Using cloud PBX pbxes.com to empower VoiP / SIP telephony
It so happened that I have several SIP numbers from different VoiP operators in different countries of the world (USA, Israel, Ukraine, Russia). At some point, having received another call with the proposal to “buy an elephant” to my Israeli number at 4 am (I am in the US, and the Israeli telemarketer, of course, does not know about this), I realized that something had to be done.
In the case of the Israeli provider, everything was simple - in the provider's personal account I changed the settings, redirecting all calls from unknown numbers (which are not in the “white list”) to voice mail from 9 pm to 7 am.
Everything would be fine, but after a while I began to receive calls with the proposal to buy a Ukrainian elephant (to Ukrainian numbers from Atlantis Telecom and Intertelecom) and a Russian elephant (to a number from Zadarma). And here it is already worse - in the personal accounts of these three providers there is no possibility to redirect the call to voice mail on a schedule or block some number. In Zadarma and Intertelecom there is no voicemail at all, with redirection, all is not well either (in Zadarma there is only unconditional redirection), in Atlantis Telecom and Intertelecom there is redirection only to telephone numbers, but not to SIP URI.
')
As a result, a solution was found using cloud PBX pbxes.com.
Who cares, the application under the cut (a lot of pictures)
Go to www.pbxes.com and create a free account. A free account has the following limitations:
1. No more than 5 trunks / external SIP records.
2. No more than 5 extensions / extensions.
3. No more than 5 incoming lines / inbound routes.
4. No more than 5 outgoing lines / outbound routes.
5. Maximum of 2000 minutes per month (incoming and outgoing in total).
6. Maximum one free account per user. pbxes is watching this closely.
Paid accounts cost from 5 euros per month and depending on the type of paid account restrictions vary. Account types can be found here www1.pbxes.com/iptel_virtual-pbx.html
The first task was to connect the Ukrainian and Russian numbers to voice mail, to ensure the redirection of calls to voice mail on a schedule and to set the redirection of calls to the SIP URI: in the USA there is a mobile operator RingPlus, which is interesting in two things:
1. It is free (lives by playing ads instead of dial tone)
2. This is a hybrid VoiP / CDMA provider: it can also be used as a regular Voip / SIP provider and can be dialed by any sip client just by calling sip .ringplus.net
Therefore, redirecting calls to the SIP URI 1234567890 sip .ringplus.net means in my case that my mobile phone with the US number 1234567890 will receive a call.
So, the procedure
1. In the newly created pbxes.com account, create an internal number
This number / extesnion will have sip login / password / sip server, you can connect to it with any sip client. Pay attention to the voicemail settings (make voicemail & directory enabled).
if you need sip uri forwarding, then the settings will be as follows (see call forwarding)
2. Create a trunk (in this example, with sip-parameters from Zadarma)
In the Register field, check Yes (Inbound and outbound calls).
3. Create an Inbound Route
The trunk name must match the trunk name specified in step 2.
In this example, all working hours are defined as 8 - 15, during these hours the calls will be redirected to the internal line 103 (it is assumed that there is a sip client registered on this line or the line is redirected), and the rest of the time the calls will go to voicemail.
4. Create an outgoing trunk. she will be the same line from Zadarma
The trunk name must match the trunk name specified in step 2
That's all. In my case, I identified several trunks and incoming lines that converge all in the same extension 103 for Ukrainian and Russian numbers (voice mail greeting in Russian) and a separate extension for Israeli number (voice mail greeting in Hebrew) plus call transfer by no answer / busy / unavailable on 1234567890 sip .ringplus.net (where 1234567890 is my American sip number from provider Ringplus).
After that, I wanted to use the Zadarma service as a calling card, considering its tariffs and a large number of access numbers.
Further pictures with minimal explanations will follow (everything is quite clear). The only note is that it’s impossible to get rid of one trunk: you need an incoming trunk (any, not necessarily Zadarma, the main thing is that you can call it from the traditional telephone network via an access number or by a direct number. In this article I used a trunk from Zadarma, but I use real-life settings from Callcentric, which gives a direct American number for free)) and an outgoing trunk from Zadarma.
Incoming trunk
Outgoing trunk
Note: the Register parameter must be No (Outbound only)
Inbound route
Pay attention to the password 1234 (replace, of course, with something more complicated) - you do not want everyone who is not too lazy to call through your account. If there is no password, then in the outbound route settings (see below) you need to define a list of allowed numbers. The problem is that it’s not a fact that your number is determined correctly (depending on the provider on the inbound route)
Outbound route
Call for health!
PS A few comments about Pbxes.com
1. A paid account allows you to define a trunk for Google Voice. Considering that GrooveIp becomes paid from February 2016, and the regular Hangouts is buggy, heavy and consumes a battery, for some it may be a solution in at least a situation if you already have a paid account with Pbxes.
2. When I try to create a trunk, sometimes I get the error “This provider requires a paid account for security reasons!”.
This happens, in particular, if the SIP Server URL has numbers (like, for example, global.ua: reg893.global.ua) and in some other cases (depends on sip headers).
The problem is solved as follows (reprinting my post with dslreports):
Steps (order is very important):
- Register any domain name if you have it already. Let's say your domain is example.com
- Let's say your sip credentials user / password / sip123.voipserver.com and you get "paid account error" with these credentials.
Create pbxes.com trunk with following credentials user / password / sip.example.com. Of course, calls will not go through (no sip registration with this domain)
- Goto your domain registrar, create subdomain sip.example.com and forward it to IP address of sip123.voipserver.com
Wait for dns propagation (from 5 mins to a couple of hours).
