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Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g.

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

Hi all, I am looking for a software package (free or not), or an inexpensive hardware device, which can route calls between an H.323 network and an MGCP-based voice service. Unfortunately, I believe (based on documentation and other forum posts -- I have not looked at the code) that Asterisk can only act as an MGCP Call Agent. Has anyone added code that can talk the 'slave' side of the

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

Has anyone tried SunRocket with Asterisk? http://www.sunrocket.com/ The $199/yr. plan seems like an excellent value, and most reviews have been favorable. However, I don't know if it is possible to obtain the SIP credentials, so one can bypass their "gizmo". Thanks, Stewart

Hi, I'm looking for a reliable, reasonably-priced, single-channel interface between * and US GSM. The VOIP GSM Gateways listed at http://www.voip-info.org/wiki-VOIP+GSM+Gateways (VoiceBlue, QUTEX) are multichannel systems, very expensive ($2500 or more). Next step down, there are various Fixed Cellular Terminal (FCT) or Fixed Wireless Terminal (FWT) devices. These typically have an FXS

> I have "callwaiting=yes" in my zapata.conf, and "Call Waiting Serv: Yes" > in the PAP2s. > However, there's sitll no callwaiting on the PAP2s. Everything else work > fine. Any ideas? Am I missing something somewhere? Hi Andy, You also need to set "CW Setting: Yes" on the User 1 and User 2 screens. Or, dial from each line, whatever you have set for

I'd like to use the VoIP service from Free with Asterisk, but am having a couple of problems. Here are some details: ADSL from Free T?l?com comes bundled with VoIP and TV services. Most users access the VoIP via the supplied Freebox, which is an integrated DSL modem, router, ATA, and media player. It is of course possible to connect the Freebox to Asterisk via an X100P or other FXO

> I have a * box with 2 nics in the following setup: > > Internet > | > 192.168.5.253 (firewall) > | > 192.168.5.xxx network (gw 192.168.5.253) > | > 192.168.5.10 (* nic 1) > 192.168.6.10 (* nic 2) > | > 192.168.6.xxx network > > The netmask for both networks is 255.255.255.0 > > The 192.168.6.xxx networks has a 48 port switch solely for the use

Hi, You need to set the DialPlan parameter to allow the proper number of digits to be collected, for all types of numbers used in your system. I believe that the factory default value would work for long numbers beginning 0011, but your unit was probably previously configured for a different environment or country. Below is an extract from the example in my H.323 firmware; I believe that

Hi, I'm testing Asterisk with a new provider. On calls to US toll-free numbers, there is no audio (calls to normal numbers are ok). In response to a valid INVITE from Asterisk, something like this is received: SIP/2.0 183 Session Progress v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea CSeq:103 INVITE i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com f:"Test User"