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with ulaw on the phones, the call doesn't complete because freeswitch is configured only with PCMA as the only preferred codec. the PSTN here supports only PCMA, just to clarify only the first Hello or what ever is spoken/transmitted in the first packet which has the huge delta is missed out...

It is changed from unassigned to PCMA - that is my inference from the wireshark RTP stream analysis.

I have a weird problem on internal calls alone, i.e between extensions, the first hello of the callee is never heard by the caller - I just did a packet capture and the first RTP packet alone has a huge delta, it also has a status message saying Payload changed to PT=8 - this message occurs on...

For any one confused about how this works. if you have a single context and if all extensions are in the same context, text messages work without any additional effort, all you need is a compatible client. if you have a multi-tenant system or multiple contexts, then messages between extensions...

Sipnetic, sessiontalk, Calls (Not suitable if you are behind NAT, works well otherwise),PortSIP (the app doesn't strip spaces on the numbers, call fails if your phone stores numbers with spaces), Zoiper free (If you are happy with not having Opus)

Is it possible to play music on hold via freeswitch when the proxy-media is set to true? In my profile I have late-neg true proxy-media true hold-music local_stream://default disable-hold false...

create a domain and add both the extensions in that domain

I think I have figured out, it is most likely due to VAD getting triggered 2021-08-17 22:22:48.731036 [DEBUG] switch_rtp.c:4413 Starting timer [soft] 160 bytes per 20ms 2021-08-17 22:22:48.731036 [DEBUG] switch_rtp.c:8810 Activate VAD codec PCMA 20ms 2021-08-17 22:22:48.731036 [DEBUG]...

Zoiper statistics are baffling, anyway if you look at the CDR information, there is packet loss (I can feel from the audio quality as well). I think something weird is going on, it may be a zoiper specific issue, I tested calling from a zoiper on a laptop to zoiper on a mobile (Device 1) (both...

RTP ports are open,even though zoiper says packet loss is 300% I don't know based on what metrics, I can hear audio, the bit rate gets reduced to half and I can feel the packet loss in the ears (with crackling sounds) (both sides use PCMA ). This was not like this earlier, I am on the same LAN...

I have 2 extensions connected to fusion PBX, no external gateway involved - both the sip clients are Zoiper clients from Android device. When mobile A calls mobile B - there is 0% received packet loss at mobile A, however at mobile B the loss as per zoiper is 300%,I can see that the received...

/var/log/messages file is filled with all the ping messages, register messages, I want to get rid of this - these are not even DEBUG/WARNING - how do I switch off this appearing in this file.

I have a setup like this a double NAT scenario. FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) Closed the issue as it is not a freeswitch or fusionpbx issue but an external route issue with the ISP

Your outbound calls has call-id of null which may be a reason 021-08-06 18:05:00.384304 [DEBUG] switch_core_state_machine.c:628 (sofia/external/XXXXXXXXXX) State INIT 2021-08-06 18:05:00.384304 [DEBUG] mod_sofia.c:93 sofia/external/XXXXXXXXXX SOFIA INIT 2021-08-06 18:05:00.384304 [INFO]...

21 603 CALL_REJECTED call rejected [Q.850] This cause indicates that the equipment sending this cause does not wish to accept this call, although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. The network may also generate this cause...

I have added a script to run fs_cli -x "sofia status profile profile_name rescan" during the ip update as a work around

in fs_cli set sofia global siptrace on - it will give you SIP messages which may throw some light.

I have fusionpbx/freeswitch running behind a NAT router with a dynamic IP which gets DDNS resolved when it changes. I have sip clients connecting from the internet to the server. Everything works well even when the IP gets updated except the external sip IP address - I have set it to...

I have the push proxy disabled, subscribe for presence & publish presence disabled as I am using the free version. Also subscribe for register unchecked. RTP for signaling and media ticked.