search for: nuvox

All, Having problems terminating to a Nuvox PRI, the tech at Nuvox is saying Asterisk is transmitting in CCITT (aka ITU?) when they're expecting (and will only accept) ANSI. The question is, is there a simple way to change this or am I stuck with rewriting code? I googled and checked the mailing list and found nothing, I could be barking...

...lly restarted on span 2 pbx1*CLI> (This is from an asterisk console started with -rvvvv.) Has anyone encountered similar behavior in the past? If so, what was the resolution? Thanks, Tim McKee configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallin...

...some > > circumstances beyond my control and I don't know a lot about > > carriers and how this all works. I am figuring it out, but > > it's a lot of trial by fire. > > > > As far as I know, we only use 1 carrier for our system. We > > have a PRI from NuVox and we use 7 channels for our asterisk > > server. So, I have a few questions: > > > > Is asterisk or the carrier causing the disconnect? > > > > Is IBM (the 800 number I am dialing) not passing the answer > > supervision or is that a function of the carrier? &g...

...er have jumped in to this asterisk project at work due to some circumstances beyond my control and I don't know a lot about carriers and how this all works. I am figuring it out, but it's a lot of trial by fire. As far as I know, we only use 1 carrier for our system. We have a PRI from NuVox and we use 7 channels for our asterisk server. So, I have a few questions: Is asterisk or the carrier causing the disconnect? Is IBM (the 800 number I am dialing) not passing the answer supervision or is that a function of the carrier? Is there a way to make asterisk not drop the call or to for...

Failed to deliver to ''menyhert@one.net'' SMTP module(domain one.net) reports: host smtp-in1.nuvox.net says: 550 5.0.0 <menyhert@one.net>... User Unknown -------------- next part -------------- Skipped content of type message/delivery-status-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/rfc822-headers Size: 476 bytes Desc: not...

Has anybody else noticed cygwin.com and sources.redhat.com being down? The machine that answers to those addresses (and sourceware.org) is refusing HTTP connections, and has been for at least 48 hours now. Anybody here know anything about it? I haven't been able to find out diddly-squat about it on the web... -J

Hi All, Interesting that I couldn't find this documented anywhere, but what happens to text that is typed after the label on the syslinux prompt? My current issue is I'm using PXE to do kickstart installations. I want the ability to do "rescue" boots also with PXE. I've gone looking at the current RH80 disk 1 which has this capability. It has a directory called

...project at work due to some > circumstances beyond my control and I don't know a lot about > carriers and how this all works. I am figuring it out, but > it's a lot of trial by fire. > > As far as I know, we only use 1 carrier for our system. We > have a PRI from NuVox and we use 7 channels for our asterisk > server. So, I have a few questions: > > Is asterisk or the carrier causing the disconnect? > > Is IBM (the 800 number I am dialing) not passing the answer > supervision or is that a function of the carrier? > > Is there a way to...

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates