My vote is for the Sipura as well; essentially I agree with all of
Brian's points. Those boxes work well once set up correctly. And, the
server and PSTN line doesn't even have to be close; PSTN line in
Washington, * server in Texas and a slow (as in 128 kbps) DSL
between them is sufficient.
Anytime I receive a landline to anything over here in AUS, it comes up as
Overseas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Malcolm-Smith
Sent: Friday, 27 May 2005 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On 5/26/05, Magnus Espeland [EMAIL PROTECTED] wrote:
Hi,
Nice work!
Is it easy to make it run without mod_perl?
No it would take a lot of changes to make it work without mod perl. I
thought a lot about converting it to a regular perl cgi. The thing is
it's a trade off between being a
Hi
all,
I've
recentrly starting to play around with *, when all I wanted is to configure an
fritz ISDN card with [EMAIL PROTECTED]
Currently
I'm stuck at the phase of what do I do with capi after everything is installed.
I'm
trying to understand how to setup incoming and outgoing
Before I start on this next step I want to see if my plan has any flaws.
I'm working on adding multi user functionality to the gui. By multi
user I mean the ability of any number of end users to create their own
configurations on the same server, without interferring with other
users.
The only
hello all,
i'm using TE110P and all configuration prosess is
done.
but after i do the 'modprobe' for zaptel and wcte11xp,
follow by ztcfg -vv and asterisk -vvvgc i have this
error.
*CLI May 27 14:44:33 WARNING[3059]: chan_zap.c:8534
pri_dchannel: PRI Error: We think we're the network,
but they
See below:
In article [EMAIL PROTECTED],
Sukardi Shahdan [EMAIL PROTECTED] wrote:
hello all,
i'm using TE110P and all configuration prosess is
done.
but after i do the 'modprobe' for zaptel and wcte11xp,
follow by ztcfg -vv and asterisk -vvvgc i have this
error.
*CLI May 27 14:44:33
Any idea how I can generate a B* message on the asterisk box
(out of order) message?
Thank you.
Cheers,
nat
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Hi All
i'm using sangoma card. connected to E1,
my wanpipe file as
## WANPIPE1 Configuration File### Date: Fri May 27 00:25:04 GMT+7 2005## Note: This file was generated automatically# by
Ronald Wiplinger wrote:
I am looking for a SIP Soft Video phone, which I
can use with Asterisk.
If you have one installed (regardless if free or
purchased) please
tell me which one, the settings in Asterisk and
your experience with it.
try eyeBeam, it works fine for me...
[]
David Phelan wrote:
Anytime I receive a landline to anything over here in AUS, it comes up as
Overseas
I asked telecom why, and they said that the standard used doesnt support longer
then 3+7 digits, so international numbers may not fit. I would still like to be
able to send an NZ number
This is the /var/log/asterisk/full
May 27 06:51:08 VERBOSE[1107]: [chan_misdn.so]May 27
06:51:08 VERBOSE[1107]: [
chan_misdn.so] = (Channel driver for mISDN Support
(Bri/Pri))
May 27 06:51:08 VERBOSE[1107]: == Parsing
'/etc/asterisk/misdn.conf': May 27 0
6:51:08 VERBOSE[1107]: == Parsing
hi there,
its working now!!
:)
thanks a lot..
now, i can call the DID number..
but i must configure to get the call go to the
voicemail because no phone connected.
thanks again..
--- Tony Mountifield [EMAIL PROTECTED] wrote:
See below:
In article
[EMAIL PROTECTED],
Sukardi Shahdan [EMAIL
Michael Stearne wrote:
Jon,
What version of PHPAGI are you using? I am starting a PHPAGI app and
want to know whether to use 1.12 or 2.0CVS.
I am using 1.12
Regards
Jon
___
How much free photo storage do you
Benjamin West wrote:
Michael,
The version, in the context of Jon's problem, was irrelevant. Jon's
problem was due to a small bug in his code, and not related to PHPAGI.
