VoIP | Virtual Number (DID) Knowledgebase - Virtual Number Provider | DID Numbers | Call Forwarding - Virtual-PhoneNumbers.com https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip Sat, 04 Jul 2020 04:47:19 +0000 Joomla! - Open Source Content Management en-gb Does Virtual Phone Numbers support h.245 tunneling on H.323? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/383-does-virtual-phone-numbers-support-h-245-tunneling-on-h-323 https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/383-does-virtual-phone-numbers-support-h-245-tunneling-on-h-323

YES, Virtual Phone Numbers support h.245 tunneling on H.323

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 17:54:55 +0000
Does Virtual Phone Numbers service Support IAX? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/384-does-virtual-phone-numbers-service-support-iax https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/384-does-virtual-phone-numbers-service-support-iax

Yes, Virtual Phone Numbers supports forwarding to IAX.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 17:56:47 +0000
How to Configure IAX with Virtual Phone Numbers IP Addresses? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/385-how-to-configure-iax-with-virtual-phone-numbers-ip-addresses https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/385-how-to-configure-iax-with-virtual-phone-numbers-ip-addresses

The following information will help you to set the IAX with the Virtual Phone Numbers IP addresses:

Create context on IAX.conf:

[46.19.209.10]
type=friend
host=46.19.209.10
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.11]
type=friend
host=46.19.209.11
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.12]
type=friend
host=46.19.209.12
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.13]
type=friend
host=46.19.209.13
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.14]
type=friend
host=46.19.209.14
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.15]
type=friend
host=46.19.209.15
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.75]
type=friend
host=46.19.209.75
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.76]
type=friend
host=46.19.209.76
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.77]
type=friend
host=46.19.209.77
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.78]
type=friend
host=46.19.209.78
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.79]
type=friend
host=46.19.209.79
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

[46.19.209.80]
type=friend
host=46.19.209.80
trunk=yes
context=from-didww
qualify=no
canreinvite=no
dtmf=rfc2833

Under 'iax_general_custom.conf.' add following lines:
requirecalltoken=no
calltokenoptional = 46.19.209.0/255.255.255.0

 

NOTE: Make sure your platform is not blocking these IPs: List of Virtual Phone Numbers IP Addresses

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 17:59:15 +0000
I Cannot Map My Number to a SIP Account https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/386-i-cannot-map-my-number-to-a-sip-account https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/386-i-cannot-map-my-number-to-a-sip-account

If you are trying to forward your number to a SIP account (e.g. DID@IPaddress  or  DID@DomainName), but the DID (virtual phone number) is still not working, please make sure that all of Virtual-PhoneNumbers.com IP addresses have been added to your system.

 
For more information about mappings containing SIP protocol, please see How to Set SIP Configuration for Virtual-PhoneNumbers.com DID Number on Asterisk?

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 21:46:49 +0000
DID number which Mapped to VoIP (SIP) is not Working https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/387-did-number-which-mapped-to-voip-sip-is-not-working https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/387-did-number-which-mapped-to-voip-sip-is-not-working

When mapping a DID number to a VoIP line which is using a SIP protocol, please make sure to enter it together with a correct country code. For example:

The DID number is 8745441221 but it is a UK number, thus has to be preceded by an international access code for UK “44”.

Its mapping then looks as follows:

SIP/64.16.210.54/448745441221 AS OPPOSED TO SIP/64.16.210.54/8745441221.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 21:47:48 +0000
SIP with firewall / NAT using Asterisk https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/388-sip-with-firewall-nat-using-asterisk https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/388-sip-with-firewall-nat-using-asterisk
Network Address Translation (NAT) is a common practice used in networks, and it doesn't play well with VoIP. Solving this problem requires an understanding of NAT, VoIP and your VoIP setup. The following focuses on the SIP protocol for VoIP using Asterisk, but problems and solutions are applicable to most other situations.

 

NAT can cause problems in several places. If one of the PBXes is behind a NAT gateway, the other PBX will not be able to contact it without some additional network setup. If one or more of the phones are behind a NAT gateway, the other phone will be trying to send audio to a non-routable address. This results in failed calls or missing audio.

The alternative to a re-invite is to have the PBX relay the voice packets between the two endpoints.

SIP client is behind a NAT gateway connecting to a server on the Internet

The client creates the translation entry for the SIP traffic when it first registers. As long as there is frequent communication between the two hosts, such as one packet per minute, the channel will stay open. The only configuration needed is to have the client use its external address in all SDP packets. On clients that support it, enable STUN (Simple Traversal of UDP through NAT), so the client can determine the external address dynamically, or enter it manually. Asterisk doesn't support STUN at this time, so all NAT configuration must be done manually. The following commands in /etc/asterisk/sip.conf set up the NAT properly:

[general]

localnet=192.168.0.0/255.255.0.0   // or your subnet

externip=x.x.x.x                   // use your address

[YOURREMOTEPEER]                   // your peer's name

nat=yes

qualify=yes                       // force keepalives

With this configuration, Asterisk uses the address defined by externip for all calls to the peers configured with nat=yes. The addition of qualify=yes causes Asterisk to test the connection frequently so that the NAT translations are not removed from the firewall. With these two commands, there always will be a communications channel between Asterisk and the peer, and Asterisk will use the outside address when sending SDP messages.

Multiple SIP phones and an Asterisk server behind a NAT gateway

Calls between the phones will work fine because NAT is not needed. For calls between you and other systems on the Internet there will be problems. Unless you register to the remote side as a client (as done in the previous example), you will not be able to receive SIP messages, so you will not be able to accept calls. Second, the address information in the call setup will point to the internal address of the phone, and the one-way audio problems mentioned previously will crop up.

The easiest solution to this is to avoid NAT entirely. If you have a public IP address available for your call server, use it. If your Asterisk server is connected to both the Internet and the internal network, the SIP port is reachable from both the inside and the outside, and the only problem is ensuring RTP flows properly. The PBX server does not need to be configured to route between the interfaces or provide masquerading; it simply needs to bridge the inbound and outbound voice calls.

As I mentioned earlier, the PBX either can stay in the voice path or get out of the way. In the latter case, the PBX tells both endpoints about each other after which the endpoints talk directly. However, Asterisk could have a call setup with both endpoints and relay the RTP packets on behalf of each endpoint. The inside host would be talking to the inside address, and the outside host would be talking to the outside address. The only configuration required to achieve this in sip.conf is to disable re-invites:

[general]

canreinvite=no              // force relaying

This configuration works well because the Asterisk server can speak freely to the Internet to send and receive calls. It also can talk to the internal phones, and by some simple bridging, completely ignore NAT.

As it turns out, this relaying behavior also is required when the Asterisk server has only a private address. The RTP ports will have to be forwarded on the firewall too. RTP chooses random port numbers based on configured limits. Before the ports can be configured, they should be limited in range. Configuring the firewall rules is much easier if the range of ports is known beforehand.

The range of ports to be used for RTP is defined in rtp.conf. The following configuration will limit Asterisk's choice of RTP ports from 10000 to 20000:

[general]

rtpstart=10000         // first port to use

rtpend=20000           // last port to use, rounded up if odd

Asterisk will need several RTP ports to operate properly. Only even ports are actually used, and disabling of re-invites causes two connections to be built per call. These ports and the SIP port must then be forwarded in by the firewall. The iptables syntax is:

iptables -t nat -A PREROUTING -i eth0 -p udp \

-m udp --dport 10000:10100 -j DNAT \

--to-destination 192.168.1.10

iptables -t nat -A PREROUTING -i eth0 -p udp \

-m udp --dport 5060 -j DNAT \

--to-destination 192.168.1.10

Replace eth0 with the outside interface of your firewall and 192.168.1.10 with the address of your Asterisk server. These rules tell the Linux kernel to translate the destination address of any UDP packets in the given range that are entering the outside interface. This must happen at the PREROUTING stage as opposed to the POSTROUTING stage, because the destination address is being translated. At this point, any SIP or RTP packet from the Internet will be forwarded to the internal Asterisk server for processing.

When a remote station makes a call to Asterisk, the SIP packet will be forwarded in because of the iptables rules. Asterisk will stay in the media stream because of the canreinvite=no command and it will use the external address of the firewall in any SDP packets because of the NAT commands. Finally, the media stream will be forwarded to the Asterisk server because of the combination of iptables RTP forwarding and port ranges defined in rtp.conf.

Up to this point, the configuration has focused on getting Asterisk working behind a NAT gateway, with some extra details to make the phones relay through Asterisk. There are, of course, more general solutions.

If you can avoid NAT in the first place, it is in your best interests to do so because it avoids all the problems encountered so far.

The Asterisk gateway can have a very restrictive firewall policy applied to it – you just need to allow UDP 5060 for SIP and whatever port range is defined in rtp.conf. In this configuration, Asterisk can contact both the internal phones and the rest of the Internet.

