Archive

Posts Tagged ‘SIP’

Online Help Site With Live Video Chat 2

January 20th, 2012 Comments off

WE need a basic site built to offer live help online.
Like “Ask a Lawyer or Accountant”

See these sites as examples:

http://www.liveperson.com/experts/business-finance/finance-accounting/?BanID=137644& gclid=CI7gnfm-gq0CFUOo4AodeAvzSQ

http://www.justanswer.com/sip/law?r=ppc|ga|1|Law|Law+Advice+%2D+2& JPKW=law%20advice%20websites& JPDC=S& JPST=& JPAD=8814752043& JPMT=b& JPNW=g& JPAF=txt& JPCD=20111112& JPRC=1& JPOP=Darren_Every9secHybrid_Test& gclid=CJGY7-O-gq0CFYPc4AodV0eewg

As you will see t…

Online Help Site With Live Video Chat

January 12th, 2012 Comments off

WE need a basic site built to offer live help online.
Like “Ask a Lawyer or Accountant”

See these sites as examples:

http://www.liveperson.com/experts/business-finance/finance-accounting/?BanID=137644& gclid=CI7gnfm-gq0CFUOo4AodeAvzSQ

http://www.justanswer.com/sip/law?r=ppc|ga|1|Law|Law+Advice+%2D+2& JPKW=law%20advice%20websites& JPDC=S& JPST=& JPAD=8814752043& JPMT=b& JPNW=g& JPAF=txt& JPCD=20111112& JPRC=1& JPOP=Darren_Every9secHybrid_Test& gclid=CJGY7-O-gq0CFYPc4AodV0eewg

As you will see t…

Support Engineer

January 8th, 2012 Comments off

Flashphoner looking for support engineer.
http://flashphoner.com

Skills
- Linux (advanced user/administrator)
- English (technical writing, spoken)
- Java – base, understanding of OOP, a short track

You will have learn this during working with us
- SIP specifications
- Java
- Working with logs

Responsibility:
- Tech support of the product
- Communicate with the customers via trucking system
- Reproduce issue in our/client environment
- Logs analysis (java, tcpdump)
- Search…

Decompile Android Apk To Get Sip Server Info For Voip

December 24th, 2011 Comments off

There’s a VoIP app on android that I’m trying to get server settings from so I can use the built-in internet calling of android instead of using the app itself. I’m thinking that it would be in the app code somewhere and should be accessible by reverse engineering the android apk. This should be fairly simple, I just don’t have the time to decompile the apk to get what I’m looking for. I already have the credentials to login with username and password so it would only be used to get the SIP serv…

Setup Full Voip Asterisk / Freepbx Server + Phonebook + Answ

December 14th, 2011 Comments off

Hi all,

I’m looking for a Sxriptlancer capable of installing a complete voip asterisk server.
We need a phonebook + answering machine + voicemail to email. We also want moh mp3 choice

disa
call forwarding
answering machine auto switch at 18h and switch off at 8 in morning.
We use grandstream phones 2000 2010 and 2020′s
5 sip trunks

8 phones.

install on debian ubuntu or FreeBSD
can you do this please let me know and your pricing.

Phones Numbers Generator + Bulk Call Center Programs Thru Pc

August 1st, 2011 Comments off

Hi,

please I am looking for a software program able to generate long list phones numbers and another one able to do or receive(outbound and inbound calls)massive auto call through pc (robot dialer)in a single click.

automated dialing software which can reach people by phone or by voice mail with personalized messages.Using computer to make calls through Voice over IP channel (SIP, SkypeOut) or through regular phone lines.

Those programs are sometimes use for telemarketing, political campaign promotions…etc

Thanks…

Predictive Dialer

July 22nd, 2011 Comments off

Our Company needs an experienced web programmer to create a web based predictive dialer.
Project Details:
1. A web based softphone. Built on Flex and capable of working using different SIP accounts.
2. Ability to call multiple numbers simultaneously.
3. When a call is “answered” the dialer will forward the call to our call agents.
4. Ability to detect busy signals, wrong number signals and answering machines.

