GABPBX

GABPBX is the Germán Aracil Boned PBX: a GPLv2 open source PBX and telephony toolkit maintained by Germán Luis Aracil Boned garacilb@gmail.com.

The project was first created in 2008 and is based on the Asterisk 1.8 codebase, later updated to the final stable Asterisk 1.8 release. GABPBX keeps the proven Asterisk architecture and APIs where they are useful, while adding new production work focused on SIP, realtime operation, codec support and carrier-grade PBX deployments.

Existing Asterisk, Digium, third-party copyright notices and GPL licensing terms are preserved in the source files where they apply.

What GABPBX Provides

GABPBX is a complete PBX platform for VoIP and traditional telephony. It includes:

• SIP telephony with the new chan_sofia Sofia-SIP channel driver.
• Legacy SIP compatibility through the same public SIP technology name.
• Dialplan applications, functions, bridging, voicemail, queues, CDR/CEL, AGI, AMI, realtime configuration and media translation.
• PostgreSQL realtime support, including cache-aware res_config_pgsql operation.
• RTP media handling, NAT support, SRTP support and session timer support.
• Opus codec support through codec_opus and libopus.
• Traditional codec support such as alaw, ulaw, GSM, G.729 passthrough, Speex and signed linear formats, depending on selected modules and local libraries.

Why chan_sofia instead of chan_sip

chan_sofia is the recommended SIP driver for new GABPBX deployments. The short version of the case:

• Threads are constant. chan_sofia spawns exactly four module-level threads at load (NUA event loop, scheduler, outbound REGISTER refresh, qualify) and never creates additional threads per peer, per call, per registered contact, or per TCP/TLS connection. chan_sip spawns one thread per TCP/TLS connection on top of its monitor thread plus transient detached threads for park and pickup operations, so a deployment with thousands of TLS endpoints carries thousands of pthread stacks and the kernel context- switch cost that goes with them.
• Lower per-entity memory. The peer and dialog structs in chan_sofia are smaller than their chan_sip equivalents because much of the per-peer configuration uses pooled string fields and because the dialog object does not duplicate state that Sofia-SIP already tracks. Registered contacts are stored in a separate compact object instead of an inline slot, so a peer with no contacts costs less and a peer with many contacts pays only for what is in use.
• Multi-contact fork ringing. When a peer is registered from more than one device on the same credentials (desk phone plus softphone, or vendor family allowlisted), chan_sofia forks the outbound INVITE to every live contact in parallel and the first device that answers wins. chan_sip does not do this natively; you have to encode the parallel ring at the dialplan layer.
• A real SIP stack underneath. chan_sofia delegates SIP message parsing, transaction handling, NAT keepalives, retransmit timers, TCP/TLS framing and SDP/SDES negotiation to Sofia-SIP, instead of carrying that logic by hand. The result is fewer hand-rolled corner cases and a clearer separation between PBX policy and SIP protocol mechanics.
• Drop-in compatibility. The channel technology stays SIP, the realtime family stays sippeers, the sip show ... and sip set debug CLI commands stay, and sofia.conf follows the sip.conf style. Most existing dialplans, AMI scripts and operator habits keep working without changes.

The high-level resource comparison, with every figure traceable to the source of each driver:

Dimension	chan_sip	chan_sofia
Threads at load	1 monitor thread (do_monitor, chan_sip.c:27246)	4 (sofia_thread, sofia_sched, sofia_reg_thread, sofia_qualify_tid — chan_sofia.c:17370-17457)
Per call threads	0 normally, +1 detached per Park / Pickup (chan_sip.c:22777, :22818)	0
Per TCP/TLS connection threads	+1 per persistent SIP TCP/TLS connection (sip_tcp_worker_fn at chan_sip.c:26924)	0 — Sofia-SIP multiplexes all transports on the NUA event loop
Per peer memory (bare, not registered)	~3-4 KB on sip_peer plus its string buffers	~2-3 KB on sofia_peer plus its pooled string field area
Per registered contact memory	inlined into sip_peer	~640 B per sofia_contact (chan_sofia.c:1112-1121), refcounted in an ao2_container on the peer
Per active call memory	~5-8 KB on sip_pvt	~3-5 KB on sofia_pvt
Multi-contact ring	dialplan only (Dial(SIP/a&SIP/b&...))	automatic when the peer has more than one live contact (chan_sofia.c:6151-6353)

Threads are the most visible scaling difference. A deployment that needs to support a few thousand registered SIP endpoints over TLS will see the chan_sip thread count grow linearly with the TLS session count; on chan_sofia it stays at 4. Memory savings per peer and per call are smaller in absolute terms but compound at deployment scale.

