I ordered one of these from Grandstream to evaluate, along with a Handy-Tone
488. They both arrived today (I'm in Australia).
After having played with the GXP-2000 for about 2 hours, my advice is to
wait. Whilst it looks and feels a lot nicer than a Budgetone (approaching
the build quality of
I'm running 1.0.5.22 (beta), and it is the best version I've found to date.
I notice .23 is also available.
http://gs-firmware.gratissip.dk/
- Original Message -
From: el Flynn [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 16, 2005 3:20 PM
Subject:
Beware of 1.0.5.23 Grandstream firmware. When I installed it, SIP
registration stopped altogether. Going back to .22 fixed things again.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I use 1.0.5.22
Can't fault it.
Don't be afraid of upgrading to a newer version, you can always downgrade
again.
- Original Message -
From: Paul Fielding [EMAIL PROTECTED]
Sent: Tuesday, January 18, 2005 2:34 AM
Subject: [Asterisk-Users] Best Grandstream firmware to use?
I've seen lots
Do some reading about contexts in *. Basically, you
want all "public" sip requests to land in a dialplan context that has no access
to PSTN, and requests from your own SER box(es) to land in another context (that
DOES have access to PSTN).
You can achieve this by adding an entry to your
Further to this, does anyone know if there is a simple way to set the party
priority in codec negotiation? (NOT the codec priority)
In other words, I want the calling (client) preferences to be considered
FIRST.
Currently, my logs show
Accepting AUTHENTICATED call from 203.89.xxx.xx:
I know that this has been a hot topic in the past, and that there are a
number of views about the overall suitability of RADIUS, which billing
methodology is best, what is the meaning of life and such...
What I would like to know is has anyone found an open-source billing
platform that
When I dial a PSTN (via * with Digium Quad-E1) number, I hear two
sumultaneous ring tones. One is coming from *, the other I assume is
from the ATA. Is there an intelligent way to get around this?
___
Asterisk-Users mailing list
Unfortunately, the polycom phone was a bad choice
from a beginner's perspective, as it loads it's firmware/config from a boot
server every time it's powered up, and won't work without it.
Get a softphone (firefly, xten) running on a PC
first (to ensure that your asterisk is OK).
Take your
Sorry If I misled people. I probably need to re-investigate the polycom
phones. Obviously I was wrong about their needing a boot server to function.
Jerry wrote:
On Mar 29, 2005, at 5:34 PM, Rod Bacon wrote:
Unfortunately, the polycom phone was a bad choice from a beginner's
perspective
I am experiencing problems with my TE410P specifically relating to the
outgoing call volume. Incoming is fine, but outgoing is way too soft.
The gain settings in zapata.conf do nothing. I have tried them at 100,
-100 and many places in between, all with no audible difference. Am I
missing
?
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
Beautiful.
Thanks to those who replied.
- Original Message -
From: Matthew Boehm [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 1:27 AM
Subject: Re: [Asterisk-Users] Zaptel Periodic Reset
Rod
SPA 2000 - Miltiple Ring Tones
Rod Bacon wrote:
When I dial a PSTN (via * with Digium Quad-E1) number, I hear two
sumultaneous ring tones. One is coming from *, the other I assume is from
the ATA. Is there an intelligent way to get around this?
I am getting the exact same thing. It sounds like
... and the grandstream phones support distinctive ring.
- Original Message -
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 10:51 AM
Subject: Re: [Asterisk-Users] Can this be done?
Maybe you should post this in the webmin forum.
- Original Message -
From: Mike Hammett [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 01, 2005 9:25 AM
Subject: [Asterisk-Users] Webmin
How do I install the asterisk module for webmin?
In case anyone's interested, I have all the Digium and Iptel lists (asterisk
and ser) in a program called LURKER. I downloaded all the archives,
populated the database and now get updates dynamically.
