "Start Video" dann sagt er: "Remote Party does not support video". Im Asterisk CLI kommt nix währenddessen :-(

Was mach ich falsch?


Gruss
Frank

 

Moin,

@Maik: meine allow Zeilen stimmen in der sip.conf, ich hatte nur hier im Postimng jeweils das "h" vergessen.

meine sip Teilnehmer habe ich mit Realtime ausgelagert in eine MySQL Tabelle. die general Einstellungen in der Datei sip.conf sehen so aus:

Code:

[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
                                ; if asterisk was compiled with OSP support.
;realm=mydomain.tld             ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet

;pedantic=yes                   ; Enable slow, pedantic checking for Pingtel
                                ; and multiline formatted headers for strict
                                ; SIP compatibility (defaults to "no")
;tos=184                        ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none
;maxexpirey=3600                ; Max length of incoming registration we allow
;defaultexpirey=1800            ; Default length of incoming/outoing registration
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
videosupport=yes                ; Turn on support for SIP video
;recordhistory=yes              ; Record SIP history by default
                                ; (see sip history / sip no history)

disallow=all                    ; First disallow all codecs
allow=h261
allow=h263
allow=h263p
allow=alaw                      ; Allow codecs in order of preference
allow=ulaw
allow=gsm
allow=ilbc                      ;
;musicclass=default             ; Sets the default music on hold class for all SIP calls
                                ; This may also be set for individual users/peers
language=de                     ; Default language setting for all users/peers
                                ; ******** CHANGE THIS TO 'DE' WHEN TIME IS RIGHT ****************

                                ; This may also be set for individual users/peers
relaxdtmf=yes                   ; Relax dtmf handling

;rtptimeout=60  ;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP activity
                                ; when we're on hold (must be > rtptimeout)
;trustrpid = no                 ; If Remote-Party-ID should be trusted
;progressinband=never           ; If we should generate in-band ringing always
                                ; use 'never' to never use in-band signalling, even in cases
                                ; where some buggy devices might not render it
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
                                ; Note that promiscredir when redirects are made to the
                                ; local system will cause loops since SIP is incapable
                                ; of performing a "hairpin" call.
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
                                ; a valid phone number
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
                                ; Other options:
                                ; info : SIP INFO messages
                                ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)

;compactheaders = yes           ; send compact sip headers.
                ; Terminate call if 60 seconds of no RTP activity
nat=no

Ich hab auf voip-info.org noch was von einem H263+ Patch gelesen. Muss ich den nachträglich installieren?

EDIT:

hab jetzt mal GENAU die Config von voip-info in meine sip.conf eingefügt. Nun bekomm ich im CLI folgende Fehlermeldung wenn ich einen Anruf mache:

Code:

Executing Dial("SIP/102-85de", "SIP/103")
Aug 24 11:39:14 WARNING[32383]: channel.c:321 ast_best_codec: Don't know any of 0x80000 formats
    -- Called 103
Aug 24 11:39:14 WARNING[32383]: channel.c:2297 ast_channel_make_compatible: No path to translate from SIP/103-3345(524288) to SIP/102-85de(4)
    -- SIP/103-3345 is ringing
    -- SIP/103-3345 answered SIP/102-85de
Aug 24 11:39:20 WARNING[32383]: channel.c:2297 ast_channel_make_compatible: No path to translate from SIP/102-85de(4) to SIP/103-3345(524288)
Aug 24 11:39:20 WARNING[32383]: app_dial.c:1319 dial_exec_full: Had to drop call because I couldn't make SIP/102-85de compatible with SIP/103-3345

Hmm was will er mir denn damit sagen? 0x8000 ist doch laut CLI Ausgabe:

*CLI> show codecs
524288 (1 << 19) (0x80000) video h263 (H.263 Video)


Gruss
Frank