That's my fault in the example - I forgot to add in the j. Anyway what
is strange is that I get my dialplan to jump to position 108, but at that
point the agent is disconnected. I thought that when falling out of the
queuetransfer context, the control would be returned to the trasferer,
Hi,
some time ago we used bristuffed Asterisk for our hfc cards cause it offered
more features (echo cancellation most important) and was quite stable...
I'm seeing now (I'm putting together Asterisk after a long time with hfc
card) that there are now 3 choices for hfc chipsets :
vISDN,
Hi. Could you help me with this warning?
channel_find_locked: Avoided initial deadlock for '0x8218ac0', 10 retries!]
I have no ideea what causes it...
It seems to appear only when i make a call...
___
--Bandwidth and Colocation provided by Easynews.com
On 08:55, Mon 02 Oct 06, Robert Rozman wrote:
Hi,
some time ago we used bristuffed Asterisk for our hfc cards cause it
offered more features (echo cancellation most important) and was quite
stable...
I'm seeing now (I'm putting together Asterisk after a long time with hfc
card) that
I've got the same question actually.
We're looking to replace CCM with * (finally.. it took me ages to
convince that * is way better), but we've got cisco 1700 2600
gateway's for the CCM in our remote offices that would have to be used
by SIP with * now.
Did anyone ever encounter or set up such
We've been using cisco 2600 gateways with asterisk for a year and
everything works fine. IOS 12.2 is installed in gateways.
-Original Message-
From: Tijl Van den Broeck [mailto:[EMAIL PROTECTED]
Sent: Monday, October 02, 2006 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
antonioInviato: sabato 30 settembre 2006 17.27A:
asterisk-users@lists.digium.comOggetto: Re: [asterisk-users] Sip
answer one side , ring other side
when i make the call ,
on the xlite side i see the call connected but for
Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones.asterisk (te110p) -- (TMS2) hipath 3550 in the future i'll connect the hipath to a telecom pri.
the pri in the hipath is
Look at your
extensionsincontext "from-zaptel" adding the s extensionsand
add immediate=yes in zapata.conf
Ciao
Da: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di
bivioInviato: lunedì 2 ottobre 2006 10.04A:
asterisk-users@lists.digium.comOggetto: [asterisk-users] Siemens
does it solve the problem with j option?Do you have autofallthrough=yes in your general section of extensions.conf ?autofallthrough: New in 1.2. From the sample extensions.conf:
If autofallthrough is set, then if an extension runs out of things to
do, it will terminate the call with BUSY,
please post your from-zaptel context in extensions.confOn 10/2/06, bivio [EMAIL PROTECTED] wrote:
Hi all,i've an hipath conneted to my asterisk box by a TE110P i can call from astersik to any hipath extension but i can't call from hipath extensions to astersik ones.
asterisk (te110p) -- (TMS2)
I want to interconnect asterisk to a siemens HiQ20 which is configured as
gatekeeker. The problem is that the HiQ20 does not accept gatekeeperrequests
and sends immediately a reject with an undefinedReason. Is there a way to
get asterisk-oh323 to skip this request?
asterisk v1.2.12.1
Hello
I connect HICOM to
Asterisk
Zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-isdn-external
;signalling=fxs_ks
;rxwink=300 ; Atlas seems to use long (250ms) winks
signalling=pri_cpe
switchtype=euroisdn
I have to setup a pbx system for a company, can someone suggest a configuration. Currently their phone bill is 1600 a monthCurrenlty 27 phone lines1/2of the calls are long distanceI'd like the savings of a voip network, but also the reliability of a pstn/pri.