In other words, I have a fully-threaded and searchable web version of the
lists without
Hello all. I am trying to architect a large-scale solution and need to
know some of the capabilities of * using realtime configuration (I have
read some docuemntation on the WIKI, but have not yet played with Realtime).
As the supporting docco is a little light-on at the moment, I'm hoping
to
This is quite interesting.
I tested calls to 2 mobiles that I knew were off, and not diverted to
voicemail. 1 with Telstra, the other with vodafone (I'm in Australia). Via
ISDN, both calls were shown as unanswered by asterisk. When the calls went
to voicemail, the call was deemed to be
Empty yer bloody mailbox...
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Can you be more specific?
What are you trying to achieve with the creation of such groups?
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:50 PM
Subject: [Asterisk-Users] how to configure groups using a sip phone
hi friends
The term RTCache has never been mentioned in the WIKI or these forums. I
assume that it's some sort of function to speed up realtime db access by
keeping transactions in RAM and writing periodically? If so, I can
understand why this would need to be flushed.
- Original Message -
From:
No.
- Original Message -
From: Alexandre Charles [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Monday, April 04, 2005 3:48 PM
Subject: [Asterisk-Users] V92 modem with asterisk
Hi everyone,
I just install Linux and asterisk on one of my pc. I
want to run some basic
Over the last few weeks/months I have been testing
phones and ATAs from Grandstream (BT101, GXP2000, 286, 488), SNOM (190), Zyxel
(Piece of Crap), Sipura (SPA-2000, SPA-841) and I personally feel that the
Sipura SPA-841 is the best value, good quality phone that I have used. I haven't
used
in zapata.
Deepak Dhiman
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon
Sent: Monday, April 04, 2005 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] how to configure groups using a sip phone
Can you
I have found that it matters what order things go in the zapata.conf file.
My echocancel and gain settings were being ignored until I moved them
further up the [channels] section.
Is this as designed, or is it a bug?
___
Asterisk-Users mailing list
Has anyone written any scripts to convert .conf files to realtime
database configs?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
thing is that it was working at some point yesterday.
Can anyone suggest a place to start looking?
Also, how do I enable debug logging so I can see the realtime info in
the * CLI or logs?
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground
:09:35 DEBUG[3672]: chan_sip.c:937 __sip_autodestruct: Auto
destroying call '[EMAIL PROTECTED]'
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 9:06 AM
Subject: [Asterisk-Users] Realtime UPDATE
I'm sorry
@lists.digium.com
Sent: Thursday, April 07, 2005 9:36 AM
Subject: RE: [Asterisk-Users] Realtime UPDATE
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Rod Bacon
Envoyé : jeudi 7 avril 2005 01:06
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Realtime
It also looks to me, on the test I just ran, that it will try calling
twice, even though MaxRetries is set to 1.
That's right. If you only want it to call once, set max REtries to 0.
-Andy
On Apr 6, 2005 5:30 PM, Ronald Wiplinger [EMAIL PROTECTED] wrote:
We use wakeup calls for reminders, but it
Bugger...
That explains it...
- Original Message -
From: Thierry Wehr [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Sunday, March 20, 2005 2:08 AM
Subject: [Asterisk-Users] No more updates of IP address and port in
I'd personally be using Ethereal to look inside the SIP messages for the SDP
info and checking the source/destination of the resultant RTP stream.
One-way audio is typical of NAT issues. Although you are running a VPN (of
sorts) I suspect that your SDP messages are getting screwed up somewhere.
I've tested about a dozen of them, and find firefly one of the best (others
have more features, but I find firefly is a good mix of
quality/features/performance). Make sure you get the third-party firefly
though, not the one that's limited to virbiage.
Try here...
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
de Rod Bacon
Envoyé : jeudi 7 avril 2005 01:06
À : asterisk-users@lists.digium.com
Objet : [Asterisk-Users] Realtime UPDATE
My problem is that upon registration, the UA's IP address and
port information isn't being written to the MYSQL
Me either...
