![]() ![]() Acc 0: Registration sentĢ2:41:49.991 pjsua_call.c Making call with acc #0 to pjsua_aud.c. Via: SIP/2.0/UDP 192.168.56.1:5060 rport branch=z9hG4bKPjjg7lshYOjbgZMpe-G9PgeT2hJqNpcRdvįrom: tag=9j1S8GzDdF6Vivl9FKqV3up5oVAenSzjĬall-ID: It9SK0FiOteFTrpnKmfLCl6jYX0fHaKAĪllow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONSĢ2:41:49.991 pjsua_acc.c. PJSUA state changed: STARTING -> RUNNINGĢ2:41:49.991 pjsua_acc.c Adding account: pjsua_acc.c. Module "mod-unsolicited-mwi" registeredĢ2:41:49.991 pjsua_core.c. ![]() 1 SIP worker threads createdĢ2:41:49.990 pjsua_core.c. ![]() Module "mod-pjsua-options" registeredĢ2:41:49.990 pjsua_core.c. Module "mod-pjsua-im" registeredĢ2:41:49.990 sip_endpoint.c. Module "mod-pjsua-pres" registeredĢ2:41:49.990 sip_endpoint.c. Module "mod-refer" registeredĢ2:41:49.990 sip_endpoint.c. Module "mod-presence" registeredĢ2:41:49.990 sip_endpoint.c. Module "mod-evsub" registeredĢ2:41:49.990 sip_endpoint.c. PortAudio sound library initialized, status=0Ģ2:41:49.975 pa_dev.c. Module "mod-pjsua" registeredĢ2:41:49.916 sip_endpoint.c. Module "mod-stateful-util" registeredĢ2:41:49.916 sip_endpoint.c. Module "mod-tsx-layer" registeredĢ2:41:49.916 sip_endpoint.c. Module "mod-pjsua-log" registeredĢ2:41:49.916 sip_endpoint.c. PJSUA state changed: NULL -> CREATEDĢ2:41:49.916 sip_endpoint.c. Module "mod-msg-print" registeredĢ2:41:49.916 sip_transport.Transport manager created.Ģ2:41:49.916 pjsua_core.c. Creating endpoint instance.Ģ2:41:49.916 pjlib. NOTICE: chan_sip.c:10460 process_sdp: No compatible codecs, not accepting this offer!Īnd the following log from the PJSIP: 22:41:49.914 os_core_unix.c !pjlib 2.4 for POSIX initializedĢ2:41:49.916 sip_endpoint.c. When I call from my softphone to Zoiper, I get the following message from the Asterisk server: = Using SIP RTP TOS bits 184 I've read that this error may occur because Asterisk does not support the codec used in my softphone, but I don't know how to solve this problem. = Everyone is busy/congested at this time (1:0/0/1)ġ92.168.56.1 is the IP-address of my softphone. When I call from Zoiper to my softphone, I get the following message from the Asterisk. This softphone can register on the Asterisk server (to make it work I replaced in the line 163 substring SIP_DOMAIN to "asterisk"), but can't make and receive calls. I used a simple example from PJSIP official site written on C ( ). I need to make a simple softphone based on the PJSIP Library to make calls via Asterisk server. ![]()
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