The SIP throughput ceiling per CPU core is similar between the two drivers: both serialise SIP request dispatch through a single thread (the monitor thread in chan_sip, the NUA event loop in chan_sofia). Scaling SIP throughput beyond one CPU core is not a goal of either driver today and is not part of the chan_sip parity surface; deployments that need that go through fronting SBCs.

chan_sofia

chan_sofia is the modern SIP channel driver for GABPBX. It is built on the Sofia-SIP NUA API and is designed as a drop-in replacement for chan_sip, while using a stronger SIP stack instead of duplicating low-level SIP parsing and transaction handling by hand.

The compatibility policy is deliberate:

• The channel technology name is still SIP.
• Existing dialplans can continue to use Dial(SIP/peer) syntax.
• The realtime family remains sippeers.
• Existing SIP CLI names are preserved, such as sip show peers, sip show peer <name>, sip show channels and sip set debug.
• Existing AMI names are preserved, including SIPpeers, SIPshowpeer, SIPqualifypeer, SIPshowregistry and SIPnotify.
• Existing dialplan functions are preserved where implemented, including SIPPEER, SIPCHANINFO, SIP_HEADER and CHECKSIPDOMAIN.

chan_sofia and chan_sip must not be loaded at the same time. They register the same public technology, CLI, AMI and realtime names. To use chan_sofia, disable chan_sip.so in modules.conf, enable chan_sofia.so, then restart GABPBX.

Example:

; /etc/gabpbx/modules.conf
noload => chan_sip.so
load => chan_sofia.so

chan_sofia Feature Map

Protocol and channel behavior:

• Sofia-SIP NUA based SIP stack integration.
• Public channel technology remains SIP.
• Inbound and outbound INVITE call setup.
• ACK, BYE, CANCEL, OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, REFER, MESSAGE, INFO, PUBLISH and PRACK handling through Sofia-SIP method dispatch.
• Blind and attended transfer handling through SIP REFER and GABPBX bridge integration.
• SIP dialog tracking through internal ao2_container registries.
• Channel state integration with GABPBX core.
• Overlap dialing support with allowoverlap.
• RFC 3261 Max-Forwards support.
• RFC 4028 session timer support.
• Per-peer and global session timer policy: session-timers, session-expires, session-minse, session-refresher.
• SIP T1 and Timer B global configuration through Sofia-SIP native timers.

Transport and listener support:

• UDP listener.
• TCP listener.
• Optional TLS listener.
• Optional WebSocket listener for SIP signaling.
• Optional Secure WebSocket listener for SIP signaling.
• Configurable bind address and bind port.
• TLS certificate directory support using agent.pem and cafile.pem.
• WSS helper behavior to derive expected WSS certificate files from TLS files.
• SRV lookup support for outbound calls.
• Configurable outbound User-Agent.
• Reverse-proxy deployment guidance for WS/WSS origin enforcement.

Drop-in chan_sip compatibility:

• Dial(SIP/peer) continues to work.
• Existing sippeers realtime family is reused.
• Existing SIP CLI command names are reused.
• Existing SIP AMI action names are reused.
• Existing SIP AMI event names are preserved where compatible.
• Existing dialplan functions are preserved where implemented.
• sofia.conf follows the sip.conf style.
• Historical spelling compatibility is preserved for selected options, such as expiry typo variants.
• chan_sofia intentionally cannot coexist with chan_sip in the same process, because both register the same public SIP names.