- Original Message -
From: Cameron Beattie [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 07, 2005 4:07 PM
Subject: Re: [Asterisk-Users] Beeps during Sip to Sip phone calls
I have a
.
-Matthew
Rod Bacon wrote:
I have playing with these, to no effect.
I am assuming that threre is indeed a bug.
Arnaud PIGNARD wrote:
Got the same problem, and i rollback to older version (before RT
cache patch).
There is a combinaison where you will get successfull update but for
me, realtime
I want to use the appradius module (specifically the cdr_radius.so)
module to dump asterisk CDR to a RADIUS server (in addition to the local
SQL database). I don't want the authorisation component, only the
CDR-RADIUS function.
I have downloaded, compiled and installed the software without
details.
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
2:55 PM
Subject: Re: [Asterisk-Users] Off Topic - Employment Opportunity -
PERL,Melbourne, AU.
Rod Bacon wrote:
Sorry if this is off-topic, but I know there's a quite a few smart people
who frequent these groups, and I was thinking that it'd be a good place
to ask.
We have an opening
It's my opinion that whilst asterisk indeed has some fax capability, it's
not a business-grade fax platform. If faxes are indeed as important to your
business as you suggest, I'd be inclinded to look for alternatives.
- Original Message -
From: Marc [EMAIL PROTECTED]
To: 'Asterisk
I don't know if what you're trying to do is possible, but the easiest way to
check would be to take a look at the raw packets on the ethernet interface
of your * server once a call is in progress. If indeed the RTP can be handed
off to the 2 endpoints, you should only see SIP traffic at your
My experience with VOIP to date has surounded SER and Asterisk, SIP and
IAX. It would appear as though I am about to be inducted into the world
of H.323 and as such I am interested in hearing from anyone who is using
Asterisk extensively in a mixed protocol environment, especially in
using
I found new firmware at:
ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip
The phone is now finally (almost) useful. Still a cheap piece of crap,
with new bugs to replace the old, but at least it sort of works now.
___
Asterisk-Users
.
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, April 12, 2005 2:45 PM
Subject: [Asterisk-Users] Zyxel P2000W Finally (Almost) Working
I found new firmware at:
ftp://ftp.us.zyxel.com/P2000W/firmware/P2000W_WJ.00.10_Standard.zip
on.)
Rod Bacon wrote:
I think there's a more sinister bug in play somewhere. The phones are
on the same LAN. It was working when I only had a single asterisk
server using the database, and seemed to stop when I added a second
server. I know this doesn't make any sense...
OK. Lemme picture
In addition to making sure that echo cancellation is enabled on the
interface(s) in question, you will also need to play with the gain settings.
Specifically, try turning down the rxgain. I dropped mine to -10.0, and the
echo disappeared altogether.
The problem was then that incoming voice was
hear me with an echo. That leads me to believe that it hs nothing to do
with the zapata stuff. It is somewhere between my SIP phone as Asterisk.
-N
Rod Bacon wrote:
In addition to making sure that echo cancellation is enabled on the
interface(s) in question, you will also need to play
The firmware at...
http://www.snom.com/download/share/snom190-3.60b-SIP-j.bin
seems to have fixed quite a few SNOM 190 bugs. Worth a try.
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria
Any ideas what bugs are fixed?
there dosn't seem to be any release notes.
-Shaun
Rod Bacon wrote:
The firmware at...
http://www.snom.com/download/share/snom190-3.60b-SIP-j.bin
seems to have fixed quite a few SNOM 190 bugs. Worth a try.
___
Asterisk-Users
I don't know about your * source code, but mine clearly states that you must
use
OpenH323 v1.15.1 and PWLib v1.8.1
- Original Message -
From: Jose R. Ortiz Ubarri [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Yes. It needs a timing source (ztdummy).
- Original Message -
From: Xu Wang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 10:34 AM
Subject: [Asterisk-Users] does meetme need ztdummy
Hello
I
Sorry for my last post. I see we are talking about different H.323 channel
software.
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 8:44 AM
Subject: Re
on by
Digium. Has anyone seen/tested this yet, or is it still vapourware?
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
I have been frustrated by a variety of zyxel issues/products and have found
the best solution for all of them lies in a cylindrical receptacle that sits
beside my desk...
- Original Message -
From: aza [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
Idle channels DO restart periodically, so this is
fine. You are correct in your assumptions that you need to change your span
numbering.
I believe it's because your telco is starting at 0
instead of 1.
I had this problem also, and had to change mine to
span=1,1,0,ccs,hdb3,crc4
Search
Conferencing (MeetMe application)
- Original Message -
From: Paul Fielding [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 12:46 PM
Subject: Re: [Asterisk-Users] ztdummy
Ok, Here's my ztdummy
Are the calls coming from SIP or PSTN?
- Original Message -
From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 3:56 PM
Subject: [Asterisk-Users] RTP not being sent by asterisk
Does maximum debugging show anything?
- Original Message -
From: trixter http://www.0xdecafbad.com; [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, April 14, 2005 3:56 PM
Subject: [Asterisk-Users] RTP not being sent by asterisk
talking crap!)
I don't know if this is relevant to your situation in any way, but it's
worth consideration.
trixter http://www.0xdecafbad.com wrote:
On Thu, 2005-04-14 at 16:15 +1000, Rod Bacon wrote:
Are the calls coming from SIP or PSTN?
from sip, and I can see packets going from sip - asterisk
Matt, can I assume from your silence that you concurr with my thinking that
realtime is in fact broken, or is it that I am using it incorrectly?
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: Matthew Boehm [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial
We have just had an ISDN PRI service commissioned here in AU (using Powertel
as provider). I have called them and ensured that we have the ability to set
Caller ID on our service to any number in our 100 number block, and I have
been assured that everything is OK from their end (unlikely). Every
SIOD ERROR: wrong type of argument to car : wholeutt
Try changing your festival.scm to the following:
(Note the extra () on the 4th last line).
(define (tts_textasterisk string mode)
(tts_textasterisk STRING MODE)
Apply tts to STRING. This function is specifically designed for
use in server
where these extra 4 characters are coming from?
Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number passed network screening
(1) '0386172169'
==
Rod
have tried all possible values for prilocaldialplan (Calling number plan)
but nothing makes any difference. The LEN field is always 14 (no matter what
string I send).
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St
Would someone mind doing an intense debug on their ISDN PRI
and see what LEN (length) the calling number field is being sent? Maybe everyone
is sending 14 characters, and my Telco is just fussier than most.
==
Rod Bacon - VOIP Systems Engineer
I know this is casting a wide net, but If you were charged
with building a large, public VOIP network with multiple PSTN gateways, the
capacity to carry a lot of traffic and bill clients accurately, what pieces (brands,
makes, models) would you use to assemble the solution? Assume that $$$
Dean, which M would you have me read? I have
been Ring The Fing M for
several months now, and have tried a number of products personally, but my very
point is that it is physically impossible to test ALL PSTN gateways, ALL
softswitches, ALL radius/billing solutions myself. I was counting
Subject: Re: [Asterisk-Users] Outbound Caller ID on PRI
Rod Bacon wrote:
Would someone mind doing an intense debug on their ISDN PRI and see what
LEN (length) the calling number field is being sent? Maybe everyone is
sending 14 characters, and my Telco is just fussier than most.
Not asterisk
I have googled until blie in the face, WiKi'd until physically exhausted
and searched through every Asterisk repository that I can find, all to
no avail...
No matter which version of SpanDSP I use, with which version of libtiff,
Asterisk, ... I simply cannot send faxes.
