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https://github.com/staskobzar/sippak
SIP command line utility
https://github.com/staskobzar/sippak
command-line-tool sip sip-client telephony utility voip voip-application
Last synced: 5 days ago
JSON representation
SIP command line utility
- Host: GitHub
- URL: https://github.com/staskobzar/sippak
- Owner: staskobzar
- License: gpl-3.0
- Created: 2018-06-10T01:54:20.000Z (over 6 years ago)
- Default Branch: master
- Last Pushed: 2024-04-09T19:40:03.000Z (7 months ago)
- Last Synced: 2024-04-15T01:05:24.154Z (7 months ago)
- Topics: command-line-tool, sip, sip-client, telephony, utility, voip, voip-application
- Language: C
- Homepage:
- Size: 3.62 MB
- Stars: 14
- Watchers: 2
- Forks: 5
- Open Issues: 2
-
Metadata Files:
- Readme: README.md
- License: COPYING
- Code of conduct: CODE_OF_CONDUCT.md
Awesome Lists containing this project
README
# sippak
## SIP command line utility
[](https://github.com/staskobzar/sippak/actions/workflows/main.yml)
[](https://codecov.io/gh/staskobzar/sippak)
[](https://www.codefactor.io/repository/github/staskobzar/sippak/overview/master)
[](https://www.codacy.com/gh/staskobzar/sippak/dashboard?utm_source=github.com&utm_medium=referral&utm_content=staskobzar/sippak&utm_campaign=Badge_Grade)
Simple command line utility for SIP protocol based on [PJPROJECT](http://www.pjsip.org/).
Can be useful for SIP administrators and developers.
## Usage examples
> Ping remote server with SIP OPTIONS message
```
sippak --color PING sip:1001@DOMAIN_OR_IP
```
> Initiate call from user 1001 with auth password MYPASS and caller ID "Alice Home" to 1002
> When remote end-point answer then sippak will hang-up the call.
```
sippak INVITE --color -u 1001 -pMYPASSWD -F "Alice Home" sip:1002@DOMAIN_OR_IP
```
> Register to user 8852 with password MYPASS, then list all contacts
> registered to 8852. Last command cancel all registations to user 8852.
```
sippak REGISTER --color -u 8852 -pMYPASSWD sip:8852@DOMAIN_OR_IP
sippak REGISTER --color -u 8852 -pMYPASSWD sip:8852@DOMAIN_OR_IP --clist
sippak REGISTER --color -u 8852 -pMYPASSWD sip:8852@DOMAIN_OR_IP --cancel-all
```
> Send MWI update to a phone using phone's IP
```
sippak notify --color --mwi=0 sip:[email protected]
```
> Publish UA "1001" status busy or has do-not-disturb activated
```
sippak PUBLISH --color -u 1001 -pMYPASSWD --pres-status=closed sip:1001@DOMAIN_OR_IP
```
> Command phone 1001 to initiate call to 1002. Not all phones support this option
> or might need extra configuration to support it.
```
sippak REFER --color --to=sip:[email protected]:12345 sip:[email protected]:5060
```
> Restart remote phone 1000. Event might depend on phone vendor. Not all phones support it.
```
sippak notify --event=check-sync sip:[email protected] --color
```
> Send SIP MESSAGE chat message to user 1000 and attach additional SIP header X-Token
```
sippak MESSAGE --body="Hello there" --header="X-Token: f7c0265e" --color sip:1001@DOMAIN_OR_IP
```
## Install
### make
Requires pjproject library and CMake 3.
```
make
sudo make install
```
CMake will require pkg-config path to libpjproject. If it has custom path
it can be provided with PKG_CONFIG_PATH variable. For example:
```
PKG_CONFIG_PATH=/usr/local/lib/pkgconfig make
```
### Ubuntu
Also, packages (rpm, deb and tgz) are available in "[dist](https://github.com/staskobzar/sippak/tree/master/dist)" directory.
Here is step by step install example for Ubuntu. Note version 2.9 of pjproject. This will install everything for deployement and development. "make test" is optional.
```sh
$ apt-get update
$ apt-get -y install build-essential automake ncurses-dev cmake libcmocka0 git \
pkg-config autoconf libterm-ui-perl libasound2-dev libalsaplayer-dev openssl libssl-dev apt-utils xterm \
curl ncurses-dev libsctp-dev libpcap-dev ca-certificates sip-tester
$ git clone -b 2. --depth 1 https://github.com/pjsip/pjproject.git
$ wget https://github.com/pjsip/pjproject/archive/refs/tags/2.9.zip
$ unzip 2.9.zip
$ cd pjproject-2.9
$ ./configure --prefix=/usr && make dep && make && make install
$ cd ..
$ git clone https://github.com/staskobzar/sippak.git
$ cd sippak
$ make
$ make test
$ make install
```
### nix package
Nix package is available in ```dist/nix/```. Inside this folder run
``` sh
nix-build
```
binary file will be available in ```result/bin```
## Try with docker without installing
```
docker pull staskobzar/sippak
docker run -ti staskobzar/sippak sippak --color --trail-dot sip:[email protected]
```
OR
```
docker build -t sippak .
docker run -ti sippak sippak --help
```
## Usage
```
sippak [COMMAND] [OPTIONS] [DESTINATION]
COMMAND:
Default commans is "PING".
