On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
I tend agree with your evaluation. Still, I was
thinking that since all these el-cheapo SOHO PBX boxes
support manual
Prathap,
That response is not sent by Asterisk. What you are most likely getting
this from is a packet capture, and what you are referring to is an ICMP
message sent as a backward notification by an intermediate router or host.
Basically, it sounds like the SIP UDP port (5060) on the Asterisk
Hi There,
I have installed an Asterisk server on Fedora Core,
I can able to run the Asterisk Server successfully.
But the problem is, my softphone(Xlite) is not getting registered with
Asterisk server. From softphone Register request were sent and the
Asterisk respond with Destination
I am using the netgear switch 24 ports and 8 ports w/ snom 360 in a
10/100 network w/ no problem.
the actual version of firmware of SNOM is 6.5.10 but the phone works w/
previews version.
Look in your network.
Bruno.
Anthony Cennami wrote:
Hello All,
I apologize for the slightly
James FitzGibbon wrote:
On 8/1/07, *Linux Lover* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
This SOHO PBX box won't interop with Asterisk
because it doesn't speak any
of the protocols that Asterisk does. This box
I tend agree with your evaluation. Still, I was
Benjamin Jacob wrote:
Ouch.
And I thought I had an answer to my query.
I totaly agree abt the long disclaimer nonsense Schmaltz, but I swear by
the powers up there, it's the admins over here at my workplace doing all
that nonsensical magic, as the mails go out. I wish i had the freedom to
Hello,
A little Off-Topic but how can you easily switch microphone and headphone
from one softphone to another on a Linux KDE platform ?
I use Skype (for incoming calls mainly) and Twinkle (for outgoing and
incoming calls).
I could't find any practical way to quickly switch audio from one
Anthony Francis wrote:
Hello good ppl,
A couple of questions for multiple pbxes
1. Is it possible to support multiple pbxes in one Asterisk box(using
contexts, etc.)?
2. Can we use the domain field in sip.conf to specify the different
domains for sip users, having one domain for each pbx?
I just
Hi Folks,
Thanks for the suggestions so far! Please keep them coming.
I plan ot summarize and post it for the record.maybe work it
into the faq somewhere, who knows.
Jeng
Steve Totaro [EMAIL PROTECTED] wrote: Chan_bluetooth is now chan_mobile and
included in
trunk/asterisk-addons.
Tom Lanyon wrote:
Hi all,
Can I ask that you please keep my personal address in the To: or CC:
in this thread as for some reason I'm only getting half of the list
emails coming through, and they're not showing up on the digium
pipermail archive either. The list archive on
Hi guys,
one of our German distributors (Allnet) has reasonable PoE switches
(price/features). They also have a distribution Channel in the US (
http://www.allnet-usa.com). They at least work pretty ok in our environment.
I could imagine that the power isn't very clean. Meaning the voltage
Tim
I have a batch of 30 that it does not affect. I also have a batch of
12 that it does effect. I like the SNOM phones and I think that I
just got a batch missing some shielding or other component. It is
_very_ noticeable when it happens.
Andrew
On 8/2/07, Tim Koehler [EMAIL PROTECTED]
Strange issue when I record a file from a phone to the asterisk
system I get a blip in the recording every 30 seconds. It's a very
small blip, but it is there.It seems like it's only if I'm
recording, not when I'm playing back that the issue happens.
My SATA drives, ETH0, and my Sangoma
I have the following:
pri box incoming/outgoing on box 1 connected through SIP to box 2.
The box 1 to box 2 has dtmfmode=rfc2833.
With this setting calls going out of box2 through box 1 the sendDTMF()
mode does not do anything.
When I change dtmfmode=info I at least hear the sendDTMF() digits.
hi,
how to add radius support to asterisk 1.4.5?
i do make menuselect and i do not see any module or option related to
radius, pam, authenticacion or similar.
any ideas?
thanks
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Telephone conversations that are being recorded, are supposed to beep
periodically, to alert/remind the
recorded person that the conversation is being recorded.
Perhaps that is what you are hearing?
joe a.
On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote:
Strange issue when I record
Matt wrote:
Strange issue when I record a file from a phone to the asterisk
system I get a blip in the recording every 30 seconds. It's a very
small blip, but it is there.It seems like it's only if I'm
recording, not when I'm playing back that the issue happens.
