Bounty
From FreeSWITCH Wiki
Bounties paid
Its just as important to know who has a history of paying and who has a history of completing tasks, so the Bounties Completed page lists those.
Alternative behavior when using P-Asserted-Identity and Privacy headers
$300 - Oyatel AS (eaf@oyatel.no) It's currently not possible to read the values of P-Asserted-Identity and Privacy headers on the A-leg of a call. This might be information that is handy on routing calls when using FreeSwitch in an SBC context. Also FS alters the Caller-Username and Caller-Caller-ID-Number based on the content of the P-Asserted-Identity.
Current behavior:
INVITE FROM ASTERISK to FS:
From: "+4711111111" <sip:+4711111111@xxx.yyyyyyy.no>;tag=as24269fa6
Contact: <sip:+4711111111@1.2.3.4>
Remote-Party-ID: "+4711111111" <sip:+4711111111@xxx.yyyyyyy.no>;privacy=off;screen=yes
P-Asserted-Identity: <sip:+4712121212@xyz.no>
Privacy: none
This information is gathered at FS with the Info app.
Caller-Username: [+4712121212] ;---- FS changes this field based on content in P-Asserted-Identity
Caller-Caller-ID-Name: [+4711111111]
Caller-Caller-ID-Number: [+4712121212] ;---- FS changes this field based on content in P-Asserted-Identity
variable_sip_from_user: [+4711111111]
variable_sip_from_uri: [+4711111111@xxx.yyyyyyy.no]
variable_sip_contact_user: [+4711111111]
variable_sip_contact_uri: [+4711111111@1.2.3.4]
Then FS removes P-Asserted-Identity header.
We would like an alternate behavior:
Maybe by setting something on a sip- profile to change from default behavior?
Alter_default_p_asserted_behavior=true
True: P-Asserted-Identity can be read from the channel variables, and is possible to export to leg-b. Caller-Username and Caller-Caller-ID-Number are not changed.
False: Like current behavior
False is default setting.
Jira ticket http://jira.freeswitch.org/browse/MODENDP-146 might contain some relevant info.
Active Directory Domain Support
$200.00 - Jimmy A. (techiegz@gmail.com) A good way to accomplish this is to ensure FreeSwitch user and group directory services inter-operates well with Likewise Open components within Linux and Unix distributions. This is because Likewise Open (which provides Active Directory authentication, management, etc for linux and unix systems on Windows enterprise networks) is starting to ship with Linux and Unix distributions according to Linux Magazine; http://www.linux-mag.com/id/5620 . Ubuntu and RedHat are the first couple of distributions to go this enterprise-ready route followed by Novell, IBM, etc. FreeSwitch should not be left out.
H.323 Support
Especially h264 (a portion of the MPEG-4 standard, referenced as MPEG-4 part 10), uses less than half the bandwidth of any other video codec. It is also specially developed for IP networks.
Mono support
*Completed, waiting for payment*
Update the existent mod_mono module to the current FS source code with basic functionality, event api, application api
- $1000 - Segtel A/S - Contact: Martin Sørensen mrs@segtel.dk
PRI STACK
Create/design a PRI stack with basic functionality and maintenance codes:
- $1000 - Voctel Communications - 705-223-2000 - Contact: Richard Cook
- We can write Pri Stack for freeswitch
- Coral Telecom Ltd +919891499321 - contact: Arun Sharma Mail arunsharma@coraltele.com
Call Timers
- NOTE: This is completed, awaiting payment of bounty
Create a paramater to BRIDGE application which will disconnect a call after X amount of time
- $100 - Wavelength Communications, Inc - 703-459-7429 -- Contact: Daniel Corbe <dcorbe@gmail.com>
- $150 EXTRA (total of $250) Ability to play announcements to the caller (callee must not hear) to warn of impending disconnect
- Must drop callee side, not caller side so that continue_on_fail type scenarios continue to process
- $100 - Nikolay Kolev - me *at* nikolay *dot* com
Continuation of a call in No-Media Mode after Successful Bridge
- NOTE: This is completed, awaiting payment of bounty
- $200 - Wavelength Communications, Inc - 703-459-7429 -- Contact: Daniel Corbe <dcorbe@gmail.com>
- A call goes into the "HIBERNATE" state if you have no-media mode enabled. This prevents the call from
matching multiple condition blocks in an extension if the outgoing call leg was BYEd (meaning a successful bridge occured).
For example, the following Dialplan works with no-media mode disabled. With no-media mode enabled, the 2nd <condition> block is never executed.
<context name="default">
<extension name="globals" continue="true">
<condition>
<action application="set" data="continue_on_fail=true"/>
</condition>
</extension>
<extension name="ld_service" continue=true>
<condition field="destination_number" expression="^1(.+)$">
<action application="set" data="no_media=true"/>
<action application="set" data="hangup_after_bridge=false"/>
<action application="bridge" data="sofia/local_profile/1231@127.0.0.1:5061" />
</condition>
<condition field="destination_number" expression="^1(.+)$">
<action application="set" data="no_media=true"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/local_profile/1456@127.0.0.1:5061" />
<action application="bridge" data="sofia/local_profile/1$1@127.0.0.1:5061" />
</condition>
</extension>
</context>
SS7 support
- $1000 contact me 516-356-1080
"FreeSWITCH Internals"
- $100 if someone can throw together a guide documenting some freeswitch internals so I can begin contributing to the project more meaningfully 516-356-1080
- Can you give an example of the internals that need documenting? There is this page: http://wiki.freeswitch.org/wiki/Documentation/Developer_Documentation
Voice & Tone Detection
- $250 (and looking for bounty pooling help from others) to add voice and tone detection capability. This is for an application that needs to determine either when someone has finished initial greeting (such as "hello" followed by silence) or when an answering machine completes the greeting and tone prior to recording a message. This capability should work for any of the codecs freeSWITCH supports. Contact Del at del_stevens@yahoo.com
- $250 more to the pool. I'm specially interested in VAD in early-media, that is, being able to detect voice activity during ringback. Contact at cesar@auronix.com
Integrate OpenSS7 into FreeSwitch
- $1000 (perhaps more) to contribute effort towards successfully integrating OpenSS7 stack into FreeSwitch. Contact Ahmed at [worldentropy@yahoo.co.uk mailto:worldentropy@yahoo.co.uk] to discuss.
(This can't happen due to OpenSS7 being GPL, It can only happen over a socket)
spanDSP + t.38 (origination, termination, & gateway) in Freeswitch
- $200 ( maybe more) to make spanDSP part of freeswitch and support faxing through g711 and t38. Contact ]technophreak[ on #freeswitch
- Looking for more contributors to make it happen.
G729 Licensing Bounty
( WE NEED 5000 license to do this )
<jay > MikeJ : Saying "WE NEED 5000 license to do this" does that mean 5000 x $50 per license ??
<MikeJ> ballpark price will be competitive with whats out there
<MikeJ> it wouldn't be over 10 per chan
for annex a+b
we are still looking at if we will do for less than that or not
50~200 licenses Voicemeup Contact ]technophreak[ on #freeswitch
250 licenses [stochastic23 at gmail.com or stochastik on #freeswitch]
50~100 licenses [Mongos on #freeswitch]
100 licenses Chance Telecom Contact [wchance@chancetelecom.net mailto:wchance@chancetelecom.net]
100~150 licenses [djin on #freeswitch]
200~400 Licenses [bcoppens on #freeswitch], supported with CNG generation and VAD.
200 Licenses [chris at fursman.com or c6burns on #freeswitch]
- TOTAL : 950~1400
Port FreePBX to FreeSwitch
Port the FreePBX WebGUI to FreeSWITCH
Based on the thread in the trixbox forums
* $1000 Tony Lewis SchmoozeCom * $1000 Ward Mundy $500 Nerd Vittles $500 PBXinaFlash Deadline 02/02/09
SIGHUP support in Windows - CDR/LOG Rotation
Provide command line ability to send event (SIGHUP). (fsctl sighup)
Action should:
* Not drop any calls/state * Rotate Master.csv file * Rotate logfiles
Fee to be paid: $75.00 Contact: Shawn -> shawnl@waterwheelnets.com
CDR_CSV Auto Rotation
Note: this is done, waiting for paiement.
Ability to AUTO-ROTATE CDR MASTER.CSV every "x" seconds/minutes whatever
* Add new conf parameter which will allow for an automatic rotation of the Master.csv every set period of time.
Fee to be paid: $75.00 Contact: Shawn -> shawnl@waterwheelnets.com
Note: I don't see the point of this. It's totally unnecessary as you can do this with logrotate easily.
Call Center using freeSWITCH
Looking for strong developer to help in implementing call center type application where use agents as XLite(SIP client) to run in call center environment and do outbound calling based on agent's presence.
- $3000 BSingh [onlyreplyme@gmail.com]
mod_capi
I would like to do the mod_capi for FreeSWITCH, Is anybody interested in contribute with a quadGSM and give to me.... :)
- contact javar on #freeswitch
Realtime FollowMe
Looking for a realtime version of the Asterisk FollowMe application. The application would need to collect prompts and numbers from a MySQL database.
- $800 CSmith [craigesmith@gmail.com]
Dialplan continuation on failed call
Allow to continue in the dialplan once the brigdged call is finished and use the call ending code as a next condition. For example : - to allow to process a failed call with bypass/proxy_media through a normal bridge with codec transcoding - to intercept 4xx/5xx message to play a friendly message to the caller
(no bounty offered atm).
mod_php updated to work with current FreeSWITCH
Any idea what sort of bounty would be required to make this happen? (and if there are other supporters of it) I am migrating a number of Asterisk systems which use PHP-based AGIs that will need to be re-developed for freeswitch and mod_php would be the easiest way of doing this.
Generate and be notified of SIP INFO messages
Completed by anthm on April 28.
Posted: April 15th, 2009
Bounty: $750; extra $500 ($1250 total) if completed before May 4th, 2009.
Bounty valid until: May 17th, 2009
Respond to: Bill Belanger - wj.belanger@gmail.com
Requirements:
2 Features required:
1: Receive an event when a SIP INFO message is received
- Register for an event using event_socket's event command
- Receive as an event through event_socket the headers and data contained in the INFO message
- Optional - Supply a way to respond to received SIP INFOs with a 200-OK
2: Create and send a SIP INFO message using event_socket
- Specify a framework to generate and send SIP INFO messages for an established call
- This framework must be able to accept custom headers, as well as data for the SIP INFO message
- Although not required, it is expected that this framework would be usable through event_socket's sendmsg command
- It is required that whatever framework is used, it is accessible through event_socket
If you decide to work on these features, please send a notification to Bill Belanger with an expected completion date.
RFC 3611 - RTP Control Protocol Extended Reports (RTCP XR) support
The goal would be to measure/collect voice quality details and report them during the live call (e.g. update channel variables which could be read through event_socket, xmlrpx, etc.) and report at the end of the call (event_socket message or/and CDR data). It has to work on win32 and linux for sure.
- $1000 eWorld Com, contact jalsot at gmail dot com
- $500 Netsip Pty Ltd, Contact jay@netsip.com.au
- more fund wanted
