• Video Length:N/A
  • Published:N/A
  • Module Location: Config > Channel SIP Settings

Text Tutorial

The QuBe Channel SIP Settings module allows you to customize the way your audio is handled as it is passed to and from the PBX through your trunk.

System Usage

  • Default Context
    • Default context for incoming calls. Defaults to 'from-sip-public'
  • Auto reject non-authorized request
    • When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password instead of letting the requester know whether there was a matching user or peer for their request. This reduces the ability of an attacker to scan for valid SIP usernames. This option is set to "yes" by default.
  • Allow or reject guest calls
    • Allow or reject guest calls (default is no). If your Asterisk is connected to the Internet and you have allowguest=yes you want to check which services you offer everyone out there, by enabling them in the default context (see below)
  • Bind IP address
    • IP address to bind listen socket to ( binds to all)
  • TCP Bind IP address
    • If TCP enabled. IP address to bind TCP listen socket to ( binds to all)
  • External IP address
    • IP address and port number to be used when talking to a host outside the NAT
  • DNS SRV lookups on outbound calls
    • Enable DNS SRV lookups on outbound calls. Note: Asterisk only uses the first host in SRV records. Disabling DNS SRV lookups disables the ability to place SIP calls based on domain names to some other SIP users on the Internet. Specifying a port in a SIP peer definition or when dialing outbound calls will supress SRV lookups for that peer or call
  • Allow Call Events
    • Set to yes to receive events on AMI when a call is put on/off hold

Local Networks

  • Network Address
    • Enter internal network address
  • Network Mask
    • Enter network mask address

NAT and Media Settings​

  • NAT
    • Asterisk may override the address/port information specified in the SIP/SDP messages, and use the information (sender address) supplied by the network stack instead. However, this is only useful if the external traffic can reach us. The following settings are allowed (both globally and in individual sections): 'no'- Do no special NAT handling other than RFC3581; 'force_rport' - Pretend there was an rport parameter even if there wasn't, 'comedia' - Send media to the port Asterisk received it from regardless of where the SDP says to send it. 'auto_force_rport' - Set the force_rport option if Asterisk detects NAT (default). 'auto_comedia' - Set the comedia option if Asterisk detects NAT
  • Direct Media
    • By default, Asterisk tries to re-invite media streams to an optimal path. If there's no reason for Asterisk to stay in the media path, the media will be redirected. This does not really work well in the case where Asterisk is outside and the clients are on the inside of a NAT. In that case, you want to set directmedia=nonat. 'yes' - Asterisk by default tries to redirect the RTP media stream to go directly from the caller to the callee. Some devices do not support this (especially if one of them is behind a NAT). The default setting is YES. If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off. This setting also affects direct RTP at call setup (a new feature in 1.4 - setting up the call directly between the endpoints instead of sending a re-INVITE). Additionally this option does not disable all reINVITE operations. It only controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting. Note that direct T.38 is not supported. 'nonat' - An additional option is to allow media path redirection (reinvite) but only when the peer where the media is being sent is known to not be behind a NAT (as the RTP core can determine it based on the apparent IP address the media arrives from). 'update' - Yet a third option... use UPDATE for media path redirection, instead of INVITE. This can be combined with 'nonat', as 'directmedia=update,nonat'. It implies 'yes'. 'outgoing' - When sending directmedia reinvites, do not send an immediate reinvite on an incoming call leg. This option is useful when peered with another SIP user agent that is known to send immediate direct media reinvites upon call establishment. Setting the option in this situation helps to prevent potential glares. Setting this option implies 'yes'. 'yes' - Enable the new experimental direct RTP setup. This sets up the call directly with media peer-2-peer without re-invites. Will not work for video and cases where the callee sends RTP payloads and fmtp headers in the 200 OK that does not match the callers INVITE. This will also fail if directmedia is enabled when the device is actually behind NAT
  • RTP Timeout
    • Terminate call if defined seconds of no RTP or RTCP activity on the audio channel when we're not on hold. This is to be able to hangup a call in the case of a phone disappearing from the net, like a power loss or grandma tripping over a cable
  • RTP Hold Timeout
    • Terminate call if 300 seconds of no RTP or RTCP activity on the audio channel when we're on hold (must be > rtptimeout)
  • RTP Keep alive
    • Send keep alives in the RTP stream to keep NAT open (default is off - zero)
  • Enable Jitter Buffer
    • Enable the use of a jitterbuffer on the receiving side of a SIP channel. Default set to "no". An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP side will be used only if it is forced and enabled

