![]() ![]() 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 ( ). M2Msoft was also one of the first on the development of specific projects for the military sector, ensuring expertise, reactivity,and customization to its clients.I need to make a simple softphone based on the PJSIP Library to make calls via Asterisk server. M2Msoft has quickly become a major actor on its domain offering more and more advanced solutions. Its range of products completely responds to the different needs of the market. Editor of voice and video over IP software since 2003, M2Msoft is developing solutions for all types of businesses. M2Msoft is the French leader for customized IP solutions. Thanks to contact M2Msoft for any information : the developer: Codecs G711A, G711U, G722 in basic version (G729 in Pro version)Ī professional WiZiN-Pro version is available, it is especially designed for organizations, professionals and VIPs, it contains advanced features such as Encryption and secured communications, G729 codec, multi-conferences, customized functions, and extended capacities (8 lines, unlimited discovery, unlimited call duration on demonstration server, etc.). Sending/receiving of SMS with conversations history Calls without server (exploration function) The incoming/outgoing calls and SMS use the history, the business directory and the personal contacts. WiZiN integrates an exploration function on the wifi network allowing the discovery of other WiZiNs’ with whom you can make calls or send/receive SMS messages without a server. WiZiN also enables the sending and receiving of SMS messages on SIP protocol. With the multi lines function, WiZiN offers the opportunity to commute lines, transfer and forward lines just like a real office phone. Interoperable with all the SIP servers of the market, it is also compatible with all the service providers and you can use it straight away. Get connected to your business server to get free calls on the Wifi or 3G/4G networks enjoying a high audio quality. WiZiN the smart SIP softphone by M2Msoft. ![]()
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