Hi,
I have a strange problem. I have a TE110p digium card.
I want to dial 19173995791 when any incoming call comes in. What is
happening is that when I dial 19173-995791. Asterisk picks up the first 5
digits assuming it is the extension and appends 212-85 (here in the
university most numbers
Eric ManxPower Wieling wrote:
What are the advantages of 9.x over the 8.x that I currently use?
I was about to ask the same question. What if my 8.x EC works just fine?
(Why expose yourself to the possibility that even the patched version
fails?)
-Stephen-
From: Arpit Mehta [EMAIL PROTECTED]
Date: Fri, 18 May 2007 02:31:22 -0400
Hi,
I have a strange problem. I have a TE110p digium card.
I want to dial 19173995791 when any incoming call comes in. What is
happening is that when I dial 19173-995791. Asterisk picks up the first 5
digits assuming it
Stephen Bosch wrote:
If you get dumbfounded responses ask to speak to someone in the
programming group (unless they are a tiny little phone company, they
will have one). If you open a ticket, it usually means they will
escalate the problem, even if the agent you are speaking with has no
idea
These are empty files. I've tried deleting this final lines and there's no
change...
_
Un amor, una aventura, compañía para un viaje. Regístrate gratis en MSN Amor
Amistad. http://match.msn.es/match/mt.cfm?pg=channeltcid=162349
Brian Capouch wrote:
Stephen Bosch wrote:
If you get dumbfounded responses ask to speak to someone in the
programming group (unless they are a tiny little phone company, they
will have one). If you open a ticket, it usually means they will
escalate the problem, even if the agent you are
Hi,
hi,
19173995791 is some number which I want to dial. 212-85- all/most of the
numbers in my workplace start with this - so I presume it has got to do
something with this.
thanks for your suggestions
regards
arpit
On 5/18/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Arpit Mehta [EMAIL
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]:
Hi all.
We have Snom phones which do have a defined key in order to drop incoming
call WITHOUT answering.
Pressing that key, a SIP/2.0 486 Busy Here message is sent back.
We have other phones (I.E. DECT Siemens C450IP,
I'm no Asterisk expert (nor Linux expert) but I am using my * box for
multiple things (transparent firewall, NAT box, samba server, poptop server)
and for a considerable time I've been running a VmWare server with a Windows XP
virtual machine up-and-running at all times! The Windows XP VM was
17 maj 2007 kl. 02.57 skrev Robert Lister:
On Wed, May 16, 2007 at 05:46:21PM -0500, Greg Oliver wrote:
If I push the response code back to the handset (Cisco 7960) then
it is even
more unhelpful as it uses the same error message for all SIP
error type
response codes: Reorder but does not
Stephen Bosch wrote:
And what do you do when they say:
We have a modern, relatively-new switch for which that sort of feature
change is a trivial click on a GUI checkbox. However, we do not have
any tariffed requirement to provide disconnect supervision. So we won't
do it for you. Goodbye.
Hi
some knows one good EUA voip provider with land lines?
thanks
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Rosli Sukri wrote:
Hi,
Has anyone on this list tested out the new SE P1 phones
(http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says
it supports VOIP, wonder if it is working with asterisk.
Nice phone, wondering what it will cost when it gets released. So to
answer your
On Thu, 17 May 2007, Vincent Delporte wrote:
Hi
To investigate the UNREACHABLE issue I'm having, I need to have
confirmation that it's OK for the Asterisk server to be behind a NAT router,
and also have clients elsewhere on the Net behind their own NAT router?
Yes, it's OK...
I know that
There is a creation for Master.csv in /var/log/asterisk/cdr-csv and its
filled with data ,but there is no pushing to mysql,asteriskcdrdb table cdr
How or what is the procedure to let the data enters mysql .
Regards
-Original Message-
From: [EMAIL PROTECTED]
2007/5/16, Stephen Bosch [EMAIL PROTECTED]:
As you mentioned, try implementing it only on
certain channels, if you can.
-Stephen-
Hi,
Unfortunately, I understood you couldn't allocate HPEC on per channel
basis.
But anyway, I agree we have to try something.
Regards
SD == Stephen Davies [EMAIL PROTECTED] writes:
SD Hi, I want to quickly mention that I've had great success with
SD running Asterisk in the under-appreciated Linux-VServer
SD environment.