How low will
2006/10/2, Marco Mouta [EMAIL PROTECTED]:
please post your from-zaptel context in extensions.confThanks to Giordano (immediate=yes) i see the first improvement now i hear the asterisk voice who says the number you digited is not in use, please check
now i'm tryng to understand how to andle the
Try to set
overlapdial=yes in your zapata, so thta whenu access to line ushould
have somethinghs of this
-- Starting simple switch on
'Zap/5-1' -- Accepting overlap voice call from '405' to
'unspecified' on channel 0/2, span 2
at this point u
can ear a continuos tone and input your dnid
try this:first: immediate=no ; otherwise what you r saying is to asterisk automatically dial when you hook up the phone!The problem is that in your context from-zaptel you are not dialing anywhere!i couldn't find you using any Dial(...)
That's why it doesn't work! try this:[from-zaptel]exten=
We were also looking at this telecomproblem as well. A major complication,with regards to recovery planning,lies in the manner in which Local Loop Unbundling occurs. Even though communication companies may carry different logos, and profess to be independent orgs, they are all/mostly, invariably,
whoa!! it works the fault was my ignorance of extension.confi modified the Marco advice in:[from-zaptel]exten= _X.,1,Dial(SIP/${EXTEN})exten= _X.,2,hangupso i can correctly call the astersik extension.
many many thanks allBivio
___
--Bandwidth and
Matthew Thompson wrote:
yum install kernel-devel
Should do the trick.
It did, thanks.
Jim.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I set up extension 120 for the first and 121 for the second. The first
one works as expected but I can't get a dial tone on the second one. I
hear a buzzing in the second port much like the first, but no dial
tone. I have power since the dtmf keys work OK. I tried changing the
exten =
Hi all
I'm having a problem getting usable quality from my Asterisk setup.
*SETUP*
2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones
in the house and 2 x FXO to receive calls and in the future faxes.
Gentoo Linux
Here is what I've done so far
(1) Moved theTDM 400p (FXS,
I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write:
Asked to transmit frame type 8, while native formats is 1024 (read/write
= 1024/1024)', where 8 = alaw and 1024 = ilbc.
If I do show translation I get this:
*CLI show translation
Translation times between formats (in
Hello,
I'm using the ChanSpy command for monitor a conversation of a channel which is
in a meetme conference. All comunications go throught voip, with some voip
phones attached to the lan and an external voip providor in order to make
external calls.
The problem is that sometimes the spy call
Is there really nobody who has any idea about this?help would be really apreciated, as otherwise we're forced to buy a conventional pbxDate: 29.09.2006 15:33Subject: attended transfer unreliable
To: asterisk-users@lists.digium.comHi,running asterisk 1.2.9 with freepbx 2.1.1, I have
Stefan Friedrich wrote:
Is there really nobody who has any idea about this?
help would be really apreciated, as otherwise we're forced to buy a
conventional pbx
Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN? There have
been a few fixes in the branch that may help.
You can
This is strange. I upgraded from an older [EMAIL PROTECTED] that was
working to the latest Tribox. I also added a A204 board, but for some
reason neither the Grandstream phone or a phone connected to the Linksys
ATA has any audio either way via the Telasip connection. Audio works OK
between
Hello,
I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbers
also via SIP through an AS5350 which has an E1 ISN PRI attached.
I have a PSTN operator number (say 012345678) routed to three SIP
extensions
Doug Lytle wrote:
Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN?
Transfer (rather, dynamic features in general) is broken in 1.2.12.1:
http://bugs.digium.com/view.php?id=7982
So you should try the version from the SVN branch.
Yours, Florian.
Hi,I am trying to integrate asterisk queues with SER.We have our queues set up in the following manner:An entry in the queue members table consts of the queue name and a SIP address.For example,queuename | member
-support-q | SIP/5558675309We have observed
1. Reset to factory defaults
2. Put registration information under Global SIP and not line 1
3. Put in IP address of Asterisk server in every field that says Proxy
4. THE TRICK: Phone number field in Global SIP must have account name, not
actual phone number.