My SATA drives,
Hi,
Has anyone ever had any problem with the TE220B card with it showing up
as four ports instead of two. I RMA'd the first one with the retailer
(Digium tech advice), but I just got another brand new card and it is
coming up as four ports again. The card identifier is showing 0420 when
I
Greetings, List.
With my current setup, I record all incoming calls to my queues. My
problem is that once a call is transferred out of a queue, recording
stops. How can I make it so recording continues even after a call is
transferred?
If you need me to post any dialplan or conf logic,
AHHA! The PRI was not plugged in (system still in testing) so the
timing was off. As soon as we plugged the PRI cable in the blips went
away..
On 8/2/07, Matt [EMAIL PROTECTED] wrote:
Strange issue when I record a file from a phone to the asterisk
system I get a blip in the recording
Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
Perhaps that is what you are hearing?
joe a.
On 8/2/2007 at 8:47 AM, Matt [EMAIL PROTECTED] wrote:
yonoko molomo wrote:
hi,
how to add radius support to asterisk 1.4.5?
i do make menuselect and i do not see any module or option related to
radius, pam, authenticacion or similar.
any ideas?
thanks
___
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Can someone point me to an agi script that will read back the asterisk
uptime, if such a thing exists?
- Mike
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
You may want to consider the multi-tenant version of Thirdlane's PBX
Manager (www.thirdlane.com).
I've been using for a long time and very happy with both single and
multi-tenant versions.
Benjamin Jacob wrote:
Anthony Francis wrote:
Hello good ppl,
Gordon Henderson wrote:
On Thu, 2 Aug 2007, Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
You really ought to qualify this with the country and
Wow! Thank you so much, James - you have certainly
clarified lots of things in my mind. You are correct
about me overlooking the feedback issue (with the
el-cheapo device). I see that I have to learn. This
world of VoIP is new and mind boggling - to me.
Thanks,
Lynn
--- James FitzGibbon [EMAIL
hi again.well i have been trying to know what is the relationship
between asterisk and stun. what i mean is, i understand that a client
requests stun server to know whether its behind a nat or not. if its not,
then its ok. if it is behind nat, then what? Now client knows what kind of
nat it is
On Thu, 2 Aug 2007, Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
You really ought to qualify this with the country and the relevant laws
that you think
Hi List;
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I know that if I place the host with static IP then no
need to register,
Hi List;
Did any one tried the H.323 module? How much it is
stable and work fine?
Regards,
ITS
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 00965 9849460
Ready
for
Bilal,
The purpose of registration is to establish a contactability/reachability
URI information in the registrar dynamically. If you have a static IP on
both ends you can nail up an IP-trusted peer session / SIP trunk. If
not, some form of registration will be required.
Registration does
Hi,
http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html
Thanks, I have already seen that document before but it did not help
much to have a better understanding to set up radius with asterisk.
In 4.3 it is written: Asterisk has been patched along
hi,
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax - asterisk1 1.2.22 -
Honestly, it's really up to the client how it handles information from
STUN.
Ideally, what will happen is that it will modify its Contact headers and
SDP information to include the STUN-discovered IP address and port. In
so doing, when it sends out a request to another server, that server
On Thu, 2007-08-02 at 08:11 -0700, bilal ghayyad wrote:
How can I configure asterisk to receive a call from
SIP end point without being registered at asterisk and
its IP address is dynamic, and authentication to be
based on the username and password or any other
string?
I think you're
On Thu, 2 Aug 2007, Rizwan Hisham wrote:
hi again.well i have been trying to know what is the relationship
between asterisk and stun. what i mean is, i understand that a client
requests stun server to know whether its behind a nat or not. if its not,
then its ok. if it is behind nat,
On Thu, 2 Aug 2007, Steve Totaro wrote:
Gordon Henderson wrote:
On Thu, 2 Aug 2007, Joe acquisto wrote:
Telephone conversations that are being recorded, are supposed to
beep periodically, to alert/remind the recorded person that the
conversation is being recorded.
You really ought to
We spent a considerable amount of time getting an A101 up and running.
Try to find out what type of switch you are connecting to. In our case,
we were working against a Nortel. For some reason, if we used ni2, it
would not work. Finally setting the switchtype to 5ess or DMS100 would
work and now
Hello all,
Can anyone help me with SAY TIME.
