voip

voip

Unable to set ringtone at receiver side, user not getting notified when getting call

Call forwarding

Opensips-cli -x command not working in opensips 3.3

[RTPEngine]Failed to init DTLS connection: key values mismatch

WatchOS 9 CallKit + VoIP

webex / TAPI integration - initiate VOIP call from TAPI-enabled application?

Telegram bot VOIP calls

freeswitch - all users list - not "registred only"

Convert OPUS stream to PCM or RTP

Nexmo-Vonage: Place call from Vonage VoIP device to phone, initiate at server

How to call from a 4g internet device

Call kit incoming call not getting triggered on Testflight build

Polycom VVX 600 VOIP phone taking almost 5 minutes to reconnect on secondary internet

How to activate the VOLTE by using my second sim card data connection

Unable to call out on VoIP number via gsm getaway mode using GoIP

Microsip softphone build problem (cannot open file 'hid.lib')

How to get graphical view of jitter, packet loss etc of an RTP pcap using tshark?

Make voip push notification work again on iOS 13

Save audio from RTP stream that contains RFC 2833 RTP events

NSInternalInconsistencyException while running VOIP enabled apps on iOS13

Freeswitch 1.8 and playback of tone_stream

Opensips suddenly crash in two-three days running

Extracting and injecting audio to an ongoing VOIP call

How many bits of codification G.711 sends in each datagrama?

Convert an audio file into a pcap with codec G722

VoIP - Send number and initiate call

Saving cdrs manually using avp_db_query in Opensips

Passing AVP to prefix core function

Parallel forking using TSILO module in kamailio

"Could not find or load main class de.fhg.fokus.hss.main.HSSContainer"- OpenIMSCore's FHoSS

Freeswitch: Check external API Before Call

How to convert string in URI

can VoIP support ussd application?

pjsip-2.7.1 Assertion failed when calling pjsua_set_snd_dev with ios CallKit

VoIP service to make a bridge between Android phone and Ubuntu server

CallKit no sound when answer second call and put first call on hold

Cisco IP Communicator - Send commands/Auto Dial

How can I pass values from the imported contacts list of an outbound campaign to the IVR Script?

how to capture VoIP phone traffic on my computer using Wireshark?

how to start a service like localphone.com

How to connect GOIP GSM Gateway with SIP Phone

grandstream UCM6104 VOIP PBX phone

Restcomm Connect USSD connected to OpenBTS over SIP

FreeSwitch outgoing call. Answering machine(fax) detection

Unable to receive call on VoIP number using GoIP 4

SIP tool to generate test traffic and measure latency

Freeswitch: play ringing tone to A-leg, while we play some message to called party B-leg

Send events from AppService to main Project

Error while requesting a transaction with CXEndCallAction

PJSip make call issues in iPhone

My Cisco phones are getting registered to other call manager

Using TLS in pjsua application

How to find out which server has less hops between user or specific IP?

Building Liblinphone for iOS

Enforce TLS connection only in Open-sip server in voip

Using ToneController to get more than one digit

Wireshark mean jitter rtp analysis

VOIP jitter on first call

Transmit Audio/Music Between Phones

How to measure voice delay between multiple hops

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