ich habe hier ein merkwürdiges Phänomen an einem GXP2200 als SIP-Client an einer Asterisk, zunächst die SIP-Daten des Telefons:
Folgendes Verhalten fällt beim Vermitteln auf:
Szenario also: GXP2200 wählt abgehend eien Rufnummer über CAPI (ISDN), die Verbindung kommt zustande. Danach versucht das GXP2200 den Anruf weiterzuvermitteln (was dank Asterisk-Dialoption T auch erlaubt ist), das geht aber schief. Statt jetzt DTMFs für den Vermittliungscode zu erhalten, gibt es den Fehler
Kennt jemand das Problem und weis eine Lösung ?
Code:
* Name : 13
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : sip_phone
Record On feature : automon
Record Off feature : automon
Subscr.Cont. : <Not set>
Language : de
Tonezone : <Not set>
AMA flags : BILLING
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromDomain : geheim
Callgroup :
Pickupgroup :
Named Callgr :
Nam. Pickupgr:
MOH Suggest :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Busy level : 2
Dynamic : Yes
Callerid : geheim
MaxCallBR : 384 kbps
Expire : 1313
Insecure : no
Force rport : No
Symmetric RTP: No
ACL : No
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: 400
DirectMedia : No
PromiscRedir : No
User=Phone : Yes
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 192.168.55.136:18284
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 13
SIP Options : (none)
Codecs : (alaw)
Codec Order : (alaw:20)
Auto-Framing : No
Status : OK (1 ms)
Useragent : Grandstream GXP2200 1.0.3.26
Reg. Contact : geheim
Qualify Freq : 60000 ms
Keepalive : 0 ms
Sess-Timers : Refuse
Sess-Refresh : uac
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : Yes
Encryption : NoCode:
[Apr 15 09:39:59] VERBOSE[10145][C-000003b9] pbx.c: -- Executing [XXXXXXX@sip_phone:36] Dial("SIP/13-000003ae", "Capi/g1/XXXXXXX/b,30,ciHT") in new stack
[Apr 15 09:39:59] VERBOSE[10145][C-000003b9] app_dial.c: -- Called Capi/g1/XXXXXXX/b
[Apr 15 09:40:00] VERBOSE[10145][C-000003b9] app_dial.c: -- CAPI/ISDN1#02/XXXXXXX-6b4 is proceeding passing it to SIP/13-000003ae
[Apr 15 09:40:01] VERBOSE[10145][C-000003b9] app_dial.c: -- CAPI/ISDN1#02/XXXXXXX-6b4 is making progress passing it to SIP/13-000003ae
[Apr 15 09:40:01] VERBOSE[10145][C-000003b9] app_dial.c: -- CAPI/ISDN1#02/XXXXXXX-6b4 is ringing
[Apr 15 09:40:01] VERBOSE[10145][C-000003b9] res_rtp_asterisk.c: > 0xb6f19b78 -- Probation passed - setting RTP source address to XXX.XXX.XXX.XXX:XXXXX
[Apr 15 09:40:03] VERBOSE[10145][C-000003b9] app_dial.c: -- CAPI/ISDN1#02/XXXXXXX-6b4 answered SIP/13-000003ae
[Apr 15 09:40:15] VERBOSE[10145][C-000003b9] res_musiconhold.c: -- Started music on hold, class 'default', on CAPI/ISDN1#02/XXXXXXX-6b4
[Apr 15 09:40:35] ERROR[10145][C-000003b9] res_rtp_asterisk.c: Received SSL traffic on RTP instance '0xb6f1566c' without an SSL session
[Apr 15 09:40:35] WARNING[10145][C-000003b9] res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up.
[Apr 15 09:40:35] VERBOSE[10145][C-000003b9] pbx.c: -- Executing [h@sip_phone:1] Hangup("SIP/13-000003ae", "") in new stackCode:
res_rtp_asterisk.c: Received SSL traffic on RTP instance '0xb6f1566c' without an SSL session