Authentication and security:

• Digest authentication with MD5.
• Digest authentication with SHA-256.
• qop=auth support.
• RFC 2069 fallback when clients omit qop.
• Nonce TTL control.
• Stale nonce challenge handling.
• Nonce-count replay protection.
• alwaysauthreject support to prevent username enumeration.
• Optional global force_invite_auth lockdown behavior.
• Per-peer insecure behavior for compatibility with trusted trunks.
• Source IP ACL support.
• Contact-header ACL support with contactpermit and contactdeny.
• Direct-media ACL support with directmediapermit and directmediadeny.
• REGISTER User-Agent locking with lockuseragent.
• AMI visibility for authentication failures.
• AMI visibility for User-Agent lock rejections.
• Local SIP blacklist for repeated invalid REGISTER/authentication attempts.
• Blacklisted IPs are silently dropped instead of receiving a SIP response.
• Default local blacklist size is 1024 entries.
• Default blacklist threshold is 5 failures.

Registration:

• Registrar support for dynamic peers.
• Multi-contact registration support.
• Configurable maxcontacts with a hard ceiling.
• REGISTER expiry bounds: minexpiry, maxexpiry, defaultexpiry.
• RFC 3261 423 Interval Too Brief handling.
• Registration state persistence to realtime.
• Optional separated sipregs realtime registration table.
• Outbound registration support through register-style peers.
• Outbound registration retry controls: registertimeout and registerattempts.
• callbackextension support for outbound Contact URI user control.
• Peer source address tracking.
• Registered contact display in sip show peer.
• Registration AMI events.
• PeerStatus AMI events for registered/unregistered state.

Realtime and database operation:

• sippeers realtime family compatibility.
• Optional sipregs family for separating registration state from peer configuration.
• Runtime peer caching through ao2_container.
• sip prune realtime CLI support.
• PostgreSQL realtime cache documentation in configs/res_pgsql.conf.sample.
• Realtime peer display keeps the source visible as realtime/static.

NAT, addressing and DNS:

• externaddr support.
• externhost compatibility.
• externrefresh behavior.
• localnet support.
• rport/comedia style NAT operation.
• nat=force_rport,comedia style peer configuration.
• directmedia control.
• DNS manager integration for peers with hostname-based host=.
• AMI DnsManagerUpdate event when DNS-tracked peer addresses change.
• matchexternaddrlocally / matchexterniplocally are parsed and displayed for migration compatibility.

Media and RTP:

• RTP media integration through the GABPBX RTP engine.
• Static RTP payload support for classic codecs.
• Dynamic RTP payload support for modern codecs.
• Opus SDP support.
• Codec preference handling through normal allow / disallow lists.
• DTMF mode support.
• RFC 2833 / telephone-event style DTMF.
• Inband DTMF detection through GABPBX DSP where configured.
• relaxdtmf support.
• Hold detection and bridge hold signaling.
• Music-on-hold interpretation and suggestion fields.
• RTP timeout, RTP hold timeout and RTP keepalive configuration.
• Direct media decisions centralized in sofia_get_rtp_peer.
• Video direct-media is intentionally kept on the relayed path today.

SRTP and encryption:

• SRTP configuration support.
• SDP crypto helper integration.
• Configurable SRTP cipher display.
• Optional per-suite fresh SRTP key policy.
• Encryption defaults are conservative: unset encryption fields behave as disabled instead of accidentally enabling SRTP.

Identity and headers:

• Caller ID name/number propagation.
• callingpres support.
• sendrpid support.
• trustrpid support.
• Remote-Party-ID and P-Asserted-Identity handling.
• usereqphone support for ;user=phone URI generation.
• Per-peer setvar support.
• Per-peer outbound SIP header= support through inherited channel variables.
• SIPAddHeader style dialplan behavior is preserved through the compatibility header path.

Call limits and usage counters:

• callcounter support.
• call-limit support.
• busylevel support.
• Busy-on-active support.
• In-use, ringing and on-hold counters.
• sip show inuse CLI.
• SIPshowpeer AMI fields for call usage.
• AMI PeerStatus events for call-limit rejection and call counter updates.

Transfers and REFER:

• SIP REFER handling.
• REFER policy through allowtransfer.
• Transfer rejection with SIP response and AMI visibility.
• AMI Transfer event integration.
• AMI TransferRejected event.
• AMI ReferProgress event on transfer progress notifications.
• Transfer mode display in CLI and AMI.

Subscribe, MWI and presence-related behavior:

• allowsubscribe support.
• Message-summary MWI SUBSCRIBE handling.
• MWI expiry configuration.
• subscribecontext is parsed, stored and displayed for compatibility.
• Unknown or unsupported subscription event packages are accepted according to current Sofia/GABPBX behavior, but full dialog/presence routing is future work.
• AMI SubscribeRejected event when subscription policy rejects a peer.

Fax and T.38 related behavior:

• T.38 fax UDPTL relay through the GABPBX ast_udptl engine.
• 4-state T.38 negotiation machine (disabled, local-reinvite, peer-reinvite, enabled) with peer-reinvite abort timer.
• SDP T.38 attribute negotiation: T38FaxVersion, T38MaxBitRate, T38FaxRateManagement, T38FaxMaxBuffer, T38FaxMaxDatagram and T38FaxUdpEC (FEC / redundancy / none).
• Per-peer t38pt_udptl configuration for enable, EC mode and maxdatagram= override.
• [general] t38_maxdatagram for the global default.
• Per-peer t38pt_usertpsource for symmetric-RTP UDPTL destination behavior on NAT'd peers.
• AST_OPTION_T38_STATE queryoption handler so applications such as SendFAX and ReceiveFAX can drive T.38 negotiation.
• AMI T38FaxNegotiation event on each state transition.
• faxdetect modes wired in: no (default), cng (DSP CNG tone detection), t38 (peer T.38 reINVITE detection), cng,t38 / yes (both). On detection the channel is async-redirected to the fax extension where the dialplan runs SendFAX / ReceiveFAX.
• t38pt_udptl=no peers stay on classic audio and the T.38 engine remains inactive for them.

CLI commands:

sip show peers
sip show channels
sip show peer <name>
sip show inuse [all]
sip show settings
sip set debug [on|off|peer <name>|ip <addr>]
sip prune realtime [peer [<name>|all|like <pattern>]|all]
sip show blacklist
sip blacklist search <IP>
sip blacklist delete <IP>
sip blacklist clear

Example sip show peer test output from a local GABPBX instance:

*CLI> sip show peer test
Setting max files open to 100000

==============================================================================
  SIP peer             : test
  Registration         : registered
==============================================================================
  Endpoint             : test@dynamic:5060 via udp
  Context / source     : killer / sofia.conf
  Contact slots        : 1 used of 6 allowed
  Media                : codecs=(opus|g722|alaw|g729) dtmf=rfc2833 nat=force_rport directmedia=No
  Calls                : 0/1 active, 0 ringing, 0 on-hold
  Qualify              : no
  Session timers       : originate, expires=3600, minse=90, refresher=uac
  Identity headers     : send=rpid, trust=No, presentation=allowed_not_screened

-- Identity ------------------------------------------------------------------
  Name                 : test
  Username             : test
  Type                 : peer
  Host                 : dynamic
  Port                 : 5060
  Transport            : udp
  Context              : killer
  Registered           : Yes
  Expires              : 360s
  Secret               : (set)
  Qualify              : no

-- Network and media ---------------------------------------------------------
  NAT                  : force_rport
  DTMF mode            : rfc2833
  Direct media         : No
  Encryption           : No
  Codecs               : (opus|g722|alaw|g729)
  Max call BR          : 384 kbps

-- Limits and features -------------------------------------------------------
  Busy on active       : No
  Max contacts         : 6 (used: 1)
  Transfer mode        : open
  Lock user-agent      : No
  Language             : es
  Default IP           : (null)
  AMA flags            : Unknown
  Subscribe MWI        : No
  Preferred codec      : No
  Ignore SDP ver       : No
  Promisc redir        : No
  Auto framing         : No
  Fax detect           : no

-- Fax and T.38 --------------------------------------------------------------
  T38 support          : No
  T38 EC mode          : FEC
  T38 max datagram     : -1
  T38 RTP source       : No

-- Timers and RTP ------------------------------------------------------------
  Timer B              : 32000
  Timer T1             : 500
  Overlap dial         : Yes
  RTP timeout          : 30
  RTP hold timeout     : 300
  RTP keepalive        : 0

-- Routing and dialplan ------------------------------------------------------
  Parking lot          : default
  User=Phone           : no
  Max forwards         : 70 (from [general])
  Mailbox              : (none)
  Outbound proxy       : (none)
  MOH interpret        : (none)
  MOH suggest          : (none)
  SRTP cipher          : (default AES_CM_128_HMAC_SHA1_80)

-- Session and identity headers ----------------------------------------------
  Session timers       : originate
  Session expires      : 3600
  Session Min-SE       : 90
  Session refresher    : uac
  Calling pres         : allowed_not_screened
  Send RPID            : rpid
  Trust RPID           : No
  Concurrent calls     : 0/1 (0 ringing, 0 on-hold)

-- Groups and source ---------------------------------------------------------
  Call group           :
  Pickup group         :
  Subscribe context    : <Not set>

-- Security and ACL ----------------------------------------------------------
  Insecure             : port,invite
  ACL                  : no
  Contact ACL          : no
  Direct media ACL     : no
  DNS managed          : no

-- Registration --------------------------------------------------------------
  Source               : sofia.conf
  Source addr          : 127.0.0.1:34549

-- Contacts ------------------------------------------------------------------
  Contact count        : 1
  Contact 1 URI        : sip:test@127.0.1.1:100
    State                : IDLE
    TTL                  : 3488s
    Source               : 127.0.0.1:34549
    User-Agent           : sipsak 0.9.8.1

AMI actions:

SIPpeers
SIPshowpeer
SIPqualifypeer
SIPshowregistry
SIPnotify

AMI events include:

PeerStatus
Registry
Hold
Transfer
ReferProgress
TransferRejected
AuthFailure
SubscribeRejected
LockUserAgentReject
RegisterIntervalRejected
RegextenOnQualifyTransition
DnsManagerUpdate
HintCreated
InsecureInviteBypass
SessionTimerRefresh
T38FaxNegotiation

chan_sofia Future Roadmap

The project is conservative about roadmap claims: features listed here are not presented as completed behavior unless the current source already wires them. They are the next logical areas for public hardening and completion.

Planned or staged work:

• WebRTC media Stage B: DTLS-SRTP, ICE and BUNDLE media integration. Current WS/WSS work is SIP signaling oriented.
• Presence and dialog event-package handling.
• Runtime subscribecontext dispatch once the presence/dialog handler lands.
• Unsolicited MWI NOTIFY support if operator demand requires it.
• Full outbound HOLD re-INVITE generation, including outbound MOH suggestion signaling.
• Full directrtpsetup early-RTP bridge behavior. The option is parsed for migration compatibility today.
• Dynamic outbound Allow header generation from disallowed_methods.
• Compact SIP header emission if Sofia-SIP exposes a native compact-header control or the project carries a Sofia-SIP patch for it.
• 3xx redirect handling for promiscredir.
• autoframing SDP ptime/framesize wire-in.
• Exact progressinband=no chan_sip state-machine parity.
• Peer eviction behavior for rtcachefriends=no and rtautoclear.
• Per-peer dynamic T1 timer adjustment based on qualify RTT.
• Text RTP QoS handling for tos_text and cos_text.
• Video direct-media support after audio/direct-media behavior remains stable.
• Better clean-reload and clean-unload behavior where Sofia-SIP thread ownership permits it. For now, restarting GABPBX is the correct operational method after changing the active SIP channel driver.

The main configuration file is:

/etc/gabpbx/sofia.conf

A detailed sample is provided in:

configs/sofia.conf.sample

Opus Codec Support

GABPBX includes native Opus translator support in codec_opus.

The module translates between Opus and signed linear audio using libopus. The current implementation supports 8 kHz and 16 kHz signed linear paths and uses Opus internally according to RFC 7587. Opus is especially useful for WebRTC, modern SIP endpoints and low-bitrate high-quality VoIP.