If SIP is not a requirement, and you are using Asterisk, consider using the IAX protocol. IAX tunnels both the control traffic and the voice traffic over a single UDP conversation that can be port-forwarded, filtered or translated easily. This method is limited to a static set of tunnels, which is sufficient if you are connecting some PBXes over the Internet or connecting to a long-distance provider.

Sometimes the above solutions are not available to you. In that case, it might be advisable to move to a full-featured SIP proxy and use Asterisk only for voice applications, such as voice mail. SIP Express Router (SER) is a powerful SIP server that handles NAT well and is used by several high-volume services, including Free World Dialup. SER's job is only in setting up calls between endpoints, so it must rely on other applications, such as specialized media proxies, to handle RTP streams if needed.

The step beyond a SIP proxy is a Session Border Controller (SBC), which is like a VoIP firewall. The SBC can intercede in either the signaling or RTP paths to add extra features, such as signaling protocol or codec translation, all while enforcing security policies. These are almost exclusively commercial products.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 21:49:25 +0000
How do I Define the CODEC which I Want to Use in Virtual Phone Numbers SIP Trunks? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/389-how-do-i-define-the-codec-which-i-want-to-use-in-virtual-phone-numbers-sip-trunks https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/389-how-do-i-define-the-codec-which-i-want-to-use-in-virtual-phone-numbers-sip-trunks

This article will help you to define the codec you would like to use with your system.

 

Please view the explanation in How to Set SIP Configuration for Virtual Phone Numbers DID Number on Asterisk? for defining the trunks in an Asterisk PBX system. Please be aware that "allow all" refers to all kind of CODEC (e.g. ILBC or G729 CODEC).
]]>
webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 21:50:59 +0000
How does my SIP Server can Identify the Calls? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/390-how-does-my-sip-server-can-identify-the-calls https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/390-how-does-my-sip-server-can-identify-the-calls

For the example purpose you have 3 numbers on your GlobexCamCalls account, as follows:

121233334444

121233335555

121233336666

And you mapped them to your SIP server with the address 1.2.3.4.

The numbers will be forwarded to your SIP server as follows:

 121233334444@1.2.3.4

 121233335555@1.2.3.4

 121233336666@1.2.3.4

Your server will identify each call by the called number (extension).

]]>
webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:04:16 +0000
Can I Use 3rd Party SIP Services such as Callcentric.com, Sipgate.com or Phonegnome.com? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/391-can-i-use-3rd-party-sip-services-such-as-callcentric-com-sipgate-com-or-phonegnome-com https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/391-can-i-use-3rd-party-sip-services-such-as-callcentric-com-sipgate-com-or-phonegnome-com

GlobexCamCalls allows DIDs to be mapped to a number of 3rd party VoIP providers. The selection of the VoIP provider is performed by using our on-line configuration interface. For more information see How can I Use 3-rd Party SIP Services Such as Callcentric.com, Sipgate.com or Phonegnome.com?

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:10:33 +0000
Is it Possible to Forward an Incoming DID Number to the SIP Server of my VoIP Provider? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/392-is-it-possible-to-forward-an-incoming-did-number-to-the-sip-server-of-my-voip-provider https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/392-is-it-possible-to-forward-an-incoming-did-number-to-the-sip-server-of-my-voip-provider

It is possible to forward an incoming DID number to the SIP Server of your VoIP provider as long as this provider is on the predefined list.

]]>
webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:17:45 +0000
Can you Recommend on a Free SIP Server that Works on Linux? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/393-can-you-recommend-on-a-free-sip-server-that-works-on-linux https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/393-can-you-recommend-on-a-free-sip-server-that-works-on-linux

You can use Trixbox or Asterisk.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:18:17 +0000
How Do I Configure my SIP Phone to Work with Virtual Phone Numbers? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/394-how-do-i-configure-my-sip-phone-to-work-with-virtual-phone-numbers https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/394-how-do-i-configure-my-sip-phone-to-work-with-virtual-phone-numbers

Virtual Phone Numbers does not support SIP registration as we do not provide termination services. If you have your own SIP server, you can accept incoming calls from Virtual Phone Numbers. You will need to use any of the VoIP Termination services, register your SIP phone with them, and forward calls from Virtual Phone Numbers to your VoIP Provider, that it will ring your device.

Please note, that the most updated list of VoIP providers, which support Virtual Phone Numbers services, you may find here.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:25:49 +0000
How to Set SIP Trunk Configuration for Virtual Phone Numbers on Asterisk? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/395-how-to-set-sip-trunk-configuration-for-virtualphone-numbers-on-asterisk https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/395-how-to-set-sip-trunk-configuration-for-virtualphone-numbers-on-asterisk

The following guide will explain how to set new DID number on Asterisk.

