I am not overly technically savy so I am asking the community if this can be done.
I am using ONSIP.com as a hosted PBX solution. ( Its easy and I like their interface.
What I would like to do is use their "bridge function" to have Twilio be the source for my SIP numbers.
I think this can be done because ONSIP describes how I can get an international number from DIDWW>com and have it "pointed" into to my system. So I thought, why not have Twilio supply me with numbers, local and international.
The real question I have though is can I make calls out on that SIP number, going through ONSIP.com to start the call but have Twilio actually make the call , (if that makes sense)
ONSIP.com charges 2.9/minute in or out bound and it appears as though Twilio is on 1/minute. so it would seem to make sense to do this if it can be done?
Here is their info on an inbound bridge
http://www.junctionnetworks.com/knowledgebase/onsip/admin-portal/apps/inbound-bridge
Thanks for any advice or input you might have on this subject.
What you need to find out is if OnSIP and Twilio both support SIP transfers.
To have a call you've started on OnSIP's server "moved" to Twilio it will require the use of SIP REFER requests, which are how call transfers are done in the SIP World. Without a transfer it means your call would end up going through both OnSIP's and Twilio's server and mean you end up incurring both sets of charges.
I suspect you'll find one or both of them don't support transfers since such functionality can be tricky for a SIP provider to implement and it's also it's not going to be in the originating provider's interests, in this case OnSIP, to provide you with a way of avoiding their call charges.
Perhaps an easier way for you would be to get an IP Phone or ATA that has two or more SIP lines and connect the first line to OnSIP for incoming calls and the second line to Twilio for outgoing calls.
For anyone who finds this now, Twilio does not support Sip Refer at the time of this post
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I'm not super familiar with VOIP.. I've been doing some research but have had a difficult time finding an answer to this question.
What I need to do, is check to make sure a phone is being forwarded to a specific number. I'm trying to determine if there is a way to do this by placing a call to it. Looking at rfc 3261, it seems the VIA header would contain the info I need if a forwarding phone just acts as a proxy, but it seems this is unlikely to work, judging from the line:
Each proxy uses the Via header field to
determine where to send the response and removes its own address from
the top.
Is it possible to tell if a call was forwarded to a specific number, simply by placing a call to it and monitoring the headers? If so, could I simply download a VOIP app, and snoop the packets with WireShark? Or will I need to create my own VOIP client?
You can use the SIP history-info header, see rfc4244 for details.
I have a question about integrating with a phone company (the Provider) using SIP.
I have a situation:
1. A call is made to a PSTN number
2. The Provider forwards the call to a SIP Gateway
3. Twilio is the SIP Gateway, so I receive an HTTP request for every new call
4. I execute my application logic
As I understand the SIP integration between the Provider and Twilio is done using SIP INVITE.
Now a have the challenge is to implement the integration using SIP REGISTER.
As I imagine, the scenario should look like this:
1. I register against the Provider using SIP REGISTER
2. A call is made to a PSTN number
3. The Provider gives me the call
4. I execute my application logic
I need to figure out what is needed in order to accomplish this:
Firstly, does this scenario make sense?
Do I need to use a PBX solution (like Asterisk, FreeSwitch) to implement SIP REGISTER and build my application on top of it?
If so, which PBX solution do you recommend and which features/modules are needed? And do I have to host it on my server?
Perhaps I don't need a PBX solution, and a library is enough as described here?
It is the Provider pushing for this way of integration and I have too little knowledge about it.
What I have figured out is that Twilio can't help me with this. So it looks like I have to take a part of solution in-house.
REGISTER is required if your terminal or terminals belong to the domain of a VoIP provider.
REGISTER records the mapping between the identity the VoIP provider gave you and the actual address and port where you will be listening for requests.
This way, calls addressed to you (sip:myuserid#voip.domain.com) will be sent by the VoIP provider to the address of record it has for you.
If you are a VoIP provider yourself (i.e. you have a sip:myuserid#myowndomain.com), then your peer voip providers will route requests to you based on DNS records or internal domain-based routing decisions. Once the call reaches you, then you can decide on how to handle it. If you are a real SIP provider, then you will have a registrar where you store the result of the REGISTER of your different users.
If you want to implement some application logic on your end, you have different options:
Easiest way is to implement a UAC/UAS, basically a terminal. Your application is the terminal, it registers with the VoIP provider and receives all your calls. You will just need the SIP stack, and you can do whatever you want with the call.
Using a PBX software. Basically it will handle normal calls for you, and the REGISTER when needed. Typically they will have APIs to perform some degree of automation/modification of the call handling.
Difference between the approaches, in the first case, you just have the protocol, so you must do everything else. In the second case, the objective is to process normal calls and they will offer you some window (smaller or greater) to see into those calls and do things with it.
I have a client company with a simple web application (Python Flask) and I need to add a phone notification functionality to it.