Build dependency:

libopus
pkg-config

On Debian or Ubuntu systems, the package is usually:

apt install libopus-dev pkg-config

Opus settings are configured in:

/etc/gabpbx/codecs.conf

Sample configuration:

[opus]
bitrate => 32000
fec => true
dtx => false

To allow Opus on a SIP peer:

disallow=all
allow=opus
allow=alaw
allow=ulaw

Runtime CLI:

*CLI> opus show

Build Requirements

GABPBX is primarily built and tested on GNU/Linux.

Core build tools:

• GCC or another C99-capable compiler with GCC-compatible extensions.
• GNU make.
• ncurses development headers for menuselect.
• OpenSSL development headers.
• zlib development headers.
• pthread and standard C library development headers.

Useful optional dependencies:

• Sofia-SIP development files for chan_sofia.
• libopus development files for codec_opus.
• PostgreSQL development files for res_config_pgsql and related modules.
• ODBC, SQLite, Radius, DAHDI, Speex, GSM and other libraries depending on the modules you select.

chan_sofia currently expects Sofia-SIP headers and libraries in the standard local Sofia-SIP install layout used by this tree:

/usr/local/include/sofia-sip-1.13
/usr/local/lib

The linked library is:

lsofia-sip-ua

If Sofia-SIP is installed in /usr/local/lib, make sure the dynamic linker can find it:

ldconfig

Full Build: chan_sofia + codec_opus

This is the complete build path for a GABPBX installation with both chan_sofia and codec_opus.

1. Install system build packages

On Debian or Ubuntu, install the base compiler stack and common development headers:

apt update
apt install build-essential pkg-config ncurses-dev libssl-dev zlib1g-dev libxml2-dev libsqlite3-dev libpq-dev unixodbc-dev

Install Opus headers and libraries:

apt install libopus-dev

Other modules may need additional libraries. make menuselect will show what is available and what is missing.

2. Build and install Sofia-SIP

chan_sofia links against Sofia-SIP:

lsofia-sip-ua

The current GABPBX make rules compile chan_sofia with:

-I/usr/local/include/sofia-sip-1.13
-L/usr/local/lib -lsofia-sip-ua

That means Sofia-SIP must be installed in /usr/local for the default build.

If you build Sofia-SIP from source:

cd /path/to/sofia-sip
./configure --prefix=/usr/local
make
make install
ldconfig

Verify that the headers and library are visible:

test -d /usr/local/include/sofia-sip-1.13
test -f /usr/local/lib/libsofia-sip-ua.so || test -f /usr/local/lib/libsofia-sip-ua.a
pkg-config --libs opus

3. Configure GABPBX

From the GABPBX source directory:

./configure

This generates the local build configuration and dependency state used by make and menuselect.

4. Select modules

Open the interactive selector:

make menuselect

Enable or verify these modules:

Channel Drivers  -> chan_sofia
Codec Translators -> codec_opus

The modules can also be enabled non-interactively after ./configure has generated menuselect.makeopts:

menuselect/menuselect --enable chan_sofia menuselect.makeopts
menuselect/menuselect --enable codec_opus menuselect.makeopts

If dependencies are missing, menuselect or the build will report them. For chan_sofia, check Sofia-SIP headers and libsofia-sip-ua. For codec_opus, check libopus-dev and pkg-config.

5. Build and install

Use the normal build and install flow:

make
make install

For a configured tree, this is also valid and convenient:

make install

make install builds pending objects first, then installs the binaries and modules.

For a first installation only, install sample configuration:

make samples

Be careful: make samples can overwrite existing configuration files under /etc/gabpbx.

6. Configure modules.conf

chan_sofia is a drop-in replacement for chan_sip, so both must not be loaded together.

