List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel - Asterisk - SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension - Asterisk -
,SendDTMF(${EXTEN})
exten = _5XXX,n,Hangup()
Thanks again,
PB
On Tue, Jun 17, 2008 at 11:15 AM, Paul Belanger [EMAIL PROTECTED] wrote:
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap
Morning list,
Was curious if it is possible to decrease the time asterisk takes to
answer an incoming call to a zaptel interface.
Example:
[Jun 11 09:33:06] VERBOSE[4489] logger.c: -- Starting simple
switch on 'Zap/2-1'
[Jun 11 09:33:10] NOTICE[4489] chan_zap.c: Got event 18 (Ring Begin)...
number!
;
;immediate=yes
---
On Wed, Jun 11, 2008 at 9:38 AM, Paul Belanger [EMAIL PROTECTED] wrote:
Morning list,
Was curious if it is possible to decrease the time asterisk takes to
answer an incoming call to a zaptel interface.
Example:
[Jun 11 09:33:06] VERBOSE[4489] logger.c
Thanks Steve,
Forgot about callerID. We are not using callerID on the lines and
have disabled it. Asterisk now answers right away.
Thanks again,
PB
Do you actually have callerID on your line? That takes about two
seconds. Try removing it and see how much faster Asterisk answers.
That
List,
Anybody have success with Digium's G729 codec and asterisk 1.6.0?
Reading
http://www.russellbryant.net/blog/index.php/2008/03/05/codec_g729-v34-builds-now-available/
is seems they are build for 1.6 and trunk. But all I could find / use
is 1.4 builds from
I noticed safe_asterisk is nolonger used from the init.d script (on
ubuntu) for asterisk-1.6.0-beta9. I'm curious if there is another
init.d script out there, or even the best way to call safe_asterisk.
Or is safe_asterisk nolonger the script of choice for starting,
restart asterisk.
One of the
G'day all,
I'm trying to come up with a quick, easy solution to have a static
inbound number in my dialplan, rotate calling 2 numbers. Example:
1st call into asterisk
exten = 1234,1,Dial(sip/,10)
exten = 1234,n,Dial(sip/,10)
2nd call into asterisk
exten = 1234,1,Dial(sip/,10)
I do link the idea of have a queue answer the calls and route to the
extensions, but will have to figure out a way to do this with have the
SIP extensions logging into the queues.
On Thu, May 8, 2008 at 1:53 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
An option to rotate between numbers is to
open /etc/asterisk/modules.conf and add the following:
load app_meetme.so
save and close file; reload asterisk
Fabio Montemaggiore wrote:
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension
http://www.unlimitel.ca
not sure if they offer DID for Detroit
Technical Support wrote:
Can anyone recommend a good IAX provider offering numbers in Toronto and
Detroit?
___
#root service asterisk start
Starting asterisk: [ OK ]
# ps aux
does asterisk show up as a process?
PB
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Cause No. 34 - No circuit available (circuit/channel congestion)
This cause indicates that there is no appropriate circuit/channel
presently available to handle the call.
http://www.telos-systems.com/?/techtalk/cause.htm
Might want to talk with your telco
BTW: Don't cross-post!
Matt
Thanks for all the replies! Looks like I was shipped the wrong
powersupply. I figured as much, cause when I first plugged it in it
took a while to boot, and started to smell something burning. :(
Time to RMA it back and get them to ship me the proper parts.
PB
Paul Belanger wrote:
Can
See comments inline!
Damon Estep wrote:
I have officially engaged in a pissing contest with the local Telco over
PRI calling name delivery.
Welcome to my world, I deal with theses guys daily! Errgiant arn't
they. We have a saying around work 'The telco is always wrong!'.
The telco
Can somebody who has a SoundPoint 501 please confirm the power adapter input /
output settings:
Input: 120V AC 60HZ 20W
Output: 24V DC 500mA
PB
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Asterisk-Users@lists.digium.com
lspci -v what output do you get? Also, what OS are you using?
Jeff Borders wrote:
I think I have a bad FXS module on my TDM400P.
Channel map:
Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
2 channels configured.
ZT_CHANCONFIG
Can you see the INVITE if you put up a trace on your gateway
(209.XXX.XXX.113)? Asterisk is not getting anything back that is why it
retransmits 5 times.
PB
OMS wrote:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f
From: 512538XXX sip:[EMAIL
Where are your calls coming from? Are you connected to the Telco or PBX?
PB
Panitaxx wrote:
Hi,
thanks for your response. here is the log of one call:
Enabled debugging on span 1
Asterisk*CLI
Protocol Discriminator: Q.931 (8) len=33
Call Ref: len= 2 (reference 72/0x48) (Originator)
What type of client (Analog, SIP, IAX, etc??). Also, is
res_indications.so loaded?
PB
Stephen J. Wilcox wrote:
Hello,
can anyone help with my problem below, searching doesnt show any results..
thanks
Steve
On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
Hi,
I'm seeing a problem where if
In you sip.conf what if you change:
register = 7771::[EMAIL PROTECTED]/7771
to
register = 7771:[EMAIL PROTECTED]/7771
PB
Jenna Cole wrote:
im using iptel.org SER proxy.
the proxy is working without authentication.
the problem is that the Asterisk is not sending a
REGISTER sip message.
Today the front page of http://www.voip-info.org/ was taken out by a
spammer. It also seem the history page for http://www.voip-info.org/
was also nuked. I've restored the best I could using google cache, but
still missing some information.
Who is an admin on http://www.voip-info.org/ and
Hello,
See comments inline
Alvaro Parres wrote:
Hi list:
I have a client that needs to connect a Asterisk PBX with a TE110P
of Digium and one Nortel Option 11.
