I’m trying to get a basic Asterisk
configuration together for ISDN incoming / outgoing calls. I have two Cisco
7905g phones working (at least talking to each other) and have purchased a
NETjet-S PCI ISDN card for routing calls to / from ISDN.
Clothes that you find hard to wear.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of hank smith
Sent: 07 June 2004 04:11
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zapata?
what is hadrware?
- Original Message -
From: "Richard Neese" <
On Sun, 06 Jun 2004 17:57:19 -0400, mag0007 <[EMAIL PROTECTED]> wrote:
Is there an interface for postgresql and asterisk?
It depends what you want to do with Asterisk & PostgreSQL, There is a
whole host of info on the mailing list archive about PostgreSQL:
http://tinyurl.com/328m8
There's also some
Hi everyone,
I want to get multiple DDI's and hunting across those DDI's in case one of the lines
is busy using analog phone lines. The system is for a large house so I want 3 x PSTN
lines. 3 x DDI's and the ability for those DDI's to be presented across all three
PSTN lines.
BT say you can'
Hello all,
Over the weekend, I setup and linked an Asterisk box at another site to the
Asterisk box here.
The phones here are a mixture of Cisco 7940/7960 and Grandstream BT-100
phones. The phones at the other end are Grandstream BT-100 SIP phones. The
Cisco phones run SIP 7.1 (upgraded last Frid
brian k. west wrote:
Its called searching the mailing list... Check the Makefile it does have
some indications of what to do on a VIA chip.
# Pentium & VIA processors optimize
#PROC=i586
also, depending on your version of GCC, there may be a bug that allows
emitting some instructions it shouldn't
Matt -
Your only tidy solution is to go for 1 ISDN line with some MSNs and an analogue line.
You will have to have an analogue line for ADSL anyway.. You will not get what you
want on analogue lines.
Rgds
Tim
Basingstoke
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECT
Drat! - The problem is we aren't on a digital exchange. We can't even get DSL here.
:-(
> Matt -
> Your only tidy solution is to go for 1 ISDN line with some MSNs and an analogue
> line. You will have to have an analogue line for ADSL anyway.. You will not get
> what you want on analogue li
Please, use the -Users lists for this kind of questions.
Now, what is the error message you get with chan_oh323?
Michael.
[EMAIL PROTECTED] wrote:
Hi Micheal,
I have some problems with chan_oh323. Asterisk seg faults and gives the
following error when loading chan_oh323. I am using the lates
You should be able to get ISDN though. I think BT provide this anywhere. Otherwise
your solution will be a Primary rate (ISDN30) with 8 channels.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt
Sent: 07 June 2004 10:19
To: [EMAIL PROTECTED]
Subje
Hi Folks,
just updated my current Dialplan available on
http://capi4linux.thepenguin.de/download/asterisk/config/
It now heavily uses ODBC database (MySQL) to hold most of the data like extensions and
incoming connection numbers. An example databasefile is also included.
The denylist has also
Gentoo has a page with all the allowed compile flags for different versions
of the via processors and gcc versions. (google is your friend :)
Zoa.
At 10:32 7/06/2004, you wrote:
brian k. west wrote:
Its called searching the mailing list... Check the Makefile it does have
some indications of what
Hi everyone
I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI
analog problem. (see [Asterisk-Users] Multiple DDI & Hunting on Analog Lines (UK) for
details) Does anyone on the list have any recent comments on reliability etc? I
would really appricated som
Tim, You're right we can get it but they can't do if for 30 days?! EEK! Don't you
just love BT. I'm considering the VOIP-Talk offering as an alternative as we have a
2mb leased line in here.
Thanks for the help
Matt
> You should be able to get ISDN though. I think BT provide this anywhere.
Hi!
I can highly recommend them, good quality and they seem to have very
competitive pricing as well.
Regards,
Maron Kristofersson
Iceland
Matt wrote:
Hi everyone
I'm interested in using the Telappliant/voip-talk offering as an alternative to my DDI
analog problem. (see [Asterisk-Users] Multip
The list price for the CON-SNT-CP7940 is £6.87 + VAT in the UK.