- Bingo! Trunk is registered and you can use it.
In the case of the Israeli provider, everything was simple - in the provider's personal account I changed the settings, redirecting all calls from unknown numbers (which are not in the “white list”) to voice mail from 9 pm to 7 am.
Everything would be fine, but after a while I began to receive calls with the proposal to buy a Ukrainian elephant (to Ukrainian numbers from Atlantis Telecom and Intertelecom) and a Russian elephant (to a number from Zadarma). And here it is already worse - in the personal accounts of these three providers there is no possibility to redirect the call to voice mail on a schedule or block some number. In Zadarma and Intertelecom there is no voicemail at all, with redirection, all is not well either (in Zadarma there is only unconditional redirection), in Atlantis Telecom and Intertelecom there is redirection only to telephone numbers, but not to SIP URI.
')
As a result, a solution was found using cloud PBX pbxes.com.
Who cares, the application under the cut (a lot of pictures)
Go to www.pbxes.com and create a free account. A free account has the following limitations:
1. No more than 5 trunks / external SIP records.
2. No more than 5 extensions / extensions.
3. No more than 5 incoming lines / inbound routes.
4. No more than 5 outgoing lines / outbound routes.
5. Maximum of 2000 minutes per month (incoming and outgoing in total).
6. Maximum one free account per user. pbxes is watching this closely.
Paid accounts cost from 5 euros per month and depending on the type of paid account restrictions vary. Account types can be found here www1.pbxes.com/iptel_virtual-pbx.html
The first task was to connect the Ukrainian and Russian numbers to voice mail, to ensure the redirection of calls to voice mail on a schedule and to set the redirection of calls to the SIP URI: in the USA there is a mobile operator RingPlus, which is interesting in two things:
1. It is free (lives by playing ads instead of dial tone)
2. This is a hybrid VoiP / CDMA provider: it can also be used as a regular Voip / SIP provider and can be dialed by any sip client just by calling sip .ringplus.net
Therefore, redirecting calls to the SIP URI 1234567890 sip .ringplus.net means in my case that my mobile phone with the US number 1234567890 will receive a call.
So, the procedure
1. In the newly created pbxes.com account, create an internal number
This number / extesnion will have sip login / password / sip server, you can connect to it with any sip client. Pay attention to the voicemail settings (make voicemail & directory enabled).
if you need sip uri forwarding, then the settings will be as follows (see call forwarding)
2. Create a trunk (in this example, with sip-parameters from Zadarma)
In the Register field, check Yes (Inbound and outbound calls).
3. Create an Inbound Route
The trunk name must match the trunk name specified in step 2.
In this example, all working hours are defined as 8 - 15, during these hours the calls will be redirected to the internal line 103 (it is assumed that there is a sip client registered on this line or the line is redirected), and the rest of the time the calls will go to voicemail.
4. Create an outgoing trunk. she will be the same line from Zadarma
The trunk name must match the trunk name specified in step 2
That's all. In my case, I identified several trunks and incoming lines that converge all in the same extension 103 for Ukrainian and Russian numbers (voice mail greeting in Russian) and a separate extension for Israeli number (voice mail greeting in Hebrew) plus call transfer by no answer / busy / unavailable on 1234567890 sip .ringplus.net (where 1234567890 is my American sip number from provider Ringplus).
After that, I wanted to use the Zadarma service as a calling card, considering its tariffs and a large number of access numbers.
Further pictures with minimal explanations will follow (everything is quite clear). The only note is that it’s impossible to get rid of one trunk: you need an incoming trunk (any, not necessarily Zadarma, the main thing is that you can call it from the traditional telephone network via an access number or by a direct number. In this article I used a trunk from Zadarma, but I use real-life settings from Callcentric, which gives a direct American number for free)) and an outgoing trunk from Zadarma.
Incoming trunk
Outgoing trunk
Note: the Register parameter must be No (Outbound only)
Inbound route
Pay attention to the password 1234 (replace, of course, with something more complicated) - you do not want everyone who is not too lazy to call through your account. If there is no password, then in the outbound route settings (see below) you need to define a list of allowed numbers. The problem is that it’s not a fact that your number is determined correctly (depending on the provider on the inbound route)
Outbound route
Call for health!
PS A few comments about Pbxes.com
1. A paid account allows you to define a trunk for Google Voice. Considering that GrooveIp becomes paid from February 2016, and the regular Hangouts is buggy, heavy and consumes a battery, for some it may be a solution in at least a situation if you already have a paid account with Pbxes.
2. When I try to create a trunk, sometimes I get the error “This provider requires a paid account for security reasons!”.
This happens, in particular, if the SIP Server URL has numbers (like, for example, global.ua: reg893.global.ua) and in some other cases (depends on sip headers).
The problem is solved as follows (reprinting my post with dslreports):
Steps (order is very important):
- Register any domain name if you have it already. Let's say your domain is example.com
- Let's say your sip credentials user / password / sip123.voipserver.com and you get "paid account error" with these credentials.
Create pbxes.com trunk with following credentials user / password / sip.example.com. Of course, calls will not go through (no sip registration with this domain)
- Goto your domain registrar, create subdomain sip.example.com and forward it to IP address of sip123.voipserver.com
Wait for dns propagation (from 5 mins to a couple of hours).
- Bingo! Trunk is registered and you can use it.
Source: https://habr.com/ru/post/274063/
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