It is better to stay with Postgres. If you don't want to loose your
business stay away from MySQL.
Oh come on, there are many reasons to use Postgres, but this is just FUD.
Just as an example off the top of my head, take a look at
http://www.livejournal.com/stats.bml (2.5 million active
Has anyone else encountered Broadvoice problems today? I was unable to
log in at all until after lunchtime. Now I can connect but any calls
ring once then get 480 Temporarily Unavailable back from Broadvoice.
It's now also impossible to call their support desk, any calls receive
a recorded
I have a problem with ringing simultaneous channels where one is IAX
and one is Zap
I have two Zap channels and a single extensions on IAX2
I'm trying to take incoming calls on Zap/1 and if not answered in 15
seconds by IAX2/100 to keep ringing IAX2 and also try another number
on Zap/2
Alexander,
I'm afraid it's POTS (X101P) and from what I have seen since I posted
this is my problem.
I wouldn't mind it hanging up the IAX2 channel and then calling it
again, but it seems that everytime the new call via Zap/2 means no
other calls are possible.
There is ISDN in the office, but I
Hi folks,
Moving on to a new install, I'm jumping straight to v1.4
Without using Priority jumping I'm wondering what the 'standard' way
to indicate to the calling party that the number the dialed is busy or
unavailable. So,if I have an entry in extensions.conf like this:
[outbound]
exten =
On 1/18/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Looks at macro-stdexten in extensions.conf.sample. Also see show
application dial
Ah, that's exactly what I was looking for - thanks.
Russell
___
--Bandwidth and Colocation provided by
Hi,
I've set up a Gizmo Project account for access on my Nokia E61 because
they work through NAT. Trouble is If I include my gizmo account in an
asterisk hunt group and I'm not connected (phone is off / outside
wireless coverage) the gizmo project always answers. Either the call
goes to voice
Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this. A
case has been upen
to supply
connections to the UK PSTN.
On 7/7/05, Michael Welter [EMAIL PROTECTED] wrote:
Russell Horn wrote:
Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls
]
type=peer
username=9479
qualify=yes
secret=
host=sip.talklite.net
canreinvite=yes
disallow=all
allow=ulaw
[2201]
type=friend
context=flat
username=albanach
secret=
defaultip=192.168.1.100
qualify=yes
type=friend
callerid=Russell Horn
host=dynamic
nat=no ; X
On 6/13/06, Jonathan Attwood [EMAIL PROTECTED] wrote:
Could your register line require attention ? (2001?)
7960xxx:[EMAIL PROTECTED]/2001 - I thought your target was 2201?
That's a good spot and I've fixed it now, but I'm sure it's not the
problem. I'm not seeing any sip traffic coming in at
Following up to my earlier post.
I'm seeing no inbound SIP traffic locally despite, apparently, being
sucessfuly reigstered with my sip provider.
sip show peers give me
Name/username HostDyn Nat ACL Port Status
2201/2201 192.168.1.100D
config is pretty basic at the moment:
[general]
format=wav
attach=yes
[default]
101 = ,Russell Horn,[EMAIL PROTECTED]
Everything else seems to work fine with inbound and outbound calling
using SIP and IAX2 trunks.
Thanks for any suggestions,
Russell
I'm able to receive but not make calls with zaptel using an X101P
connecting to Asterisk with an Xlite client. My client has context = flat
in sip.conf and extensions number 8919
In extensions.conf I've got:
[home]
; Line 1
;
exten = 8919,1,Dial(${PHONES1},20,Ttm)
exten =
Is it possible using asterisk to allow someone to dial in and remotely
change where their call is forwarded to?
For example, I'm working from home so I want my calls to go to 555 1234,
now I need to go out for a bit so I'd like to phone the office and using
DTMF tell the asterisk PBX to now
Since Thursday evening my asterisk box has been failing to register with
broadvoice. I haven't changed any of my config files in the last week.
Can anyone suggest anything?
Asterisk is reporting:
*CLI Aug 28 16:15:17 NOTICE[6150]: chan_sip.c:3914 sip_reg_timeout:
Registration for '[EMAIL
Hi,
We are using MozIAX as a softphone with a USB headset and are making
outbound calls using IAX with ulaw encoding to our voip provider.
We're running asterisk 1.4
Users are complaining that the ring tone generated by asterisk is much
louder than the voice call once connected. They are having
Hi,
Incoming calls to one of my lines are set to ring two internal lines
and simultaneously start ringing my cell phone. Something like this:
exten = s,1,Dial(SIP/2201SIP/2202IAX2/[EMAIL PROTECTED]),90)
The internal lines 2201 and 2202 will both see the callerID for the
incoming call, but my
On Nov 17, 2007 8:13 PM, Robert Lister [EMAIL PROTECTED] wrote:
I think your carrier has to permit you to set callerID to something that is
not one of your numbers in the range you have been allocated.
I know I can set a different caller ID - I'm just not sure how in
Asterisk I would set the
On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote:
From what I have seen in the past asterisk should pass along the CID
automatically. As some one else already mentioned. It can be your ITSP. You
can always set the CID with Set(CALLERID(num)=1234567890).
Asterisk does pass the caller ID
Hi,
I've a DID number that gets passed to three internal phones and a cell
phone via my outbound IAX trunk. If the cell phone is off or out of
coverage, its voice mail captures the call.
What's the best way to avoid this? Is there a recommended way to force
the cell phone user to press 1 before
On Thu, Nov 5, 2009 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote:
You can dial the cell like this
Dial(DAHDI/1c/w5551212) instead of
Dial(DAHDI/1/w5551212)
Danny - thanks, however I think that's a feature of DAHDI. My outbound
trunk is IAX.
I don't think that's a standard feature of
On Tue, Apr 24, 2012 at 6:17 PM, John Novack
jnov...@stromberg-carlson.org wrote:
Voip.ms is high quality, handles number ports and supports both IAX2 and SIP
2 different pricing plans, and their costs range from 4.95 to 7.95 per month
depending on the rate center for one plan, and less with
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