Maintaining:

Asterisk Billing System,

Astersik PBX

Developing:

Custom applications using Asterisk or Freeswitch

How to apply:

send email with CV to: backoffice@winet.ch

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How to apply:

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* Full Time Job, 4-7 Years Experienced needed

*Configuration, installation and integration of Asterisk

* Development of SIP Client in C#.NET

* Maintenance of VOIP system

* Knowledge and experience with VOIP Technology and business

* knowledge and experience of A-Z Terminations

* Experience with ASTERISK A2Billing Software

* Strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used to calling stored procedure from MSSQL server). Modification of existing AGI will be required.

*Strong debugging.

*Working knowledge of databases preferable MS SQL 2008 server and also basic understanding of stored procedure.

*Knowledge of configuration of Audiocodes (VoIP Gateway) like Mediant 1000.

*Experience in configuration of SIP/IAX Phones.

* Knowledge of integration PSTN, T1/E1 lines with Asterisk.

* Strong Understanding of Linux server (preferably CentOS).

*Familiar with .Net Architecture, Web Services using SOAP

*Solid understanding of web-based architectures and programming, including data-driven e-commerce websites.

*A portfolio of previous telephony projects you’ve been a part of, great if you can demo it for us.

*Project organizational skills

You can work fulltime from home,

Permanent job but based on first year’s performance.

The company is based in Japan and hence you may visit if needed but the working location is your own location

How to apply:

Mail to aabusiness1@yahoo.com

I have my FreePBX setup so that incoming calls goto IVR and after the user selects option ‘4′ their call is sent to DISA where they get a dialtone and dial a number. This trunk that I have setup will allow only 1 channel/simultaneous call.

I would like to have additional callers into this IVR option ‘4′ be put on hold with custom MOH and then get pushed through DISA (through the trunk I have setup) once a channel is available.

Also, I would like absolutetimeout() to restrict this call time on this trunk to 3 minutes.

How to apply:

Please email

I am implementing one dialer type of application.

In which i am first dialing one source number and sending it to conference and then starting dialing the different destination numbers.

i have used meetme application of asterisk for this as i dont want to disconnect the main source number.

Both numbers are being originated using AMI and in the context i have used MeetMe with the specified room.

So, as soon as they got answered, they are being sent into the conference and they can hear each other and talk.

Everything goes fine with the above mentioned settings.

But, what i am trying to achieve is, I want to dial the other number from the meetme, so the source number can also hear the ringing sound of the other phone. So, the call must be originated from conference and get into conference as soon as the channel is originated.

For this i have used channel redirect function of phpagi-asmanager.php to send the dialing channel of destination number to conference. By doing this i can hear the ring sound, but when the call is being answered, the first user of conference is not able to hear the voice of second user.

I checked the asterisk CLI and debug the issue, And i found that the channel which we sent into conference without answer is having state “Down”. Can this be the issue of voice ?

Can anyone please advise me to fix the issue of sound with this configuration?

How to apply:

Contact me through my mail id shrikant.soni@globussoft.com

We are looking for Asterisk Developer to work within our VoIP and Data Convergence group; you will be responsible for leading development of the next generation of VoIP applications and features. As a member of the development team, you’ll be involved in all stages of the software development cycle, with particular focus on design and architecture.

Required Skills & Qualifications:

• College or University degree.

• Excellent understanding of VoIP telephony with a particular focus on SIP and the related protocols and standards.

• 2 years of hands-on experience with Open-Source VoIP Platforms Asterisk and Freeswitch, SipX, OpenSER, etc.

• Experience in troubleshooting and fine-tuning of data networks for VoIP using wireshark, tcpdump, ngrep, etc.

• Significant experience with Linux, including administration and optimization of high performance server-type applications. Also should have experience with maintaining a Subversion code repository to support the complete software development process.

• 5 year experience in programming in Linux environments, using any number of the following: Perl, C, C++, Python, AWK, Bash, etc.

• Good analytical and troubleshooting skills.

• Excellent written and verbal communications.

Assets:

• Familiar with .Net Architecture, Web Services using SOAP, and experience with building database applications MySQL, MSQL, PostgreSQL etc

• Experience with the integration of Microsoft Dynamics CRM with an Asterisk environment.

