I'm getting these messages from this Asterisk version:
Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h
Sep 19 07:11:18 WARNING[99391]: chan_sip.c:2084
sip_new: Unable to allocate channel structure
Sep 19 07:11:18 NOTICE[99391]: chan_sip.c:7523
handle_request: Unable to create/find channel
When I kill the a
when I searched for _**6XX the site returned:
An error occured in a database query!
Context:
File/tiki-searchresults.php
Url
/tiki-searchresults.php?words=_**6XX&where=pages&search=go
Query:
SELECT COUNT(*) FROM tiki_comments c, tiki_pages p
WHERE c.objectType = "wiki page" AND
p.pageName=c.o
Hey anyone know once you've bought a DID from DIDx
under the Purchased DIDs tab you can click a link for
one of your numbers that says '0 (SIP)'
and when you click that like there's SIP and a space
or the URL and AIX and a space for the URL
What do I put here?
How do I setup Asterisk to accept
I am getting this on the console once people call in
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"sipline|3044") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"SIP/3044|20|
Here is the full "transaction"
-- outgoing agentcall, to agent '1001', on
'Local/[EMAIL PROTECTED],1'
-- Called Agent/1001
-- Executing Macro("Local/[EMAIL PROTECTED],2",
"sipline|3044") in new stack
-- Executing Dial("Local/[EMAIL PROTECTED],2",
"SIP/3044|20|t") in new stack
-
features.conf is devoid of #
the queue doesn't have h in it.
only have tT
--- Kevin Bockman <[EMAIL PROTECTED]> wrote:
> Crystal Stream, Incorporated wrote:
> > I am getting this on the console once people call
> in
> >
> >-- outgoing agentcall, to
Here is my extensions.conf file for debugging
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http://mail.yahoo.com
extensions.conf
Description: 3949034846-extensions.conf
___
--Bandwidth and Colocat
How do I combined these in and out wav files on the
fly through asterisk to where I hear the whole
conversation and only have one wav-file
(i.e. :
agent-1001-asterisk-478-1127389080-17-in_out.wav)
agent-1001-asterisk-478-1127389080-17-in.wav
agent-1001-asterisk-478-1127389080-17-out.wav
After "-- IAX2/NuFone/3 is making progress passing it
to SIP/3044-bcd0" I'm getting a "Busy" tone and it's
not even connecting the call.
-- Executing Macro("SIP/3044-bcd0",
"outvoip-2|1800759") in new stack
-- Executing SetCIDName("SIP/3044-bcd0", ""X X X"|a")
in new stack
-- Executing Se
g 11
> 8663xx3 is that correct? I had an issue last
> week with passing to many
> digits to my provider and the call would hang up
> immediately.
>
> You could also turn debugging on for this so we can
> get a better log.
>
> iax2 debug peer nufone
>
&
1) What do these two notices mean?
Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
198.22.67.70, request '[EMAIL PROTECTED]' does not
exist
Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
66.234.228.170, request
Hello,
How would I setup where I call into my number and
press say 911 and then it would ask for a pass and
would accept it and then would prompt for a number so
I could call out of my number on the road?
Joshua
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Hello,
I have a block of 25 DIDs and have 10 phones on the
network. I want when a person tries to call out for *
to pick a number for the CIDN and I want to make sure
that the number isn't duplicated while it's in use.
Joshua
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Hello,
How do I make sure the G.729 codec is being utilized
fully and not just as a passthru? I've registered it
and followed the install instructions
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http://mail.yahoo.com
_
gainst your licenses 'cause the
> phones are now the only
> devices in the conversation. you can still find in
> my phones' status
> menu what codec it has negotiated for the current
> call. When asterisk
> drops from the loop for me, my phones remain on
> g729.
&g
Here is my extensions.conf file. Things have been left
out or changed to protect the innocent.
Why isn't it working when I call from the outside that
when I press 124 it repeats the menu and doesn't
initiate DISA correctly to dial out?
[general]
static=yes
writeprotect=yes
[globals]
[voicepulse-
ot right. There
> should be small numbers there, like 10-20ms in my
> experience. I'm not
> familiar with wy "show g729" doesn't work if the
> g729 codec is
> registered, however
>
> Crystal Stream, Incorporated wrote:
> > show g729 doesn&#
Just to recap my other three questions:
1) I have a block of 25 DIDs and have 10 phones on the
network. I want when a person tries to call out for *
to pick a number for the CIDN and I want to make sure
that the number isn't duplicated while it's in use.
2) What do these two notices mean?
Oct 4
Hello,
I have VPC (Voice Pulse Connect) and NuFone for
providers and I have setup modules.conf with the
registered (Digium) G.729 Codec such as:
load => codec_g729a.so
load => res_crypto.so
With both sip/iax2 configuration disallow=all is first
and then allow=g729 is next
(allow=ulaw,allow=alaw,a
s if I remember
> right) have the priority
> configured to place ulaw above g729? It's in your
> ipmid.cfg (if you do
> use polycoms) or maybe your sip.cfg if you've
> upgraded to SIP 1.5.2 and
> migrated your config files
>
> Crystal Stream, Incorporated wr
Ah It was a typo. It should work now! L:)
--- Erik Slooff <[EMAIL PROTECTED]> wrote:
>
> written by Crystal Stream, Incorporated
> > Here is my extensions.conf file. Things have been
> left
> > out or changed to protect the innocent.
> > Why isn't it worki
Hey I just setup service with IPComms and they are
telling me to setup such as this:
iax.conf:
[IPCommsNet]
type=user
host=69.15.xxx.xx
context=voicepulse-in ;(changed by me)
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=gsm
When I'm calling once of my numbers it's giving me
t
this isn't working
[IPComms-in]
exten => s,1,Noop(${DATETIME} ${CALLERID})
exten => s,2,SetCallerID(${CALLERID})
exten => s,3,Answer
exten => s,4,Goto(main-menu,s,2)
exten => s,5,Hangup
What I have is a block of 20 DIDs and I want to accept
calls from all of them.
It would be way to freaking comp
CURRENTLY in GLOBALS I have
IPCCIDN01 - IPCCIDN20 set to their respective numbers
How do I need to change the IPComms-in context to do
what you say
--- Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Crystal Stream, Incorporated
> [EMAIL PROTECTED] wrote:
> > this isn't
Hello How do I fix this
[IPComms-in]
exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID})
exten => ${IPCCIDN01},2,SetCallerID(${CALLERID})
exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID})
exten => ${IPCCIDN02},2,SetCallerID(${CALLERID})
exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}
what does this mean and how do I fix it?
Channel 'SIP/3044-80e1' sent into invalid extension
'573486' in context 'redial-from-local', but no
invalid handler
Here is the
[redial-from-local]
_91NXXNXX,1,Macro(redial,${EXTEN})
_91NXXNXX,2,Congestion
_9NXXNXX,3,Macro(redial,${EXTEN})
I'm getting:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type:
IAX Subclass: REJECT
Timestamp: 9ms SCall: 8 DCall: 2
[66.98.180.77:4569]
CAUSE : No authority found
Under the DIDx number I'm putting IAX:
[EMAIL PROTECTED]/1567252(IAX)
where N is the rest of
You could have your customers call in and enter all of
that -- then give them a confirmation number and they
could fill out the rest online.
--- trixter aka Bret McDanel <[EMAIL PROTECTED]>
wrote:
> I am interested in hearing some user experiences of
> anyone using a
> merchant account. The cons
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