Phone Calls Terminating After About ~6 Seconds

All of a sudden phone calls are terminating after about 5 or 6 seconds.

I ran one module update this morning for the cert manager. Attempted to get a SSL let’sencrypt cert, but was unsuccessful. Yesterday we had an issue where our public IP changed and I had to change the VoicePulse setup to ipAuth and updated my PBX trunk accordingly. I tested that setup yesterday and had no issues.

Anybody have insight to what could be going on? If there is a certain log I should supply can you point me in the direction of how to obtain it? I’m no expert.

Turns out the only thing not working as it is intended to do so is incoming calls. They terminate after 6 seconds. Even if they go to voicemail. VoicePulse ran a trace and found that they were receiving a BYE message.

Here is a log:

[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] WARNING[18411][C-0000002e]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:41] ERROR[18411][C-0000002e]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:41] ERROR[18411][C-0000002e]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:41] ERROR[18411][C-0000002e]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] WARNING[18434][C-0000002f]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2017-04-13 14:24:49] ERROR[18434][C-0000002f]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:49] ERROR[18434][C-0000002f]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:49] ERROR[18434][C-0000002f]: res_pjsip_header_funcs.c:520 func_write_header: This function requires a PJSIP channel.
[2017-04-13 14:24:58] WARNING[3194]: chan_sip.c:4076 retrans_pkt: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2017-04-13 14:24:58] WARNING[3194]: chan_sip.c:4100 retrans_pkt: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
freepbx*CLI>

Outgoing calls and calls from extension to extension work as intended.

how is your router configured? you need to make sure that the voicepulse ip addresses are whitelisted in your router and that the traffic from those ip’s router directly to your pbx.

My voicepulse endpoint points to my public static ipv4. My router forwards port 81 (I believe for UCP) to PBX IP.

I can’t find anything about how to whitelist. Yesterday an IT specialist added a Linksys E4200 router running Tomato off our Comcast Business gateway which then leads to the LAN switch with the PBX setup. The public IP of the Comcast gateway was originally the same at it’s gateway address, so adding this Linksys modem allowed him to change the Public IP (by one number); he was here because of issues with a VPN setup. So I updated the IP with VoicePulse with the new one and flipped it to ipAuth=true.

I’m still at a loss.

You have got to forward a lot more than port 81. You need the signaling port of voicepulse (default is 5060), plus the media (rtp) ports (typically udp 10000-40000). Otherwise when voicepulse initiates an inbound call your pbx will never see it. outbound calls open the ports but inbound cannot do that so you have to make sure all the ports you need are forwarded from your router to the pbx. also make sure that you setup the firewall on the pbx and whitelist the voicepluse ip addresses in both the firewall and intrusion detection (fail2ban)

I disagree. A proper carrier that is smart can handle connecting with no ports opened. We do this in SIPstation all the time.

VoicePulse had me set the UDP port to 5160. Is this correct?

And I’m with Tony on this… I never had to forward ports or whitelist anything previously (~6 months of “successful” setup).

Here’s a trace if it helps.

The invite comes from your provider, your PBX sends back the 200 OK, but the PBX never receives the ACK from the provider acknowledging the OK. Since it is highly unlikely that the provider is not sending the ACK, there is a misconfig of your PBX or your router that is interrupting this critical step.

Do you have the external IP address set properly in Settings, Asterisk SIP Settings?

Lorne, that was it!

THANK YOU ALL SO MUCH FOR YOUR ASSISTANCE!

Where can I send the pizzas for you all?!

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