The following guide will explain how to set new DID number on Asterisk. In the following explanation we will use DID number 972775654199 for example, this DID number will be forwarded to the internal SIP extension 100.

Your Asterisk configuration files should look as follows:

sip.conf:


[46.19.209.10]
host=46.19.209.10
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.11]
host=46.19.209.11
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.12]
host=46.19.209.12
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.13]
host=46.19.209.13
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.14]
host=46.19.209.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.15]
host=46.19.209.15
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.75]
host=46.19.209.75
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.76]
host=46.19.209.76
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.77]
host=46.19.209.77
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.78]
host=46.19.209.78
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.79]
host=46.19.209.79
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.209.80]
host=46.19.209.80
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.10]
host=46.19.210.10
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.11]
host=46.19.210.11
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.12]
host=46.19.210.12
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.13]
host=46.19.210.13
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.14]
host=46.19.210.14
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.15]
host=46.19.210.15
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.75]
host=46.19.210.75
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.76]
host=46.19.210.76
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.77]
host=46.19.210.77
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.78]
host=46.19.210.78
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.79]
host=46.19.210.79
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all

[46.19.210.80]
host=46.19.210.80
dtmfmode=rfc2833
dtmf=rfc2833
type=peer
context=from-virtualphonenumber
insecure=very
nat=never
allow=all


extensions.conf:
[from-virtualphonenumber]
exten=> 972775654199,1,Dial(SIP/100)

]]>
webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:35:40 +0000
How Can I Generate Ring Tone Using Asterisk System? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/396-how-can-i-generate-ring-tone-using-asterisk-system https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/396-how-can-i-generate-ring-tone-using-asterisk-system

You can generate the ring tone by adding/changing the following strings:

exten => s,1,Answer
exten => s,n,Dial(SIP/andrej,,r)

]]>
webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 22:55:41 +0000
Are the VirtualPhone Numbers (DID) services compatible with Asterisk? https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/397-are-the-virtualphone-numbers-did-services-compatible-with-asterisk https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/397-are-the-virtualphone-numbers-did-services-compatible-with-asterisk

Virtual Phone Numbers supports VoIP protocols used by Asterisk, including IAX.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Sat, 25 Oct 2014 23:17:33 +0000
The List of Virtual Phone Numbers IP Addresses https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/1402-the-list-of-virtual-phone-numbers-ip-addresses https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/1402-the-list-of-virtual-phone-numbers-ip-addresses
Make sure your platform is not blocking these IPs
 
USA:
46.19.209.10
46.19.209.11
46.19.209.12
46.19.209.13
46.19.209.14
46.19.209.15
46.19.209.75
46.19.209.76
46.19.209.77
46.19.209.78
46.19.209.79
46.19.209.80

In addition, RTP media may arrive from 46.19.209.0/25, please allow the range in your firewall.

 
Europe:
46.19.210.10
46.19.210.11
46.19.210.12
46.19.210.13
46.19.210.14
46.19.210.15
46.19.210.75
46.19.210.76
46.19.210.77
46.19.210.78
46.19.210.79
46.19.210.80

In addition, RTP media may arrive from 46.19.210.0/25, please allow the range in your firewall.

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Wed, 28 Jan 2015 06:32:37 +0000
General SIP information https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/1446-general-sip-information https://www.virtual-phonenumbers.com/virtual-did-numbers/call-forwarding-number-mapping/voip/1446-general-sip-information

SIP addresses

We originates calls to customers from the following endpoints:

  • 46.19.209.14:5060 (for US PoP)
  • 46.19.210.14:5060 (for EU PoP)

These IP addresses correspond to our routing cluster exit nodes.

RTP addresses

Our system sends RTP packets from the following subnets:

  • 46.19.208.0/21

with RTP port-range: 10000-32767

RTCP

We transmit and receive RTCP packets from port = rtp_port + 1 (as recommended in RFC3550). Additionally, we are able to receive RTCP packets on the same port as RTP, considering utilizing RTCP conflicts avoidance payloads (payload types 72-76).

Supported codecs

  • G.711 A-law/U-law
  • G.729
  • G.723.1
  • L16
  • G.726-16/G.726-40/G.726-32/G.726-24
  • G.721
  • GSM
  • Speex

DTMF transport methods

DTMF signaling is supported as follows:

By default RFC 2833 enabled

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webmaster@globexcam.com (Virtual Phone Number Admin) VoIP Mon, 08 Aug 2016 12:01:27 +0000