Actually the Nortel Option 11 have a AMI E1 card. With it the have
problems of clock sync.
Is the Nortel the CPE or
with the PRI it's going to be easy all the work ??
Only one question the Nortel guys here, say that they need one more
clock to have a PRI card, is this correct
On 8/5/05, Paul Belanger [EMAIL PROTECTED] wrote:
Hello,
See comments inline
Alvaro Parres wrote:
Hi list:
I have
How about posting the output from the console? version of asterisk,
zaptel, etc.
Also, have you checked out http://www.voip-info.org/wiki-Asterisk+cmd+Dial
quote
Return codes
If all the called channels are busy, Dial will exit with a return code
of 0 and will continue in the current context
check in modules.conf:
load=res_indications.so
is it there?
Bernie Courtney wrote:
indications.conf reads as follows
[general]
country=us
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion =
Olle,
Awesome! Now that everybody know your aiming for September 1 for
Asterisk 1.2, I'm sure will make it.
Come' on Asterisk community, step up to the plate!
PB
Olle E. Johansson wrote:
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
See inline comments:
Peter Svensson wrote:
What span is your clock source? A TE405P card can only operate in one
clock domain at a time. I.e. the same clock will be used on all of them.
Not correct, I actual have span 1 connected to my telco and span 2
connected to a Norstar PBX. See
Ryan Williams wrote:
I understand how CID works and how you must set CID when dialing out on
a PRI and how the phone company sets the name.
I was wondering how this works in regards to inbound calls. I have a pri
and I get the number that the caller is coming from but I do not get the
name.
Are your problems with incoming calls to your PRI or outgoing calls?
Are the calls being dropped or just not hitting your asterisk box?
PB
JOAO CARLOS MOURA wrote:
Hi All
Help.
We are using a T1 with Paetec Telecom in the Miami area, with a Digium card
into our Asterisk
software, and in
Oh you but I did, was not impressed. So, I sent them a friendly email
(hehe) asking WTF? What burns my ass, is they used a reply address of
[EMAIL PROTECTED]
PB
Jay Milk wrote:
Got an email this morning with the subject Welcome to Gizmo Project.
I didn't sign up with those yokels. Anyone else
See comments inline.
David Stude wrote:
Hi all,
I'm currently gearing up for a possible PBX replacement project using
Asterisk, and I'm just breaching the iceberg of information that's
available. I typically like to have something thick with pages in front
of me. Mahler's book was the first
Evening all,
Just got my first PRI got event: HDLC Abort (6) on Primary D-channel of
span 1 error message. Our production box has been up for ~2 month. We
are Asterisk 1.0.9 with Slackware 10.1. Now I have search the lists
from this message and hear all the problem. Everything from asterisk
Hello,
I think there maybe an issue with my refer transfers. See below or attached:
No. TimeSourceDestination Protocol Info
1 0.00192.168.1.2 192.168.1.5 SIP/SDP
Request: INVITE sip:[EMAIL PROTECTED], with session
Hello,
Quick Diagram:
Telco-PRI - Asterisk - Norstar PRI - Norstar PBX
(DMS100) (TE405P) (DMS100)
|
|
V
Cisco 7960G
(SIP)
I'm trying to change the Origination Number on my outgoing PRI, and running
into a weird
Hello list,
From time to time, I get the following warning in my message log.
Jun 23 15:56:40 WARNING[559]: PRI: XXX Missing handling for mandatory IE 12
(cs0, Connected
Number) XXX
Should I be concerned? To my knowledge I have not had an problems because of
it, but if
somebody can give me
Derrick,
Thanks for the ideas. I have since removed any USB/Firewire/un-needed
hardware from loading in the MOBO BIOS and recompiled the kernel to boot.
However I still seem to have the same problem.
Here is some more information.
# lsmod
Module Size Used byNot tainted
unknown unknown
GNU/Linux
Thanks inadvance,
---
Paul Belanger (mailto:[EMAIL PROTECTED])
Technical Support Specialist
Cisco Certified Network Associate
Pronexus Inc. - A Powerful Voice in Communication Solutions
---
Tel: 613.271.8989 ext. 516
---
Paul Belanger (mailto:[EMAIL PROTECTED])
Technical Support Specialist
Cisco Certified Network Associate
Pronexus Inc. - A Powerful Voice in Communication Solutions
---
Tel: 613.271.8989 ext. 516
Fax: 613.271.8388
http://support.pronexus.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Enable debugging to see the reason:
CLI sip debug
quote
A UAS rejecting an offer contained in an INVITE SHOULD return a 488 (Not
Acceptable Here) response. Such a response SHOULD include a Warning
header field value explaining why the offer was
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Ya, it has been a little slow for me today too.
PB
Matthew Boehm wrote:
| Hey guys, I sent an email to the list at 2:57PM central. I just now see it
| on the list, and its 3:23PM.
|
| Anyone else experience this? I am sending this email at 3:24PM
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello all,
I was looking for some information about using Asterisk to convert an
incoming H.323 call to and outgoing SIP call. Is this possible?
PB
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird -
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I have been successful in getting Digest authentication to work with my
Mitel 5055 IP Phones, however I'm wondering if Asterisk still supports
Simple authentication? I know it has been depreciated in the RFC, but I
have some phones with don't
: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Belanger
|Sent: Wednesday, January 12, 2005 1:06 PM
|To: asterisk-users@lists.digium.com
|Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
|
| Hello all,
|
| I was looking for some information about using Asterisk to convert
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