Closed derekcrpd closed 4 months ago
Sorry for posting the same image twice. This was supposed to have been the second image, which confirms that the feature code is correct.
Hi @derekcrpd , Can you please give a try again by upgrade all modules to latest
Updated all modules. Even rebooted the system. Still receiving the same error.
We haven't seen any real details or data from all this. Show a verbose call log to show how it is being handled.
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-0000003d to use for Request msg INVITE/cseq=241 (rdata0x7f9b8429cb98)
[2024-05-28 09:02:54] DEBUG[1751]: chan_pjsip.c:2904 chan_pjsip_session_begin: 1000
[2024-05-28 09:02:54] DEBUG[1751]: chan_pjsip.c:2908 chan_pjsip_session_begin: Direct media no glare mitigation
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:3976 new_invite: 1000
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4065 new_invite: 1000: Call (TLS:<
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4825 session_inv_on_tsx_state_changed: 1000 TSX State: Proceeding Inv State: INCOMING
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4445 print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4459 print_debug_details: The state change pertains to the endpoint '1000()'
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4464 print_debug_details: The inv session still has an invite_tsx (0x7f9b382de758)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4470 print_debug_details: The UAS INVITE transaction involved in this state change is 0x7f9b382de758
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4474 print_debug_details: The current transaction state is Proceeding
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4476 print_debug_details: The transaction state change event is TX_MSG
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4481 __print_debug_details: The current inv state is INCOMING
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:5014 session_inv_on_tsx_state_changed: Nothing delayed
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4245 session_on_tsx_state: 1000 TSX State: Proceeding Inv State: INCOMING
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4249 session_on_tsx_state: Topology: Pending: (null topology) Active: (null topology)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4254 session_on_tsx_state:
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:769 handle_incoming_sdp: 1000: Media count: 1
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:795 handle_incoming_sdp: 1000: Processing stream 0
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:832 handle_incoming_sdp: 1000: Using audio-0 for new stream name
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:876 handle_incoming_sdp: 1000: Using new stream 0:audio-0:audio:sendrecv (nothing)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:495 ast_sip_session_media_state_add: 1000 Adding position 0
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:541 ast_sip_session_media_state_add: Creating new media session
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:584 ast_sip_session_media_state_add: Setting media session as default for audio
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:589 ast_sip_session_media_state_add: Done
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:929 handle_incoming_sdp: 1000: Negotiating incoming SDP media stream 0:audio-0:audio:sendrecv (nothing) using audio SDP handler
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_sdp_rtp.c:1501 negotiate_incoming_sdp_stream: 1000
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_sdp_rtp.c:1515 negotiate_incoming_sdp_stream: Incompatible transport
[2024-05-28 09:02:54] ERROR[1751]: res_pjsip_session.c:937 handle_incoming_sdp: 1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:937 handle_incoming_sdp: 1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:960 handle_incoming_sdp: 1000: Handled? no
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4591 handle_outgoing_response: 1000: Method is INVITE, Response is 488 Not Acceptable Here
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4610 handle_outgoing_response: 1000
[2024-05-28 09:02:54] DEBUG[1751]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '<
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4825 session_inv_on_tsx_state_changed: 1000 TSX State: Completed Inv State: DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4445 print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4459 print_debug_details: The state change pertains to the endpoint '1000()'
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4464 print_debug_details: The inv session still has an invite_tsx (0x7f9b382de758)
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4470 print_debug_details: The UAS INVITE transaction involved in this state change is 0x7f9b382de758
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4474 print_debug_details: The current transaction state is Completed
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4476 __print_debug_details: The transaction state change event is TX_MSG
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4481 print_debug_details: The current inv state is DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4847 session_inv_on_tsx_state_changed: Disconnected
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4245 session_on_tsx_state: (null session) TSX State: Completed Inv State: DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4254 session_on_tsx_state:
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4110 new_invite: 1000
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4201 handle_new_invite_request: Request: Session: 1000
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:4319 session_on_rx_request: (null session) Handled request INVITE ? yes
[2024-05-28 09:02:54] DEBUG[1751]: chan_pjsip.c:2925 chan_pjsip_session_end: 1000
[2024-05-28 09:02:54] DEBUG[1751]: chan_pjsip.c:2928 chan_pjsip_session_end: No channel
[2024-05-28 09:02:54] DEBUG[1751]: res_pjsip_session.c:2912 session_destructor: 1000: Destroying SIP session
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip/pjsip_distributor.c:502 distributor: Searching for serializer associated with dialog dlg0x7f9b382e6c58 for Request msg ACK/cseq=241 (rdata0x7f9b84148f18)
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip/pjsip_distributor.c:471 ast_sip_get_distributor_serializer: Calculated serializer pjsip/distributor-0000003d to use for Request msg ACK/cseq=241 (rdata0x7f9b84148f18)
[2024-05-28 09:02:54] DEBUG[1751]: netsock2.c:170 ast_sockaddr_split_hostport: Splitting '<
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4825 session_inv_on_tsx_state_changed: (null session) TSX State: Terminated Inv State: DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4445 print_debug_details: Function session_inv_on_tsx_state_changed called on event TSX_STATE
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4457 print_debug_details: inv_session 0x7f9b38303448 has no ast session
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4467 __print_debug_details: The inv session does NOT have an invite_tsx
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4470 print_debug_details: The UAS INVITE transaction involved in this state change is 0x7f9b382de758
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4474 print_debug_details: The current transaction state is Terminated
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4476 print_debug_details: The transaction state change event is TIMER
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4481 print_debug_details: The current inv state is DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4837 session_inv_on_tsx_state_changed: Session ended
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4245 session_on_tsx_state: (null session) TSX State: Terminated Inv State: DISCONNCTD
[2024-05-28 09:02:54] DEBUG[1750]: res_pjsip_session.c:4254 session_on_tsx_state:
[2024-05-28 09:02:58] DEBUG[1762]: res_pjsip_registrar.c:1377 check_expiration_thread: Woke up at 1716901378 Interval: 30
[2024-05-28 09:02:58] DEBUG[1762]: res_pjsip_registrar.c:1384 check_expiration_thread: Expiring 0 contacts
No, that is a debug not a call log. You don't need debug output for this. So do asterisk -rvvvvvvvvv and don't turn on debug.
