asterisk expert needed cli function to add call leg rev3
$250-750 USD
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已发布超过 3 年前
$250-750 USD
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Using asterisk 12.6.0 / FreePBX 12.0.1rc29
I will not be able to provide server access. You must use your own server for testing.
Part 1:
I need a method to add a leg to a call through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script.
For example, 1231231212 from the outside calls inward, it rings through a ring group, and ext 500 picks up.
I want to have a button on my crm call a php script to transfer the call that is ongoing between the outside number and ext 500, to another internal ext 100, or even external phone number. By the end of the function being completed, there will be a joined called from ext 500, the external number, and ext 100. The user at ext 500 will end the call by hanging up, while the call between the external number and the ext 100 continues.
In some types of failure, a graceful error would be nice.
The second internal extension should get the original caller id information passed through as is.
I will handle the crm integration.
When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address.
If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui.
Part 2:
I need a method to transfer a call to a wav file through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec script.
For example, extension 123 internally, calls an external number. 123 starts the call, and is tranfered to the remove voicemail. 123 would click on a button that transfer the call to a wav file, which is played, and then the call is ended.
In some types of failure, a graceful error would be nice.
I will handle the crm integration. I will provide properly formatted wav files.
When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address.
If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui.
Hi, i am skilled voip developer. I have done numbers of project with php and asterisk. it's not too complex job, but there are many small details have to be done to make your scripts working fine. contact me for discuss details.
Hi,
I have done that for you before and can do it again, the second point is almost the same working, we just need to send the call to a different dial plan,
kindly let me know if we can discuss the project,
regards,
Hello,
Upon reading the job details I would say that all the required skills English (US) and Asterisk PBX fall under my skills. I work on freelancer full time and I believe I can do this job if I get all the detailed requirements. Please check my portfolio as well: https://www.freelancer.com/u/AwaisChaudhry?w=f
Looking forward to your response. Awais
Hello, I'm a voip tech with over 10 years of experience working on the level2/NOC department for a US company; I'm familiar with many SIP devices and applications such as softphones, ATAs, IP-phones and PBX systems like Asterisk/FPBX. I read your project description and I can build this solution for you, the way that I'm idealizing you'll just need to implement your CRM to trigger a php file and pass the extension triggering the app and the destination number (instead of the IP address). You may contact me at any time to discuss more on this project.
Thank you.
Greeting From Ukraine
whnta i propose
1- upgrade asterisk to 13 version
2- use ari to get active calls and then transfer the call to another number or voicemail