Hi,
I have successfully made the call. However, the error log is still recorded in the log and the call reception time is quite slow, I don’t know the cause. I appreciate any help (I use GRANDSTREAM UCM6202)
You have 1 active call
Current call id=0 to sip:[email protected] [CALLING]
>>> 11:46:09.457 pjsua_app.c .Turning sound device -99 -99 OFF
**11:46:39.919 tcpc0x7fe4a9052508 TCP connect() error: [code=120110]: Operation timed out**
**11:46:39.919 tsx0x7fe4a90686b8 Temporary failure in sending Request msg INVITE/cseq=2804 (tdta0x7fe4a9061a88), will try next server: Operation timed out**
**11:46:39.919 pjsua_app.c SIP TCP transport is disconnected from xxx.xxx.xxx.xxx:5060: Operation timed out [status=120110]**
11:46:40.241 pjsua_app.c ........Turning sound device -99 -99 ON
11:46:40.241 pjsua_app.c .....Call 0 state changed to EARLY (180 Ringing)
11:46:43.184 pjsua_app.c .....Call 0 state changed to CONNECTING
11:46:43.185 pjsua_app.c .....Call 0 state changed to CONFIRMED
11:46:45.296 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
11:46:45.296 pjsua_app_common.c ......
[DISCONNCTD] To: sip:[email protected];tag=12ad81cb-4556-4dd2-9aa0-80f1a4b7c3bd
Call time: 00h:00m:02s, 1st res in 31785 ms, conn in 34729ms
#0 audio PCMU @8kHz, sendrecv, peer=xxx.xxx.xxx.xxx:18952
SRTP status: Not active Crypto-suite:
RX pt=0, last update:00h:00m:01.838s ago
total 103pkt 16.4KB (20.6KB +IP hdr) @avg=63.5Kbps/79.4Kbps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.709 1.750 1.250 0.362
TX pt=0, ptime=20, last update:never
total 90pkt 14.4KB (18.0KB +IP hdr) @avg=55.5Kbps/69.4Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
i used this a lot when i had google translate, but now i use nabu casa tts and i need to set the language and gender in my calls- but how to do this in this addon?. i dont have any config in configuration.yaml when i use nabu casa. i set the tts in the addon config and it works but i need to set the gender and language.
Thank you very much sdesalve for your really handy addon and all the hard work, you’ve put into creating this. It’s working perfectly with my Fritzbox and is currently used in Fire Alert-Automations, as well as in a self created “Panic Mode”.
Is there any way in showing the CLIP-number to the callers, when i am calling someone through the addon? The number seems to be always supressed, even though i set up the FritzBox to generally showing the numbers for outbound calls. When using the SIP-Number with other calling tools, the number will be shown. Is it a configuration thing in the addon itself, or is it the wrong place to look into in finding it out?
Thanks in advance and stay safe.
Chris
Thanks for the fast reply.
I just tried out all of the constellations, but of no luck. When used with a trailing plus the addon refuses to start at all based on the following error: “pjsua_app_config.c Error: invalid SIP URL ‘sip:+49301234567’ in local id argument”
All the other number variants were used by the addon and with a test call through the dev tools according to the protocol of the addon it was in the state of “CALLING”, but on fritzbox-side there were no sip-registrations active, or any call attempts visible.
This time it accepted the plus in the id, but nevertheless no calls went through to my phone.
I again tried out all the ways possible how to write my number, but no combination worked in the end.
Thanks for your help so far sdesalve. I really appreciate your support.
I actually would be intrigued to know, if i’m the only one here with a surpressed caller id (in combination with a fritzbox).