With just that I receive the following when I attempt to activate it: ERROR[72748]: res_pjsip_session.c:937 handle_incoming_sdp: 1000: Couldn't negotiate stream 0:audio-0:audio:sendrecv (nothing)
This is a codec issue. What codecs are being used on the device and what codecs does the PJSIP endpoint have allowed?
Just the basic ones that are enabled by default... ulaw, alaw, gsm, g726, and g722. The endpoint supports g711u, g729a, g722 and gsm. But is a FreePBX feature code being sent all the way through the trunk? As far as the system recording that are attached to the call flow... I tried changing them back to and from default to see if that made any difference, though it did not.
Also forgot to mention. The phone itself supports g722 and g726
Ok, feel a bit stupid on this one. The issue had to do with SRTP encryption. It was not enabled on the line that was attempting to set the call flow control. Once it was set to "enabled and forced" on the phone then everything works fine. msanthosh18, thank you for putting up with me on this one. :)
FreePBX Version
FreePBX 17
Issue Description
I am using a FreePBX17 install that is less than 1 week old. I have set a flow control and the BLF works as expected. However, when I push the button (which in fact sends the expected *280 to Asterisk), I receive a 488 error. I can manually change the status of the flow control through FreePBX, just not with my Grandstream phone using either the designated button, or the designated feature code. The set up is identical (I believe) to the setup I was using under FreePBX16 which worked fine.
I noticed this in the log: Using new stream 0:audio-0:audio:sendrecv (nothing) Out of an abundance of caution I tried removing my custom audio (set through System Recordings) and set it to "default"; however, same issue.
Operating Environment
FreePBX 17; Asterisk 21.0.2
+-------------------+------------+-----------------------------------+------------+-----------+ | Module | Version | Status | License | Signature | +-------------------+------------+-----------------------------------+------------+-----------+ | amd | 17.0.1 | Enabled | GPLv3+ | Sangoma | | announcement | 17.0.2.1 | Enabled | GPLv3+ | Sangoma | | api | 17.0.1.2 | Enabled | AGPLv3+ | Sangoma | | arimanager | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | asterisk-cli | 17.0.2 | Enabled | GPLv3+ | Sangoma | | asteriskinfo | 17.0.1 | Enabled | GPLv3+ | Sangoma | | backup | 17.0.5.29 | Enabled | GPLv3+ | Sangoma | | blacklist | 17.0.1.2 | Enabled | GPLv3+ | Sangoma | | builtin | | Enabled | | Unsigned | | bulkhandler | 17.0.4 | Enabled | GPLv3+ | Sangoma | | calendar | 17.0.4.13 | Enabled | GPLv3+ | Sangoma | | callback | 17.0.2.1 | Enabled | GPLv3+ | Sangoma | | callforward | 17.0.1.4 | Enabled | AGPLv3+ | Sangoma | | callrecording | 17.0.3.6 | Enabled | AGPLv3+ | Sangoma | | callwaiting | 17.0.3.3 | Enabled | GPLv3+ | Sangoma | | cdr | 17.0.4.13 | Enabled | GPLv3+ | Sangoma | | cel | 17.0.2.7 | Enabled | GPLv3+ | Sangoma | | certman | 17.0.3.10 | Enabled | AGPLv3+ | Sangoma | | cidlookup | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | conferences | 17.0.3.2 | Enabled | GPLv3+ | Sangoma | | configedit | 17.0.1.1 | Enabled | AGPLv3+ | Sangoma | | contactmanager | 17.0.5.9 | Enabled | GPLv3+ | Sangoma | | core | 17.0.9.57 | Enabled | GPLv3+ | Sangoma | | customappsreg | 17.0.1 | Enabled | GPLv3+ | Sangoma | | dashboard | 17.0.4.2 | Enabled | AGPLv3+ | Sangoma | | daynight | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | dictate | 17.0.1.2 | Enabled | GPLv3+ | Sangoma | | directory | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | donotdisturb | 17.0.2.2 | Enabled | GPLv3+ | Sangoma | | dynroute | 17.0.3 | Enabled | GPLv3+ | Sangoma | | extensionroutes | 17.0.1 | Enabled | Commercial | Sangoma | | extensionsettings | 17.0.1 | Enabled | GPLv3+ | Sangoma | | fax | 17.0.3.3 | Enabled | GPLv3+ | Sangoma | | featurecodeadmin | 17.0.2 | Enabled | GPLv3+ | Sangoma | | filestore | 17.0.2.12 | Enabled | AGPLv3 | Sangoma | | findmefollow | 17.0.4.7 | Enabled | GPLv3+ | Sangoma | | firewall | 17.0.1.25 | Enabled | AGPLv3+ | Sangoma | | framework | 17.0.15.22 | Enabled | GPLv2+ | Sangoma | | hotelwakeup | 17.0.1.6 | Enabled | GPLv2 | Sangoma | | iaxsettings | 17.0.1 | Enabled | AGPLv3 | Sangoma | | infoservices | 17.0.1 | Enabled | GPLv2+ | Sangoma | | ivr | 17.0.6 | Enabled | GPLv3+ | Sangoma | | languages | 17.0.1 | Enabled | GPLv3+ | Sangoma | | logfiles | 17.0.3.1 | Enabled | GPLv3+ | Sangoma | | manager | 17.0.5 | Enabled | GPLv2+ | Sangoma | | miscapps | 17.0.3 | Enabled | GPLv3+ | Sangoma | | miscdests | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | music | 17.0.4 | Enabled | GPLv3+ | Sangoma | | outroutemsg | 17.0.1 | Enabled | GPLv3+ | Sangoma | | paging | 17.0.3 | Enabled | GPLv3+ | Sangoma | | parking | 17.0.2.4 | Enabled | GPLv3+ | Sangoma | | phpinfo | 17.0.1 | Enabled | GPLv2+ | Sangoma | | pinsets | 17.0.3.2 | Enabled | GPLv3+ | Sangoma | | pm2 | 17.0.3.2 | Enabled | AGPLv3+ | Sangoma | | presencestate | 17.0.2.2 | Enabled | GPLv3+ | Sangoma | | printextensions | 17.0.1.2 | Enabled | GPLv3+ | Sangoma | | queueprio | 17.0.1.4 | Enabled | GPLv3+ | Sangoma | | queues | 17.0.1.9 | Enabled | GPLv2+ | Sangoma | | recordings | 17.0.2.2 | Enabled | GPLv3+ | Sangoma | | restapps | | Not Installed (Locally available) | Commercial | Sangoma | | ringgroups | 17.0.2.4 | Enabled | GPLv3+ | Sangoma | | sangomaconnect | | Not Installed (Locally available) | Commercial | Sangoma | | sangomartapi | | Not Installed (Locally available) | Commercial | Sangoma | | setcid | 17.0.1.2 | Enabled | GPLv3+ | Sangoma | | sipsettings | 17.0.6.7 | Enabled | AGPLv3+ | Sangoma | | sipstation | | Not Installed (Locally available) | Commercial | Sangoma | | smsplus | | Not Installed (Locally available) | Commercial | Sangoma | | soundlang | 17.0.4.1 | Enabled | GPLv3+ | Sangoma | | sysadmin | 17.0.1.83 | Enabled | Commercial | Sangoma | | timeconditions | 17.0.1.16 | Enabled | GPLv3+ | Sangoma | | tts | 17.0.1.1 | Enabled | GPLv3+ | Sangoma | | ttsengines | 17.0.1 | Enabled | AGPLv3 | Sangoma | | ucp | 17.0.4.15 | Enabled | AGPLv3+ | Sangoma | | userman | 17.0.6.22 | Enabled | AGPLv3+ | Sangoma | | vmblast | 17.0.2 | Enabled | GPLv3+ | Sangoma | | voicemail | 17.0.5.16 | Enabled | GPLv3+ | Sangoma | | voipinnovations | | Not Installed (Locally available) | Commercial | Sangoma | | weakpasswords | 17.0.1 | Enabled | GPLv3+ | Sangoma | | webrtc | 17.0.2.1 | Enabled | GPLv3+ | Sangoma | +-------------------+------------+-----------------------------------+------------+-----------+
Relevant log output