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SIPclient configuration

Different SIP clients (hardware SIP phones, ATAs and software ones) have different configuration screens, but all have a common set of configuration parameters. In the table below, username and password are your 9-digit long SIP username and the password shown in "VoIP accounts" menu in customer portal.
Configuration parameterValue
SIP server (or proxy, or domain)sip.callwithus.com
SIP proxy (or "Outbound Proxy")leave blank
STUN serverstun.callwithus.com
Username (or User ID)username
Passwordpassword
Auth name (or Auth ID)username
Display NameYour name
Register (or Send registration request)Yes
G729a Codec Name (for buggy Linksys/Sipura/Cisco ATAs)G729. The default codec name in those adapters is set to non RFC compliant "G729a" and might not work with our service, go to Admin/Advanced/SIP menu in ATA settings to change the codec name.
RTP Packet Size (for buggy Linksys/Sipura/Cisco ATAs)0.02
Registry Expiry (or Registration interval)120 sec (2 minutes) if your SIP client is behind NAT router.

If your ATA is behind a NAT router, then the router configuration is very important. If your router "supports" SIP ALG (or SPI), disable that feature ASAP. See more details in this wiki article. Our servers implement "server side solutions", but SIP ALG in faulty routers breaks it.

Here are the examples of SIP clients configuration (thanks to our users!)

If your SIP client is behind a firewall, make sure your firewall does not block UDP port 5060 and the range of UDP ports used for voice transmission (see your firewall and SIP client documentation for details).

Asterisk PBX configuration

sip.conf
[general]
;canreinvite=no ;if your asterisk box is behind a NAT router
register => username:password@sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=username
secret=password
qualify=no
insecure=invite
extensions.conf
exten=>_X.,1,Dial(SIP/callwithus/${EXTEN},60)
exten=>_*X.,1,Dial(SIP/callwithus/${EXTEN},60)

If you forward incoming DID calls to external URI, the calls could come from west.callwithus.com or east.callwithus.com, make sure your settings and dialplan support multiple hosts, see FAQ for details.


Caller ID setup

We suggest you to add your real telephone number (starting with the area code, 5557772222 for example) to caller ID ("Add Caller Id" menu in your control panel). When you place a call this real number will be shown to the called party. Caller ID spoofing and/or call center and autodialer calls are not allowed with our service. We do not guarantee reliable CID number delivery to all destinations.

DID destination setup

You can set up your DID to forward incoming calls to PSTN or cell phone line or to any VoIP destination. See more details in FAQ.
 

Fine print: All prices are final, there are no bogus fees and unfees. Period. Only SIP devices that have already been created can be connected to sip.callwithus.com to make calls. Please ensure you only use devices approved by you (Please do not try and connect using two tin cans and a piece of string as we do not yet support this, but we may support this in the future, the work is in progress and preliminary results are positive). Callwithus.com monthly subscription charge of $0 must be paid in advance and does not include tax of $0 which also must be paid in advance. You will be billed an activation fee of $0 plus tax and this must be paid in advance. Calls made incur tax at the rate of 0% each month and must be paid in advance. On cancellation of the service you will be charged a one time disconnection charge of $0. Additional features will be billed at the additional rate of $0 per call. All **YOUR** rights reserved.

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