asterisk Guru needed ASAP

Hello Warren, I hope you are doing well. The best way to know the reason of bad quality is "wireshark" ... If you like then I can capture a wireshark log for 10 - 20 minutes when calls are in progress .. Wireshark is the only tool to know about issues in RTP (Voice Streaming). Regards, Aneel

Hi my name is Ambiorix. Share you SIP configuration file and also the cli ouput and I will diagnose your issue and fix it

Hi, I have been working with Cisco VoIP since 2001 and Asterisk and freepbx ,trixbox, elastix since 2007. I'm very new to the site so trying to bid as low as possible and get better reviews as much as possible. That being said I would like to have a little chat about your project to better understand what kind of audio issues you are having. are you not having audio at all? are you having one way audio? kindly let me know if we can discuss about it. regards,

Hello dear sir, i have a lot experience with asterisk, so i hope my skills will be useful for you. Please contact me if you interested in. Thanks.

Hello, Please take a look with your nat port. You can fix that issues. In case you still cannot fix that problem, please contact me. Thank you, Vu Le

Hello I have very good experience with asterisk and all major asterisk based application. I can help you in fixing the audio issue in your server. Thanks

**Expert Asterisk Engineer** Hi there, I can resolve your issue with asterisk in a bit, just ping me. Take a look at my profile to see amazing reviews by others.

I'm a professional with experience of over 10 years in management of Linux platforms, information security, Cisco equipment, asterisk and web applications Certificates: ASTERISK DCAP, LINUX LPI, RED HAT RHCSA, CCNA and CCNP, CEH EC-Council

i am aasif working on asterisk more than 6 years. If its one way audio issue then you have to set NAT settings in asterisk server. If it is voice issue then it can be either your internet or provider route issue and one another things its can be hardware issue like RAM or memory cache is not realeasing Hoping that you will give me a chance

Hi, I am Janko from Serbia, I am not a guru but I am Digium Certified Asterisk Administrator, and I maintain several dozen Asterisk instances, therefore I may help you with your issue, however I would certainly need more details in order to give proper estimates on how long would it take and whether we would need additional resources.

Good day! I work with the asterisk and PBX systems over 8 years. Also have expirience in Freeswitch I created a lot of call centers that are working successfully for several years. I have my own call center and my own server on which i install the hybrid Vicidial\Goautodial and FreePBX\a2billing. It is configured and ready to go.I can show you it to the test. I integrate Vtiger\Sugar CRM etc. I install and configure KVM hypervisors and work with virtual machines I have experience in supporting operators and customer of call center. I managed call center for more than 3 years. My last job was: Creating PBX system with billing, call center and CRM modules for US Provider I think I can provide you with good and quality services.

I am 15 years experienced senior software developer. I have mostly worked on the VOIP projects. i have designed hybrid VOIP PBX. I have used Asterisk, i have written modules for it and configured. I used b2bua at the past and i used opensips too.

hi, I have good experience of asterisk. I have word on IVR server and I have also have knowledge of codec . so I can do your task easily

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Hi! I'm am expert developer and an Asterisk admin/installer. I had many experiences installing and mantaining some asterisk pbx amd a call center. Maybe I could help you to solve this issue. Take a look to my profile. Thank you.

I have been working as asterisk engineer for last 3 years. Also have configured asterisk from sorce and did all type of functianlity like Inbound/Outbound DID call, voicemail , call forward , call transfer , call park. Also i do have hands on experience in ejabberd server as well

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