Working awesome so far, thanks Dave
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06, Luca Corti
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
(Polycom, Linksys, Grandstream, Snom, etc.). I dial public PSTN numbersalso via
My mistake sorry for last postOn 10/2/06, Marco Mouta [EMAIL PROTECTED] wrote:
when you want to dial something via ZAP interface (to PSTN world) you should use dial(ZAP/)On 10/2/06,
Luca Corti
[EMAIL PROTECTED] wrote:Hello,I' using asterisk as a PBX for a dozen of SIP phones of various makes
First impressions:
1. Audio is decent on the cordless, not as good as a Panasonic but quite
usable.
2. Cordless range is awesome. Went 150 metres away, 2 buildings over,
through probably a dozen walls. No problem.
3. Possible for handset to independently originate and terminate calls while
base
I've got an interesting situation where I am running Asterisk 1.2.10 with
the chan_sccp2 implementation. The system crashes periodically, and each
time I get similar looking results when using gdb on the core files.
It looks almost as if someone is transferring a call to someone's voicemail
and
On Mon, Oct 02, 2006 at 10:43:44AM +0800, Steve Underwood wrote:
Jay R. Ashworth wrote:
On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote:
Well, fax detection isn't entirely reliable anyway. Even if you assume
that your fax detection feature and operation is flawless in properly
On Mon, Oct 02, 2006 at 01:33:43PM +1000, RR wrote:
does anyone happen to know of a good utility or CLI tool to convert
prompts into a g.726 format? I tried using the convert utility in (*)
but it doens't like G.726. I understand I can just hunt around the net
for it, but if someone knows one
On 9/30/06, Tim Panton [EMAIL PROTECTED] wrote:
On 29 Sep 2006, at 19:20, Yu Safin wrote:
Hi, I am a salesman currently using asterisk to contact my customers.
So far, I have asterisk connected to two PSTN analog lines where I
only receive phones calls.
Then, I have asterisk connected to a
Jay R. Ashworth wrote:
My personal experience is that I've never seen a consumer-grade fax
machine with send-CNG turned off, and I don't *think* I've ever seen
one on which there was a knob *to* turn it off; I would be less sure
about fax modems -- those may have a knob, but I would expect it
I'm confused with something, maybe someone can explain to me. if your currently on a pri and are considering moving over to VOIP, that means you would have to purchase a t1 or fractional t1 for a your voip connections. but then, voip connections aren't as reliable as PRI. so then you would
stan ford wrote:
I'm confused with something, maybe someone can explain to me.
if your currently on a pri and are considering moving over to VOIP,
that means you would have to purchase a t1 or fractional t1 for a your
voip connections.
but then, voip connections aren't as reliable as PRI.
Hi guys,
I'm tryng to estabilish a trunk with an Alcatel 4200 Pabx. In asterisk i'm using a Digium TE110P as E1.
The span is ok with green led, but when pabx make calls to asterisk, i received this error:
asterisk*CLI
!! Unexpected Channel selection 3
-- Accepting call from '3069' to
If
reliability is the issue, then use the PRI *first* then failover to
VoIP.
If
cost savings are the issue, use VoIP then have a 2nd VoIP provider to fail over
to, and no PRI.
In
either scenario, inbound call routing is thorny, some guys that provide both PRI
and VoIP can route calls
Hello, I have a problem with an adit 600 and a T400P card. This equipment was in a shelf for 2 years and when we connected an install it asterisk everything worked fine. But then we started receiving complaints that a person pick up their phone and will hear some other conversation. It happens
I set up mine with the web interface but I notice that some settings can
only be made by config files. Do you know how to extract the current
config file from the phone?
Here's how I set up the web interface:
Authentication Name: aastra480_1
Password: password
BLA Number: blank
Line Mode:
On 2006-10-02 04:02:56 -0700, James Harper
[EMAIL PROTECTED] said:
I'm getting messages like 'WARNING[10263]: chan_sip.c:2552 sip_write:
Asked to transmit frame type 8, while native formats is 1024 (read/write
= 1024/1024)', where 8 = alaw and 1024 = ilbc.
If I do show translation I get this:
Sorry! I think 1.2.12 had the bug I was referring to.