Every time I ask to say time, it gives me wrong time.
I want the system to say time, what ever I give to say.
Is it possible?
Cheers,
Nitesh
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Hi there,
I have use the H.323 module that comes with asterisk-addons and i
consider it (so far) VERY stable for my needs.
Im talking about 10,000 minutes at month , + or - , and never had a
crash or something bad about it.
Personally, i recommend it,
--
J. P.
rakh at slackware-es dot
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones (configured for RFC2833). Am I the only
There is a strong possibility that the problem is on your side. Are you
using a cable or dsl? What are your download and upload speeds? Are you
doing any kind of traffic shaping?
You will not get a guarantee of QoS from any provider. They cannot
control what is happening on your end or what
John Meksavan wrote:
Asterisk Users,
I recently ran into some problems with the quality of service with
Teliax. This occurred on August 1, 2007 with a dropped outbound call,
audio quality isse on the callee side- not hearing me well on callee
side, and sending DTMF tones (configured
At 09:23 AM 8/2/2007, you wrote:
I recently ran into some problems with the quality of service with
Teliax. This occurred on August 1, 2007 with a dropped outbound
call, audio quality isse on the callee side- not hearing me well on
callee side, and sending DTMF tones (configured for
On Tue, 31 Jul 2007, Steve Kennedy wrote:
What if the radio is on in the background when I make a call ? is that
rebroadcasting ? kind of gets blurry on the definitions there.
That's not as you're listening to it and not trying to rebroadcast.
I've not been following this thread closely,
Has anyone ever had any problem with the TE220B card with it showing up
as four ports instead of two. I RMA'd the first one with the retailer
(Digium tech advice), but I just got another brand new card and it is
coming up as four ports again. The card identifier is showing 0420 when
I do
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ira
Sent: Thursday, August 02, 2007 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Teliax Quality of Service
At 09:23 AM 8/2/2007,
Hi Alex;
Kindly find my answers below preceeded by ( * ).
Bilal,
The purpose of registration is to establish a
contactability/reachability URI information in the
registrar dynamically.
* What is the URI?
If you have a static IP on both ends you can nail up
an IP-trusted peer session / SIP
Forums wrote:
You may want to consider the multi-tenant version of Thirdlane's PBX
Manager (www.thirdlane.com).
I've been using for a long time and very happy with both single and
multi-tenant versions.
Benjamin Jacob wrote:
Anthony Francis wrote:
Hello good ppl,
A couple of
Greetings,
We have a handlewelcome.agi script which handles every new caller. For
every new call we play a welcome message and ask the caller to enter a
four digit code .. something on the lines Welcome... please enter the
four digit number
Our asterisk java agi script calls a function
On Thu, 2 Aug 2007, William Moore wrote:
If you are only *SENDING* calls to asterisk and not receiving, you do
not need to send a registration. You only need to send a
registration if you want to *RECEIVE* calls from asterisk.
That's not necessarily true. Asterisk isn't going to just
On Thu, 2 Aug 2007, Alex Balashov wrote:
On Thu, 2 Aug 2007, William Moore wrote:
If you are only *SENDING* calls to asterisk and not receiving, you do
not need to send a registration. You only need to send a
registration if you want to *RECEIVE* calls from asterisk.
That's not
* I was asking if the endpoint send a call, and it has
a username and password typical to that configured in
SIP.conf file, then should this end point being
registered or not?
If you are only *SENDING* calls to asterisk and not receiving, you do
not need to send a registration. You only need
Shivaram U wrote:
Greetings,
We have a handlewelcome.agi script which handles every new caller. For
every new call we play a welcome message and ask the caller to enter a
four digit code .. something on the lines Welcome... please enter the
four digit number
Our asterisk java agi
I have read and followed as much as I can find but I am missing something.
What I want to do is get as much as I can running from mysql and keep the
*.conf files for static things. So I have setup a SIP users/peers table in a
mysql database and I have populated it with a few peers. I have
Hi all,
First, this is not my first PRI/T1 Asterisk deployement. Did several
with Bell, Telus, AllStream, Rogers but this is my first with Videotron.