The main requirement is that the app should call users, play a certain sound file and accept some tone input ("Hello! This is an automated message from your WebApp account. You have a meeting with $John today at $5pm. Please press 1 to confirm").
The other requirement is that the solution should be relatively cheap and fast to market.
I have done some research already and it seems that there are a few consequent steps to achieve that:
Set up an Asterisk or a FreeSwitch server;
Set up a SIP account;
Write some business logic for the Asterisk server which allows to make calls and play sounds via a SIP account;
Write an API at the Asterisk server and expose it to the Python Flask web app.
Do I miss something here? Can any of the steps be omitted anyhow? Can I do it simpler?
the fastest way to get it working is to use one of the cloud voice services with speech synthesiser. Here's a short list to check out:
Twilio
Tropo
Plivo
Here I listed some details.
Those services charge you per minute, plus you may have to pay some monthly fee.
If you want to run an independent and standalone service, I would recommend FreeSWITCH instead of Asterisk. It's got reach integration possibilities and API. You will need to read the FreeSWITCH book in order to understand how it works and how to build your service.
I agree with Stanislav Sinyagin on the cloud based solutions, but I would add one more, Voxeo Prophecy. Tropo is from Voxeo, but they have offered Prophecy as a solution for a lot longer and it supports the open standards CCXML and VoiceXML. The advantage of CCXML for outbound notification applications is you have a lot more control of the notification process.
The Prophecy platform has excellent call progress analysis (CPA) which will allow you to determine whether a machine or a human answered and handle the call accordingly. For example, it does not make sense to ask a machine to "...press one to confirm". Instead you may want to leave a message that provides a call back number for the user to confirm with after they have listened to the voice message. The CPA can be used to leave a message on a machine at the correct time (when the greeting message has stopped) so that you do not get clipped messages in the voice mail. CPA will also allow you to provide detailed reports on who was notified and for those that did not it can tell you whether it was a bad number (received a SIT tone), a modem or fax answered, or ring-no-answer (pretty rare these days). These type of details can factor into your retry process for failed notifications.
The other advantage to using Prophecy and open standards is your application will be portable to other IVR systems that are VoiceXML/CCXML compatible if you ever want to migrate. Tropo, Twilio, and Plivo all use proprietary API's which does not allow you to move your applications to other services. Prophecy is also available as a software solution so that if you want to take it out of the cloud you can run it on premise. You can get a two port version for free to try it out.
There is excellent documentation on developing outbound notification systems on Voxeo's developer site. Take a look at the CCXML documentation in section F on Outbound Dialing.
Not sure which development languages you are familiar with, but if you are used to ASP.NET MVC there is an open source project called VoiceModel that makes it easier to develop VoiceXML applications. The other advantage of VoiceModel is that you develop your application once and it will run on any VoiceXML compatible platform and Tropo. They are currently working on adding outbound notification support in this project that will work for both Tropo and VoiceXML.
Third party solutions listed are your easy choice. Running your own asterisk is also suitable for what you want to do, but i think for only this much it would be overkill, from an operational perspective.
In asterisk, you can originate a call that has the 2 variables you need with an (basic-authenticated) HTTP request. You will also need some settings and a tiny dialplan. Setting up the SIP account is easier or more difficult, depending on the documentation from the provider. Most of them have detailed documentation for configuring asterisk (not so much so for freeswitch). Keeping the damn thing alive is what's gonna get to you :)
I have one simple question which stuck me.
I have FreePBX server with SIP trunk to ISP. ISP Dedicated one PSTN number. So all internal SIP Extension go out with the same number. Does it mean that at any time just one Extension can call outside?
Thank you.
Best Regards
Ask that question to your provider.
Usualy no limit, if no special limit on provider's side.
Just a hint. It would depend on your VoIP provider the number of simultaneous calls they allow you to do with one account, you can try SIP Trunk providers such as Switch2Voip Call Center VoIP they allow unlimited concurrent calls with very low rates.
Actually We are doing thesis work where we need to make 10 voip phones which are SIP based connected with each other.So they can call and talk among each other.Also we want to add video calls access.Another question is it possible video calls on SIP.
SIP already has built in reliability measures, most of which are specifically to cope with unreliable transports such as UDP. You should read the section in the SIP RFC on Transactions to gain an understanding of how it works. One aspect missing from the SIP RFC is reliability for provisional responses and the supplementary RFC3262 deals with that.
SIP is agnostic to the type of sessions, such as voice or video, it sets up so yes it can be used to set up video calls. There are heaps of readily available SIP softphones around that already provide video, one example being x-lite.
To make it reliable you need to emulate the following two features:
For Calls
You need to sequence the packets.
One end needs to tell the other end that a sequenced packet is missing if this happens, and you probably want take jitter into account -- i.e., wait a small amount of time before you request a missing packet.
For protocol commands
You need to ackknowledge command packets -- if a command is not acknowledged it has to be sent again.