Marty
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-Original Message-
From: Douglas Garstang
Sent: Friday, September 29, 2006 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Queue AddQueueMember()
-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent:
On 2006-10-01 05:28:24 -0700, Andy Green [EMAIL PROTECTED] said:
Hello,
Can anyone point me in the right direction to source a WiFi SIP handset =
that
can also connect to a Bluetooth headset.
I have a requirement for a hands free warehouse/distribution centre =
setup
using such devices and
On Mon, Oct 02, 2006 at 01:14:45PM -0400, Steve Glaus wrote:
stan ford wrote:
I'm confused with something, maybe someone can explain to me.
if your currently on a pri and are considering moving over to VOIP,
that means you would have to purchase a t1 or fractional t1 for a your
voip
Hi group,Can anyone help out in selecting the right codec to download from the digium site.Im using an AMD Sempron 2800+ CPU speed 1.6 GhzThanks in advanceDan
___
--Bandwidth and Colocation provided by Easynews.com --
Thanks Marco! I found NVFaxDetect before getting around to your post. It
works a treat! Good call! no pun intended
Marco Mouta
[EMAIL
How
does one do this?
Thanks,
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
It all boils down to this: If they don't send a tone I won't get the fax.
Its like my email now with DNS blacklists enabled. If they have a dial-up
ADSL account they can't send me mail as my server denied them. Different
technology, same problem. Whatever they invent next will, more than
I wouldn't first presume that there is any law that states you have to run VoIP over a PRI. Other technologies exist such as SDSL - with some providers speaking of crazy prices such as £65pcm for2Mbs5:1 contention ration and £100 for 1:1 (U.K) - should be even less in the states if that's where
if i went with an SDSL line, don't those lines hook up to a common point, the DSLAM?i do like this idea of faling over not to a pri but another cheaper high speed line.adebayo omo-dare [EMAIL PROTECTED] wrote:I wouldn't first presume that there is any law that states you have to run VoIP
[EMAIL PROTECTED] wrote:
It all boils down to this: If they don't send a tone I won't get the fax.
And I certainly understand this approach. However, there are some
situations where this is simply not suitable - where missing a fax costs
money. Take, for example, the real estate
I am trying to set a minimum expiry time. I have the latest trixbox
installed and I have added minexpiry=60 in sip.conf. However my sniffer
shows my client can still get any expiry that it wants without getting
any rejection the server
I tried to apply a patch I fond from digum but it is
You can trick their machine into sending tones. The following code with
send tones that a terminating fax machine would normally respond with.
This will even force really old G2 fax machines to respond.
indications.conf:
faxrec = !2100/2600,!0/10,!1850/2600
[custom-fax-did]
exten =
Have you tried
passwd-amp or passwd-maint from the command line?
George
On 9/30/06, Tom Vile [EMAIL PROTECTED] wrote:
login as root and type help-aah and you will see a list of commands to
change the admin password.
On 9/30/06, Jim Lynch [EMAIL PROTECTED] wrote:
I'm looking for the
On Sun, Oct 01, 2006 at 03:55:45PM +0100, Marco Mouta wrote:
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)
Are you sure that your invalid context is correctly written?
I've never heard about this pattern match _X!
As far as i know the wild card
I have my extensions.conf set up as follows:
exten = _Z.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07862nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten =
Hello,
I am trying to run Tormenta 2 Quad E1 (non-Digium clone) card on one of my
asterisk box. I don't know why the card is not taking any interrupts:
CPU0 CPU1
0:10996711067142IO-APIC-edge timer
1: 9 0IO-APIC-edge i8042
8: 1
Interesting trick!
On the down side, won't sending this tone be pointless? If the receiver is
not sure a fax is calling, then he will BEEP every caller (even voice
calls). If the receiver is sure a fax is calling, why play the tones?
