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Hi all! This is my VoIP network scheme H323EndPoint ----- --- GW H323/SIP-IN -- -- SIP Phone | | (Sipquest) | | | | | | | | | | H323EndPoint --------- GK1 ---- GK2-| |-- SER ---- SIP Phone | | | | | | | | | | | | H323EndPoint ----- --- GW H323/SIP-OUT-- -- Asterisk as Voicemail (Sipquest) In calls between SIP to H323 endpoints it works fine . I have a problem in calls between H323 endpoints with asterisk voicemail functionality. In case of not response, the call is forwarded to asterisk voicemail by SER Router but I obtain the following error: -- Executing Set("SIP/X.X.X.X-085340d0", "LANGUAGE()=es") in new stack -- Executing SetCallerID("SIP/X.X.X.X-085340d0", "331223") in new stack -- Executing VoiceMail("SIP/X.X.X.X-085340d0", "[EMAIL PROTECTED]") in new stack -- Playing 'vm-theperson' (language 'es') -- Playing 'digits/3' (language 'es') -- Playing 'digits/3' (language 'es') -- Playing 'digits/1' (language 'es') -- Playing 'digits/2' (language 'es') -- Playing 'digits/2' (language 'es') -- Playing 'digits/2' (language 'es') -- Playing 'vm-isunavail' (language 'es') Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor: Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback: Failed to write frame == Spawn extension (default, 331222, 3) exited non-zero on 'SIP/172.25.92.153-085340d0' The channels has RTP activity because I hear the voicemail message Someone has an idea to arrange this problem Thanks in advance! |
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