Yes! CommPeak is happy to offer a rewarding partners program where IT professionals can join as VoIP reseller, VoIP carrier or VoIP provider.
Make use of our carrier-grade VOIP service and call quality to create a residual business for yourself!
Yes, at Pricing -> Termination, click on Download as CSV.
CommPeak’s termination service will work over any type of Internet connection that has relatively low and stable latency. Wireless Internet connections sometimes suffer from fluctuating latency and packet loss, both of which would cause sub-optimal audio quality. Most satellite Internet connections will have issues with delayed or choppy audio due to the inherent amount of latency.
Our recommendation is that you check the ping times to our POPs. We recommend an Internet connection with ping times below 230ms.
For best sound quality and, assuming you have the bandwidth available, we recommend G.711u. However, you can still maintain an excellent voice quality and lower bandwidth usage with codecs such as G.729.
We support the following codecs: G.729a, G.711u/a, GSM, ADPCM, iLBC
Most definitely! We offer a complete voice termination service in practically every country. We strive to provide our customers with the best possible quality routes and reliable all-around service. For this reason we are the No. 1 choice of termination for VoIP providers, SMBs & enterprises, calling card operators and more business segments.
CommPeak passes the customer’s caller ID to its carriers at all times. When configuring a SIP account (Setup -> SIP accounts -> Edit account), you can choose if you wish CommPeak to pass caller ID, restrict it or override it with pre-configured setting at the SIP account configuration. You can configure the account to respect caller ID based on SIP header “From”, “P-Asserted-Identity” or “Remote-Party-ID”.
Call origination refers to calls initiated by a calling party using a telephone exchange of PSTN, and reaching a VoIP endpoint. Call termination refers to the reverse process, whereby a call is initiated by a VoIP caller with a PSTN endpoint. CommPeak proudly offers carrier-grade voice termination, voice or voip origination solution, toll-free DIDs and wholesale SMS, among others.
We only provide DNS A records to point to our SIP proxies. sip.commpeak.com will provide different results based on GeoIP lookup.
We do not support establishment of IPSec with customers at this point, however, we are planning to offer this in the near future. If you require it, please contact us.
Yes, CommPeak currently supports SIP TLS and SRTP.
CDR records are accessible through the customer portal and are updated almost instantly, within minutes of call completion.
CommPeak has a large network of globally distributed POPs. We recommend making a ping test from your own network where the media servers are located to find the closest POP to your network: – uswest.sip.commpeak.com – useast.sip.commpeak.com – tokyo.sip.commpeak.com – singapore.sip.commpeak.com – brazil.sip.commpeak.com – ireland.sip.commpeak.com. You can always use sip.commpeak.com which will attempt to determine the POP that is closest to you.
Due to technical limitations, it will not always return the best results for you. We are aware that not all SIP proxies support DNS-based configuration of peers, which is why we will notify you via email in case we plan to change an IP address for a POP. However, we strongly advise to use DNS based configuration if possible.
CommPeak utilizes a virtual IP at each of its POPs for high availability. However, in case the entire local network (AS network) goes down, it may be useful to use multiple POPs for higher redundancy.
CommPeak SIP Termination is a proxy service. We do not absorb media. Your media is routed directly to the closest demarcation point of the carriers.
CommPeak supports both IP ACL and password based authentication. When creating new account at Setup -> SIP Accounts, you will be able to choose which authentication type you desire per sip account.
CommPeak accepts E.164, for example: 14087658080, 442074938000.
Any SIP compatible software or hardware is supported including: Asterisk, FreeSWITCH, OpenSIPS, Snom, Linksys, 3CX, FreePBX, Trixbox, Cisco, eyeBeam, Bria, Zoiper and many more.
We accept the following payment methods:
Wire transfer – $100 (USD) minimum per transaction. Please make sure you elect to pay all of your bank’s related charges/fees. Not doing so will result in these charges be deducted from the original amount wired.
PayPal Express Checkout – No limit to the amount we will accept on condition that the account is FULLY VERIFIED & ADDRESS CONFIRMED. We require a government issued photo ID and utility bill (for example, gas bill, electricity bill, cable bill). This needs to be done for each new PayPal account you would like to use.
Credit card – Payments will be accepted up to $500 in any 24 hour period. We require a copy of the front and back of the card (blanking out the middle 8 digits) and a copy of a government issued photo ID. Both must be emailed or faxed to us at [email protected]. This needs to be done for each new card you would like to use.
Here are our billing increments by location:
Other countries: 1 seconds initial; 1 second increments.
Billing increment is the way we (and the telecom industry) calculate our rates in order to bill your calls. For example, if you call the Unites States for 10 seconds, you will be charged for 12 seconds (call minimum is 6 seconds, and 2(x)6 seconds since this is a 6 second increment call), instead of the full minute.
If you call Mexico for 10 seconds, you will be charged for 1 whole minute (60 seconds minimum call charge, 60 second increments).
Before jumping right in to our free trial, we recommend that you explore our full range of industry-grade A-Z VoIP termination solutions. Once you have registered for your free VoIP trial, you will be given a small amount of credit to test the service ($0.20 USD which gives you approximately 30 call minutes).
There is a minimum top-up value of $25. This minimizes our admin expenses, allowing us to keep rates low.