Recent content by tconn

It would be easier if you sent us your config or even the logs... <Sip.conf> register => 2145551212:pw@sip.broadvoice.com:5060 [sipBroadvoice1] type=peer user=phone host=sip.broadvoice.com fromuser=2145551212 fromdomain=sip.broadvoice.com secret=pw username=2145551212 insecure=very...

Do I get a pink star? :)

Try mimicking my config on the register line.

Yup, host= name or IP: [labcm33] type=friend host=172.16.0.1 context=labcm33 ;<- calls in from CM goto this context in extensions.conf disallow=all allow=ulaw allow=alaw nat=no canreinvite=no qualify=yes

Mine does calls in both directions. Vonage's marketing department makes medium to large businesses shy away. They way they have things priced doesn't scale unless you use less than 500 minutes a month. Check out Broadvoice's plans (unlimited for $30 I think?). Like any other decently support...

I'll try the trace. The calls completes when you hits transfer the first time, but it won't successfully transfer the call. I'll dig into the trunk-to-trunk transfer settings also.

Definity G3 Rev 11 (CM1.3) I have noticed when calls are transferred, you cannot complete the transfer if the destination extension is listed in the uniform dialing plan table. I have redirects setup so that 4 digit extensions are redirected to their new 5 digit extension which is on a newer...

Anyone know if returning off-warranty no-support dead cards is worth the time or money? I've got a TN464 with a burnt spot on it, and another that just doesn't work. Should I chuck them in the trash or is there a reasonable repair channel for these cards?

Hmm.. No replies. Has anyone seen a way to troubleshoot something like this? I see the CID go out the trunk to the Definity, but when the Definity routes the call, the CID is being removed.

I've got Trunk group 2, which is multiple PRI to the LEC. CID works great in both directions. I've got trunk group 4, which is a TIE line PRI to my VoIP device (not avaya) which is able to send caller-id to the definity and have it show up correctly when internal calls are made from the VoIP...

I'm able to dialout, just not having much luck dialing in... I suspect my firewall is not allowing rtp through.

Error 500 isn't a "unreachable" type error. This is a response from the webserver saying its not able to produce the webpage you requested. Probably an internal error or something bad... Sorry I can't give you more specific info. you might check your event viewer to see what its reporting.

I will probably end up using a asterisk server in the middle. I'm not having much luck doing any trunking from the CM...

Is there anyone out there who has successfully configured a sip-trunk from a Cisco Callmanager to a Vonage softphone account? Anyone?

Never mind, found a vendor willing to do a month to month @ $25 per phone.