Just spoke with the person taking the order and on top of the standard
settings (switch, coding,...) she asked me about data rate (56k or
B-chans should be 64k. That is a strange question indeed.
Thanks,
Steve Totaro
Andre Courchesne - Consultant wrote:
Hi all,
First, this is not my first PRI/T1 Asterisk deployement. Did several
with Bell, Telus, AllStream, Rogers but this is my first with Videotron.
Just spoke with the
On 8/2/07, Jay Moore [EMAIL PROTECTED] wrote:
With my current setup, I record all incoming calls to my queues. My
problem is that once a call is transferred out of a queue, recording
stops. How can I make it so recording continues even after a call is
transferred?
If you need me to post
Greetings,
i've been posted a message to this list in july, which had one response.
Thanks for that idea! Unfortunately asterisk is only a hobby, and did
not have much time dealing with the problem since. My original letter
was long, i wouldn't post it again, the archive url is
Steve Totaro wrote:
B-chans should be 64k. That is a strange question indeed.
For PRI, agreed. This is, however, a common question when provisioning
channelized T1 services, since the B channels on robbed-bit T1's are
really only 56K since the lowest bit is robbed for signalling.
Forrest W Christian wrote:
Steve Totaro wrote:
B-chans should be 64k. That is a strange question indeed.
For PRI, agreed. This is, however, a common question when provisioning
channelized T1 services, since the B channels on robbed-bit T1's are
really only 56K since the
It seems the problem with Unicall and Nextel is also present in
Asterisk 1.2 and not only in 1.4. I decided to downgrade from 1.4.9 to
1.2.23 so the customer could have CID and calls from Nextel but today he
told me that they cannot receive any calls from Nextel, they get a busy
tone
Here is a log with level 255 when a Nextel phone tries to call in:
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ]
Aug 2 15:38:18 WARNING[32670]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/1
On 8/2/07, John Meksavan wrote:
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones
Andre Courchesne - Consultant wrote:
Hi all,
First, this is not my first PRI/T1 Asterisk deployement. Did several
with Bell, Telus, AllStream, Rogers but this is my first with Videotron.
Just spoke with the person taking the order and on top of the standard
settings (switch, coding,...) she
Hi list,
I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS
5.
I am having trouble to make my simple IVR extension work, here is relevant
config:
zapata.conf
context=incoming
signalling=fxs_ks
channel = 4
context=internal
signalling=fxo_ks
channel = 1
-
Hi,
Does anyone have any experience with the PhonixEQ T100P card?
I wanted to know if it works fine with Asterisk without much of an issue.
Thanks for your comments.
TE100P 1 Port T1/E1 ISDN PRI Interface Card
datasheet http://store.phoniceq.com/datasheet/te100p-datasheet.pdf TE100P
offers
Dnia 2007-08-01, o godz. 11:47:42
Jason Adams [EMAIL PROTECTED] napisał(a):
Hi, All,
I have a question about agents and queues. Right now we have about 4
queues in our system. Some agents are in multiple queues. Our main
queue is for technical support and it's by far our busiest queue
Asterisk Users,
In my setup, I have a T1 service with McleodUSA and I am using the SIP
protocol. I am considering switching back to analog lines because quality
of service outweighs the cost savings at my work.
Any good SIP providers out there?
From: Baji Panchumarti [EMAIL
Any good SIP providers out there?
It really depends where you are. We're serving pretty much only Los
Angeles and Seattle rather than the entire US, and thus by focusing
our efforts on those limited markets we can achieve pretty good
quality and reliability. Servers are 15 ms away, less
Hello,
I want to get rid of bunch of useless notices in the logs when the hint
is not found, does setting the hint to noop for everything breaks anything?
exten = _X.,hint,NoOp
So far it did what I wanted.
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Steve Totaro wrote:
I knew someone would have an explanation that makes sense. I have
NEVER done anything but PRI from the Telco. Wouldn't the question of
signaling and switchtype negate the need to ask for data rate?
Yes. But these are probably telco ordering droids, meaning that all
Hi, Erik,
Never heard of call-by-call trunking.
Are you in Minnesota? What carrier are you using?
--Don
Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Thanks to all that responded so quickly. It was helpfull to me and I
hope other that will be asked the same question by telcos.
