Still no good. Here is what I have now. It looks
like the problem is in my set and VoicemailMain statements.
exten = 8600,1,GotoIf($[${CALLERID(num)} = 2001]?50:2)
exten = 8600,2,GotoIf($[${CALLERID(num)} = 2002]?50:3)
exten = 8600,3,GotoIf($[${CALLERID(num)} = 2003]?50:4)
exten =
)
exten =
8600,51,VoicemailMain(${CALLERID(num)}|s)
exten = 8600,52,Hangup
From: Keith O'Brien
Sent: Sunday, September 24, 2006
12:11 PM
To:
'asterisk-users@lists.digium.com'
Subject: RE: 1.4 Beta 2 Config
Problem
Importance: High
Still no good. Here is what I have now. It looks
I just upgraded from 1.2.12.1 to 1.4 beta 2 and am having a problem
resolving an issue with the following configuration. The
logic below worked fine in 1.2 but seems to be broken in 1.4 beta 2. The
statements 50 and 51 dont seem to properly reassign the caller id to 2000
or some other 4
Still no good. Here is what I have now. It looks
like the problem is in my set and VoicemailMain statements.
exten = 8600,1,GotoIf($[${CALLERID(num)} = 2001]?50:2)
exten = 8600,2,GotoIf($[${CALLERID(num)} = 2002]?50:3)
exten = 8600,3,GotoIf($[${CALLERID(num)} = 2003]?50:4)
exten =
I am having some
trouble implementing OR login in the GotoIf statement. I have
followed the examples in the Wiki and I still am getting a syntax
error.
Essentially I want
to screen for CallerIDs set for "Anonymous" OR "Unknown Caller". If
either of these are true I want to send it to
Yup, that did the
trick. Thanks.
--
Hi! Try
exten = 5000,2,GotoIf($[$["${CALLERIDNAME}" =
"Anonymous"]|$["${CALLERIDNAME}" = "Unknown Caller"]]?3:5) intead of
exten = 5000,2,GotoIf($[$["${CALLERIDNAME}" = "Anonymous"] |
$["${CALLERIDNAME}" = "Unknown Caller"]]?3:5) Deleting spaces
Does anyone know if it is
possible to run both Asterisk and SER on the same box? I am
looking to use SER as the SIP proxy while sending SIP calls to a local Asterisk
processfor vmail.I am assuming that I would have
to change Asterisk from listening on 5060 to some other port to make this
If you are running a Cisco VoIP
gateway you can send a 0dBm 1000Khz test tone into or out of a voice port
with:
"test voice
portport#inject-tone network 1000hz"
to measure the tone do
a:
"sh call active voice
brief"
Another common problem that
causes echo in networks is not setting
Yes, if you hear echo then the
source of the echo is from the far end equipment. This is due to the
fact that delay is what makes echo noticeable. Think of yelling in a small
room. Your voice bounces off all of the walls and returns to your ears but
you probably don't hear any echo since
The Cisco part number for spare
handsets is CP-HANDSET= (list price is $15) You should be able
to purchase this from any Cisco Reseller or Cisco directly. To order
from Cisco direct:
http://www.cisco.com/en/US/partner/ordering/or13/or8/o25/ordering_ordering_tool_launch.html
or 1 800
553
I am pretty sure that there are
no IAX providers that offer CallerID name but wanted to double check with the
list in case something has changed recently. Is anyone aware of an
IAX provider that offers incoming CallerID name?
Is there a technical limitation
within IAX which is preventing
I suspect that what you are hearing is the VAD/silence
suppression kicking in and out. Unfortunately, I have the same complaints
from my users and I have been unable to determine a way to disable silence
suppression. VAD=no seems to have no effect in IAX. Also running CVS head
from about 2
Yes, but the 7970 relies on
the very latest SCCP protocol version which was introduced with Call Manager
4.0. As far as I know the SCCP module for Asterisk doesnt support the
latest SCCP version so I dont think it will work. The 7960 works with
the older versions of SCCP. I have yet to
EM is analogue, not digital...
Not true. You can have EM signaling on a T1
CAS interface. Likewise T1 CAS interfaces can also be setup for FXS
and FXO signaling. This is just how the robbed bits
communicate. With EM wink before a remote switch sends a call
to a local switch it send a
As far as I am aware there isnt a way for * to
receive/send audio to a multicast group. There needs to be a way
for Asterisk to tell the phone which ip multicast group to join in order to
receive the page. This method varies by vendor. I know that
with Cisco ip phone multicast paging
Has anyone figured out how to power a Digium TDM 400P card
in a Dell 1750 server? I opened the server and noticed that there is no
access to 4 pin power to power the card. Is there some sort of adapter that I
need to power the Digium card in a Dell Server? I see that the 1750 is listed
on
Yes, in most cases
this would work. Unfortunately, this wont work as
there are no 4 pin outputs from the Dell power supply to attach the
splitter. Dell rack mount
servers dont use 4pin power at all. The
hard disks are hot swap scsi so they dont use 4 pin power.
Also the power supplies are
Ok, makes sense. I obviously have one of the newer TDM400P
cards since it does have the power connector. However, I only plan on using
FXO ports. Will the card work without external power if there are only FXO
ports?
Thanks
Old TDM400P cards did not
have a power connector. The
Essentially
its because * has been architected to send an rtp packet after
receiving a packet. If * never see's and incoming rtp
packet, then it won't send an rtp packet (which usually contains some amount of
audio). Thus choppy audio in one direction.
So why cant * just play
It is trying to download its firmware. You need to setup a TFTP Server. Also be aware that the 7970 only supports SCCP not SIP. Further, the * implementation of SCCP doesnt support the latest version of SCCP which is required for the 7970. I dont see how it would work at all with *.Hi
It's
more than that, from what I know a *missing*
RTP packet could be
'silence'
(vad) or it could be 'network related' (jitter). * not seeing
a packet
doesn't always mean it was vad, it might mean your network had
a split
second (subsecond) hiccup that caused the packet to disappear
Does anyone know where I can find the Asterisk Python interpreter?
The one referenced here:
http://www.sineapps.com/news.php?rssid=173
It seems like the site
http://vox.groovy.net/moin/PyAsterisk
is down
___
Asterisk-Users
QOS for the PIX isnt available until the 7.0 release
which is currently in beta.
--
Hi ListJust a little bit OT, but then again perhaps an information that could be ofgreat value for a lot of administrators !!Does anyone have experience with how to setup VoIP QoS for
It doesnt have anything to do with VP Connect as the problem exists with other carriers that I have tried. Further, VP Connect is actually handling the call properly. In the test calls I am making, I am intentionally disabling callerid by hitting *67. VoicePulse correctly forwards the call
For what it is worth I am experiencing the exact same problem with the latest CVS. I have tried numerous IAX providers and the problem follows so it isnt the provider. Are you running stable or CVS?I think that you hit it on the head, the fact that audio is being sent prior to an IAX ANSWER
I have been having problems getting PrivacyManager to work
correctly. Right now I am running the 1/21/05 CVS but I have been unable to
get this to work on asterisk-stable either.
You can see from the debug below that the inbound call is
arriving via IAX2 and both the CALLING NUMBER and
We're trying to PQ (Priority Queue) packets on a Cisco using ACL's.
---
You do not want to use PQ for
voice QOS. You will still receive far too much
jitter. Instead configure LLQ which was specifically designed for
voice scheduling on an interface. Aside from
I have a similar
setup. To make it easy and get the best of both worlds, have the
Linux softphones (SIP or IAX) register to Asterisk. Keep the
physical phones registered to CM. From there setup a dialplan on
both Call Manager and Asterisk to relay calls between the two
systems. For
Thanks but I already tried their
knowledgebasethey dont have any configs for OpenAccess only
VoicePulse Connect. Also, tried calling their support and they were unable to
assist in getting * working with OpenAccess only VoicePulse Connect.
Does anyone have * working using SIP and
I did finally get through to
them. However, after troubleshooting for about ½ hour they were at a loss as
to why it wasnt working. I sent them my sip.conf file and
extensions.conf file and they didnt notice any problems.
What you are saying doesnt
make sense. VoicePulse Open Access
I highly suggest you work on
getting either the RTC or USB driver loaded
to
provide timing if you don't already have a PSTN card for that job.
I am having a similar problem. Can someone point me to the
procedure to install these virtual drivers for timing. I searched the wiki
but
The URL you are looking for
is:
http://www.voip-info.org/wiki-Asterisk+timer
Thanks. After reading through the notes I checked my server
(Dell 1750) and noted that it uses a USB OHCI interface so the first option
doesnt appear to be an option. Also it indicates that the second
spurts and I am assuming will also minimize
the impact of not having a ZAP timing source.
Is there a way to disable VAD in *?
Thanks again.
-Original Message-
From: Martin List-Petersen [mailto:[EMAIL PROTECTED]
Sent: Saturday, December 18, 2004 8:20 PM
To: Keith O'Brien
Cc: [EMAIL
http://www.voip-info.org/wiki-RTP+Silence+Suppression
http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html
So I am confused. The first says that VAD is supported in RTP.
Ok, I know that. The second is kinda ambiguous and seems to imply that * doesnt
support
I did something like this
with some IF logic and changed the callerid to the appropriate callerid for the
mailbox number. Granted I am sure that there is a more eloquent
approach to this but it works for me.
I am shopping around for an IAX provider that provides both
outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX.
http://teliax.com/
Does anyone have any experience with them? What codecs do
they support? Inband or 2833 dtmf? Voice quality and
I am having some problems getting TelIax service to
work with *. Outbound calls work just fine. When I try an inbound call the
phone rings and there is no audio. Upon further investigation iax2
show channels indicates that the codec is unknown The
provider confirmed that they are set for
If your vmail is connected
via serial to your PBX it is most likely using SMDI for MWI which isnt
supported by Asterisk. I seem to remember that this was submitted
as a feature request with a bounty tied to it.