Voipswitch Expert Needed 3

July 7th, 2011 Comments off

Hello, i have just setup my voipswitch server
i have installed all the modules, and setup.. i want an expert in voipswitch to check out if all the modules is working properly, like rates e.t.c
Basically i need these items Bellow to be setup
1 dialing Plan
2.signup with DID numbers
3.plan packs
4.automatic provisioning of all known SIP phones e.t.c
5.calling card setup
6.call back setup

you will be responsible to make sure everything works smoothly
I already have account with call terminators

I repeat, I need an expert in voipswitch

thanks

tony

Voipswitch Expert Needed 3

July 4th, 2011 Comments off

Hello, i have just setup my voipswitch server
i have installed all the modules, and setup.. i want an expert in voipswitch to check out if all the modules is working properly, like rates e.t.c
Basically i need these items Bellow to be setup
1 dialing Plan
2.signup with DID numbers
3.plan packs
4.automatic provisioning of all known SIP phones e.t.c
5.calling card setup
6.call back setup

you will be responsible to make sure everything works smoothly
I already have account with call terminators

I repeat, I need an expert in voipswitch

thanks

tony

Voipswitch Expert Needed 2

June 29th, 2011 Comments off

Hello, i have just setup my voipswitch server
i have installed all the modules, and setup.. i want an expert in voipswitch to check out if all the modules is working properly, like rates e.t.c
Basically i need these items Bellow to be setup
1 dialing Plan
2.signup with DID numbers
3.plan packs
4.automatic provisioning of all known SIP phones e.t.c
5.calling card setup
6.call back setup

you will be responsible to make sure everything works smoothly
I already have account with call terminators

I repeat, I need an expert in voipswitch

thanks

tony

Voipswitch Expert Needed

June 24th, 2011 Comments off

Hello, i have just setup my voipswitch server
i have installed all the modules, and setup.. i want an expert in voipswitch to check out if all the modules is working properly, like rates e.t.c
Basically i need these items Bellow to be setup
1 dialing Plan
2.signup with DID numbers
3.plan packs
4.automatic provisioning of all known SIP phones e.t.c
5.calling card setup
6.call back setup

you will be responsible to make sure everything works smoothly
I already have account with call terminators

I repeat, I need an expert in voipswitch

thanks

tony

Phone/data-entry Assistant

June 16th, 2011 Comments off

ExcellenceOnTime is one of the largest buyers on oDesk. We need to validate about 6000 data records by calling each one (using a VOIP account that we will be providing; you will need a headset and a piece of SIP-compatible software that is free on the web), entering data, and validating the work of others.

http://bin.nu/arq

Categories: Data, Phone, SIP, VoIP Tags: , , , , , ,

Install Vici Dialer On Lamp System

May 27th, 2011 Comments off

Install Vici Dialer to Cent OS 5.4
Configure Vici dialer to connect to SIP Trunk

Website Sip Client

May 20th, 2011 Comments off

I am seeking a programmer who is able to write a website-based SIP client to handle inbound calls. It will preferably have user-selectable mic/speaker (but default recording and playback devices are an option here), support auto-answer (configurable to be user selectable or forced), and sending of DTMF tones. I will need to be able to read the status of this from a the webpage (ie. connected:123456; status:waiting;) by polling it in the background with javascript preferably. Support for G.711 and G.723.1 should be configurable.

It must run on Windows XP/Vista/7 and OSX/Linux would be an added bonus. Java and flash are available to work with.

Seems to me java has a SIP API so it might be the best route. I’m not opposed to exploring the use of Adobe Flash but I don’t believe it supports it natively and might be too complicated to get setup. Please leave a comment indicating what you’re thinking this should be built in.

Preference will be given to those who can demonstrate they have a solid understanding of my needs and are able to complete this project without issue. I don’t expect you to support the software once it is built but if you’re available for future consulting please advise. Full source code (where available) will be required upon successful completion of this project (sloppy coders beware.)

Nimbuzz Clone

November 9th, 2009 Comments off

Fring or Nimbuzz clone (www.fring.com . www.nimbuzz.com),SIP mobile dialer
I need customized Fring/Nimbuzz clone to work with my asterisk based server (SIP) and to can work in most of the mobile phones (windows CE , Symbian ,Iphone and other old phones) via 2G 3G 4G GPRS or WIFI connection
-making and recieving a Voip calls
*Showing the SIP phone number
*Showing the Balance
*Showing the rate of the minute for the dialed number
*to be able to use phonebook
-sending and recieving SMS
-IM
-very important is to block changing the ip of the SIp server in order to work just with my server