Andre Courchesne
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asterisk-users mailing list
To
On Thu, 2 Aug 2007, Forrest W. Christian wrote:
The order got kicked because I didn't specify whether or not I wanted
EM and which type of em (immediate, wink, etc) I wanted.
Are you serious? Which ILEC is this?
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel
Hi Carlos,
I suggest you download spandsp-0.0.3pre22.
(http://www.neuwald.biz/files/spandsp-0.0.3pre22.gz)
I don´t know why , spandsp after that uses digits 1,2..8,9,A,B,C,D,E,F
instead of 1,2,..,9,0,A,B,C,D,E. So, do you get F digits that are
incompatible with mfcr2 .
Luis A P Barbosa.
On 8/2/07, Forrest W. Christian [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
I knew someone would have an explanation that makes sense. I have
NEVER done anything but PRI from the Telco. Wouldn't the question of
signaling and switchtype negate the need to ask for data rate?
Yes. But
Has anyone set up Speechphone (Mandi) directly with Asterisk and not used an
ATA? If so, could you share how you did it?
TIA
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asterisk-users mailing list
To UNSUBSCRIBE or
On 8/1/07, Erik Anderson [EMAIL PROTECTED] wrote:
I've been working with a telco for the past two days trying to get a
PRI span up and running. This is a small-ish telco and I get the
feeling they don't do this very often. Anyway, they specified a
pretty standard setup: ni2 switchtype, esf
I think one critical aspect to explore here is, what exactly is meant by
drop the DS3 service to redundant back-ups?
SONET protection switching inside their transport core should not impact
your DS3s or voice-bearing T1s if it is within the 50 ms tolerance of
the standard. I imagine some
I have a customer installation with an Adtran DS3 mux. The DS1's go into my
Asterisk servers that run IVR/Call recorders. The DS3 provider is Qwest, and
they tell me that they routinely drop the DS3 service to redundant back-up's
and that this is a common practice that happens thousands of
On Thu, 2 Aug 2007, [EMAIL PROTECTED] wrote:
That is the back-up we are talking about here, the call loss is %100
when this happens.
Ah, I see. So, if I understand you correctly, what you appear to be
saying is that somewhere between your Asterisk box and your Adtran mux
this is not
I'm battling from a position here where I don't have a different DS3 to play
with, and I don't have a differnet mux. I'm being leaned on completely with the
argument that everyone else does this without any service interruptions. I'm
asking this group for the secret.
I have run out of
That is the back-up we are talking about here, the call loss is %100 when this
happens.
Alex Balashov [EMAIL PROTECTED] wrote:
I think one critical aspect to explore here is, what exactly is meant by
drop the DS3 service to redundant back-ups?
SONET protection switching inside
That is exacly what is happening. The 50ms interruption is disturbing
everything up to the chan_zap level, even though I have supressed the yellow
alarms.
Date: Thu, 2 Aug 2007 23:58:11 -0400 (EDT)
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial
Can someone here tell me why a switchover at the SONET level CAN disturb my
DS1? From the beginning, I though that carrier and messages were contained in
this specification.
Alex Balashov [EMAIL PROTECTED] wrote:
On Thu, 2 Aug 2007, [EMAIL PROTECTED] wrote:
That is the back-up we
marek cervenka wrote:
hi,
i have problem with pass-through faxing
with this scenario
hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen
virtual) - linksys ATA
i can fax to fax2mail on hylafax
but after upgrade asterisk2 to 1.4 faxing is not working
hylafax(iaxmodem) - iax
The problems started when we weren't getting the dtmf codes properly.
ie. if we type 4000500600 we werent getting the dtmf digits as is , we
were getting wrong dtmf codes like 405600 something to that
effect. This was without any changes to the system. We later moved the
asterisk server to
Might want to start by proving out your DTMF by just sending the calls
to something like VoiceMailMain().
When going into the voicemail system, see if you can reliably get DTMF
to work while entering mailbox numbers and password and moving around
the VM system..
At first glance it sure sounds to
Andrew Joakimsen wrote:
PLEASE tell me who that carrier is. I work with an inept company that
doesn't even know what ANI and CPN mean. Well our ANI and CPN are one
and the same. A bunch of inbred hicks somewhere in Alabama.
The underlying carrier is actually really clueful (Qwest the LD
90 matches
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