I just want to do an AOL here: me too! Linux-vserver is great for
asterisk, although we will probably
Anyone? some debugging help would be good otherwise I'm in the process of
building a new asterisk box and hoping that getting away from Trixbox will help
me
CentOS 4.4
Asterisk 1.2.18
Zaptel 1.2.17
X100p (until I'm happy to swop the box into production then I will use my
Sangoma A200)
Hi,
I don't want multiple instances of Asterisk. My goal here is to make
Asterisk and its Zaptel hardware run nice on a machine that is not
dedicated and also hosts VM. I had lot's of problems with Xen, as the
host runs a modified kernel that has apparently issues with the
interrupt handling
I will try VMWare then. Does it support AMD-V or Intel VT virtualization
at processor level for speedup? If so, I guess there is at some point
some kernel code (module like KVM or special kernel like Xen) that could
provoke some similar problems with zaptel interrupts. Could there be
some
lenz wrote:
I would try one of the two things:
1. adding a hint for the Local/[EMAIL PROTECTED] channels
2. using the = for queue members
member = Agent/1001
member = Agent/1002
member = Agent/1003
Does this change anything?
Hi Lenz
thanks for your suggestions.
I tried them both
Hi,
I am using asterisk 1.4 with Digium TDM22B card. My system is running well
except CALLERID. I have tried all options for cidsignalling, cidstart but no
luck.
Btw, I am living in Malaysia. From google web site, I found some one having
the same problem with new version of asterisk but not in
Can someone help me use Asterisk to provide SIP trunking to another PBX?
I'd like to insert an Asterisk platform between a network PRI and an
existing PBX. Incoming calls would (based on DNIS in a SQL table) go to PBX
A on a PRI channel or PBX B as SIP. Outgoing calls from either PBX would
select
Hi,
Someone who has had experience with shoretel VoIP systems, can you please
give me a run down of how Asterisk is either better or worse? I am
completely unfamiliar with Shoretel systems, but someone had suggested we
look into them. I said, you bring your Shoretel features, and I'll show you
Hello Aslay,
In some country, this feature is a paid option from the TELCO side.
In France the analog lines have not this feature enabled in standard, only
the digital lines .
Are you sure that it's actualy available in your case ?
Best Regards,
Francois BERGERET,
France.
-Message
Hi all,
There is a case in which the call counter is not set to zero for a sip peer
(incoming call). Here is the scenario.
Dialplan:
exten= 1,1,Dial(SIP/U1)
exten= 1,2,Gotoif($[${DIALSTATUS}=ANSWERED]?:10,1)
exten= 1,3,Hangup
exten= 10,1,Voicemail()
If a user just registered with my asterisk
After spending a couple of hours on the phone with Digium support,
monkeying with battery threshold and several other settings, we figured
out that the line was too noisy. I worked with our local telco to
resolve that, and it's fixed!
Thanks for all the suggestions; I'm learning *so* much about
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP
addresse for a few seconds, audio goes blank obviously, and after about
30-60 seconds get the same IP addresse back and resumes the call. This shows
that call was not dropped but phone lost connection with the server,
Does it loose it's IP address consistently and at a designated
interval. I've seen something similar in a number of various cases
where it was always an issue
of the 'client' device blocking the DHCP renew traffic. But when it
went into rebinding it would drop $filter and allow the dhcp
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP
addresse for a few seconds, audio goes blank obviously, and after about 30-60
seconds get the same IP addresse back and resumes the call. This shows that
call
was not dropped but phone lost connection with the server,
Brian Capouch wrote:
Stephen Bosch wrote:
And what do you do when they say:
We have a modern, relatively-new switch for which that sort of feature
change is a trivial click on a GUI checkbox. However, we do not have
any tariffed requirement to provide disconnect supervision. So we won't
As others have said, it does sound like a DHCP issue.. you can try
increasing the lease time, or giving it a static IP address.
On 5/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP
addresse for a few seconds, audio
I'm kind of curious on this one myself. Any direct comparisons online? I
wasn't really paying much attention to them until they announced the
Salesforce.com tie-up.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
Zeeshan Zakaria wrote:
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses its
IP addresse for a few seconds, audio goes blank obviously, and after
about 30-60 seconds get the same IP addresse back and
The phone is probably renewing it lease with the DHCP. Set your
Quoting Doug Lytle [EMAIL PROTECTED]:
Zeeshan Zakaria wrote:
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses
its IP addresse for a few seconds, audio goes blank obviously, and
after about 30-60 seconds get the same IP addresse back and
The phone is probably
On 5/18/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
Hi,
Recently I've noticed on a customer's GXP-2000 phone that it loses its IP
addresse for a few seconds, audio goes blank obviously, and after about
30-60 seconds get the same IP addresse back and resumes the call. This shows
that call was
Hi,
I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet
card (with ports in PTP mode). I noticed a long delay when making
outbound calls, more precisely between (taken from Asterisk CLI)
Called 1/X/s and mISDN/1-u43 is proceeding passing it to
SIP/8-5486
I
If I use Asterisk to initiate two call legs with a callfile, dialing
the channel and setting the extension to an AGI that dials another
channel, and both dial by SIP connection to a switch that allows only
G.729, do I need a G.729 codec running on Asterisk? Do I need 2?
And if I
1 of our customers records all phone calls and needs to be able to be
played back via a searchable web app. I tried ARI but it is very
limited.
Anyone have any ideas?
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
(Note: resending with proper Subject)
If I use Asterisk to initiate two call legs with a callfile, dialing
the channel and setting the extension to an AGI that dials another
channel, and both dial by SIP connection to a switch that allows only
G.729, do I need a G.729 codec running on
Olivier wrote:
2007/5/16, Stephen Bosch [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
As you mentioned, try implementing it only on
certain channels, if you can.
-Stephen-
Hi,
Unfortunately, I understood you couldn't allocate HPEC on per channel
basis.
But anyway, I
MessageHello,
My TELCO already enable the service. I also verify by connectiong the PSTN line
direct
to a phone with LCD display. I can see the caller number displayed on the LCD
screen.
I believe that the root problem is the setting on the zapata.conf
(cidsignalling, cidstart), which I am
Hi,
I'm trying to get a TE212 working on a Dell PowerEdge 1850 running
Debian etch using the latest release of libpri (1.4.0), zaptel (1.4.2.1)
and asterisk (1.4.4). The initilization of the Octasic echo canceller
seems to fail when the wct4xxp module is loaded.
[...]
VPM450: echo cancellation
Anselm Martin Hoffmeister wrote:
Am Donnerstag, den 17.05.2007, 10:40 +0200 schrieb [EMAIL PROTECTED]:
Hi all.
We have Snom phones which do have a defined key in order to drop incoming
call WITHOUT answering.
Pressing that key, a SIP/2.0 486 Busy Here message is sent back.
We have other
Thanks for all the replies. I've updated client's router firmware, which was
very old. Now I've put MoH on for the whole day and see if it happens again.
MoH was doing the same thing, i.e. going down and coming back up. When it
went down, on the screen you could see message saying 'No IP', and
Hi Eric,
We used a quick hack to Asterisk Queue/CDR Log Analyzer
(http://www.micpc.com/qloganalyzer/) to turn the Uniqueid field into a
hyperlink to the .WAV file for the call. We only use this internally for
our call centre, it is not customer ready as all call details are
revealed and not
13 apr 2007 kl. 16.45 skrev Brian Jones:
I've encountered a similar problem with Cisco equipment. The Cisco
proxy was not replying to Asterisk with an ACK after * sent an OK.
Since version 1.2.14, * was changed so that not receiving an ACK to
an OK is considered a FATAL error.
The
On 5/18/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Do these sets not have a Do not disturb button?
For sets that do, what is the behaviour when one presses the DND button
I have some Aastra's (a mixture of 480i's, 9112i's, and 9133i's) and
whenever I program a key as do-not-disturb, I can
Hello every body,
kindly i make one phone which is in java applet and there is no generate any
DTMF signal at client side only beep tones is hearing but not generate DTMF
at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
Thank you.
On May 18, 2007, at 9:53 AM, Francois Deppierraz wrote:
Hi,
I'm trying to get a TE212 working on a Dell PowerEdge 1850 running
Debian etch using the latest release of libpri (1.4.0), zaptel
(1.4.2.1)
and asterisk (1.4.4). The initilization of the Octasic echo canceller
seems to fail when
Physical connection is fine because customer's computer is also connected
through the phone's inline ethernet port. When phone was losing IP, computer
was still working fine on the Internet.
But after the changes I made earlier today, as mentioned previously, it
seems to be working fine so far.
Hi,
Sounds like a flaky physical connection to me.
Try new patch cables (DO NOT EVER use homemade patch cables, they cost
too much) and a different jack if in an office environment.
If that doesn't fix it, try locking the switch port and/or the phone
port speed/duplex settings and turn off
You've got 2 possibilities :
1. If you want something which is channel or phone type independant and
works for analog phones, for instance,
you've got to use AMI to tell Asterisk to hangup this specific call.
2. If you accept something which depends on phone type, you've got to look
in phone
I believe this is a Asterisk console bug, but thought I would run it through here first. I can get Asterisk into a tight loop 100% of the time. Here is what I know...
First I have verbosity set to 20 in the asterisk.conf file. I believe that has some bearing on this issue for two reasons. One it
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if
there's no dial-tone, it doesn't seem to detect this?
Using a TDM400 with UK settings.
Steve
--
NetTek
gaurang sheladiya wrote:
Hello every body,
kindly i make one phone which is in java applet and there is no generate
any DTMF signal at client side only beep tones is hearing but not
generate DTMF at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
gaurang sheladiya wrote:
Hello every body,
kindly i make one phone which is in java applet and there is no generate
any DTMF signal at client side only beep tones is hearing but not
generate DTMF at the back end side.
so plz if anyone know that DTMF generation proccess then plz reply me.
On May 18, 2007, at 11:50 AM, Steve Kennedy wrote:
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if
there's no dial-tone, it doesn't seem to detect this?
On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote:
On May 18, 2007, at 11:50 AM, Steve Kennedy wrote:
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE) it
Can anyone recommend a 4-port ATA (preferably IAX2, but SIP would be okay).
TIA for any recommendations/experience.
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Steve Kennedy wrote:
I'm trying to get zap fallback to VoIP working. I dial the zap channel
and if it fails I want to then try another route.
If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if
there's no dial-tone, it doesn't seem to detect this?
Using a TDM400 with UK
Lol - I think everyone starts on budgetones then moves to ciscos or
polycoms etc.
BTW I have 2 x 3line IP500's that I'm about to ebay this weekend if
anyone in NY or near Manhattan wants to buy them.
Just upgraded to the 6 line versions.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL
I fought this for a bit when I found if the file Master.csv didn't exist, it
wouldn't create it on it's own. I created an empty csv file, CDR started
writing.
Ken
From: [EMAIL PROTECTED] on behalf of Khaled Chehab
Sent: Thu 5/17/2007 10:50 AM
To: [EMAIL
i would force a timer on it..
dial (blah,30)
maybe that would bypass , maybe not..
i actually think it wont..
another example of this problem is DNS
echo '1.2.3.4your.favorite.itsp' /etc/hosts
then Dial(SIP/[EMAIL PROTECTED])
DNS failing will BLOCK the call indefinitely...
On
Do you have any specific experience with astmanproxy?
Can anyone give me an idea on number of simultaneous connections this
can legitimately handle with ease?
This has been around for a while by the looks of it but I haven't heard
about it before.
Regards,
Dean Collins
Cognation Pty Ltd
Hi all,
I was wondering if there is any IAX2 sniffer and decoder. Wireshark
can decode and play RTP streams using G.711, and Cain Abel decodes
and plays any kind of RTP stream. But I didn't find anyone that can
decode IAX2 streams.
Any programs??
Regards,
--
Diego Quintana a.k.a. RouterMaN
Matt Florell wrote:
On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have 50-100+
clients connected to the AMI on full output mode. Running a system
On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have 50-100+
clients
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, 18 May 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk manager interface stability
Thank you for the quick response. Do I need to create a route to the
other machine? like a trunk?
greetz, Tim
2007/5/17, JR Richardson [EMAIL PROTECTED]:
[mappings]
priv = dundi-priv-canonical,0,SIP,${IPADDR}/${NUMBER},nopartial
priv =
Is there a way to know who picked up a call using *8? A customer wants
to know if someone is picking up their calls when they are not at their
desk.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Use the cdr's, who wont know who but at least which phone did it.
Carlos Chavez wrote:
Is there a way to know who picked up a call using *8? A customer wants
to know if someone is picking up their calls when they are not at their
desk.
On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, 18 May 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Sweet thanks Matt.
If there are any developers in Manhattan (or very nearby) who have
experience with Astproxy and are looking for sweat equity ownership in a
new Asterisk application get in touch. Also looking for someone with ROR
UI skills but I might already have that role filled.
Cristian López F.
Integración y Tecnología - Terra Chile
Phone: (56 2) 330 6966 movil: 56-92401759
E-mail: [EMAIL PROTECTED]
Este correo y su contenido solamente interesan a las personas autorizadas
de TERRA NETWORKS CHILE.
Si usted fue receptor de este correo por error, por favor no lo tome
Disclaimer at the bottom still looks ridiculous even in Spanish... LOL
Wiley E. Siler
Director of Information Technology
4110 N. Scottsdale Rd. Ste 110
Scottsdale, Arizona 85251
(480) 296.0260
(866) 320.2083 ext. 1003
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
www.education2020.com
On Fri, 2007-05-18 at 16:42 -0600, Anthony Francis wrote:
Use the cdr's, who wont know who but at least which phone did it.
I tried following the CDR but if I dial extension 4000 and extension
4002 picks up the call using *8 the CDR says that extension 4000
ANSWERED the call. It does
Matt Florell wrote:
On 5/18/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
On 5/16/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Matt Florell wrote:
The issue has more to do with the sheer amount of data passed to the
client from within the Asterisk application when you have
hi there!
I have a couple phones connected to a sipura ata and
if I go into *- IVR, I press options on the regular
phones and it all works fine and dandy.
then I connect an xten softphone, a new extension in
my dialplan, I dial the ivr, * asks me to dial
something to go through it, I press keys
On Fri, May 11, 2007 at 11:06:35AM -0400, Mike wrote:
Hi,
I have a question of using 2 SIP providers. Let's say I have provider A and
provider B, and I would like my calls to go to A, and then B if A wasn`t
available
What would be really cool, but require special code in the chan_sip
Matt Florell wrote:
On 5/18/07, Dean Collins [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, 18 May 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Dean Collins wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt Florell
Sent: Friday, 18 May 2007 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk manager interface
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