Hi Benjamin,
Actually I would say it was more to do with my lack of understanding
with how Asterisk AGI worked and my
Thx Scott,
However, and from what I can read, chan_sccp is not really meant for
server-to-server trunks, yet...
Am I wrong ?
BR, - Patrick -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Herrick
Sent: jeudi, 26. mai 2005 04:24
To: Asterisk
When a call comes into our office, it rings multiple Cisco 7912's (in
Asterisk the dial command is just Dial(SIP/101SIP/102SIP/103...)
Then when somebody answers the call, all the other phones stop ringing
(of course). The only problem with this is that all the other phones
then display
Message: 3
Date: Thu, 26 May 2005 23:01:28 -0300
From: miguel [EMAIL PROTECTED]
Subject: [Asterisk-Users] AreskiCC
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
I'm tring to dowload the AreskiCC but the
www.areski.net is out of
Title: Message
Is any one willing to make their TFTP server
available and try this upgrade for me?
Razza26 May 2005 18:43'Status messages' simply shows SEPMACaddr.cnf
for info, 'Firmware versions' shows
'Application Load ID' P003G302 and 'Boot Load ID' PC030301
Ray
Christopher
Hi,
I`m looking for SIP SoftPhone for debuging some situations in
my SIP VoIP network.
Requirements:
- it mustn`t be registered with any
registrar/proxy/anything
- it must be able to send INVITE msgs without registration, so
it must accept characters @ and .
- it must be able to receive
Hi
I am using CVS-HEAD-05/09/05 and astcc.
The call sometimes does not hang up at all, does not even get the
warning notices, as it should, since the L option specifies that the
caller gets played a message like 1 minute before the call is
disconnected, and then the call should end.
Its
;
; Music on hold class definitions
;
[classes]
default = /var/lib/asterisk/mohmp3
;loud = mp3:/var/lib/asterisk/mohmp3
;random = quietmp3:/var/lib/asterisk/mohmp3,-z
;unbuffered = mp3nb:/var/lib/asterisk/mohmp3
;quietunbuf = quietmp3nb:/var/lib/asterisk/mohmp3
; Note that the custom mode cannot
barney wrote:
I`m looking for SIP SoftPhone for debuging some situations in my SIP
VoIP network.
Requirements:
- it mustn`t be registered with any registrar/proxy/anything
- it must be able to send INVITE msgs without registration, so it must
accept characters @ and .
- it must be able to
-Original Message-
From: Michiel van Baak [mailto:[EMAIL PROTECTED]
Sent: 26 May 2005 20:22
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID)
Anton,
My script is not connecting to the manager interface.
The php script is
If, after you've changed your config to pri_cpe and you still get the
same messages (only saying they think they're the cpe too), ask your
provider to see if they have any interface loops running on their
switch. On many devices, you can set a loop on the interface that won't
show up as a
hello. i'm new with asterisk but so far managed to make it work with
SIP, IAX and even with 2 asterisk servers. Now i tried to configure it
for H323 but it didn't work. i used net-meeting from microsoft. when i
try to call from h323 to sip asterisk didn't say anything. probably the
h323 client
In a pure voip envoirnment which uses a single codec say ulaw across all its
phones can asterisk support 5000 voip sip phones on a dual / single xeon with
1 gb ram. If all the phones support reinvite (Send RTP stream directly to
each other).
Or would I need more than 1 system to support 5000
depends on how much they're calling each other... if they're all using
reinvites, ulaw, and only calling each other, I don't see why that box
wouldn't be able to handle it, you just won't have any fault tolerance.
