![]() ![]() Channel PJSIP/siptrunk-00000000 swapped with into 'simple_bridge' basic-bridge ![]() Channel left 'simple_bridge' basic-bridge Channel PJSIP/zoiper-00000001 joined 'simple_bridge' basic-bridge requested media update control 26, passing it to PJSIP/zoiper-00000001 Channel joined 'simple_bridge' basic-bridge Executing "PJSIP/zoiper,20") in new stack Channel PJSIP/siptrunk-00000000 joined 'simple_bridge' basic-bridge Executing "CHANNEL(hangup_handler_push)=hdlr,s,1(args)") in new stack PJSIP/siptrunk-00000000 answered answered > 0x7f896400e130 - Strict RTP qualifying stream type: audio ![]() PJSIP/siptrunk-00000000 is making progress passing it to is making progress Called PJSIP/siptrunk-00000000 is making progress passing it to 0x7f896400e130 - Strict RTP learning after remote address set to: some_ip:16886 Via: SIP/2.0/UDP 192.168.19.187:5060 branch=z9hG4bK0691a84aįrom: "" tag=ac7e8a2bfaf900d4617722f7-46c21df8Īllow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBEĬontent-Type: application/x-cisco-alarm+xmlĬontent-Disposition: session handling=requiredĪt this point I am completely out of ideas, having tried various things that I found in this forum (changing, etc.) without success. Error processing 1262 bytes packet from UDP 192.168.19.187:5060 : PJSIP syntax error exception when parsing 'Request Line' header on line 1 col 1: Meanwhile, the Asterisk logs show the following: ERROR: pjproject: : sip_transport. On the phone's status screen, it just keeps repeating: However, I have not been able to get the phone to register. I created two SIP extensions and confirmed I was able to connect to them with Zoiper / Linphone and make calls back and forth. I installed the Asterisk/FreePBX distro, and enabled TCP and UDP pjsip, both on port 5060. Most of the ones I tried did not work and resulted in an error entry in the phones status log, but I got one to work - and by "work" I mean that the phone acknowledges it without error. ![]() I set the TFTP server address manually on the 8841, then spun up a TFTP server on the Pi to serve a file (attached below) gleaned from the various ones floating around this forum. pjsip with UDP and TCP transports enabled on port 5060, TLS on 5061.I am trying to connect an 8841 phone to a vanilla FreePBX install running on a Raspberry Pi. I'm experienced with devops but new to the telephony / SIP world. ![]()
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