MD
-Original Message-
From: [EMAIL PROTECTED]
Hello - I have a small PBX setup for testing, and have put two small business accounts (from within my organization) on the PBX to see how things work out. I have two trunks two outbound routes setup and am using Teliax as my ITSP (two accounts; one for each account (different billing)). Outgoing
Try running the echo test from both the house side and the co (outside)
side. That will let us know where the problem is.
Post results.
Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Monday, October 02,
I've only used this on dedicated fax numbers.
I noticed that some fax machines didn't send tones and Asterisk didn't
detect.
They just sat there and looked at each other.
After playing the tones, the fax machines started sending and it worked.
James Taylor
www.metrotel.net
- Original
I updated a batch of Polycom IP501 phones and an IP601 to the 2.0.1
firmware to get the new NAT keep-alive feature and the ability to watch
more than a handful of buddy contacts but it appears to have broken the
buddy-watch feature. Is anyone seeing this? Anybody know if it's a
Polycom problem
If you don't need the Bluetooth bit then I can recommend someone to you.
Drop me an email for more info
-Original Message-
From: Martin Joseph [mailto:[EMAIL PROTECTED]
Sent: Tuesday, October 03, 2006 1:47 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: WiFi SIP
Jay R. Ashworth wrote:
On Mon, Oct 02, 2006 at 10:43:44AM +0800, Steve Underwood wrote:
Jay R. Ashworth wrote:
On Sun, Oct 01, 2006 at 02:58:37PM -0700, Lee Howard wrote:
Well, fax detection isn't entirely reliable anyway. Even if you assume
that your fax detection feature
I'm using the ChanSpy command for monitor a conversation of a channel which is
in a meetme conference. All comunications go throught voip, with some voip
phones attached to the lan and an external voip providor in order to make
external calls.
The problem is that sometimes the spy call can hear
On Tue, Oct 03, 2006 at 08:44:16AM +0800, Steve Underwood wrote:
So, y'know, that assertion gets made a lot.
What's the turn rate of fax machines in the market? 3 years? 5? CNG
tones are *well* over 10 years old, no?
What relevance does that have to CNG? It was a feature of the original
Yes, we saw the same problem. We opened a ticket with our reseller, who
escalated it to Polycom. Here's what the reseller said...
We escalated this up to Polycom. They said that they had seen a problem with
the display not updating with Asterisk, and they are going to see if your issue
is a
I did the same thing with the Polycom's - upgraded all mine from 1.6.x to
2.0.1 but I had great success and no problem with the buddy watch / presence
feature --- if anything, it works a little better.
Whats your mac-address-directory.xml configuration file look like? Did
you make any changes to
I had some weird flaky-ness after upgrading to the latest. Did a format
file system and let it reload from scratch. Works like a charm.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Higginbotham
Sent: Monday, October 02, 2006
Install went fine. No troubles other than this and it'd be minor if one
of the reasons for the update wasn't to expand the number of buddies
allowed on the IP601+sidecards we're adding for the attendant. Ugh...
Anyway, directory entries haven't changed:
?xml version=1.0 standalone=yes?^M
!--
I just received my first TDM2400 card I tried searching and couldn't
find anything on this.
I have 2 FXO modules with this card, it came with one modlule in the
slot marked as slot 6, so I put the other in slot 5. Since I don't
have an Amphenol connector/cable and a 66 block at the moment I can't
I am looking for advise on tools/techniques/metrics that are commonly
used to measure quality of device/end-points. Any pointers will very
helpful. Thanks.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
There are a few things to look at.
First off, you have a lot of wildcard testing that is probably throwing
the dial plan off. For example, you have the following:
exten = _07956nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten = _07879nn,1,Dial(${OUTBOUNDTRUNK}/${EXTEN})
exten =
On 10/3/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
It seems unreasonably difficult to get a list of the supported formats,
but does sox (http://sox.sourceforge.net/) do what you need?
Cheers,
-- jra
hey Jay, thanks but I am not sure what to tell sox as my output format
to be. I must admit,
80 matches
Mail list logo
