An alternate would be to install RasPBX (Debian 10) on the raspberry , then install snapd and THEN snap install home-assistant.
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Sip trunk can send but not receive calls from SPA3102
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Call Rates Announcement of provider suppressed by freepbx
Yes, you’re right.
The issue here is that this specific provider does only signal 180 Ringing (not 183 Session Progress), which therefore leads to the phone not opening the media channel. Instead the phone plays local ringback.
Can anyone confirm that this is non-standards compliant by the provider?
I was able to mitigate this by setting inband_progress=yes manually for all my extensions in pjsip.endpoint_custom_post.conf like this:
[100](+)
inband_progress=yes
This forces asterisk to always send 183 Session Progress even if it didn’t receive that from the provider, which in turn, leads to the local phone opening the media channel.
Setting it on the trunk doesn’t help, and unfortunately it is not possible to set it for extensions through freepbx. 
Is there anyway to set this automatically for all extensions?
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Internal calls not working
I’m sure you know, but U&D is unsupported. Every single request for help needs to start out with this fact.
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Sip trunk can send but not receive calls from SPA3102
Default for FreePBX is pjsip on 5060 and chan_sip on 5160. Your softphones are pjsip and 3102 trunk is chan_sip.
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Sip trunk can send but not receive calls from SPA3102
Oops, Dial Plan 2 should be
(S0<:landlinephonenumber@192.168.1.201:5160>)
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Sip trunk can send but not receive calls from SPA3102
Do I need to change the trunk configuration (port=) ?
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Sip trunk can send but not receive calls from SPA3102
Nevermind it works now thanks for help.
And @dicko I don’t really wan’t to port all of my existing configuration + I really like docker
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Call Rates Announcement of provider suppressed by freepbx
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Auto Answered Calls in CDRs
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Sip.linphone.org
This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.
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Suspicious activity
All items discussed have been addressed and, of course, something else is broken. Outbound alls get all circuits busy.
We changed the SIP port, it is port number NNNNN in the allowed ranger, I checked.
Now we get all circuits busy. The calls are not getting to the SIP provider at all.
Changed the SIP port with the provider, changed it in:
CONNECTIVITY-TRUNKS - SIP SERVER PORT and
SETTINGS - ASTERISK SIP SETTINGS - CHAN PJSIP SETTINGS - UDP - PORT TO LISTEN ON.
How can I upload a pcap file?
//pastebin.freepbx.org/view/68f3e6a9 ](//pastebin.freepbx.org/view/68f3e6a9)
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Can I send multiple commands on speedial?
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Cisco 7975 and FreePBX
I’ll need to check that, however I do not think that is how opnSense works, the traffic to and from freePBX and AD is set to be explicitly allowed, which means it won’t matter how many requests are made, they’ll all be put through. opnSense on the other hand does not do well with UDP traffic send out on one port and returned on another, which if I am not mistaken is what the rport setting changes.
So it is very likely that whatever router you have just handles this, so the toggling the rport setting would not make a difference, or if it is opnSense then you have some sort of NAT setup that compensated for this.
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Adding file to backup
Thank you so much! I did not see this before.
I could change it so that it reads from its own config file.
Will the module somehow be backed up?
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Adding file to backup
You install this module and then that appears in your list of modules to backup to select
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Call Confirm Issues?
Seems to work ok for followme and ringgroups
Seems that it’s queues that’s affected
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ALL CIRCUIT BUSY pjsip log
2020-06-05 17:21:32] VERBOSE[9683][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:32] ExecIf(“PJSIP/204-00000002”, “0?Set(DIAL_TRUNK_OPTIONS=)”) in new stack
[2020-06-05 17:21:32] VERBOSE[9683][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:33] Set(“PJSIP/204-00000002”, “HASH(__SIPHEADERS,Alert-Info)=unset”) in new stack
[2020-06-05 17:21:32] VERBOSE[9683][C-00000004] pbx.c: Executing [s@macro-dialout-trunk:34] Dial(“PJSIP/204-00000002”, “PJSIP/13034640011@F1Systems,300,Tb(func-apply-sipheaders^s^1,(1))M(send-obroute-email^13034640011^3034640011^1^1591399292^^204)”) in new stack
[2020-06-05 17:21:32] ERROR[5959] res_pjsip.c: Endpoint ‘F1Systems’: Could not create dialog to invalid URI ‘F1Systems’. Is endpoint registered and reachable?
[2020-06-05 17:21:32] ERROR[5959] chan_pjsip.c: Failed to create outgoing session to endpoint ‘F1Systems’
[2020-06-05 17:21:32] WARNING[9683][C-00000004] app_dial.c: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
[2020-06-05 17:21:32] VERBOSE[9683][C-00000004] app_dial.c: No devices or endpoints to dial (technology/resource)
What is going on here?
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Dongles and prefixes
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ALL CIRCUIT BUSY pjsip log
Before even attempting the call, Asterisk decided that the trunk was unavailable. Registration (if used) may have failed. Otherwise, qualify failed (no reply to OPTIONS requests sent out).
In addition, the Outbound Route did not specify any backup trunks, so the overall call failed.
Earlier (perhaps much earlier) in the log, you should see errors relating to registration failure and/or host unreachable. You can troubleshoot this with pjsip logger or network packet captures.
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ALL CIRCUIT BUSY pjsip log
I dont see that. In fact it looks like it is “REACHABLE”
[2020-06-05 17:45:26] VERBOSE[15607][C-0000000a] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
[2020-06-05 17:45:27] VERBOSE[15607][C-0000000a] res_agi.c: <PJSIP/204-00000009>AGI Script sangomacrm.agi completed, returning 0
[2020-06-05 17:45:27] VERBOSE[15607][C-0000000a] pbx.c: Executing [s@crm-hangup:8] Return(“PJSIP/204-00000009”, “”) in new stack
[2020-06-05 17:45:27] VERBOSE[15607][C-0000000a] app_stack.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/204-00000009’
[2020-06-05 17:45:27] VERBOSE[15607][C-0000000a] app_stack.c: PJSIP/204-00000009 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
[2020-06-05 17:45:36] VERBOSE[5959] res_pjsip/pjsip_configuration.c: Endpoint F1Systems is now Reachable
[2020-06-05 17:45:36] VERBOSE[5959] res_pjsip/pjsip_options.c: Contact F1Systems/sip:14280612@us-west-or.sip.flowroute.com:5060 is now Reachable. RTT: 58.664 msec
[2020-06-05 17:45:54] VERBOSE[5879] asterisk.c: Remote UNIX connection
[2020-06-05 17:45:54] VERBOSE[15732] asterisk.c: Remote UNIX connection disconnected
[2020-06-05 17:45:54] VERBOSE[5879] asterisk.c: Remote UNIX connection
[2020-06-05 17:45:54] VERBOSE[15738] asterisk.c: Remote UNIX connection disconnected
[2020-06-05 17:45:54] VERBOSE[5879] asterisk.c: Remote UNIX connection
[2020-06-05 17:45:54] VERBOSE[15740] asterisk.c: Remote UNIX connection disconnected
[2020-06-05 17:45:54] VERBOSE[5879] asterisk.c: Remote UNIX connection
[2020-06-05 17:45:54] VERBOSE[15742] asterisk.c: Remote UNIX connection disconnected
[2020-06-05 17:45:57] VERBOSE[5879] asterisk.c: Remote UNIX connection
[2020-06-05 17:45:57] VERBOSE[15938] loader.c: Reloading module ‘res_odbc.so’ (ODBC resource)
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