If you provide more information about the project, I'm sure you'll get
some
Hi,
Asterisk will work, but in your situation I think it's better use a SIP
proxy for that (SER for example http://www.iptel.org/ser) which is
really meant for this purpose
Niels.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vikram
Rangnekar
Sent:
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
I am interested in G729 because the internet in my
country is very
Hi
I am trying to get 2 incoming SIP accounts working from 2 different
providers. One is sipgate.co.uk and the other is voipuser.org. If I load
the Register command seperate they will both register phone and incoming
works. If I try to load them both only sipgate registers. Anybody got
any
Would it be possible to use spandsp to answer calls from normal
analogue modems? I'd like to create a software modem bank in a similar
fashion to my spandsp-supported fax bank.
If this works I would then try to adapt it to provide me with a ppp session.
Thanks,
Wallace
Hi,
Ya I know about SER but I wanted to specifically know about asterisk in this
envoirment and its always great to stress test asterisk any bugs that comeup
would help make asterisk better.
+++ [EMAIL PROTECTED] [27/05/05 12:13 +0200]:
Hi,
Asterisk will work, but in your situation I think
Hi Miguel,
Try to use a PUBLIC PROXY outside BRAZIL.
I don´t know exacly what´s happening ... but
routing direct from brazil doesn´t reach Areski´s domain.
I´ve contacted it before , and he doesn´t know why !
Regards,
-Jefferson Carvalho
miguel wrote:
I'm tring to dowload the AreskiCC but
Daniel Eboa wrote:
what i want is be able to authenticate user before they connected to my
asterisk box.
users can be registered with asterisk, but i want that each time a user want to
place outgoing call, he is first authenticate, and then authorize to place the
call through the asterisk
On May 27, 2005 06:12 am, chawki hammoud wrote:
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
It sounded more or
On May 27, 2005 03:20 am, Nathaniel Angelo A. Torres (247talk) wrote:
Any idea how I can generate a B* message on the asterisk box (out of order)
message?
Easy. Do *not* have an exten = line in your PRI incoming context that
matches the number you want SIT to be played for. At least for Bell
On May 27, 2005 01:47 am, Vij wrote:
The above command always shows zero value for jitter. (Actually, only rtt
and kpkts are non-zero). The behaviour is the same even for
cross-continental calls.
Post your iax.conf without passwords.
Also, are there any native bridges going on on either
When I try to transfer calls with a Polycom 500, (blind or not), the
digits don't display properly on the screen, and the call is just put on
hold. I also can't park a call for the same reasons. Is there
something funny with sending digits to this polycom? I'm sure this is
something silly.
hello,
I am in test time about my Asterisk server.when my server is running, the
client bob can't connect to it. the server say :
chan_iax2.c:3865 register_verify: No registration for peer 'bob'
chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.204
How can I resolve this
Mailing List wrote:
Any chance it's the phone mentioned here?
http://voxilla.com/voxstory134.html
She's on screen, and you're looking at the *phones*? shakes head
--
Adrian Chapman
Director
Trivas Ltd
Business on the Move
Mobility - Messaging - Infrastructure - Security - Remote Access
07796
See below...
At 07:29 PM 5/26/2005, you wrote:
It's
not the licenses, that's like 10% of the problem. One can always
buy licenses... But the 16 man hours I wasted waiting for the
across-town overnight shipment vastly outweighed the cost of the
licenses.
To mitigate risk, why didn't you ask to
SteveK and Andrew,
Thanks a
lot for the suggestion. It helped. We didnt know that jitterbuffer wont
be enabled with sip endpoints. forcejitterbuffer=true solved the
problem.
Thanks again,
Vijay AshishOn 5/27/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On May 27, 2005 01:47 am, Vij wrote: The
On May 27, 2005 08:15 am, Karl J. Vesterling wrote:
It's cheaper to pay the $85.00 to send the equipment than it is to pull
someone off the assembly line at KTE to act as the gopher.
Fair enough.
Yup... Had that... Got the tracking number when I got on the flight, but
it didn't scan in
chawki hammoud wrote:
I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm, G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?
I am interested in G729 because the internet
Hi,
I want to use the sip extension 105 as the voicemailbox number.
