Experto en WEBRTC Tenemos un servidor asterisk y un servidor web queremos hacerlo correr en HTML5 con un cliente que tenemos hecho. You have to be a Expert in WEB-RTC we need to connect an app to asterisk via sip we already have a client and server we need to get alive.
...in CLI: (the lag is between these two lines) Sequence: 1. Line Posts in CLI: 0xb7503018 -- Probation passed - setting RTP source address to [url removed, login to view] 2. call shows connected, times has started, but cant hear dialed party for 1st 4 seconds, until next line below posts to CLI 3. When this line posts, then we hear other party: 0xb7503018
I am using Express Talk VoIP Softphone in my Windows PC (WIN7 OS)But the Sip is not connecting properly all time Need Guidence to connect quicky all time to sip account
I have a custom andriod voip client done using linphone. the job was not 100% completed. I need to: 1- fix the logo and 2- turn off access to settings , 3- turn off the sip address when dialing a number, and 4- make the app recognize only the last 5 digits of the phone number. 5- Publish to google play. I only have the app no source code
...contra un Server Asterisk. Actualmente lo tenemos configurado pero con algunos problemas como que solo algunos puertos se registran y otro no. /////////////////////////////////////////////////////////////////////////////// We are looking for someone who has the ability to configure a CISCO VG224 gateway using SIP with an Asterisk server. Currently
...for this TASK. Novice please not APPLY. Not the problem, but the complexity of these codes working as one. We have an outbound dialer that works when used manually to call a number with no lag upon connection. Our situation, is that since we moved from Twilio to WebRTC the dialing from a LIST function, seems to have random on connect issue
...channel -Channel performance status. -RSSI (signal strength) of each channel. -The duration of the current call. -Activity and statistics for each SIM-card. -Rapid notification of problems via Skype, SMS or email. 5: Built-in SIP server to work with SIM card bank. We can show you the cases of the similar services which work with GoIP equipment
I need someone to configure a SNOM 710 phone so that the openvpn will communicate with my openvpn server. My openvpn server is working and I have other devices connecting remotely. If you can config the sip settings that is an added bonus.
To invoice customers, by using templates stored in MySQL Database. People get to the module, by signing in through an Apache Web Server. Some of the information comes from our Asterisk PBX. The Module also sends Packing slips, Reminders, etc.
...are answering the phones via SNOM telephones, we would like in our [url removed, login to view] make it so that Zoiper will be integrated. It should be possible to answer calls, call up numbers that are in our system just by one click on zoiper via our system. When an employee is on a break then the employee should be able to put the phone on DND, IF
Hi nikessl, I noticed your profile and would like to offer you my project. I need to setup a call center in Kathmandu. I'm in urgent need of person with skill of asterisk customization and deployment. You can reach me here 9801113237 or ghanshyam at sarathi dot cab. Please let me know if you're interested Regards, Ghanshyam
...requirement, no source code build] I need a FreeSWITCH (FS) configuration for the following functionality: 1. Interconnect internal endpoints 2. Link with an existing Asterisk PBX 3. Link with an external SIP trunk for incoming and outgoing calls 4. Configure basic CDR, voicemail, IVR and conference 5. Configure presence/BNL 6. Configure chat
...an experienced Linux Administrator with Asterisk Voip experienced. Min. 5 years hands on experience. Mandatory skills: Centos7,6 Asterisk Optional Skills Expeience with 5 Softswitch Asterisk, Freeswitch, Kamailio,Opensips,vicidial,A2billing, Free pbx, elastix, PRI & PSTN card installation, call centre setup, Advanced IVR, Vmwire Esx,