Chris
- Original Message -
From: "Nik Martin" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, June 06, 2004 2:29 AM
Subject: Re: [Asterisk-Users] Configuring cisco 7940
>
>
> Tony Hoyle wrote:
>
> >
> > That's n
From: Habiyakare
Aimable [mailto:[EMAIL PROTECTED]
Sent: Monday, June 07, 2004 11:49
AM
To:
'[EMAIL PROTECTED]'; 'gt'; '[EMAIL PROTECTED]'
Subject: Problem with Asterisk PRI
forwarding to SER
Hi all,
I have a problem. We have a phone system setup like this:
SIP phone
Hi all !
We have 3 NTBAs which are all going to our existing PBX. Our areacode is
06003 and our DDI enabled number 9141.
I want to exchange that PBX with Asterisk, but still struggle to get it
working.
My CAPI.CONF is currently like this:
[general]
nationalprefix=0
internationalprefix=00
rxga
Holger Schurig wrote:
Hi all !
We have 3 NTBAs which are all going to our existing PBX. Our areacode is
06003 and our DDI enabled number 9141.
I want to exchange that PBX with Asterisk, but still struggle to get it
working.
My CAPI.CONF is currently like this:
[general]
nationalprefix=0
interna
Hi Matt,
I use voiptalk via my DSL connection. It seems to work very well.
Originally I was using the connection in Sip mode, but had problems with
DTMF, I could only get it to work on outgoing calls, or incoming if I
changed mode, but not both. I switched to using IAX last week, which they
set up
Hi -
Anyone using zaphfc cards (bri-stuff-0.0.2) on a Portugal Telecom BRI service? I am
getting loads of errors and the audio is very distorted all the time.
System is a Dell PIII/933 MHZ with SCSI. Here in the UK we have perfect audio with a
Compaq PIII/500 MHZ SCSI system. Files are identi
Thanks, but that post doesn't cover the red alarms... Which is probably the
most important error message in the log.
When I said Middle of the night, I mean various times in the night.. 10pm,
2am, 3am, etc. It also happens in the middle of the day... 10am, 1:30pm,
etc. Its all over the map.
The
thanks everybody for your answers,
I finally decided to change my pbx configuration to:
- 1 x TE410P/TE405P quad T1
- 3 x TA750 CB (60 FXO)
- echo cancellation will be used on fxo channels
- up to 600 registered sip phone
- using ulaw as default codec for all connections
-
> At least for PTMP (Mehrgeräteanschluss) the MSN does not contain the
> areacode.
Thanks for the info. Somewhere in the list archive I saw a hint to use the
areacode as well. Now tried it without and had the same (non)success:
-- Executing Dial("SIP/hschurig-ced5", "CAPI/9141:0170811|60|T
>
> I'm interested in using the Telappliant/voip-talk offering as an
> alternative to my DDI analog problem. (see [Asterisk-Users] Multiple DDI
> & Hunting on Analog Lines (UK) for details) Does anyone on the list have
> any recent comments on reliability etc? I would really appricated some
> po
Chris Glover wrote:
I use voiptalk via my DSL connection. It seems to work very well.
Originally I was using the connection in Sip mode, but had problems with
DTMF, I could only get it to work on outgoing calls, or incoming if I
changed mode, but not both. I switched to using IAX last week, which t
> BTW: at a different NTBA (which is a PTMP for an ISDN router) I can call
> out and receive calls, so the chan_capi + fcpci + capi chain is known to
> work.
Did you configure your CAPI driver to run in PTP mode? (not PTMP mode)
Regards,
Julian Pawlowski
___
Duane [EMAIL PROTECTED] wrote:
> Chris Glover wrote:
> > I use voiptalk via my DSL connection. It seems to work very well.
> > Originally I was using the connection in Sip mode, but had problems with
> > DTMF, I could only get it to work on outgoing calls, or incoming if I
> > changed mode, but not
> Chris Glover wrote:
>
> > I use voiptalk via my DSL connection. It seems to work very well.
> > Originally I was using the connection in Sip mode, but had problems with
> > DTMF, I could only get it to work on outgoing calls, or incoming if I
> > changed mode, but not both. I switched to using IA
Duane wrote:
Chris Glover wrote:
I use voiptalk via my DSL connection. It seems to work very well.
Originally I was using the connection in Sip mode, but had problems with
DTMF, I could only get it to work on outgoing calls, or incoming if I
changed mode, but not both. I switched to using IAX last
The Cisco 7960 has a softkey called DND which when
pressed as the phone is ringing will sack the call to
voicemail. If you where using Cisco CME or CM you
can forward all calls to Vmail via CLI or GUI.
Kurt
__
Do you Yahoo!?
Friends.
> Did you configure your CAPI driver to run in PTP mode? (not PTMP mode)
No, I didn't. Which one would this be:
# lsmod
capi 17728 0
capifs 3436 0 [capi]
fcpci 531008 1
kernelcapi 29344 2 [capi fcpci]
capiutil
On Mon, 2004-06-07 at 07:07, Gary Franczyk wrote:
> Thanks, but that post doesn't cover the red alarms... Which is probably the
> most important error message in the log.
>
> When I said Middle of the night, I mean various times in the night.. 10pm,
> 2am, 3am, etc. It also happens in the middle
On my 7905s I can configure a "voicemail" number, which in turn
activates a "Messages" softkey. When pressed, it goes to that exten,
which in turn is configured to go to VoiceMailMain. Great, works like a
charm. However, the phones also have a "go to voicemail timeout" after
which, the phone divert
On Mon, 7 Jun 2004, Kurt wrote:
>
> The Cisco 7960 has a softkey called DND which when
> pressed as the phone is ringing will sack the call to
> voicemail. If you where using Cisco CME or CM you
> can forward all calls to Vmail via CLI or GUI.
It does? I have 7940s, and the DND is buried deep in
Hi all.
I'm looking to set up a fax via email service so that users can email a
specific mailbox and receive fax's to a specific mailbox. Can this be done?
I've had a look an SpanDSP and I think that's what I want but I'm not sure.
Cheers
Matt
___
As
Thanks to everyone for their comments :-)
Very useful
Matt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer
Sent: 07 June 2004 13:41
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voip-talk?
Duane wrote:
> Chris Glover wrote:
>
>> I
> Did you configure your CAPI driver to run in PTP mode? (not PTMP mode)
Hmm, I guess you brought me on the rigth track.
# cat /proc/capi/controllers/1
name fritz-pci
io 0xA800
irq 5
type A1
class10
ver_driver 3.11-02
ver_cardty
I'd do three separate 2 U Rackmounts, loaded with ASUS MB's with SATA raid
controllers, onboard nic and video
3.0 gig 800mHz FSB p4's
Slackware Linux, latest 2.4 kernel
1 gig ram
2 120 gig sata drives each server, in raid 1
1 t100p each
1 TA750 connected to each box
YOU NEED REDUNDANCY WITH THAT
Stephen Rosebush wrote:
I just got myself a GS HandyTone and it works great, it was a breeze to
setup. My only issue is I seem to be hearing a humming noise on the line
when I am in calls..
You have a short (or leak) to ground somewhere in the analog line.
S Besch
___
Daniel Jimenez wrote:
Hi all.
I've ordered a TDM400P with 4 FXO, but after using my X100P I'm thinking
about returning the TDM400P because of bad echo issues. If I do get the
echo issues I'll look at digital options.
My question: Is anyone using ISDN (BRI) in the states? I've heard
ISDN4LINUX d
On Mon, 2004-06-07 at 07:10, shabanip wrote:
> thanks everybody for your answers,
> I finally decided to change my pbx configuration to:
>
> - 1 x TE410P/TE405P quad T1
> - 3 x TA750 CB (60 FXO)
> - echo cancellation will be used on fxo channels
> - up to 600 registered sip phone
>
Nik Martin wrote:
I'd do three separate 2 U Rackmounts, loaded with ASUS MB's with SATA raid
controllers, onboard nic and video
3.0 gig 800mHz FSB p4's
Slackware Linux, latest 2.4 kernel
1 gig ram
2 120 gig sata drives each server, in raid 1
1 t100p each
1 TA750 connected to each box
YOU NEED RED
Brian Capouch wrote:
FYI to all you Grandstream users out there. I just fetched and
installed the 1.0.5.0 firmware, and it appears they have removed the
option to either do or not do SIP registration. Now it appears that one
is going to register with the server specified in the "SIP Server"
f
> The modules are from Linux 2.4.26, fcpci is from 03.11.02.
I remember that it's not possible to have an AVM Fritz card on an PTP mode ISDN line.
I think cards with HFC chipset are able to do so. Of cause you could also use an
active card with CAPI driver ;-)
Regards,
Julian Pawlowski
Daniel,
no you are not stupid. It's just that very few people have had a BRI
experience in the US. The only CAPI card with support for NI-1 is the
Eicon DIVA Server (single BRI or four BRI). They work with NI-1 BRIs
and chan_capi. And chan_capi supports the active echo cancelation on
the Eicon car
On Monday June 7 2004 09:22, Daniel Jimenez wrote:
> No one has any comments on this? No recommendations, or "you are stupid
> for trying that" or anything?
How about, "I'm interested too?" I've had an ISDN line for ages, but I've
always used a Terminal adapter. Now that my office is looking in
Brian Capouch wrote:
Tomas Prybil wrote:
Brian Capouch wrote:
FYI to all you Grandstream users out there. I just fetched and
installed the 1.0.5.0 firmware, and it appears they have removed the
option to either do or not do SIP registration. Now it appears that
one is going to register with th
... might as well use hylafax.
Iain
--On Monday, June 7, 2004 2:15 pm +0100 Matt <[EMAIL PROTECTED]> wrote:
Hi all.
I'm looking to set up a fax via email service so that users can email a
specific mailbox and receive fax's to a specific mailbox. Can this be
done? I've had a look an SpanDSP and I
>
> Would this mean three separate voicemail systems? Why not diskless
> servers with /var on an nsf mount from a file server?
>
> --
Good point, a fourth server with the sata raid subsystem would offer a much
more efficient and administratable system.
__
Daniel Jimenez wrote:
Daniel Jimenez wrote:
Hi all.
I've ordered a TDM400P with 4 FXO, but after using my X100P I'm
thinking about returning the TDM400P because of bad echo issues. If I
do get the echo issues I'll look at digital options.
My question: Is anyone using ISDN (BRI) in the states? I'
Stephen R. Besch wrote:
I know that GS monitors this list. I just hope that they take this
seriously. Breaking the no register option is a really serious, and
idiotic, mistake. They really are obliged to fix this as soon as
absolutely possible.
Some fairly serious flaws in either asterisk or th
Hi,
you can use the AVM Fritz card in P2P mode, if you use the new mISDN
capi layer with chan_capi. You dont have to rely on a kernel-tainting
module anymore.
Of course it makes sense to use a hfc-pci card instead, since it will
provide your box with zaptel timing and echo cancelation!
best regar
I have a GS BT102 which when receiving calls that include DTMF tones
only have short clipped beeps.
Is asterisk not passing the DTMF info on to the phone or is the tone not
bing generated by the phone?
I am trying to check that I am getting the DTMF for an application where
the length that a key
[EMAIL PROTECTED] wrote:
> On Mon, 7 Jun 2004, Kurt wrote:
> > The Cisco 7960 has a softkey called DND which when
> > pressed as the phone is ringing will sack the call to
> > voicemail. If you where using Cisco CME or CM you
> > can forward all calls to Vmail via CLI or GUI.
> >
> It does? I have
On Sun, Jun 06, 2004 at 04:25:32PM -0400, Tim Sailer wrote:
> On Tue, Jun 01, 2004 at 07:49:29PM -0500, Yelson Vivas wrote:
> > Hi everybody
> >
> > i have a problem trying to connect an incomming phone call from pstn to my
> > (soft phone) iaxcomm, the phone rings but when i try to answer the ca
Iain Stevenson wrote:
... might as well use hylafax.
Yes, well, that requires using modems and having Asterisk send the audio
back in/out as analog. It would be really fantastic if someone could
come up with an app for Asterisk that emulated a Class 1 FAX modem and
allowed Hylafax to talk to it
> I remember that it's not possible to have an AVM Fritz card on an PTP
> mode ISDN line. I think cards with HFC chipset are able to do so. Of
> cause you could also use an active card with CAPI driver ;-)
I read something like this in the mailing list archive, but they were
referring to isdn4lin
On Mon, 2004-06-07 at 07:14, Duane wrote:
> exten => _.,1,Dial(IAX2/username:[EMAIL PROTECTED]/number)
As far as I can tell Asterisk will not use iax.conf if you do it that
way. (specifically if you use @host rather than @iaxconfentry)
--Eric
--
Eric Wieling * BTEL Consulting * 504-899
Hi !
I've fetched a spare AVM B1 card from the cellar, and installed it. After
"modprobe b1pci" I did "capiinit" and capiinit moaned about a missing
t1.b4.
So I search the web and found one at
http://www.avm.de/ftp/cardware/b1/x_misc/ddi/. When I now look at the
controller, I finally see p2p-
At 07:52 AM 6/7/2004, you wrote:
The Cisco 7960 has a softkey called DND which when
pressed as the phone is ringing will sack the call to
voicemail. If you where using Cisco CME or CM you
can forward all calls to Vmail via CLI or GUI.
Kurt,
What version of SIP firmware are you running on the 7960
I do not think this is the case, I e-mailed the company I ordered it off
of and they're going to send me a new device, the thing is that I've
tried many different analog telephones, tried many different firmware
versions, tried a changable voltage adapter I have on the unit, I hear
the buzzing
Hi all,
can Asterisk be used as a videoconference server or the like when
enabling 'videosupport=yes' ? if so, how do I use it? is there any
recommended SIP/Video-client for both Windows and Linux?
Thanks,
Martin
___
Asterisk-Users mailing list
[EMAIL
Scott Nelson wrote:
On Monday June 7 2004 09:22, Daniel Jimenez wrote:
No one has any comments on this? No recommendations, or "you are stupid
for trying that" or anything?
How about, "I'm interested too?" I've had an ISDN line for ages, but I've
always used a Terminal adapter. Now that
> you can use the AVM Fritz card in P2P mode, if you use the new mISDN
> capi layer with chan_capi. You dont have to rely on a kernel-tainting
> module anymore.
Thanks for this info. About to change the cards ...
> Of course it makes sense to use a hfc-pci card instead, since it will
> provide yo
Il 10:34, martedì 01 giugno 2004, Chris Bond ha scritto:
> Are they any issues still with hyperthreading processors, I've read and
> been told by a few people to make sure its disabled in bios if I want to
> use * on a hyperthreading machine.
>
> Kind Regards,
> Chris Bond
I'm using asterisk on hyp
Hello Holger,
I guess that you must configure your /etc/capi.conf
options = p2p..
Bye
Felix
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Holger Schurig
> Sent: Monday, June 07, 2004 5:04 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-
Nik Martin wrote:
http://www.ams.net/public/products/product_info.cfm?Product_ID=7993
This was from a google search, but it's a little high.
Search for con-snt-7940
CON-SNT-7940 UK gives no results.
A search for CON-SNT-CP7940 UK gives a single result
(triointernational.com)... they look cheap bu
Matt wrote:
Hi everyone,
I want to get multiple DDI's and hunting across those DDI's in case one of the lines
is busy using analog phone lines. The system is for a large house so I want 3 x PSTN
lines. 3 x DDI's and the ability for those DDI's to be presented across all three
PSTN lines.
BT sa
Hi!
> Some fairly serious flaws in either asterisk or the GS iLBC code as
> well. If I call BT to BT using iLBC with asterisk playing proxy etc,
> often one end can't hear the other, although it sounds fine on the end
> that can hear.
Interesting - I hadn't yet figured out that this was relate
I can't speak for "general" cases, but I know when I've tried to set
videosupport=yes, my as5300 can no longer speak w/*. I wonder if it can
be set per peer - haven't tried that...
jeremy jones
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> M
Am Mo, 2004-06-07 um 17.33 schrieb Holger Schurig:
> > Of course it makes sense to use a hfc-pci card instead, since it will
> > provide your box with zaptel timing and echo cancelation!
>
> Still waiting on this card.
>
> Hehe, eventually I'll even get your quad-card. But beforehand I have to
>
Me too, in fact it's just a little better with HT on.
MATT---
-Original Message-
From: Diego Ercolani [mailto:[EMAIL PROTECTED]
Sent: Monday, June 07, 2004 11:34 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hyperthreading?
Il 10:34, martedì 01 giugno 2004, Chris Bond ha scritt
Stephen R. Besch wrote:
Any other features you've empirical found out but that?
I note the tones with 1.0.5.0 are all files >64Kb.
the 1.0.5.0 version anyway. It hasn't fixed any of the outstanding
issues (at least those related to use with "*", or added any really
useful functionality.
Two thi
I compiled libtiff version 3.6.1 and spandsp and spandsp version k. When
trying to load asterisk I get the folloein error:
Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading
module app_dtmftotext.so failed!
Ouch ... error while writing audio data: : Broken pipe
[EMAIL PROTECTED] ro
Hi!
> Did you try out the new ring tones? One of them contains a regular ring,
> followed by a voice announcing the caller id of the calling party. VERY
> neat. It seems the ring tones can contain not only sound, but also
> either code to be executed, or a flag to announce the caller id.
?? Wh
Julien Levi wrote:
DDI on analog lines is more complicated and I believe there is a set-up
charge of over £1000 (see
http://www.serviceview.bt.com/list/current/docs/Exch_Lines.boo/1228.htm
) If you must have multiple numbers you'd be better off with an isdn line.
To emulate a couple of DDI numb
I have a SIP phone (Cisco 7960) registered to 2 * pbx, is there anyway to control
which * pbx will be used for making calls? I know by default the cisco will use
and I want to change that.
thanks
micko
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
ht
With the newer firmware versions of the Granstream, just wondering what
people suggest/recommend for the orders of codec for greatest compatibility?
Have had a few issues recently with one-way audio when calling to a ciso ata
186 from my BT-100.
Regards,
Ian.
---
Outgoing mail is scanned but n
On 7-Jun-04, at 10:10 AM, Manuel Marin Garcia wrote:
I compiled libtiff version 3.6.1 and spandsp and spandsp version k.
When
trying to load asterisk I get the folloein error:
Jun 7 10:15:03 WARNING[16384]: loader.c:408 load_modules: Loading
module app_dtmftotext.so failed!
did " /usr/lib/asteri
Hi Greg,
Thanks for the fromuser= and fromdomain= hints. Adding those cleared
up the problem and we can make and receive calls now.
However, as with Voiceglo, BroadVoice seems to ignore DTMF on
incoming calls. It does send DTMF on outgoing calls. I've tried all
three dtmfmode values in sip.conf lea
Klaus-Peter Junghanns wrote:
> you can use the AVM Fritz card in P2P mode, if you use the new mISDN
> capi layer with chan_capi.
That's a good idea in principle but in my opinion it's sometimes too buggy at the
moment so productive usage is not recommendable. I made those experiences when I tried
I use GS and have no problem with iLBC. as for the registry problem I have
talked to rich and he is looking into it...
On June 7, 2004 10:41 am, Duane wrote:
> Stephen R. Besch wrote:
> > I know that GS monitors this list. I just hope that they take this
> > seriously. Breaking the no register
Richard Neese wrote:
I use GS and have no problem with iLBC. as for the registry problem I have
talked to rich and he is looking into it...
At least 2 others are having the same issue...
--
Best regards,
Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globa
Ian Pilkington wrote:
With the newer firmware versions of the Granstream, just wondering what
people suggest/recommend for the orders of codec for greatest compatibility?
Have had a few issues recently with one-way audio when calling to a ciso ata
186 from my BT-100.
Are you using iLBC at all?
--
B
it seems that the use of renivite in sip peer configuration is very much
dependent on sip endpoint
have read of what seems defnite "no no" when using the cisco ata 186
it seems eminently preferable from a networking / performance view for the
media data transfer to be between the two endpoints an
Hello
I have the follow situatuion:
< ISDN >
|
|
V
E100P
|| IAX2 / g729A || T100P
| Asterisk1 |- - - - - - - - - - - - - - > | Asterisk2 | - - - - -
-> |--|
| | | | | Zhone|
-
Hello,
I am relatively new to Asterisk and I need to
compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed
it in the codecs directory, but apparently Asterisk needs a rather different
library than the one provided from ITU.
As I've seen in the mailing list archi
line 1 is always default for calls when a line isn't selected prior to
dialing. Best bet would just be reverse the order you have them on the
Cisco line 1 as primary line 2 as secondary.
On Mon, 2004-06-07 at 12:57, Dragan Mickovic wrote:
> I have a SIP phone (Cisco 7960) registered to 2 * pbx, is
Checkout http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
and http://www.voip-info.org/wiki-Asterisk+t+extension
You could use extention t, which is reached after dial times out.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon Brow
My office is investigating using an Asterisk PBX and also going to a VOIP
provider for our main phone connections, but one of the tricky things is that
we need to have outbound and inbound modem calls (fax too).
I see a lot of talk about faxes but no mention of modems on this list. I get
the i
Hello All,
I am battling a problem I don't know how to fix... Here is the
scenario:
Fractionated T1 with 1-6 channels voice, 21-24 channels data.
Comes into a box with Digium T100P, splits off data channes with HDLC,
to devicec pvc0,
This works fine, but I have a routing problem getting to my m
exten => _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _NXXNXX,3,Congestion
The above will attempt to dial out your Zap interface first. If that
fails, it will dial out using "username" for the username and the
password, IP address
I would use:
exten => _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _NXXNXX,3,Congestion
exten => _NXXNXX,102,1,Busy
exten => _NXXNXX,103,1,Busy
That way if number you dial is busy it will not immediately try dialing
the same
Scott,
I haven't done *any* research into this but.. I'm using a Sipura2000 and
I have my laptop modem attached to it. I'm forcing alaw or ulaw encoding
on the line. As soon as the modems begin to "train" I get what sounds
like "static" instantly "come on". Very unusual. Not sure what the
source of
Not sure I understand what all you are doing here, but you need
your gateway to be on the same subnet as your main ip address.
Why do you assign 2 ip's to your pvc?
If ping's are allow, sometimes it helps to use traceroute to
see the path a packet is taking.
On Mon, 2004-06-07 at 15:21, Michael
Hallo Holger Schurig
On Mon, 7 Jun 2004 15:17:17 +0200 you wrote:
> > Did you configure your CAPI driver to run in PTP mode? (not PTMP mode)
>
> Hmm, I guess you brought me on the rigth track.
>
> # cat /proc/capi/controllers/1
> name fritz-pci
> io 0xA800
> irq
Grandstream Phones do have a bug where they sometimes send an UDP packet to
port 0 instead of port 5060. The effect is that you see 5 times the error
message
WARNING4101: chan_sip.c:601 retrans_pkt: Maximum retries exceeded on call
[EMAIL PROTECTED] for seqno 22209 (Critical Response)
on the as
Richard Neese wrote:
I use GS and have no problem with iLBC. as for the registry problem I have
talked to rich and he is looking into it...
Who is "rich?"
B.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/ast
Hallo Holger Schurig
On Mon, 7 Jun 2004 17:08:53 +0200 you wrote:
> Hi !
>
> I've fetched a spare AVM B1 card from the cellar, and installed it.
> After "modprobe b1pci" I did "capiinit" and capiinit moaned about a
> missing t1.b4.
Dont use modprobe, use "capiinit start" !
Configu
1 - 100 of 165 matches
Mail list logo
