Displaying 7 results from an estimated 7 matches for "glomph".
Did you mean:
glimp
2004 Dec 06
3
PRI/Zap premature dialing problem
...considered
'complete' it will build the setup frame and transmit it to the terminating
system.
Watch the output of 'pri debug span 1' when the originating system is
dialing different numbers to see this in action.
Hope that helps.
> -----Original Message-----
> From: Jerry Glomph Black [SMTP:asterisk-users@glomph.com]
> Sent: Monday, December 06, 2004 5:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] PRI/Zap premature dialing problem
>
> I know this may be RTFM flamebait, but I've spent a lot of time on the...
2004 Oct 03
0
FW: Broadvoice
...**:XXXXXXX@sip.broadvoice.com/9999
;;
; Providers
[bvoice]
type=friend
canreinvite=no
username=847*******
fromuser=847*******
callerid=847*******
secret=*******
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=inband
insecure=yes
Thanks!!
Zac
-----Original Message-----
From: Jerry Glomph Black [mailto:glomph@glom.ph]
Sent: Sunday, October 03, 2004 12:53 PM
To: Zac Amsler
Subject: Re: [Asterisk-Users] Broadvoice
No problems here. Inbound/Outbound both fine.
Their NAT behavior is pretty bad, I'm sure your
problem is probably related to that.
On Sun, 3 Oct 2004, Zac Amsler wr...
2006 Jan 31
1
Polycom IP301: Pass-through ethernet port unusable?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Jerry Glomph Black
> Sent: Monday, January 30, 2006 11:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port
> unusable?
>
> Have just done a deployment of 45 of these puppies.
>
> They are doing their...
2005 Jul 21
11
IAX over HTTP
For work environments where you only get HTTP or HTTPS access, what is
the feasibility of doing IAX over HTTP?
I have heard of projects such as stunnel.
Has anyone tried something like this?
I did a quick search but didn't come up with much.
2004 Jul 27
1
asterisk <-> stanaphone?
I had a working 2-way SIP connection running until about 2 days ago, now my
outbound calls are promptly blocked with a "403 Forbidden" error. Inbound
still functions OK.
Perhaps they are fingerprinting
and blocking Asterisk access (I hope not). They do not answer their support
mail, or questions on their own forum.
I'm sure there are other Asteriskers out there who have
2004 Dec 08
1
Leadtek BVA8051 / Sipphone.com CallInOne with Asterisk?
I have a lot of experience, all of it pretty good, with various Sipura products,
Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into
Asterisk as clients. Good sound quality, great reliability.
I've tried two of the units named in the subject line, and frankly I'm
frustrated. Calls usually start out OK, but within a brief period the sound
goes totally to
2004 Sep 24
0
Calling to Broadvoice via Linux MASQ (NAT)
I just signed up for Broadvoice, and used a similar network configuration that
I have on stanaphone, voipjet, and others.
My asterisk box is behind a vanilla Linux masquerade (netfilter/ipchains)
firewall. The SIP and IAX services have been working fine in both directions
for the other SIP termination services.
The Broadvoice inbound service worked immediately. (which to me is odd, inbound