Categories: Programming Tags: , , , , , ,

Fring Or Nimbuzz Clone

November 6th, 2009 Comments off

Fring or Nimbuzz clone (www.fring.com . www.nimbuzz.com),SIP mobile dialer
I need customized Fring/Nimbuzz clone to work with my asterisk based server (SIP) and to can work in most of the mobile phones (windows CE , Symbian ,Iphone and other old phones) via 2G 3G 4G GPRS or WIFI connection
-making and recieving a Voip calls
*Showing the SIP phone number
*Showing the Balance
*Showing the rate of the minute for the dialed number
*to be able to use phonebook
-sending and recieving SMS
-IM
-very important is to block changing the ip of the SIp server in order to work just with my server

Categories: Programming Tags: , , , , , ,

Voip Sip Project

October 12th, 2009 Comments off

Sipme.me
Mobile Client – SRS

Client Target Device:

Basic SIP steak installer and management client which will make sure sipme service is defined on top of os and will manage it according to the defined in the Feature set section.
Feature Set
PJSIP – Open Source SIP Stack Client

Application:

R – 1. Application will provide sip in integrated way with telephony services of operating system, such as Incoming/Outgoing/Missed calls logs SMS and voicemail messages.

R – 2. SIPME client will be confront to OS user profile settings such as:
 Silent
 Meetings
 General
 User custom profile.

R – 3. Allow initiating calls from regular phone dialer / or contact lists

R – 4. The system will remove domain part of the caller id.

R – 5. The client will enable only service from sipme service and not any other sip provider.

R – 6. The application will be opened automatically when handset is turned on.

R – 7. Application will not allow user to turn it off and will always run in the background.

R – 8. sipme service indicator implementation on handset status bar

Networking:

R – 9. WiFi / 3G smart management

Always prefer WiFi over 3g – If service disconnected
a. if WiFi available
i. connect to WiFi
b. else
i. connect to 3G – if 3g fail connect again
WiFi available criteria
- no security / known WiFi network
- reception is above XX%
- WiFi has internet access.

R – 10. kept alive / refreshing the net – tcp , udp

R – 11. Listen to the network 3G data traffic counters and warn user on exceeding data network usage limitation according to – Billing day at operator – Size of internet bundle of the mobile subscriber.

R – 12. Auto disable service over 3G when roaming to other carriers – user define.

R – 13. Client will support working from port 80 in order to bypass potential firewall issues.

R – 14. The system will enable working from fallback barriers;

i. Auto Fallback to EDGE by handset will not break the connectivity to the Register Server of SIP, if it does than the client will re-register to the service.

SMS:

R – 15. SMS over IP implementation
i. Receive into SMS inbox
ii. use regular SMS mechanism to send SMS over IP to sms getaway

Telephony & Communication:

R – 16. Default barrier for outgoing call, will be internet call via SIPME service.

R – 17. Supported codec’s: AMR, G.729 A/B , G.711 ,pcmu ,pcma ,ilbc ,cn and more.

R – 18. Client will indicate of a new voicemail in incoming SMS.

Happy Bidding!

Voip Sip

September 28th, 2009 Comments off

Hi,
We need to develop a sip client for symbian , windows mobile and other os
pjsip is Possible
WiFi / 3G smart management
integrated way with telephony services of operating system, such as Incoming/Outgoing/Missed calls
initiating calls from regular phone dialer / or contact lists
Full srs can delivered

Cti For Mac Os For Sip_phones

August 24th, 2009 Comments off

We need a simple CTI-Client which communicates with the phone directly, communication with the pbx is not needed.

We need this features/functionality:
- call handling (answer,end,hold,transfer)
- dialing from MAC OS adressbook
- dialing from mac os filemaker-database (command development is sufficent gui/mask/field in database will be created by us)
- namelookup from interal/central/phonebook and mac os adressbook(xml-format; template is available)

More features like callerlist(missed,lastdialed,answered) are welcome but not mandatory.

No preferred scriptlanguages. Sourcecode must be delivered to us and must bechangeable by us.

Cti For Mac Os For Sip-phones

August 24th, 2009 Comments off

We need a simple CTI-Client which communicates with the phone directly, communication with the pbx is not needed.

We need this features/functionality:
- callhandling (answer,end,hold,transfer)
- dialing from MAC OS adressbook
- dialing from mac os filemaker-database (command development is sufficent gui/mask/field in database will be created by us)
- namelookup from interal/central/phonebook and mac os adressbook(xml-format; template is available)

More features like callerlist(missed,lastdialed,answered) are welcome but not mandatory.

No preferred scriptlanguages. Sourcecode must be delivered to us and must bechangeable by us.

Cti For Mac Os For Sip

August 24th, 2009 Comments off

We need a simple CTI-Client which communicates with the phone directly, communication with the pbx is not needed.

We need this features/functionality:
- callhandling (answer,end,hold,transfer)
- dialing from MAC OS adressbook
- dialing from mac os filemaker-database (command development is sufficent gui/mask/field in database will be created by us)
- namelookup from interal/central/phonebook and mac os adressbook(xml-format; template is available)

More features like callerlist(missed,lastdialed,answered) are welcome but not mandatory.

No preferred scriptlanguages. Sourcecode must be delivered to us and must bechangeable by us.

Categories: Programming Tags: , , , , ,

Voip Softphone

August 11th, 2009 Comments off

I need a voip softphone;

I will use this softphone with my label but for other sip proverders.

firstly It should login from my web page database (mysql).
then it will take sip server adress, user name and password from same database.

then it can make sip calls.

thera are open sourcec softphone like http://www.qutecom.org/
also there are free softhones http://www.zoiper.com/

I want a softphone just to make phone call
no chat
no video
or other extra properties
only with my label and
it will login from my datebase and it will take sip settings from my database.

Asterix / Trixbox Troubleshoot

June 24th, 2009 Comments off

Hi,
We need an experience asterix / trixbox phone server administrator to help us fix the issue below. Either by remote administration or live chat skype/msn etc.

Please let us know -
1) your experience with trixbox
2) cost to fix issue below.

Thanks,
Mitch

#############ISSUE##############
I am having an issue with an openvox A400P card with 4 FXO modules in a trixbox 2.8.0 system.

When an incoming call comes in the sip phones will ring then stop ringing, record a missed call, realise the call is coming in and pick it up for a few seconds then drop again.

it usually picks up and drops 4 or 5 times in the time of 1 incoming ring.

I watched the logs in real time to see what is happening and found that it was detecting a “red alarm” when the call dropped.

[Jun 24 15:53:56] VERBOSE[3726] logger.c: — SIP/102-09fe3c20 is ringing
[Jun 24 15:53:56] VERBOSE[3726] logger.c: — SIP/102-09fe3c20 is ringing
[Jun 24 15:54:02] WARNING[3726] chan_dahdi.c: Detected alarm on channel 2: Red Alarm
[Jun 24 15:54:02] VERBOSE[3726] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on ‘DAHDI/2-1′ in macro ‘dial’
[Jun 24 15:54:02] VERBOSE[3726] logger.c: == Spawn extension (ext-group, 602, 21) exited non-zero on ‘DAHDI/2-1′
[Jun 24 15:54:02] VERBOSE[3726] logger.c: — Hungup ‘DAHDI/2-1′
[Jun 24 15:54:04] NOTICE[3332] chan_dahdi.c: Alarm cleared on channel 2
[Jun 24 15:54:05] VERBOSE[3730] logger.c: — Starting simple switch on ‘DAHDI/2-1′

It then clears the red alarm and starts ringing after a couple of seconds just to do it all over again

Originally I thought this issue was with only 1 phone line, but over the last day it has affected all of the lines at least once.

I have struggled to find any info on the forums and google around this, I did find a few people saying that dahdi has only recently started checking the alarms, so im hoping there is a way to disable this check or any other way to solve the issue.

Thanks,
Mitch

P2p Voip

June 5th, 2009 Comments off

Hello,

This is more of a development test project so the end result does not have to be an entirely pretty polished interface, but a working example. (very low bandwidth use is important, only for voice on VOIP not music, so keep it as low as possible using lowest codecs)

The goal is to create a P2P VOIP appliction in visual basic, that will also allow IM and file transfers (small pictures). The system should have an address book, and open channels which are essentially like radio channels. as well as the VOIP ability to call an individual.

So the system would work as follows, when started there would be a default ‘Channel’ which is like a big confrence room that each peice of software is set to. So that when you communicate through VOIP on that ‘Radio Channel’ like a CB radio everyone can hear that. There should be the ability to create an unlimited amount of ‘Channels’ with any name required. so for example an event could be given a new channel and it could either be an open channel that anyone can join. or a passworded channel.