• Demonstrable history of technical leadership driving small development teams to deliver high quality software on time and on budget.

• Experience with call-processing architectures, including event-driven finite state machine(FSM) and multi-threading programming techniques.

• A portfolio of previous telephony projects you’ve been a part of, we would love to see a demo of them.

• Project organizational skills.

What We Offer

• How about the opportunity to join a fast-growing company and be part of a fun and dynamic team. You will be working directly with some of the best professionals in the world.

How to apply:

Please send your resume and cover letter to resumes@zimbra.gcinfotech.com. In your cover letter please reference the position you are applying for and provide a short introduction on why you believe you would be a good fit for the position.

Configuration, installation and integration of Asterisk for various applications like hosted PBX, IVR.

Experience in writing dial plan using extension.conf as well as extensiosn.ael.

Working knowledge of various PBX Manager like Trixbox, Druid, Thirdlane.

Strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used to calling stored procedure from MSSQL server). Modification of existing AGI will be required.

Strong debugging.

Working knowledge of databases preferable MS SQL 2005 server and also basic understanding of stored procedure.

Knowledge of configuration of Audiocodes (VoIP Gateway) like Mediant 1000.

Experience in configuration of SIP/IAX Phones.

Knowledge of integration PSTN, T1/E1 lines with Asterisk.

Strong Understanding of Linux server (preferably CentOS). Knowledge of Samba and ADS are highly preferred.

Basic knowledge of .NET will be highly preferred. (You may required to make small web panel using .NET application for asterisk context).

How to apply:

send email with resume to oj@achieveip.com

Asterisk Developer

Job Description:

We are looking for Asterisk Developer to work within our VoIP and Data Convergence group; you will be responsible for leading development of the next generation of VoIP applications and features. As a member of the development team, you’ll be involved in all stages of the software development cycle, with particular focus on design and architecture.

Required Skills & Qualifications:

• College or University degree.

• Excellent understanding of VoIP telephony with a particular focus on SIP and the related protocols and standards.

• 2 years of hands-on experience with Open-Source VoIP Platforms Asterisk and Freeswitch, SipX, OpenSER, etc.

• Experience in troubleshooting and fine-tuning of data networks for VoIP using wireshark, tcpdump, ngrep, etc.

• Significant experience with Linux, including administration and optimization of high performance server-type applications. Also should have experience with maintaining a Subversion code repository to support the complete software development process.

• 5 year experience in programming in Linux environments, using any number of the following: Perl, C, C++, Python, AWK, Bash, etc.

• Good analytical and troubleshooting skills.

• Excellent written and verbal communications.

Assets:

• Familiar with .Net Architecture, Web Services using SOAP, and experience with building database applications MySQL, MSQL, PostgreSQL etc

• Solid understanding of web-based architectures and programming, including data-driven e-commerce websites.

• Demonstrable history of technical leadership driving small development teams to deliver high quality software on time and on budget.

• Experience with call-processing architectures, including event-driven finite state machine(FSM) and multi-threading programming techniques.

• A portfolio of previous telephony projects you’ve been a part of, we would love to see a demo of them.

• Project organizational skills.

What We Offer

• How about the opportunity to join a fast-growing company and be part of a fun and dynamic team. You will be working directly with some of the best professionals in the world.

How to apply:

Please send your resume and cover letter to resumes@zimbra.gcinfotech.com. In your cover letter please reference the position you are applying for and provide a short introduction on why you believe you would be a good fit for the position.

We are looking for an experienced Asterisk developer with a background in C# for custom development of our system.

Experience with the following:

1. Computer-assisted dialing

2. Automated telephone call recording

3. Dot Net C#

How to apply:

Please contact me via email with your applicable experience at kallred@primary-intel.com.

We are customizing and integrating our Asterisk system with our in-house developed software solution.

We are looking for an experienced Asterisk developer with experience in the following areas:

1. Computer assisted telephone dialing

2. Telephone call recording functionality

3. Some .Net C# experience (to help with the integration efforts)

Hours and location are very flexible.

Pay will be commensurate with experience.

How to apply:

Please email me at kallred@primary-intel.com. Include your experience that you feel would be applicable.