Use:

noload => chan_sip.so
load => chan_sofia.so
load => codec_opus.so

7. Configure sofia.conf

Start from the provided sample:

install -m 644 configs/sofia.conf.sample /etc/gabpbx/sofia.conf

Minimal peer using Opus:

[general]
context=default
bindaddr=0.0.0.0
bindport=5060
allowguest=no
realm=gabpbx

[phone100]
type=peer
host=dynamic
secret=change-this-secret
context=internal
disallow=all
allow=opus
allow=alaw
allow=ulaw
dtmfmode=auto
nat=force_rport,comedia
directmedia=no
maxcontacts=3
call-limit=3

8. Configure codecs.conf for Opus

Start from the provided sample:

install -m 644 configs/codecs.conf.sample /etc/gabpbx/codecs.conf

Useful Opus defaults:

[opus]
bitrate => 32000
fec => true
dtx => false

9. Start and verify

Start in console mode:

gabpbx -vvvc

Or connect to a running instance:

gabpbx -r

Verify module loading:

*CLI> module show like sofia
*CLI> module show like opus
*CLI> sip show settings
*CLI> sip show peers
*CLI> opus show

Expected results:

• chan_sofia.so is loaded.
• codec_opus.so is loaded.
• sip show settings reports the Sofia-SIP channel configuration.
• opus show reports bitrate, FEC, DTX and active encoder/decoder counts.

If chan_sofia.so does not load, check:

• chan_sip.so is not loaded.
• Sofia-SIP is installed in /usr/local.
• /usr/local/lib is visible to the dynamic linker.
• ldconfig has been run after installing Sofia-SIP.

If codec_opus.so does not build or load, check:

• libopus-dev is installed.
• pkg-config --cflags opus returns include flags.
• pkg-config --libs opus returns linker flags.

Short Build Summary

Use the normal source build flow. Do not edit generated configure or make scripts for a normal build.

./configure
make menuselect
make
make install

make menuselect is optional, but recommended. It lets you select modules and see missing dependencies before the build.

For a configured tree, make install is also the normal deployment target: it builds pending objects and then installs them.

For a first installation only, sample configuration files can be installed with:

make samples

Be careful: make samples can overwrite existing configuration files.

Basic Runtime

Start GABPBX in foreground console mode:

gabpbx -vvvc

Connect to a running GABPBX instance:

gabpbx -r

Run a single CLI command:

gabpbx -rx "core show version"
gabpbx -rx "module show like sofia"
gabpbx -rx "sip show peers"
gabpbx -rx "opus show"

Load chan_sofia from the CLI:

*CLI> module load chan_sofia.so

After changing SIP channel drivers, restart GABPBX instead of relying only on a module reload, because chan_sofia and chan_sip own the same public SIP names.

Minimal chan_sofia Peer Example

[general]
context=default
bindaddr=0.0.0.0
bindport=5060
allowguest=no
realm=gabpbx

[phone100]
type=peer
host=dynamic
secret=change-this-secret
context=internal
disallow=all
allow=opus
allow=alaw
dtmfmode=auto
nat=force_rport,comedia
directmedia=no
maxcontacts=3
call-limit=3

Dialplan usage remains compatible with existing SIP dialplans:

exten => 100,1,Dial(SIP/phone100,30)

Realtime Notes

chan_sofia uses the same sippeers realtime family as chan_sip.

Example extconfig.conf style:

sippeers => pgsql,general,sippeers

Registration state can also be separated into a sipregs realtime family when configured. See:

doc/realtime_sipregs.txt
doc/realtime_sippeers.txt
configs/res_pgsql.conf.sample

res_config_pgsql includes cache support and UDP cache invalidation support. The sample configuration documents the available settings.

Security Notes

• Do not expose SIP services to the public Internet without firewalling, strong secrets and abuse controls.
• Disable guest calls unless they are intentionally required.
• Use strong peer secrets.
• Keep system time synchronized with NTP; SIP registration and digest authentication are time-sensitive.
• Review sofia.conf.sample before deploying TLS, WS or WSS listeners.
• Do not load both chan_sip.so and chan_sofia.so at the same time.

License

GABPBX is free software distributed under the terms of the GNU General Public License Version 2. See LICENSE and COPYING in the source tree.

GABPBX preserves existing Asterisk, Digium and third-party notices where they apply. New GABPBX work is maintained as part of the GABPBX project by Germán Luis Aracil Boned garacilb@gmail.com.