- freeswitch-biz@lists.freeswitch.org - FreeSWITCH Mailing Lists

[Freeswitch-biz] Freeswitch + Icecast conferencing
by Josh Horton 22 Jun '15

22 Jun '15

Hello, I need an application that will broadcast a conference bridge to thousands of people. I initially built an asterisk system with the ices() module + icecast but ran into reliability issues. I would like to use Freeswitch and mod_shout for this. Before I spend a lot of time and $ in development, I thought I would see if anyone on this list already has this solution for sale. Thank you, Josh

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[Freeswitch-biz] Sipp call Test Fail
by Jeevan Dhami 18 Jun '15

18 Jun '15

Dear Sir Please help me , when i was sending 1200 simultaneous call to freeswitch using sipp it was working fine. but i tried to increase simultaneous call (>1200 ) there was freeswitch not respond well. -- *Jeevan Singh Dhami* *(9650541245)*

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[Freeswitch-biz] DID partner for Australia
by Tamas Jalsovszky 28 Apr '15

28 Apr '15

Hello, I'm looking for a DID partner with call termination capabilities in Australia. FreeSWICTH on the other side preferred :) Due to latency questions I'm interested in partners having direct POP in Australia. Regards, Tamás ps.: Call volume is pretty small at the moment.

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[Freeswitch-biz] CSRP v0.3 release
by Alex Balashov 09 Mar '15

09 Mar '15

Hello, Evariste Systems announce with pleasure the newest major release of the Canonical SIP Routing Platform (CSRP), a SIP trunking service delivery platform based on Kamailio and PostgreSQL. CSRP combines a high-performance Least Cost Routing engine with a sophisticated and flexible business layer to provide an advanced "Class 4" solution for origination and termination routing, rating and accounting to the ITSP industry. It is used by a range of SIP operators, from conversational SIP trunking providers to wholesale termination companies specialising in high-volume traffic, as well as application providers such as call broadcasters and hosted IVR companies. The v0.3 release provides many long-awaited improvements, including: - An improved CPS (Calls Per Second) rate-limiting mechanism. - Pseudo-asynchronous call processing for improved CPS performance and throughput. - Global termination ANI blacklist. - PDD-sensitive failover across termination vendor routes. - Redis-based LRN and CNAM dip caching (to replace Memcached). - Full topology hiding support. - Integration of optional outboard SBC/B2BUA component. - Many API and GUI improvements in functionality and usability, including better online help and navigation. - Signalling-based dead peer detection/keepalive support. At Evariste, we take pride in our established reputation for high-value vendor support and deep technical expertise. Moreover, we offer CSRP at a flat licence fee irrespective of number of channels, calls, hosts, ports, etc. If you are interested in learning more about CSRP, please direct enquiries to sales(a)evaristesys.com or give us a call at +1-678-954-0670. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/

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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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[Freeswitch-biz] Freeswitch / LUA IVR Developer Position
by hr＠singo1.com 23 Jan '15

23 Jan '15

Singo Solution provides hosted software contact management solutions for a number of industries - including call centers, utilities, telecom providers, education, and medical. Singo has an opening for part time contracting position as a Freeswitch/LUA IVR Developer. This will be a remote position. Task will be given, developer can accept or deny and be paid hourly at half hour intervals. Required skills include: § Fluent knowledge of Linux (in particular, Debian 7.0) and MySQL § Extensive knowledge of Freeswitch & LUA 5.2 § Ability to interpret and quickly understand call flow and IVR logic. § Ability to make recommendations to prevent and handle user input errors. Send your resume to <mailto:hr@singo1.com> hr(a)singo1.com for immediate consideration. Thanks!

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[Freeswitch-biz] Business opportunity - need a hosted FS instance w/ GUI
by Vance Gorke 22 Jan '15

22 Jan '15

Business opportunity -- I'm looking for a Freeswitch Hosted solution with an Admin interface to handle about 1 MM short duration calls per month. SIP in SIP out to my carriers. - IVR support - DTMF routing - IVR self-administration optional but desired - basic reporting interface/dashboard based on user login a must - Self admin call routing -- additional detailed requirements can evolve Please provide your estimated costs for this service. 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] Freeswitch assistance
by Yaron Samuel 12 Jan '15

12 Jan '15

Hi Guys, I am after a recommendation for someone who can help me migrate my Freeswitch installation to a new server. Also need a Ruby on Rails expert to look after my billing system Thanks! Yaron

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