Basic question here...for making calls in/out of Twilio, between PSTN numbers and SIP endpoints (e.g. PBX), it seems you could accomplish this using 1 of 2 methods:
Twilio SIP (using TwiML for translating between PSTN call and SIP call)
Elastic SIP Trunking
I'm wondering what the main differences in these methods are. It seems that with Twilio SIP you have use of TwiML-based applications, and with SIP Trunking you do not...is that the only difference?
ELASTIC SIP Trunking by Twilio is VoIP Carrier option with Global reach such that Load balancing / geographical redundancy and every other interconnect pain point is handled by Twilio and you can just focus on your application and not worry about interconnect partners and call termination in different countries.
I believe, after having built the partnerships to do the call terminations for applications that were making calls using TwiML, they decided to open the infrastructure to applications that dont necessarily have to be written over TwiML.
Related
I am trying to understand if there is a way we can transfer/redirect active call from Twilio to Microsoft Teams user.
Incoming call to Twilio will be a SIP audio call.
MS Teams will act as SIP Gateway/Server which can receive call from Twilio and Teams user will see the incoming call.
I was reading few of the articles on Twilio:
https://www.twilio.com/docs/voice/tutorials/call-forwarding-java-spark
https://www.twilio.com/docs/sip-trunking/call-transfer
Does Twilio and MS Teams supports this feature using the APIs?
As you will have read in the SIP trunking call transfer article, you can absolutely transfer a SIP call from Twilio to another SIP endpoint.
I have no experience of that with MS Teams, but it appears that you can make SIP calls to Teams. Though some of the documentation points out that you may need specific hardware that supports Teams.
I have a phone number registered in Twilio that I wanted to use for both a Twiml application and an Elastic SIP Trunk (connected to Asterisk). The idea is that inbound calls hit the Twiml app first and then can be forwarded to the Asterisk server if needed, while outbound calls just go via the SIP trunk. (The reason it needs to be a SIP trunk instead of simply using SIP Registration with Programmable Voice is because that is the only way to have E911 support for outbound calls.)
Twilio support told me that it is not possible to use the same number for both.
Because of that limitation, my current plan is to use two Twilio phone numbers. My published phone number will go to the Twiml application, and a second number that I will not give out will go to the SIP Trunk. (Twilio allows number spoofing of other numbers on your account, so I will have the Asterisk server pretend to use my primary number for outbound calls instead of using the second private number.)
In order for this to work, I need to be able to transfer calls from my Twiml app to Asterisk and from Asterisk server back to the Twiml application. The former is easy: just use <Dial> with a SIP URL that points to the trunk. The latter is what I need help with. (I also want to do this in case someone does manage to call the second number - I want them to be redirected to the Twiml app.)
As far as I can tell, the only way for me to transfer calls back into my Twiml application is to forward the call from the Asterisk server back to my public number. The problem is that I think this will look like an outgoing+incoming call and I will get double-billed for these minutes. I'm already paying for another number, and I really don't want to have to pay extra for the minutes too.
Is there a better (or "official") way to transfer a call back to the Twiml app? Or am I wrong about Twilio seeing (and billing) this as two calls?
It is not clear why you cannot use the Twilio number for both a Twiml application and an Elastic SIP Trunk (connected to Asterisk). Did they indicate why?
Just don't assign that particular number to your Elastic SIP Trunk and you should be able to assign it to your TwiML application for inbound calls and use a when you want to forward the the call to your Asterisk PBX.
For outbound calls, you can have you Asterisk PBX send calls with that number as CallerID to your Elastic SIP Trunk Termination URI.
For E911 calls from the Twilio Elastic SIP Trunk, you should have a number associated with your Elastic SIP Trunk, enabled for Emergency Calling, so when 911 calls are placed, the CallerID of that number is used for outbound calls and calls can be returned to that number should the connection get disconnected.
If you did go the second route you mentioned, can you have your Asterisk Server send the call to a Twilio Programmable Voice SIP Domain, maybe have a Dial Plan defined so an Asterisk prefix digit sends calls out this different trunk. Not sure this will work (since mixing Elastic SIP Trunking with Programmable Voice in this manner) but one idea. Your Asterisk server will remain in the call path.
I have the next situation:
A call is made to a PSTN/DID number of my phone company and that I can use with proprietary SIP cell phone app or SIP client application like Zoiper / Linphone configuring with SIP, user and password.
I cannot configure this service to forward a call to 3rd party service SIP URI. I've already asked it to my PSTN provider and they say that it is impossible.
The question is:
How can I make Twilio (or maybe another service) register as SIP client with user and password (like extension/endpoint ) to receive a call from PSTN and forward it to other SIP URI or phone number?
I know that asterisk / some cloud pbx can do it like it was previously free account at pbxes.com.
Have you considered porting your number to a carrier that will allow you to forward inbound calls to a specific SIP URI you configure or register for calls to that URI, given your current carriers constraints?
For example, Twilio has some documentation here on porting a number.
International Porting
Porting a number To Twilio
I understand, based on your country, this may not be an option.
Twilio does not have a way to register as a SIP client with another provider. It does offer the ability of using a Twilio hosted number (and thus the comment on porting) to register a SIP client against or forward a PSTN call to a SIP User Agent.
I know there are some twilio experts here and would really appreciate it if someone could answer a question for us and if so, please let me know if you do freelance work.
Our website offer clients to purchase numbers, which are twilio numbers, and we forward the calls and SMS to their original number, while doing demographics, call recordings and marketing. The number on which call was received is important to us. We want to use Thinq LCR to reduce cost. But Thinq wants us to port the twilio number to them. If we port the twilio number to Thinq, will the existing twilio services break? and on which routing profile will we forward the numbers after porting, if twilio number has been ported to thinq, there's no twilio number to forward anymore. And will we need to change all the code to work with the new Thinq API as twilio is out of the game now?
Thank you!!!
Chip :)
From what I understand of your problem, you should port your existing Twilio DIDs to ThinQ and switch to provisioning new DIDs from ThinQ directly, going forward.
Once the DIDs are visible in the dashboard (i.thinq.com), you would configure them all to route to a Twilio SIP domain that you will need to create. See https://www.thinq.com/thinq-voice-origination-with-twilios-bring-your-own-number-byon-service/ for instructions on how to do this.
With this setup, people trying to reach your clients would dial the DIDs controlled by ThinQ. ThinQ would send the SIP calls over to the Twilio SIP domain which would then interact with your server's callbacks to handle the call.
Your callback would use the appropriate Twilio API (REST or TwiML) to dial the client's actual phone number via SIP so that it goes over your ThinQ VoIP account for lower costs (e.g dial to sip:#wap.thinq.com?thinQid=&thinQtoken= )
I have a situation where I need to connect 2 persons together using Twilio Elastic SIP Trunking.
So, I have two mobile phone numbers, I need to call first person and when he answer an incoming call, I need to call second person and when he answer an incoming call, connect them together.
Those 2 persons could be in different countries. So, is it possible to use Twilio's Elastic SIP Trunking to accomplish this purpose?
Twilio developer evangelist here.
I believe this is a follow on from your previous question. I recommend you read my answer there.
If you have two people with separate phones, then you are not looking to use Elastic SIP Trunking. That feature is for when you have a PBX and SIP phones and you are looking to connect them to the PSTN.