Keith
Date: Tue, 7 Dec 2004
12:58:11 -0500
From: George Herndon
Granted I havent
really hammered on the Asterisk version of SCCP. With this being said, is the
SCCP project for Asterisk dead? Why is the Asterisk version of SCCP so
unstable and crappy? Is there any effort underway to improve the support of
SCCP for Asterisk? This would be great to be
No you dont have to use SIP. You can also use the SCCP channel on * with Cisco phones.
Message: 16
Date: Sat, 4 Dec 2004 12:45:53 +0200
From: "Walid Azab" [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and Cisco IP Phones
To: [EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type:
I am having a problem with the Vmail Web app and was hoping someone
has some advice. I have been able to install vmail.cgi.
I am able to bring up the vmail web page and login with out problems. However,
I never see any vmails listed even when there are vmails present in the
directory
/(\w+)\s+.*$/$1/;
Anyone have an idea what variable isnt
initialized??
From: Keith O'Brien
[mailto:[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004
3:06 PM
To: '[EMAIL PROTECTED]'
Subject: No Files Seen via vmail.cgi
I am having a problem with the Vmail Web app and was hoping
I am actually able to see
the drop down list without a problem. BTW, vmail.cgi is:
http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi
Should I submit a bug
request??
Keith
Message: 10
Date: Thu, 02 Dec 2004
16:08:30 -0500
From: Sean Cook
[EMAIL PROTECTED]
Subject: Re:
I get the following. Thanks for your help.
SCRIPT_NAME = /cgi-bin/vmail.cgi
SERVER_NAME = 10.1.1.9
HTTP_REFERER = http://10.1.1.9/cgi-bin/vmail.cgi
SERVER_ADMIN = [EMAIL PROTECTED]
HTTP_ACCEPT_ENCODING = gzip, deflate
HTTP_CONNECTION = Keep-Alive
REQUEST_METHOD = POST
CONTENT_LENGTH =
http://www.voip-info.org/wiki-SCCP-HOWTO2
-
Message: 1
Date: Wed, 1 Dec 2004
08:12:18 -0500
From: Rodney Acosta Coya
[EMAIL PROTECTED]
Subject: RE:
[Asterisk-Users] cisco 7902g
To: 'Asterisk Users Mailing
List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Message-ID:
Yes, you can use SCCP registration to * for 7902 support.
Date: Tue, 30 Nov 2004
11:28:08 -0500
From: Rodney Acosta Coya
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
cisco 7902g
To:
'[EMAIL PROTECTED]' [EMAIL PROTECTED], 'Asterisk
Users
Mailing List -
Non-Commercial Discussion'
I am assuming that the TOS values directly map to DSCP
values in the ip header. Is this a correct assumption? If so, can someone
tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media
with EF(101110)? So would the setting for AF be TOS=46?? Is it
I am assuming that the TOS values directly map to DSCP
values in the ip header. Is this a correct assumption? If so, can someone
tell me the correct setting to set call control packets with a DSCP of AF31(011010) and media
with EF(101110)? So would the setting for AF be TOS=46?? Is it
I would like to use * as a standalone voicemail system. As
such I need it to be able to outdial a certain extension for MWI-ON and another
extension for MWI-OFF
Is there anyway to get * to automatically dial an extension
when a voicemail is left and another extension when the mailbox is
Also, one other item. I need
to also set the CallerID of the outgoing call to the extension of the mailbox.
Thanks
From: Keith O'Brien
[mailto:[EMAIL PROTECTED]
Sent: Saturday, November 27, 2004
9:54 PM
To:
'[EMAIL PROTECTED]'
Subject: VoiceMail Outdial?
I would like
I am trying to get Cisco Unity Voicemail working with * and
am having a slight problem. Essentially the setup is as
follows:
Broadvoice---asterisk-Unity SIP VMail
Unity needs the following from * :
In the Diversion header, the
extension of the called party
In the Diversion
Yes, I have both Call Manager and Call Manager Express integrated with *.
Prior to Call Manager 4.0 you would need to perform an H.323 integration
with *. As of CM 4.0 Cisco supports SIP trunking so this would be the
preferred method of integration. This config is on http://www.voip-info.org
: Re: [Asterisk-Users] Google newsgroup or Forum setup.
Top-quoting. Argh.
On Monday 29 September 2003 12:16 pm, Keith O'Brien wrote:
I'll offer one better. Why don't we mirror all of the maillist
posts to a forum. That way both parties are happy. Those that want a
forum can use a forum
It doesn't matter. The session target in a cisco voip-dial peer always
has to point at the far end. In the case of H.323 this would still be the *
box. Pointing this at the local router eth0 interface definitely is not
correct.
-Original Message-
From: [EMAIL PROTECTED]
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