![]() ![]() To diagnose this further would require see the pjsip log output. Feel free to contact us with support questions or for more information on whitelabel solutions. Since your example seems to not provide the port information it should work. Zoiper is an easy to use sip video softphone, with excellent voice quality and easy to setup. ![]() 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("") The service is free to use based on a fair-use policy and federates with publicly reachable SIP and XMPP domains. Server: fritz.box Address: fd00::2665:11ff:fef9:ec51 SIP2SIP is a real time communications service for audio, video, presence, chat, file transfer and multiparty conferencing based on WebRTC, SIP, XMPP and related protocols. 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. Check the relevant headers for port 5060.I am using PJSIP for voice calling. (With luck, your issue may be different from the above, in which case there may be an easy solution. Or, connect through a VPN so the endpoint does not appear to be behind NAT. If you are using a non-standard port for security reasons, reverting to port 5060 but with more restrictive firewall rules might be a workaround. I don’t know whether this bug has ever been properly reported, or even whether it is in Asterisk or in pjsip (I’m reasonably certain that it is not in FreePBX). You can useĪt the Asterisk console to see if any Via or Contact headers sent by Asterisk incorrectly show port 5060. Several users have reported issues with the combination of TCP transport, non-standard bind port and Asterisk behind NAT. I doubt that this issue is related to DTMF. ![]()
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