09-12-2020 04:14 PM
Greetings all. I'm experienced with devops but new to the telephony / SIP world. I am trying to connect an 8841 phone to a vanilla FreePBX install running on a Raspberry Pi.
Configuration
Issue description
I set the TFTP server address manually on the 8841, then spun up a TFTP server on the Pi to serve a sepmac.cnf.xml file (attached below) gleaned from the various ones floating around this forum. 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 installed the Asterisk/FreePBX distro, and enabled TCP and UDP pjsip, both on port 5060. I created two SIP extensions and confirmed I was able to connect to them with Zoiper / Linphone and make calls back and forth.
However, I have not been able to get the phone to register. On the phone's status screen, it just keeps repeating:
[timestamp] Error updating locale [timestamp] Error updating locale [timestamp] VPN not configured [timestamp] SEPxxxxxxxxxxx.cnf.xml(TFTP)
...over and over.
Meanwhile, the Asterisk logs show the following:
[2020-09-12 23:16:41] ERROR[1185]: pjproject: <?>: sip_transport. 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: R?h?V????????8.19.189 SIP/2.0 Via: SIP/2.0/UDP 192.168.19.187:5060;branch=z9hG4bK0691a84a From: "" <sip:@192.168.19.187>;tag=ac7e8a2bfaf900d4617722f7-46c21df8 To: <sip:192.168.19.189> Call-ID: ac7e8a2b-faf900d4-0d22d180-744c4a6f@192.168.19.187 Date: Sat, 12 Sep 2020 20:43:42 GMT CSeq: 1000 REFER User-Agent: Cisco-CP8841/10.3.1 Expires: 10 Max-Forwards: 70 Contact: <sip:@192.168.19.187:5060;transport=udp> Require: norefersub Referred-By: "" <sip:@192.168.19.187> Refer-To: cid:3cad1cf0@192.168.19.187 Content-Id: <3cad1cf0@192.168.19.187> Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Content-Length: 509 Content-Type: application/x-cisco-alarm+xml Content-Disposition: session;handling=required <?xml version="1.0" encoding="UTF-8"?> <x-cisco-alarm> <Alarm Name="DeviceTLInfo"> <ParameterList> <String name="DeviceName">SEPAC7E8A2BFAF9</String> <String name="IPv4Address">192.168.19.187</String> <String name="IPv6Address"></String> <String name="CTL_Signature">Not Installed</String> <String name="CTL_TFTP_Server">N/A</String> <String name="ITL_Signature">Not Installed</String> <String name="ITL_TFTP_Server">N/A</String> <String name="StatusCode">6</String> </ParameterList> </Alarm> </x-cisco-alarm> -- end of packet.
...over and over.
At this point I am completely out of ideas, having tried various things that I found in this forum (changing <transportLayerProtocol>, etc.) without success. Any pointers or suggestions would be very much appreciated.
Thanks!
09-12-2020 05:02 PM
Okay, looking at the Asterisk log suggested to me that the issue was the garbled junk at the beginning of the "Request Line" request. I thought that might be related to UDP so I changed the transportLayerProtocol to TCP (1), and I can now connect to Asterisk.
I am now having two new issues:
1. I can dial into the 8841's extension from a Linphone client. The 8841 rings and I can answer it with the Answer softkey, which puts the call on speakerphone. However, if I pick up the handset I can neither hear nor talk.
2. I cannot call the Linphone extension (102) from the 8841. When I dial that number I get the following error in the Asterisk log:
[2020-09-13 01:00:21] NOTICE[16847]: res_pjsip/pjsip_distributor.c:676 log_failed_request:
Request 'INVITE' from '"Anonymous" <sip:Anonymous@192.168.19.189>' failed for '192.168.19.187:50640'
(callid: ac7e8a2b-faf90010-415e8743-07315e62@192.168.19.187) - Failed to authenticate
As before, any suggestion much appreciated!
04-13-2022 12:55 PM
Changing <transportLayerProtocol> to 1 worked for me too (after enabling TCP in PJSIP). Thank you!
09-12-2020 05:17 PM
Okay, well: the answer to #1 was "plug the handset in correctly".
As for #2, I enabled Anonymous Inbound SIP calls and Allow SIP Guests in the Asterisk settings, which allowed me to make calls to other extensions. But seems like I should configure authentication from the 8841 if that is possible in the xml file.
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide