I have a lot of folks asking me about an auto-negotiating WLAN phone
supposedly being brought to market by Hop-On, which is touted to carry an
MSRP of $40 Press photos (stock art) of the device shows it looks almost
identical to devices from Zyxel and UTStarCom.
I am trying to explain to folks
You can set, in the linksys, the codec G729 for your line. In the Linksys
also set only to use that codec. This can be done at the admin page of the
line you use in the linksys. Also do that in the asterisk for your device.
First buy the license from Digium.
Then you will use less bandwidth and
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, June 28, 2005 6:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] RTP session between two end users
Erdem HAKİ wrote:
HI!
I configured asterisk to send all outgoing calls to our Gateway. I noticed
when asterisk sends call to gateway that he represents all calls as
asterisk and not as callerID(number of sjphone client registerd to
asterisk).
Can anyone give me an example of such configuration?
Thank you
Howdy,
Am Dienstag, den 28.06.2005, 09:01 +0200 schrieb vdasilva:
Hello
I have asterisk running in Red Hat 9 with a cheap HFC card on i4l. I have
choppy sound problems sometimes, and echo problems often. I am using a 2
port Grandstream ATA, Grandstream BT and a Grandstream GPX-2000
I
That would probably be me.
You could use a lot of different things to do the testing,
one would be the tcl script in your asterisk/contrib/scripts directory,
some more can be found in the beginning of this presentation:
http://astertest.com/astricon_performance.ppt
We started some callgenerator
Thanks for your help Bernard, it's realy useful web site, but i also want to
know limits which depens on hardware of the box. Any practical experience?
Thanks again :-)
Erdem HAKI - [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Hi,
CRC errors are caused by bit errors on layer 1. In most cases this is
a cable issue. Did you try replacing the cable from the NT1 to the
quadBRI? How long is that cable?
However if only 1 of the 2 B channels are working then you might
have your BRI lines get checked or try a different ISDN
Hi,
what signalling does the telco run on those lines?
best regards
Klaus
Am Dienstag, den 28.06.2005, 19:02 +0200 schrieb Doug Reid - Stormcorp:
Hi all
Correction on my last mail, I found that line 1 both channels work
but on line 2 none work.
I have 2 BRI ISDN lines coming in on port
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:
Hey gang,
I've been able to use sipp to produce some call volume on our
asterisk
server. The server has no problems handling 50 simul calls. But
then again,
no RTP is being done. I tried to use the rtp echo ability of sipp
but that
* Hamish Whittal ([EMAIL PROTECTED]) ha scritto:
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now looing at sipXphone seem to be picking up that it is not that
stable, but perhaps someone
On 29 Jun 2005, at 04:51, Matthew Boehm wrote:
Hey gang,
I've been able to use sipp to produce some call volume on our
asterisk
server. The server has no problems handling 50 simul calls. But
then again,
no RTP is being done. I tried to use the rtp echo ability of sipp
but that
hi, all:
I have two phones, one is SIP/200, another is IAX2/203. Now, i use
IAX2/203 call to SIP/200, sometime CallerID display is 203(at phone
SIP/200), sometime display is 200. Is this a bug? Please help me!
Sorry my english.
Li Yuqian
___
Thanks for the reply. But how do you troubleshoot which application is
the culprit. Any ideas ?
I am using FEDORA 3.
Rgds
T.E.Yap
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That looks right, the database is being updated properly. The last call
lasted 9 seconds and cost you 1.5c so it should show up in the database.
You did create a 2-digit card called '21' right?
- Original Message -
From: Juan Luis Moyano [EMAIL PROTECTED]
To: Asterisk Users Mailing
Hi all,
on sourceforge.net I added the fixup release 0.5.3 of
chan_capi-cm driver.
The changes from 0.5.2 to 0.5.3 are:
- voice data queue (send buffer) fix
- fix for CVS-HEAD of Asterisk (Thanks to Frank Sautter)
I have tested this version with Asterisk 1.0.7, 1.0.8 and HEAD(2005/06/28).
Have
Hi,
Need to implement hunting (create a hunt group so my
subscribers can have a single GSM number for access to
me)of GSM SIMs on a GSM bank independent of the Telco
for the SIMs.
Anyone got an EXACT idea how to do this?
Thanks,
Latex.
On Tue, 28 Jun 2005, Luis Vazquez wrote:
Hello all,
I'm having problems getting chan_capi 0.3.5 to work well with an Eicon Diva
Server card using using the driver from linux kernel both 2.6.10 and 2.6.12
(vanilla versions).
Have you tried the chan_capi-cm version from sourceforge ?
I have
The reason for the problem is clear below, theASTERISKCDRDB database is being updatedinstead of theASTCCDB database which holds the cdrs and BILLCOST.
How can this problem be corrected???
3 Query UPDATE cards SET used='0' WHERE number='58767059' 3 Query UPDATE cards SET inuse='0' WHERE
Hi,
-Original Message-
Need to implement hunting (create a hunt group so my
subscribers can have a single GSM number for access to
me)of GSM SIMs on a GSM bank independent of the Telco
for the SIMs.
Anyone got an EXACT idea how to do this?
If you want 1 GSM number that can access
I'm currently unable to register with Teliax's server via IAX2 and can't
reach them via either of their phone numbers. Their website is up and I
have logged a support incident.
Is anyone else experiencing the same problems? Having been caught up in the
Broadvoice fiasco a couple of months
I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4
port Eicon Diva card. All works fine, but i'd like calls from the PBX to
Asterisk to show the Caller ID name and not just the number. I know this
information is being presented by looking through the ISDN trace for the
Hi all,
I've been trying to get meetme working for a while now (complie problems
- will probably try again later on another machine) but have given up
and started looking at alternatives.
I've managed to get app_conference compiled and installed - show modules
shows its there in asterisk,
I'm currently unable to register with Teliax's server via IAX2 and can't
reach them via either of their phone numbers. Their website is up and I
have logged a support incident.
Is anyone else experiencing the same problems? Having been caught up in the
Broadvoice fiasco a couple of months
I'm testing NVBackgroundDetect with Sipura-300 and I get this error:
rtp.c:505 ast_rtp_read: Unknown RTP codec 100 received21
Does anybody know what is it?
--
#Joseph
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I really hate to have to make a post like this, but I feel I have little
choice but to relay to the group my experience with Teliax, and explain why
I recommend against using them as a primary Voip- PSTN provider. I hope
that a letter like this will inspire companies like Teliax to work harder at
Lets not jump the gun here..one failed iax registration does not a
bankrupt company make...
(p.s., yes my registrations are not getting responses either)
Mark
On 6/29/05, Rich Adamson [EMAIL PROTECTED] wrote:
I'm currently unable to register with Teliax's server via IAX2 and can't
reach
Try voip-co2.teliax.com to register with. And read my other letter I
suppose. This domain is apparently working as of 4:30, but have had the
same problem since 1:30 AM PDT.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On
Does anyone have anything +/- to say about TeleSIP? They appear to have
local DIDs where I live and all comments on the wiki indicate they are
reputable..
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Rich
Hi there...
I've to setup an Asterisk system for a small office, I haven't done one of
these in at least a year and was wondering if someone could just let me know
what sort of phones are doing well these days.
It just needs 9 phones in the office, for general use, no fancy things
required for
If astccdb exists, go to the database configuration
page [Configure] and change the database name to the
correct one. You may have to set up permissions on
this database if it wasn't set up before. If it
doesn't exist, use the 'Create Database' button to
create a new one.
--- Ade Agbero [EMAIL
An ethereal trace indicates the IP address is active, but it is not
responding to iax packets (registration). So, either their asterisk
app has failed or they have folded their tent as well.
I am sure it's just a crashed server, wait an hour and let the support
people deal with it.
--
You would be better using extensions_custom only because of the fact that
when you restart ampportal, it will overwrite extensions_additional with
what ever it has stored in the Database.
I've actually taken to adding the code that I build onto what AMP
generates into the database. For
On Wed, 29 Jun 2005 08:15:20 -0400
Chris Mason (Lists) [EMAIL PROTECTED] wrote:
An ethereal trace indicates the IP address is active, but
it is not
responding to iax packets (registration). So, either
their asterisk
app has failed or they have folded their tent as well.
I am sure it's
bash-3.00# cat musiconhold.conf | more
;
; Music on hold class definitions
;
[classes]
; Christian Rock.NET
;default = quietmp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/
;loud = mp3:/var/lib/asterisk/mohmp3-empty,http://209.51.128.160:5112/
; Cleft in the Rock Radio (TESTING)
Title: Message
I receive this error
on the asterisk console and it is pretty much ALWAYS coming
up.
Sometimes there will
be a break where it does not display.
We had our PRI
provider test the lines and they claim that there is no signalling
problem.
It doesn't matter if
there are no
Hi,
I am planning to try Asterisk and would like some guidelines on the size of
machine I need. Is there a page somewhere with some suggestions?
Kevin Roche
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On Sat, Jun 25, 2005 at 07:58:24PM -0500, Greg Oliver wrote:
That works well. You may also want to make sure your compatibility
matrix between Asterisk/OH323/PWLIB/GNUGK is right - incompatibilities
cause more issues than I care to talk about. The GNUGk web site has the
best matrix to
Title: [Asterisk-Users] Problem with Connecting PBX to Asterisk
The framing is ESF/B8ZS. But I have had some luck and have gotten to the point where when dialed from Asterisk, the digits reach the telrad switch and the DID that I have configured in the telrad switch works and rings the right
Kevin Roche wrote:
Hi,
I am planning to try Asterisk and would like some guidelines on the size of
machine I need. Is there a page somewhere with some suggestions?
http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware
Doug
___
On Tuesday 28 June 2005 23:15, Rich Adamson wrote:
I cannot get this thing to work. Anyone know of any tricks?
Call digium support; its free.
Well technically it's not free. You just paid for support in the price of the
card (of all their cards)...
-A.
It's up and running again now. I just found it a little disconcerting not
to be unable to reach their support numbers during the outage.
Malcolm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Wednesday, June 29, 2005 8:15 AM
On Wed, 29 Jun 2005, louis g wrote:
I have an Asterisk server connected to ISDN2 lines off a PBX (Avaya) using 4
port Eicon Diva card. All works fine, but i'd like calls from the PBX to
Asterisk to show the Caller ID name and not just the number. I know this
information is being presented by
Looks like 9 out of 10 calls are failing on voipjet at the moment (at
least terminating to South Florida numbers). Keep getting message
that says number can not be completed as dialed. Anyone else seeing
this?
On 6/15/05, Pedro [EMAIL PROTECTED] wrote:
Couple of days. Apparently the new US
Unfortunately no. Someone say the press release and bugged me about it
as well but I haven't seen anything that would indicate they plan on
doing anything more than parting with carriers with large rollouts of
these phones. That MSRP seems too good to be reality too.
-- William
-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db. I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system.
This is an
exten = 901,1,Conference(Internal Test Conference/S/1)
Looks like it does the job...
Mark Benson wrote:
Hi all,
I've been trying to get meetme working for a while now (complie
problems - will probably try again later on another machine) but have
given up and started looking at
On 15:54:12 June 29, 2005 Armin Schindler [EMAIL PROTECTED] wrote:
On Wed, 29 Jun 2005, louis g wrote:
I have an Asterisk server connected to ISDN2 lines off a PBX
(Avaya) using 4 port Eicon Diva card. All works fine, but i'd like
calls from the PBX to Asterisk to show the Caller ID name
Hi,
the situation: A call rings at extension 123. My own extension is not in
the same call- or pickupgroup for that extension.
Is there a way to route the ringing extension 123 to my phone?
Thanks,
Kib
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-Original Message-
From: Steve Prior [mailto:[EMAIL PROTECTED]
Here is the text of the last 2 bug comments by MikeJ (who I
would assume
closed the bug).
text snipped
I think there are three issues here:
1. The bug was originally filed as a feature request for a feature that
would
hello,
i've an asterisk box which is connected to an E1/PRI via a TE110P card.
incoming calls from mobile phones where the number is transfered as a
whole block work fine, but when dialing from an analog or ISDN line to
the asterisk box there is a timeout of about 3-5 seconds.
originally my
http://www.gnugk.org/compiling-gnugk.html
Also, the reqs for the included 323 channel and gnugk differ on
versions. I have unreliably gotten them both to run on the same box
with 100% reliability. Outbound calls transcoded from SIP - 323 -
Gnugk - CCM - MGCP - PRI get dropped from DRQ after 2-4
No I do not hear any clicking sound. Some calls come in perfect, and others
come in with some echo and sometimes artifacts, which I think might be caused
by jitter. Also it is mostly inbound calls that I have the problem with. If you
didn't have any echo, just clicking, would you possibly
What does zttool say? Do you have any IRQ issues or anything?
--
Tom
On 6/29/05, Michael Blood [EMAIL PROTECTED] wrote:
I receive this error on the asterisk console and it is pretty much ALWAYS
coming up.
Sometimes there will be a break where it does not display.
We had our PRI
I have been unable to get it to pickup sip-sip calls but if an
incoming zap rings I can hit *8# and it works.
My config is the same as yours:
zapata has callgroup = 1
and in sip.conf I have
pickupgroup = 1
I'm also using Grandstreams.
t o n y
On 6/28/05, Robert Woodcock [EMAIL PROTECTED]
On 6/29/05, Joe Murray [EMAIL PROTECTED] wrote:
Is anyone on this list using and audiocodes FXO gateway? I have
Asterisk(1.07 on OS X) setup and working fine, including SIP phones
and IAX2 phones - I can make outbound calls just fine and receive
inbound calls just fine. However, I can't seem
Hi,
if you are using the QSIG protocol for the interconnection between
Asterisk and the PBX, I have maybe a solution.
for the X100P you are using Zapata driver of asterisk. (with the
switchtype QSIG right?)
But for the eicon you use the capi module?
Caller Name within QSIG is standardized as
I have indeed already done so - I use G729 quite a bit since I travel alot
in adverse conditions. Interesting thing is, I can get less choppy audio
sometimes from my Vonage device using (what I suspect to be) Ulaw, while
either ulaw or G729 will still give choppy response at that moment from
I am not sure about E1 but it _should_ be the same. The Dialed Number is
usually transferred in 'a whole block' as the Telco passing the call to you has
already routed that call to you. What type of switch are you connected to??
Could your switch be expecting a ACK of some sort from *??
I am having some
trouble implementing OR login in the GotoIf statement. I have
followed the examples in the Wiki and I still am getting a syntax
error.
Essentially I want
to screen for CallerIDs set for "Anonymous" OR "Unknown Caller". If
either of these are true I want to send it to
the database exists because that is where the cards\PINs are stored, without the card\PIN I can not make a call, so the database exists, the permissions issue also may not be valid because when I set a connection charge the connection charge is recorded as billcost, but the cost of the call is not
Hi folks,
how could I play an announcement to the calling party as soon, as the
called party picked up. I would like to deploy an asterisk in an
environment, where a premium rate support-number is offered to customers
which do not want to pay a monthly support contract. In Germany, you are
On Wed, 2005-06-29 at 09:18 -0500, Jeremiah Millay wrote:
No I do not hear any clicking sound. Some calls come in perfect, and others
come in with some echo and sometimes artifacts, which I think might be caused
by jitter. Also it is mostly inbound calls that I have the problem with. If
you
um
do I paste the below info in to a file and name it something?
this looks really odd.
from what my screen reader is reading to me it looks like to be some sort of
script file or something
- Original Message -
From: Huddleston, Robert [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List
One might also conclude that during the outage the support people were
focusing on getting the system back up and were not near phones. At
least that is what I would bet on. Just a thought considering how most
of the smaller ITSPs seem to work.
Cheers,
Wiley
-Original Message-
From:
From email I just rec'd from Teliax:
Wed. 6/29/05 3am-6am Service Outage on voip-co1
'This morning starting at approximately 3am we experienced an unexpected
outage on proxy voip-co1. The outage was the result of a thread collision
between the proxy and it\\\'s database cluster. During this
On Wed, 2005-06-29 at 10:05 -0400, David Brodbeck wrote:
[snip]
2. I believe there are quite possibly two seperate bugs conflated in that
one item. There's the recording format problem (compressed formats are at
-6 or -10 dB compared to uncompressed) and possibly also a TDM-specific
recording
Hello,
I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that
comes with asterisk. I tried to place a call from h323 device into asterisk.
in extensions.conf, I routed the call to my sip phone. The sip phone was
already registered with asterisk. all the signaling looks ok to me.
I use them and I have another friend with them so far they are okay, support
is awesome, not any outages thus far and have been with them for about 3
weeks, not sure if they support iax or not, they do allow biod, prices are
good.
hth
- Original Message -
From: Chris Coulthurst
I would appreciate if someone can help me figure out what could be the
problem in receiving the digits from the telrad switch/pbx.
When you dial from the telrad, do you see any information being generated
on the asterisk CLI? You may have to increase verbosity of the console by
starting with
What experience can be shared about installing and running
the OrderlyQ application?
I have a bunch of queues set up and want to start adding
some additional apps and this one looked promising but I have very little java
experience and it doesnt seem to be running properly.
Jason
Go get app_intercept from www.pbxfreeware.org
/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”
On Jun 29, 2005, at 9:16 AM, Tony Nichols wrote:
I have been unable to get it to pickup sip-sip calls but if an
incoming zap rings I can
1 Master phone for a receptionist. Is there an easy way at the moment for
one of these bigger phones (cisco or whatever) to view the status of the
various lines etc? Some phone with an expansion board maybe?
Steve,
Flash Operators Panel is a very good tool for a receptionist if they
have a PC
Why not play the message BEFORE you call the Dail application. This
would also give the caller a chance to terminiate the call by hanging up
BEFORE your techs even get the call..
Hint: use the playback application
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hello,
I have the following situation:
I have a PRI with 200 DID numbers and I have set up
200 sip extensions that matches the last 4 digit of
the corresponding DID numbers so that when any of the
200 DID number is called, asterisk can pass the call
to the respective sip extension. Incomming has
I have tried increasing it to about verbosity level 11. Even then no sign of
digits coming in. My telrad technician also came in and checked everything
and certified the telrad is sending the digits as he switched cable on the
T1 card with another card (connected to Telco) and showed that the
Hi! Have you tried
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous]|$[${CALLERIDNAME} = Unknown Caller]]?3:5)
instead of
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
?
Giorgio
Keith O'Brien wrote:
I am having some trouble
Keith O'Brien wrote:
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
One too many $s?
exten = 5000,2,GotoIf($[${CALLERIDNAME} =
Doug
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Hi! Try
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} =
Anonymous]|$[${CALLERIDNAME} = Unknown Caller]]?3:5)
intead of
exten = 5000,2,GotoIf($[$[${CALLERIDNAME} = Anonymous] |
$[${CALLERIDNAME} = Unknown Caller]]?3:5)
Deleting spaces before and after ANd or OR logic worked for me.
Giorgio
If you need a fast solution put two gotoif
statements in a row, one to check for the first condition, another to check for
the next, you can leave out the redirect If the condition is not matched so it
just goes to the next priority.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL
I am assuming that you mean Telasip?
Don't expect to get any numbers ported over to them.
I have never been able to get anyone on the phone.
Can't say that I have had any technical issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
On Wed, 29 Jun 2005, Christian Händel wrote:
Hi,
if you are using the QSIG protocol for the interconnection between Asterisk
and the PBX, I have maybe a solution.
for the X100P you are using Zapata driver of asterisk. (with the switchtype
QSIG right?)
But for the eicon you use the capi
I have my systems running on ulaw, alaw or GSM. No other codecs. Myself I
even prefer the ulaw because of the quality.
I will look tomorrow a little bit further in the Linksys as I have 2 of them
here to test and so far I am very happy with them.
I will play a bit around with the settings and
Yes I was just reading that TeleSIP and Telasip are often mistaken, and was
just editing my dialplan for my mistakes!
When you meen porting numbers, I assume you are talking about LNP? If so,
not a problem for me anyway.
Chris Coulthurst
[EMAIL PROTECTED]
|-Original Message-
|From:
The CallerID that is seen by others on calls originating from your
PRI is set by your PRI provider; you have no control from Asterisk
about this as it gets overridden by the provider. You must contact
your carrier and ask them to set the CallerID for all PRI lines to
the desired
Chee Foong Chiew wrote:
I have a PRI with 200 DID numbers and I have set up
200 sip extensions that matches the last 4 digit of
the corresponding DID numbers so that when any of the
200 DID number is called, asterisk can pass the call
to the respective sip extension. Incomming has been
fine.
That would have been understandable, but their phone lines both gave 'number
unavailable' tones. I suppose this was because their lines use their own
service.
Malcolm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: Wednesday, June 29,
On Wed, 2005-06-29 at 10:40 +0200, Filippo Carone wrote:
* Hamish Whittal ([EMAIL PROTECTED]) ha scritto:
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now looing at sipXphone seem to
Hey federico,
I have it working in a rhe 3.0 start asterisk in debug and see what it spits
out probably a config issue.
/usr/sbin/asterisk -vv -g -dd -c
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: carlos at
Jason Kawakami wrote:
What experience can be shared about installing and running the
OrderlyQ application?
I have a bunch of queues set up and want to start adding some
additional apps and this one looked promising but I have very little
java experience and it doesn’t seem to be running
How does Asterisk calculate "BILLCOST", it appears the program for calculating BILLCOST may be wrong. Wherecan I locatethe program\file.Juan Luis Moyano [EMAIL PROTECTED] wrote:
Has anyone noticed that the primary key in the cdrs table is cardnum? soit won't record more than the first call made by
Seth Remington wrote:
On Wed, 2005-06-29 at 10:40 +0200, Filippo Carone wrote:
* Hamish Whittal ([EMAIL PROTECTED]) ha scritto:
Hi Folks,
I am wanting advise on a good soft-phone on Linux. I have looked at
Gnophone but cannot seem to get it to compile under debian sarge. I am
now
needhelp trying to figure out
why calls hang when using multple ports on Sangoma card.
we have 1 quad card with 3 T1 ports
configured, Port1 acts as connection to teleco (to our T1 PRI)
port 2 connects second system and
routes calls to port1
port 3 is Asterisk
pbx
calls all go in and out
Here is my experience in this area. Using asterisk on public IP no
nat, and no firewall. Polycom and Sipura clients inside NAT.
The sipura seems to be much more stable with almost everything, in
terms of asterisk being able to connect to it. I'm not using qualify
in sip.conf, but enabled them on
asterisk 1.0.7-r1 stable just came out on Gentoo but dtmfmode=inband is
still broken.
The work around is to use rfc2833
Was it fixed in ver. 1.0.8 ?
--
#Joseph
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You may also want to do some packet captures when you experience the
problem for both the Linksys and the Vonage ATA to see what they do
differently..
-Greg
On Wed, 2005-06-29 at 17:59 +0200, Marcel van Kaam, Fonetica wrote:
I have my systems running on ulaw, alaw or GSM. No other codecs.
The Lucent has fairly new cards in it. We just had firmware upgraded to 11.0.2
I believe. I'm thinking it is a configuration issue either in asterisk or the
lucent. Just wondering if anyone is running SIP between asterisk and a Lucent
TNT successfully without any echo or problems of that
Bryce Chidester wrote:
The CallerID that is seen by others on calls originating from your
PRI is set by your PRI provider; you have no control from Asterisk
about this as it gets overridden by the provider. You must contact
your carrier and ask them to set the CallerID for all PRI lines to
Ummm are you sure about this... I've seen people outpulse on PRI before
It's dependent on the carrier - was my understanding.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bryce Chidester
Sent: Wednesday, June 29, 2005 12:28 PM
To: Asterisk Users
Chee Foong Chiew wrote:
Hello,
I have the following situation:
I have a PRI with 200 DID numbers and I have set up
200 sip extensions that matches the last 4 digit of
the corresponding DID numbers so that when any of the
200 DID number is called, asterisk can pass the call
to the respective
1 - 100 of 203 matches
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