I get this error when making a call on vicidial. *********** WARNING: chan_sip.c:21126 handle_response_invite: Received response: "Forbidden" from '"80" <sip:+18xxxxxxxxxx@[url removed, login to view]>;tag=as0c9f627d' *********** On Twilio site Debugger i get this error this. ********************* Error - 32204 MESSAGE Invalid Caller
We have blog articles/content that will need Infographics Design will be based on the c...you have samples and good reviews, we will hire for multiple projects. Examples of infographic subjects include: 1. Internet Options (DSL, Cable, Fiber) 2. VoIP options (hosted versus SIP trunking) 3. SaaS solutions for SME's Thank you for checking in.
Hi Malay P., I would like to discuss a project with you, I can't find anyway this system won't create a whole job :| I'm looking for a SIP/RTP Proxy with Repacketization I have a system already in place which spits out 30-45ms packets, I need them 20ms or less when they head off to the provider. Can you let me know what you're thinking that would cost
...Debian with postfix and Dovcot My epost clients is Microsoft outlook 2015 I work a lot with customers or elsewhere. I experience up that email my not work because SIP / network admin has sprret port 25 (SNMP) to that I must use their SMTP server to send mail. I've tried to open my server with 2 ports. 25 and 9800 it did not solve the problem
I am looking for a teacher to give CISCO Courses, English speakers, Cisco CCNA CCNP Juniper JNCIE JNCIP JNCIS JNCIA interested people send the course that are able to teach online.
...on a monthly retainer basis. I already have the server running and your first job would be to secure it / put in place everything required to prevent BRUTEFORCE attack of my SIP, SSH and other usual attacks on those type of servers. Once this is done, I will pay you monthly to do maintenance on my server and from time to time a few adjustments.
Needed: Experienced PHP programmer for VoIP needed, knowledgeable in telecom programming particularly GoIP sim bank gateways, GoIP API programming, Asterisk A2Billing and FreePBX, and Cisco. Description: Create an API to control the actions of a GoIP sim bank gateway by Call Progress Tones. GoIP sim gateway is located in a third world
Necesitamos unos textos pero de alguien que sepa sobre telecomunicaciones, SIP TRUNK, Servidores dedicados, IVR, reconocimiento de voz. A la persona escogida le indicamos nuestro producto y debe diseñar unos textos TECNICOS sobre lo que necesitamos. Es importante que la persona conozca sobre que se trata y hable de ello con propio conocimiento y tecnica
Hello, We are in a need of a secure and encrypted VoIP client 1st for Android, 2nd for iOS. The VoIP client may be using Open Source libraries, and has to be based upon ZRTP (up to) SRTP protocols. The protocols and technology can be used from open source libraries. Other details which has to be customized are - guidelines; - Sandbox application
We need a skilled sales representative who will support our company in selling and buying VoIP routes. Voip experience highly preferred. Perfect written English, Above average spoken English needed. Experience with CRMs. Budget is for the first month of work. Approximate work time 1-3 hours per day. Monday-Friday.
[url removed, login to view] I installed an Asterisk - freepbx pbxip 14, I need: ....[url removed, login to view] I installed an Asterisk - freepbx pbxip 14, I need: .- create 2 extensions .- synchronization between crm and asterisk .- configure trunk sip .- to configure DiD
...CDMA VoiP gateway. This project will be separated in two parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application
We use PASCOM as asterisk system , we want to configure it for making outbound calls, using IPCOMMS as SIP. We are having issued doing that, first we could make only 1 call per time now we can't even make a call. We need someone who can debug this issue through reading logs and find the solution. Please bid only if you have experience in these requirements