History viewer for incoming, outgoing, missed calls and recordings. If you are looking for a solid feature rich and professional VoIP app this is it!This app has a lot of great features like, call recording, multiple accounts, Address Book integration. This is the Ferrari in VoIP applications. Try with the Lite version to be sure it's to your liking. It's early days for the product, but is getting consistently better. The call quality, stats, logging and "uptime" more than compensate. The UI is overall quite clunky compared to others - for example, the history view needs a little too much interpretation, there's no "redial" and there's no simple hotkey for "quick, dial me a number". The multi-way calling is great, but hold is a little flaky. Compared to the competition, it scores about 98% uptime compared to about 80% for the next-best. This is important to me since I have to rely on this for important incoming calls. Uptime for me is a combination of registration, stay-alives, non-dropped calls and other operations that keeps the phone ready to ring and keeps the call active. Generally, most softphones are great with outbound calls, but Blink Pro wins on incoming call "uptime". To diagnose this further would require see the pjsip log output.I use this as my main "desk phone" on my Mac after failed attempts with iSoftphone, Zoiper, Telephone and others. Since your example seems to not provide the port information it should work. Sip2Sip also support STUN setup, so I would also setup the STUN settings on the account as well: cfg.stun_srv_cnt = 1 Ĭfg.stun_srv = pj_str("") If they ever add more then it would become more useful. You could also take it further to support failover, but it looks like sip2sip doesn't have multiple sip servers in there DNS SRV record so it will not be used currently. use results to fill in the outbound_proxy Static void resolver_cb_func( pj_status_t status, void *token, const struct pjsip_server_addresses *addr) Pjsip_resolve(resolver_, pool, &host, token, resolver_cb_func) Host.type = PJSIP_TRANSPORT_UDP // if using UDP, see pjsip_transport_type_e Host.flag = PJSIP_TRANSPORT_DATAGRAM // is using UDP, see pjsip_transport_flags_e Status = pjsip_resolver_create( pool, &resolver_ ) If you want a more robust solution, you need to use pjsip's SRV resolution functions to resolve srv record e.g: "_sip._" and then set the outbound_proxy records with the result. cfg.outbound_proxy_cnt = 1 Ĭfg.outbound_proxy = pj_str("sip::5060") If you want a "quick and easy" setup, what you want to do is set the outbound_proxy to "". "What we've been suggesting is to implement the failover mechanism in the application layer." What PJSUA will not support automatically is failover support, they say: In PJSUA it will only do DNS SRV lookup if you don't provide the port number in the SIP will try to do a DNS SRV record lookup first then fail over to DNS A/C name will only do DNS A/C name lookup. " the SIP device must always perform DNS lookups as defined in SIP standard RFC3263 (NAPTR + SRV + A DNS lookups)" If you read the Sip2Sip device configuration page it states that: I noted that we need to use outbound proxy in sip2sip called "".īut confused how can i used in pjsip code. Status = pjsua_acc_add(&cfg, PJ_TRUE, &_acc_id) char cfg_reg_uri = "sip:" Ĭfg.cred_info.realm = pj_str(cfg_cred_realm) Ĭfg.cred_info.scheme = pj_str(cfg_cred_scheme) Ĭfg.cred_ername = pj_str(cfg_cred_username) Ĭfg.cred_info.data_type = PJSIP_CRED_DATA_PLAIN_PASSWD Ĭfg.cred_info.data = pj_str(cfg_cred_password) i saw log in there wasn't log of outgoing or incoming call. Registration is OK But Call is not connected. When i used our server, everything fine i.e.
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