When i initiate a call to the number 105 from my polycom 105 i only get
a call on new line to phone.
But i want in this moment is the voicemailmenu which ask me for my password.
How can this be done with the polycom phone?
Thank you very much for that debugging information Tony. That's exactly what I
was looking for, I just didn't know how to ask for it. You understand Noobian
very well ;). It lead me right to the problem.
Unfortunately, it was not a dazzling problem, it was a hard (for me) to spot
typo in
Is call waiting supported on an analog incoming line? I have a customer
that has a line with call waiting that wants to go to Asterisk, but
wants to keep the call waiting. If it is, how would I set it up in
asterisk.
Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161
Has anyone seen this problem: I recently purchased new 7960g's with
sccp firmware on it. When I try to upgrade the firmware to SIP the
phones seem to be making malformed tftp requests. The logs shows:
sending NAK (4, Request not null-terminated) to 192.168.0.55
Thanks in advance,
Josh Laroff
Yes it is possible, if a call comes in it will beep on the extension
that is speaking to the first call.
Keep in mind individual extensions can also turn off call waiting so you
need to check both pbx and extension settings.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
SetCDRUserField doesn't appear to be working in the 5/25/05 CVS - Anyone
else having this problem?
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Dear Kapejod List
i have notice a problem using the bristuff-0.2.0-RC8e and the option
priindication=passthrough, after a lot of incoming/outgoing call in a day
seems to have oneway audio from to ISDN zaphfc (Cologne Chip Designs GmbH
ISDN network controller [HFC-PCI]). If i remove
Hi,
apenon apenon wrote:
We have installed asterisk and using with many small ata. Now there is
an AS5300 outside a PSTN PBX which makes termination. Now I want
asterisk to handle only some of the calls with SIP and asterisk from
5300. There is no problem at asterisk configuration but in AS5300
On Fri, May 27, 2005 9:48 am, Adam Collard wrote:
Is call waiting supported on an analog incoming line? I have a customer
that has a line with call waiting that wants to go to Asterisk,
but wants to keep the call waiting. If it is, how would I set it up in
asterisk.
We have callwaiting=yes
I will be installing [EMAIL PROTECTED], if that helps.
Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim
I posted this to the -dev list and have recieved no reponse so I'm gonna try
the -users list.
I'm trying to bring ast-ax-snmp up to date with current CVS. It's a
small patch against chan_zap and pbx, but also has a loadable module
component. That module is compiled against some NET-SNMPD
Hi,
Someone released a succefull interconnection in H323 with WTL equipement
?
I'm trying to do that with an IPNx. But get dead air.
With chan_oh323 it's fine, all works. With chan_h323 = dead air.
The configuration is GW to GW.
This is my configuration from h323.conf:
Is it possible to have a Dial my name menu in Asterisk?
AZ
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I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] = (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, PRI Signalling signalling
--
Good day list,
I have been fighting echo problems on my PRI card.
Everything is working great outbound, however inbound calls have
echo.
I have found the issue but need help in fixing it.
During outbound calling zap show channel 18 shows that the
call is
I tried it a while, its impossible.
(Well you can get it to work, but not in a stable way)
Zoa.
Laurent Tostain wrote:
Hi,
Someone released a succefull interconnection in H323 with WTL equipement
?
I'm trying to do that with an IPNx. But get dead air.
With chan_oh323 it's fine,
I have a Grandstream GSX-2000 with ..
Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3
I tried to do an HTTP update from the Grand Stream web site...
After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of
calls when there is such high latency? Can anyone comment?
Thanks,
Waldo
may be something like this?
[macro-sipextens]
exten = s,1,SetVar(voicemail=${ARG1})
exten = s,2,Dial(SIP/${ARG1},40,r)
exten = s,3,GotoIf($[${DIALSTATUS} = NOANSWER] ? 66 : 3)
exten = s,4,GotoIf($[${DIALSTATUS} = BUSY] ? 68 : 4)
exten = s,5,Playback(iss_invalid_sipexten)
exten =
Waldo Rubinstein wrote:
I'm planning on setting up some remote agents and before doing so, I
did some simple PING tests to measure latency. The average latency I
got was 250ms. Does anyone have experience in terms of quality of calls
when there is such high latency? Can anyone comment?