PING Send OPTIONS packet to destination.
PUBLISH Send PUBLISH events and status. Default document is 'pidf' and default event is 'presence'.
SUBSCRIBE Send SUBSCRIBE request. Default event is 'presence'
NOTIFY Send NOTIFY request. Default event is 'keep-alive'
REGISTER AOR contacts list, register or unregister.
REFER Send REFER method outside dialog. Implements click-to-dial scenario as in RFC5359 #2.18.
This command requires parameter --to for Refer-To header.
MESSAGE Send MESSAGE method with text. SIP instant messaging.
INVITE Initiates and handles INVITE session. After session is confirmed (200) sends BYE.
OPTIONS:
-h, --help Print this usage message and exit.
-V, --version Print version and exit.
-v, --verbose Verbosity increase. Short option can be repeated multiple times.
Long option can have value. Example: --verbose=6
-q, --quiet Silent or quiet mode. Mute sippak.
--ns=LIST Define DNS nameservers to use. Comma separated list up to 3 servers.
Can be defined with ports. If ports are not defined will use default port 53.
For example: --ns=8.8.8.8 or --ns=4.4.4.4:553,3.3.3.3
--color Enable colorized output. Disabled by default.
--trail-dot Output trailing dot '.' at the end of each SIP message line.
--log-time Print time and microseconds in logs.
--log-level Print log level: ERROR, INFO etc.
--log-snd Print log sender file or module name.
-P, --local-port=PORT
Bind local port. Default is random port.
-l, --local-host=HOST|IP
Bind local hostname or IP. Default is first available local inface.
-u, --username=USER
Username part in Authentication as well as in Contact and
From header URI. Default is from destination URI.
-p, --password=PASS
Password for authentication.
-c, --contact=CONTACT
Custom contact header value. Must be valid SIP URI. For example: sip:[email protected]:123
If not set, contact header is generated automatically.
-F, --from-name=DISPLAY_NAME
Display name in From header. Default is empty.
-t, --proto=PROTO
Transport protocol to use. Possible values 'tcp' or 'udp'. Default is 'udp'.
-X, --expires=NUMBER
Expires header value. Must be number more then 0.
--pres-status=STATUS
Presence status for PUBLISH command. STATUS value can be 'open', 'closed' etc.
If this parameter is not defined or invalid, will use 'open' status.
--pres-note=MESSAGE
Presence note message string for PUBLISH command.
-C, --content-type=TYPE
Publish or notify content type. TYPE values can be pidf, xpidf or mwi.
Note: XPIDF implementation is not complete in pjproject.
-E, --event=EVENT
Presence event header for subscribe, publish or notify methods.
EVENT values can be "presence", "message-summary", "keep-alive" etc.
For convinence, there is an alias "mwi" can be used for "message-summary" event.
For PUBLISH and SUBSCRIBE method default value is 'presence'.
For NOTIFY method default value is 'keep-alive'.
-M, --mwi=N,N,N,N
Voice messages list. Comma separated list of numbers.
List of messages new,old,urgent_new,urgent_old.
List from 1 to 4 members. Not set members will be initiated with 0.
--mwi-acc=ACCOUNT
Voicemail account for message-summary body. If not set then destination URI used.
--clist
Flag for REGISTER method to get list of contacts registered for AOR.
--cancel-all
Flag for REGISTER method to cancel all registrations for AOR.
--cancel
Cancel registarations for REGISTER or session for INVITE.
When used with REGISTER method then cancels contact registration for AOR.
Contact field can be set with --contact option or will be generated.
When used with INVITE will cancel session in early state.
--to=SIP_URI
Parameter for REFER command to setup Refer-To header value.
--body=TEXT
Parameter for MESSAGE command to setup message body text.
--codec=LIST
Set codec or codecs to use during INVITE session. Value can be
a single codec name or comma-separated list of codecs. By default
sippak will use codec g711. Codec names are case-insensitive.
Following codec names can be used:
speex,ilbc,gsm,g711,g722,ipp,l16,amr,silk,opus,bcg729,all
When set to "all" will try to setup all available codecs for media.
--rtp-port=PORT
Port to use for media streams and negotiate with SDP.
-A, --user-agent=STRING
Set User-Agent SIP header value.
-H, --header=HEADER
Add custom header to request. Multiple custom headers (up to 12) can be added.
Parameter value must contain header name and value separated by colon.
Examples:
--header="X-Foo: bar", -H X-Foo:bar, -H "X-Assert: sip:[email protected]"
-R, --proxy=PROXY
Add proxy to request. Multiple proxies (up to 12) can be added.
The first proxy will be used as outbound proxy where the request will be sent.
All additional proxies will be added as Route headers to the request.
Supported methods are INVITE, REGISTER and PUBLISH.
Examples:
--proxy=sip:sip.com:2585, -R sip:10.23.24.100:6060;lr
```