February 2015 - freeswitch-biz@lists.freeswitch.org

[Freeswitch-biz] REQUEST FOR QUOTATION
by Nik San 18 Aug '15

18 Aug '15

Hello, I need someone to build me a system in order to sell VoIP, including FAX and SMS all in one, and of course the web interfaces for the admin and the users. Whoever is interested please send me over your contact details and also any information you have related to previous similar work you have done. I will get back to you with more details. Although much discussion needs to be done in order to decide on the final cost, I need you to inform me of your lowest amount you require for such a project. At least I will have an idea of your level of quotation and we will not waste each other's time. Don't forget to mention your preferred currency. Thank you, Nik

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[Freeswitch-biz] FreeSwitch-1.2.3 Version Issue Realted to HOLD Functionality
by Dhrupal Tilava 19 Feb '15

19 Feb '15

Hi All, We are using FreeSwitch Version 1.2.3. In this version we are facing the issue related to HOLD Functionality. Issue is when caller send the call hold request at that time freeswitch is not taking any action and call is not hold. But When Callee send the call hold request at that time freeswitch is taking the proper action and call is hold. We want to resolve this issue in this freeswitch version only. So Kindly Share your suggestion or send me answer on dhrupal.tilava(a)panamaxil.com. Regards, Dhrupal ----- Original Message ----- From: freeswitch-biz-request(a)lists.freeswitch.org To: freeswitch-biz(a)lists.freeswitch.org Sent: Friday, January 23, 2015 11:15:23 PM Subject: Freeswitch-biz Digest, Vol 16, Issue 1 Send Freeswitch-biz mailing list submissions to freeswitch-biz(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz or, via email, send a message with subject or body 'help' to freeswitch-biz-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-biz-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-biz digest..." Today's Topics: 1. Re: Sangoma looking for lead web and systems engineer (jay binks) 2. CSRP v0.2.1 (LCR and trunking platform) released (Alex Balashov) 3. Freeswitch assistance (Yaron Samuel) 4. Billing and LCR for FusionPBX (Luis Daniel Lucio Quiroz) 5. SS7 gateways for sale , Mediant 1000, 2000, 3000 (Dumi Voip) 6. Re: Freeswitch assistance (Luis Daniel Lucio Quiroz) 7. Business opportunity - need a hosted FS instance w/ GUI (Vance Gorke) 8. Freeswitch / LUA IVR Developer Position (hr(a)singo1.com) _______________________________________________ Freeswitch-biz mailing list Freeswitch-biz(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz

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[Freeswitch-biz] hosted solution to distribute calls on a % basis
by Vance Gorke 05 Feb '15

05 Feb '15

Hosted solution to accomplish the following I need to send about 500 calls a day to a sip address, (~5000 mins per day) I then need those calls sent to 3 - 6 US DIDs on a round robin approximate % basis. eg. 500 calls in DID 1 would receive approximately 33% ~165 -- in any given hour about 20 calls DID 2 would receive approximately 33% ~165 -- in any given hour about 20 calls DID 3 would receive approximately 33% ~165 -- in any given hour about 20 calls I'm happy to pay a hosting fee, and a rational termination fee per min. I need this solution ASAP thank you. -- Vance Gorke Great Oak Media 25 Broad Street, 3rd Floor New York, NY 10004 Phone 84-GREATOAK (844-732-8625) Email vance(a)greatoakmedia.com

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[Freeswitch-biz] text+ is looking for a Voice Engineer
by Voice Jobs 04 Feb '15

04 Feb '15

Location: Los Angeles, CA The Voice Over Internet Protocol (VoIP) Engineer will be responsible for designing, analyzing and implementing projects with the Voice Carrier infrastructure. The VoIP Engineer will work closely with the operations and engineering teams to deploy our SIP-based telephony solutions to a multitude of vertical markets. The position requires working with both hardware (in our data centers) and software to provide a Carrier Grade Voice cloud-based solution while interoperating with customer requirements. Responsibilities *The VoIP Engineer will work with internal and external NOC staffs to identify and mitigate escalated (tier 3) call completion issues *The VoIP Engineer will work individually and in a team environment with operations and engineering to drive new initiatives and complete existing projects for VoIP PBX offerings *Supports the existing network infrastructure and its need for security, stability and effective use of existing and future technology *Provides documentation and planning for all VoIP and infrastructure work *Delivers new projects using strong interpersonal, written and oral communications skills *Conducts research into networking issues and products as required *Ensures the stability, integrity, source control, change control and schedule for both existing and new products *Works with operations to resolve any ongoing implementation issues related to gateways and existing legacy systems. Requirements: VoIP Engineer Must have experience with: *SIP (OpenSIPS or FreeSWITCH) 3-5 years with VoIP 3-5 years supporting SIP G.711 in large / enterprise scale production environments 3-5 years with Unix Linux networking *Strong engineering and operational experience in service provider networks *In-depth understanding of SIP, RTP and RTCP interworking *Experience configuring and maintaining Open Source SIP Server Software including patching, upgrades, logs, debugs and traces (OpenSIPS or FreeSWITCH) *In-depth understanding of Open Source PBX features, configuration, deployment and support *Understanding of network server policies, routing network elements and collocation modeling *Ability to perform SIP debugs, capture and analyze SIP and RTP to isolate and resolve issues *Understanding of MOS Score, Jitter, Delay, Echo, Gaps, and how to troubleshoot these occurrences *Hands-on experience configuring routers and switches to maximize SIP / RTP pass through *Experience working directly with carriers and trunk providers *In-depth understanding of switching for a telephony standpoint into the PSTN *Must possess strong organizational, verbal, written and documentation skills *In-depth understanding of legacy telephony and legacy PBX interworking *Experience with IP PBX Trunking Media Gateways in NAT Environments Please send your resume to: voicejobs(a)gogii.net

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