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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 proxy (or "Outbound Proxy")leave blank
Username (or User ID)username
Auth name (or Auth ID)username
Display NameYour name
Register (or Send registration request)Yes if you expect incoming calls (have a DID with us).
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, to fix the problem go to Admin/Advanced/SIP menu in the ATA settings and change the codec NAME to G729. No letter "a" at the end.
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 and you expect incoming calls (have a DID with us).
Use IPv6No if your network is not IPv6 ready.

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

;canreinvite=no ;if your asterisk box is behind a NAT router
register =>


If you forward incoming DID calls to external URI, the calls could come from or, 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 country code, +15557772222 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. If standard routes do not deliver CID right, then switch to Premium or PSTN tariff plan in "User settings".

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 our FAQ.

Fine print: All prices are final, there are no bogus fees and unfees other than specified at Services page. Period. Only SIP devices that have already been created can be connected to 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). 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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