Check the documentation for the Directory application on
http://www.voip-info.org .
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, May 27, 2005 10:54 AM
To: asterisk-users@lists.digium.com
Subject:
I've got three GS 100 Phones with same problem.
Some lights.
Some no lights.
Some garbled display.
I would welcome suggestions for a resurrection.
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
On May 27, 2005 10:57 am, Ronald Hartmann wrote:
Actual Hookstate: Onhook
What version of Asterisk is this? CVS HEAD from about an hour ago does not
show hookstate for anything but FXS interfaces, and certainly not for PRI.
-A.
___
Asterisk-Users
Define happily?
Jitter is obviously important, but latency is too. For day-to-day
business calls, 250ms is a little high. Both parties will definitely
notice it. In my experience you will find yourself talking over one
another quite often. Even with 100ms this continues to happen from time
to
hello everyone,
I just had a thought on this subject why not create a daemon process on
the Client PC That registers its self and What phone the user is
connected. An AGI script could monitor the progress and when answered
could send a push to the registered daemon which would push a link to
the
The person on 617 is unavailable --- Why
*CLI
-- SIP Seeding peers from Astdb: '617' at [EMAIL PROTECTED]:6990
for 3600
-- Executing Dial(SIP/601-f18a, SIP/617|60|tr) in new stack
-- Called 617
-- Got SIP response 480 Temporarily Unavailable back from
192.168.250.107
--
On 27/05/05, Max W Blackmer Jr [EMAIL PROTECTED] wrote:
hello everyone,
I just had a thought on this subject why not create a daemon process on
the Client PC That registers its self and What phone the user is
connected. An AGI script could monitor the progress and when answered
could send a
Not sure I understand your meaning. You have a phone with 105 as the
registered extension but you want to dial 105 and get voicemail?
Lets assume that *98 is your voicemail extension.
If you dial *98 from any phone it should ask for the extension and
password.
So extension *98 looks something
On May 27, 2005, at 6:50 AM, Andrew Kohlsmith wrote:
On May 27, 2005 03:20 am, Nathaniel Angelo A. Torres (247talk) wrote:
Any idea how I can generate a B* message on the asterisk box (out of
order)
message?
Easy. Do *not* have an exten = line in your PRI incoming context that
matches the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
MF Hulber
Sent: Thursday, May 26, 2005 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Size of extensions.conf
Although you may not see it
At 08:59 AM 5/27/2005, you wrote:
[ snip for brevity ]
I just wanted to clarify ... this isn't a voipsupply.com problem at all,
but
rather a courier screwup... which happens anywhere and at anytime...
right?
TWO screw ups in the shipment.
1.) It was shipped to the Bill-To address. Since there
I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the
On 27/05/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
The person on 617 is unavailable --- Why
Maybe he's in the bathroom?
The condition being reported is coming from the UA on the end of the
SIP call - is there a DND setting or something there?
Peter
--
Peter Bowyer
Email: [EMAIL
It will be about 100 phones at about 20 locations all within
about 4 miles of each other.
Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres reliably, and using Ethernet repeaters every hundred
If you have the money, I would recommend the Cisco 7900 series, but if
you need cheap phones, go with sipura. I can get you all you need if you
want. The Sipura phones run about $100.
Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC
brandt Milczewski wrote:
I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping
On Fri, 27 May 2005, Mark Elkins wrote:
I tried to do an HTTP update from the Grand Stream web site...
You upgraded the firmware over the Internet? You are braver than I am. I
would have used a local http server.
Is there a magic re-incarnation routine ?
(Power on whilst holding down some
Wireless bridges??
