9/4/2023 0 Comments Zoiper pjsipDEBUG channel.c: Channel PJSIP/6001-00000002 setting write format path: ulaw -> ulaw DEBUG res_pjsip/pjsip_resolver.c: Target '192.168.0.7' is an IP address, skipping resolution DEBUG res_pjsip/pjsip_resolver.c: Transport type for target '192.168.0.7' is 'UDP transport' DEBUG res_pjsip/pjsip_resolver.c: Performing SIP DNS resolution of target '192.168.0.7' DEBUG res_pjsip_session.c: Applied negotiated SDP media stream 'audio' using audio SDP handler DEBUG channel.c: Channel PJSIP/6002-00000003 setting write format path: ulaw -> ulaw DEBUG channel.c: Channel PJSIP/6002-00000003 setting read format path: ulaw -> ulaw DEBUG rtp_engine.c: Copying tx payload mapping 101 (0x7f4da803f928) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Copying tx payload mapping 3 (0x5602af20f618) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Copying tx payload mapping 0 (0x5602af20f5c8) from 0x7f4e04ce6df0 to 0x7f4da807ee88 DEBUG rtp_engine.c: Setting tx payload type 3 based on m type on 0x7f4e04ce6df0 DEBUG rtp_engine.c: Setting tx payload type 0 based on m type on 0x7f4e04ce6df0 DEBUG res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f4da807ecb0' DEBUG res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7f4da807ecb0' Here is the full log after the call goes through: DEBUG res_pjsip_session.c: Applying negotiated SDP media stream 'audio' using audio SDP handler 192.168.0.2 in both clients as host then everything is perfect. However, when I put both clients behind the NAT use local ip i.e. Problem I am having is when one client calls the other, the call goes through, but no audio is present in either client. I have configured my router to forward 5060/UDP and ports 10000-20000/UDP as well. Another Client is an iPhone running on 4G network. My Asterisk and one of the clients using Zoiper Softphone are behind NAT. Zoiper is also available as an OEM / whitelabel solution license and can be customized on demand.īe sure to configure ios to allow notifications for incoming calls for Zoiper and provide access to the contact list.I have just installed and configured Asterisk 17 in a desktop PC running Ubuntu 18.4 * The 3rd party patented codecs h264 and g729 are available as optional in app purchase. Please check user's Account Setting to disable the auto-renewal.
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