search for: pipecall

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxx...

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I guess I didn't give enough detail in my last message, so here's as much as I've done so far: 1. I've reconfigured to network to non-NAT (was 1:1 NAT) so there's no rewriting going on. 2....

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel He...

...mbination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as: [voiptalk] type=peer secret=xxxxx username=xxxxxxx host=voiptalk.org [pipecall] type=peer secret=xxxxx username=xxxxx host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to au...

I'm using Asterisk to registry several DDI's to a sip proxy (pipecall.com). Everything works fine apart from several times a day my firewall (zywall70) reports a UDP port scan attack from the pipecall sip proxy. I can't seem to work out why this should be. All I could think was that the sip registry was expiring and causing some strange probing from the proxy, is...

...guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter? [...] [pipecall] type=peer secret=xxxxx username=xxxxx host=sipproxy.pipecall.com The first one works OK - I can dial out with no problems. The second one needs an extra field for the authuser - when I try to dial out I just get: May 17 01:03:45 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to aut...

...ike to get the same thing for the UK, so that UK customers can call them as a local call AND they can dial out UK numbers as local calls. The obvious side benefit would be that US employees could call UK customers and vendors as a local call as well. I've looked at Telappliant, VoipTalk and PipeCall so far, and I'd like to get some feedback before going with one or the other. I'd be grateful for any opinions on the quality of (these, other) services, how responsive they are to problems, and if they are as easy to setup with Asterisk as TelaSIP. Recommendations are appreciated, of...

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris