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[SR-Users] One way audio on calls from pstn
https://users.openser.narkive.com/e3vV1zdQ/sr-users-one-way-audio-on-calls-from-pstn
2. OK. Outgoing calls from kamailio users to PSTN work also very well. 3. Not OK. Incoming from PSTN side calls have only one way audio. I tcpdump'ed kamailio box and found, that pstn provider sends RTP. packets to kamailio IP in case of answered call. I guess that rtpproxy is not active in case of pstn call.
SDP Information Change during Call Setup - one way audio ...
https://community.asterisk.org/t/sdp-information-change-during-call-setup-one-way-audio/16685
Polycom Phone --> asterisk --> OpenSER --> Gateway (Audiocodes) --> PSTN --> Phone. Our challenge: we just upgraded two of our servers to 1.4.10.1 (we have two other’s still running on 1.2.9.10) and are having trouble with one-way audio (the Polycom can hear downstream media but the upstream path cannot hear the Polycom).
How to solve VoIP one way audio- step by step.
https://www.voipmechanic.com/voip-one-way-audio.htm
Solve One Way Audio-Step by Step. One way audio where one side or one party can hear the other, but not reverse, is typically indicative of something stopping either the outbound or inbound audio from reaching the receiving party. One way audio can be caused by network issues; NAT (Network Address Translation) issues or firewalls, so finding the cause …
kamailio - 1 way audio only when registering to OpenSIPs ...
https://stackoverflow.com/questions/58124882/1-way-audio-only-when-registering-to-opensips-in-front-of-asterisk
Long time Asterisk user but fairly new to OpenSIPs. I have a SIP phone working with audio both directions when registering to and receiving calls directly from Asterisk. The same phone works with 2 way audio if I register to OpenSIPs and receive a call from OpenSIPs but only IF the call originated from somewhere OTHER than our Asterisk server.
One-way audio in some queue calls - General Help - FreePBX ...
https://community.freepbx.org/t/one-way-audio-in-some-queue-calls/4592
One-way audio in some queue calls. General Help. justincase. 2014-05-31 18:30:10 UTC #1. Hello all. ... We are directly connected to a lucent box for RTP and OpenSER for sip. It seems evident that something between * and the lucent box is getting confused. All sip transactions on working and non-working calls are the same. /Justin.
troubleshooting:nat - Kamailio (OpenSER) Wiki
https://www.kamailio.org/dokuwiki/doku.php/troubleshooting:nat
Audio only works one way, when connected with client behind nat. Answer. install mediaproxy module. install mediaproxy dispatcher. install mediaproxy server. Follow the install procedure and try using this openser.cfg : # # ----- global configuration parameters ----- #debug=3 # debug level (cmd line: -dddddddddd) # ...
sip - kamailio imsdroid one way audio is clear the other ...
https://stackoverflow.com/questions/15174574/kamailio-imsdroid-one-way-audio-is-clear-the-other-way-just-wind-and-jitter-soun
kamailio imsdroid one way audio is clear the other way just wind and jitter sound. Ask Question Asked 8 years, ... Audio heard in IMSdroid is OK. Audio heard in windows linphone is just wind sound and jitter (or train sound, when I speak louder it gets louder) ... Browse other questions tagged sip voip sip-server kamailio openser or ask your ...
OPENSER Integration –- USD$200.00 | Netgate Forum
https://forum.netgate.com/topic/2647/openser-integration-usd-200-00
This scenario does not allow for the RTP ports to be translated properly and ends up with the external phone having no audio at all or at best one-way audio when placing or receiving a call. This issue is solved when the SIP server is paired with OpenSER as a …
Anoty: How to Configure OpenSER: SIP Registar, SIP …
https://anoty.blogspot.com/2008/11/how-to-configure-openser-sip-registar.html
By not forking and logging to STDERR one can run openser directly to gather information about an invalid configuration directive or perhaps why a specific module isn't loading as expected. However to start OpenSER using openserctl you will need to set fork=yes and log_stderror=no. listen=1.2.3.4 alias=1.2.3.4 port=5060
'Re: [Kamailio-Users] kamailio and rtpproxy-no audio' - …
https://marc.info/?l=openser-users&m=122533442006644
Below please find my kamailio.cfg file. But this is what is happening: 1- ext 101 [IP 76.109.183.2] is able to send audio to kamailio's IP \ [65.111.185.187]but it does not reach ext 100 IP [76.109.15.75]. 2- ext 100 does not \ send RTP at all. 3- I still see the private IPs in …
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