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Friday, 27 May 2005 1:18 PM
To: 'brandt Milczewski'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users]
Well, that will be pretty preferential
As stated before, I love the Polycom IP500. I think it is just a great
phone for less than $200.
It configs easily once you get used to the config file and Polycoms have
great speaker phones.
Many love the Ciscos... Admittedly a beautiful phone but
On Fri, 27 May 2005, Colin Anderson wrote:
It will be about 100 phones at about 20 locations all within
about 4 miles of each other.
Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres
I thought he meant that as well but I hope that what will occur is that
there is DSL somewhere already that can be utilized.
That conflicts with the 'old town PBX' scenario as well though.
So, assuming there is DSL already, that even makes you wonder why bother
if a phone line already exists and
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not suppose
to sound that way. Most Carriers want the calls to be in g711u so
thats why I use G711u otherwise I want to save money on bandwidth. G729
on Asterisk
Have you placed a group=1 in zapata.conf ?
For a trial, you can use: exten = 203,1,Dial,Zap/1/onetelephonnumber
Aitor a écrit :
I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] =
On Fri, 27 May 2005, Mike Clark wrote:
brandt Milczewski wrote:
I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be
I've got the same issue with a Swissvoice IP10S SIP phone. I couldn't
find much information with this issue, but it seems to appear because
Asterisk does not support variable length for g.729 (don't ask me what
it really means). Anyway, it is recommanded to disable the silence
suppression,
I am using version 1.4.1.0040 and 2.6.1.0003. I have read about newer
versions out there, how can I get them? I'm having an intermittent issue
regarding DHCP with one of my phones, and I recall when I loaded 1.4.1 on
this one phone, it failed once, then it succeeded the second time.
I am
Michael D Schelin wrote:
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not
suppose to sound that way. Most Carriers want the calls to be in
g711u so thats why I use G711u otherwise I want to save money on
Title: Message
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cory
AndrewsSent: Thursday, May 26, 2005 6:33 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] VoiPSupply Dot Com
Karl,
On May 27, 2005 01:27 pm, Michael D Schelin wrote:
I have used G729 and it sounds almost as good as G711U. The problem is
the way Asterisk uses it. It does not sound robotic and it's not suppose
to sound that way. Most Carriers want the calls to be in g711u so
thats why I use G711u otherwise
Peter Bowyer wrote:
On 27/05/05, Ronald Wiplinger [EMAIL PROTECTED] wrote:
The person on 617 is unavailable --- Why
Maybe he's in the bathroom?
No, I am testing phone (not in the bathroom ;-) )
The condition being reported is coming from the UA on the end of the
SIP
Title: Message
May 28 01:58:32 WARNING[6821]: chan_sip.c:1830
sip_write: Asked to transmit frame type 256, while native formats is 4
(read/write = 4/4)May 28 01:58:33 WARNING[6821]: channel.c:2126
ast_channel_make_compatible: No path to translate from SIP/3901506-efd7(4) to
Title: Message
anyone got any ideas
on this?
TDM H323
Gateway SIP
Inbound H.323 call
'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and
attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting
channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) native
Is pstn call waiting working on a Digium TDM-400 with FXO ?
Configuration in zapata.conf:
callwaiting=yes
callwaitingcallerid=yes
callprogress=yes
If an incoming call happens while the FXO channel has a call in progress,
and the call is routed to a FXS channel (which has callwaiting=yes in
On 5/27/05, Karl J. Vesterling [EMAIL PROTECTED] wrote:
At 08:59 AM 5/27/2005, you wrote:
[ snip for brevity ]
I just wanted to clarify ... this isn't a voipsupply.com problem at all,
but
rather a courier screwup... which happens anywhere and at anytime... right?
TWO screw ups in
I've configured SIP softphone to work with asterisk n it's working fine.
but i m unable to connect my netphone IP phone I've connected my phone
to LAN and assigned an IP address to it but how can i make call... plz
tell me step wise.
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