Audio Codecs​

  • ulaw
  • alaw
  • gsm
  • g726
  • g726aal2
  • adpcm
  • slin
  • lpc10
  • speex
  • ilbc
  • g722
  • testlaw

RTP Settings

  • Start RTP Port
    • Enter the RTP port to start
  • End RTP Port
    • Enter the RTP port to end
  • UDP checksum on RTP
    • Whether to enable or disable UDP checksums on RTP traffic
  • No “end” DTMF timeout
    • The amount of time a DTMF digit with no 'end' marker should be allowed to continue (in 'samples', 1/8000 of a second)
  • Time between rtcp reports (ms)
    • Milliseconds between rtcp reports(min 500, max 60000, default 5000)
  • Enable strict RTP protection
    • Enable strict RTP protection. This will drop RTP packets that do not come from the source of the RTP stream. This option is enabled by default
  • RTP sockets probation packets
    • Number of packets containing consecutive sequence values needed to change the RTP source socket address. This option only comes into play while using strictrtp=yes. Consider changing this value if rtp packets are dropped from one or both ends after a call is connected. This option is set to 4 by default
  • ICE Support
    • Whether to enable or disable ICE support. This option is disabled by default
  • STUN server
    • Hostname or address for the STUN server used when determining the external IP address and port an RTP session can be reached at. The port number is optional. If omitted the default value of 3478 will be used. This option is disabled by default. e.g.
  • TURN server
    • Hostname or address for the TURN server to be used as a relay. The port number is optional. If omitted the default value of 3478 will be used. This option is disabled by default. e.g.
  • TURN server username
    • Username used to authenticate with TURN relay server
  • TURN server password
  • Password used to authenticate with TURN relay server

Audio and Video Settings

  • Non-standard g726
    • If the peer negotiates G726-32 audio, use AAL2 packing order instead of RFC3551 packing order (this is required for Sipura and Grandstream ATAs, among others). This is contrary to the RFC3551 specification, the peer should be negotiating AAL2-G726-32 instead
  • T38 Fax
    • This setting is available in the [general] section as well as in device configurations. Setting this to yes enables T.38 FAX (UDPTL) on SIP calls; it defaults to off
  • Video Calls Support
    • Whether to enable or disable Video Calls support. This option is disabled by default

Video Codecs

  • h264
  • mpeg4
  • vp8
  • h263p
  • h261
  • h263


  • Notify Ringing
    • Control whether subscriptions already INUSE get sent RINGING when another call is sent (default: yes). NOTE! This feature is used by panel
  • Notify Hold
    • Notify subscriptions on HOLD state (default: no). Turning on notifyringing and notifyhold will add a lot more database transactions if you are using realtime. NOTE! This feature is used by panel

Registration Settings

  • Registration Timeout
    • Retry registration calls timeout (default: 20)
  • Registration Attempts
    • Number of registration attempts before we give up 0 = continue forever, hammering the other server until it accepts the registration Default is 0 tries, continue forever
  • Registration Maximum Expiry
    • Maximum allowed time of incoming registrations (seconds)
  • Registration Minimum Expiry
    • Minimum length of registrations (default 60)
  • Registration Default Expiry
    • Default length of incoming/outgoing registration
  • SIP Advanced Settings​
    • Set any option here – all options defined here will override options set above

  • Click the Submit button to save configuration
Channel SIP Module Screen