sip

sip

CORE_PCM_MODULE is not loaded when I start FreeSwitch service

Multiple contact headers in 200OK response for a Register request

How to get the audio stream from PJSIP when there is no audio hardware device

Supported: histinfo in SIP INVITE

UAC Behaviour in PRACK implementation

SIP Session Tracking

Android VoIP application: PARTIAL_WAKE_LOCK vs ignoring Battery Optimizations (doze whitelisting)

PJSIP - Freepbx - Trunk Registration Rejected

No audio with openmeeting sip call

Should a SIM Card and Cell Phone Produce an Authorization Response after Recieving "401 Unauthorized"?

Add multiple sip registrars to one phone

Why Jain SipDialog didn't increase localSequenceNumber after it sent reply for Subscribe request?

Can't set ttl in sdp as described in RFC 2327

No Connection in SessionDescription

Low average call duration

How to configure Kamailio with SEMS

Is it possible to detect if the callee forwards the call to a specific number?

SIP - connection

ICMPv6 Protocol Unreachable (Protocol Not Supported)

Android app for switching the network from 2G to 3G

RTCP enabled or Disabled - RR and RS in SDP

connect call from local number to termination provider

Configure sip account in FRITZ!Box

TLS Configuration OpenSIPS

Catch audio stream in freeswitch

wrong value for parameter wsServers

How do I forward fisrt n calls to one phone number and then next n calls to another phone number in sip voip?

Is there a way to tell from SIP SDP to check if PBX is using inband dtmf?

IMS Droid calee is Ringing but screen is black

Oracle SBC bridge 2 SIP calls

how to find from RTP stream that it belong to which sip call?

What is the difference between tel: and sip: uri in SIP?

Initiating call and receiving call in web browser using freeswitch

re-transmission of re-invite over UDP

Who may initialize "refresher"?

Can a SIP call's media address changed mid-call?

Flexisip installed On centOs 7 but can not register to it

SIP REGISTER getting 500 server internal error - 63

Voicemail detection on FreeSWITCH

Fusionpbx login fails after IP address change

How to interpret SDP payloads of SIP responses?

SIP ACK request re-transmission on transport error

Register Multiple Sip client accounts on same client PC and same server

Freeswitch server getting INVITE request in ngrep but nothing in fs_cli

Cannot send RTP by SIPp tool

Freeswitch spandsp.conf.xml Tone settings

Multiple -Re-invites from UA with incremented Cseq Handling

get status of SIP invite of Freeswitch

What is the difference between the normal INVITE and the INVITE on hold in a session using SIP protocol?

Knowing the type of VoIP application carried by H.323

How to remove presentity users entry in kamailio server?

Transfer Nexmo DID using a voip provider

Liblinphone SIP Error: 403 Phone Check Failed

SIP Timer inside core

In SIP INVITE, why ACK is called a transaction

In a SIP INVITE Method Getting 200 Response continuosly

EWS and Skype for Business/Lync meeting info

Can not integrate TeleStax Restcomm in MetaSwitch Clearwater

Restcomm Docker Installation Issues

Processing FMTP line above RTP line in a SIP SDP body?

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