When someone joins that channel, any IM message / Broadcast message which is not specific to one person gets sent to everyone on that channel through P2P, as with file transfers. The system should be encrypted.

In addition there should also be the ability to have an address book, so if users were not on a channel, or they wanted to talk to a specific person they could use that VOIP call that would be direct to that person, not in the confrence call.

Another feature would be the ability to ‘listen’ into multiple ‘Confrence’ calls. So in essence there would be multiple radio channels much like a CB and the software can be told to monitor one channel, which might light up on the screen when talking comes up on that one, and have the radio set to monitor and talk on another channel.

One other feature is there would be two versions of the software, one mobile and another a server like desktop version. So the mobile software can all commuincate to eachother even if the server is not there, but the mobile can also communicate with the desktop software which would have more messaging abilities, and more administrative abilities in setting up mobile nodes and the likes.

Another important feature is the ability to updates to the software to be distributed automatically from the desktop version.

In terms of software, the mobile versions would all be on a big LAN, and desktop versions most likely on WAN so some ability to communicate would be important between the two, even if the mobile versions send a register command to the WAN desktop servers to get lists of who is on, radio channels, information and then update that information so that if connection is lost they can still function.

Any questions please use Public Message Board and i can clarify and give diagrams and pictures and the likes.

Keepiong budget to a minimum is important as well as its a trial idea build not a finished project, and i would need copies of all source code and visual basic project files so i could modify that if needed.

You can use existing open source VB solutions if they help achieve the end goal, but basically i want the end application to be all in one, not an application that interfaces to an existing softphone or something, it needs to be in the app.

Any suggestions let me know, Additional information submitted:

06/04/2009 at 7:33 EDT:
One other important feature would be to allow for at least in desktop software logins with different permissions (eg use, settings, supervisor, guest) and that all the desktop consoles have an ability to link up automatically with each other and the mobile software somehow.

In addition a transactions database should be maintained within the software so that logs of messages / files being sent are kept as well as calls being made etc so stats can be provided and logs are kept in a database of some sort.

06/04/2009 at 7:51 EDT:
System needs to be true VOIP P2P, so would be able to function without a server being present. But if a server (being any desktop software) was present, it would detect it and get all channels, settings, information and retain that in case the server disappeared.

Cant rely on the server though, each software node needs to be able to broadcast to discover both LAN and WAN nodes and list those and their status and information in like a buddy (or present nodes) list, not have to enter IP addresses. Kind of like DHT i think.

06/05/2009 at 5:13 EDT:
Could use one of the open source P2P SIP projects possibly provided they can be used in Windows or converted to windows and provide the functionality i need (including NAT traversal)

Listed below are some P2P SIP projects that might be worth having a peek at to use as part of the project.

http://sipdht.sourceforge.net/sipdht2/index.html
P2P SIP using DHT2

http://www1.cs.columbia.edu/~salman/peer/download.html
Open P2P IM / VOIP system

Either way it needs to be meeting the requirements of above, and i guess idiot proof in that it cant require servers, or users to enter IP addresses, should just work.

Whilst it cant require servers, it can download information from desktop systems to get more information about other nodes, both in internal and external networks, but it should keep copies of those tables so if the desktop nodes are not present, it has the ability to find nodes in the private network, or on the WAN.

I guess it is really client-less, all nodes become super-nodes in the DHT sense.

Cti For Mac Os For Sip

June 3rd, 2009 Comments off

We need a simple CTI-Client which communicates with the phone directly, communication with the pbx is not needed.

We need this features/functionality:
- dialing from MAC OS adressbook
- dialing from mac os filemaker-database (command development is sufficent gui/mask/field in database will be created by us)
- namelookup from interal/central/phonebook and mac os adressbook(xml-format; template is available)

More features like callerlist(missed,lastdialed,answered) are welcome but not mandatory.

No preferred scriptlanguages. Sourcecode must be delivered to us and must bechangeable by us.

Categories: Programming Tags: , , , , ,

Voip/sip Answering Machine Mac

April 8th, 2009 Comments off

I need a simple answering machine which stores recorded calls in seperate audio files. It must connect to an existing SIP account and the script must run on Mac OSX and should be done in Python, PHP or Perl. Maybe a modified version of http://divmod.org/trac/wiki/ShtoomProject

No GUI is necessary, we can modify conf files ;-)

Bear