I can receive faxes
@lists.digium.com
Sent: Wednesday, February 23, 2005 6:28 PM
Subject: Re: [Asterisk-Users] SpanDSP - Still can't send
On Wed, 23 Feb 2005, Rod Bacon wrote:
No matter which version of SpanDSP I use, with which version of libtiff,
Asterisk, ... I simply cannot send faxes.
Did you remember to add the caller option
'Zap/1-1'
- Original Message -
From: Peter Svensson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 24, 2005 8:38 AM
Subject: Re: [Asterisk-Users] SpanDSP - Still can't send
On Thu, 24 Feb 2005, Rod
option. Don't confuse caller with
sender. The machine which makes a call can be either the one sending a
FAX, or one picking up a FAX by polling. That is the reason the caller
option exists.
Regards,
Steve
Rod Bacon wrote:
My understanding is that this is only required when using it inside
There are so many possibilities here...
First, get hold of every whitepaper that you can find on NAT Traversal in
SIP, so you at least understand the issue.
In my case, with Grandstream phones, I set them to use STUN, and make sure
that they use a dynamic port.
Your ultimate solution will be
their parameters. That seems to have disappeared from the source
code of both rxfax and txfax. I just put it back in.
Regards,
Steve
Rod Bacon wrote:
From my personal experience, the 'weird ideas' come from a lack of
consistent documentation. The caller option, and the use thereof is not
clearly explained
I agree. The following commands may also be of use...
.
.
exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
.
.
- Original Message -
From: Greg Hill [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I've been playing with a variety of them over the past month.
The 'candidates' I've got down to are X-Lite, Firefly and SJphone. They all
have strengths and weaknesses, and all behave differently behind my firewall
(STUN client differences?). So far, I am happiest with the performance of
I too am having the same problem with CVS from last
night. From my debugging, * never attempts to start MOH. Anyone else found
this?
- Original Message -
From:
Krystian Filiks
To: asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 1:46
PM
Subject:
I am just playing with a SNOM 190. Overall, I'm
very impressed with the quality of the unit and the feature set. I am running
the latest firmware (snom190-SIP 3.57u) and
the asterisk CVS from last night (1/3/05).
The only problem that I've encountered so far is
with Call Forwarding, which
.
- Original Message -
From: Nils Ohlmeier [EMAIL PROTECTED]
To: Rod Bacon [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Sent: Tuesday, March 01, 2005 10:03 PM
Subject: Re: [Asterisk-Users] SNOM Call Diversion
Hello,
the Call Forwarding Always was not working in that firmware version
SIPURA and SNOM phones to it as their "MOH server", and everything works sweetly
(notwithstanding a bug in the SNOM firmware).
- Original Message -
From:
Rod Bacon
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, March 01, 2005
I have the openline card with a recent CVS and I can't seem to get CallerID
to work.
Debugging shows;
vpb/1-1: New call for context [local_pstn]
Caller ID disabled
vpb.conf is...
[general]
type = v4pci
cards = 1
[interfaces]
echocancel = on
board = 1
txhwgain = 12
txgain = 12
context =
Some more info, I now have it attempting ID at least.
vpb/1-1: New call for context [local_pstn]
Using VPB Caller ID
CID record - start
-- Executing SetMusicOnHold(vpb/1-1, default) in new stack
-- Executing Dial(vpb/1-1, sip/1232|20|tTwW) in new stack
-- Called 1232
--
There are 2 issues here. Firstly, the timing source
you are looking for is ZTDUMMY. It's in the zaptel source but you may need to
edit the Makefile and uncomment it (remove the # from in front of ztdummy) in
order for it to be compiled and installed. Once you've got it compiled, Just
I have not seen this myself, but I assume it's doen with the sendurl and
sendtext dialplan commands. The PC would need a SIP client installed that
could accept text/url data, then the diaplan (or AGI script, most likely)
could perform the necessary lookup, send the data to the correct PC
Please ignore my stupidity (once more). It turns out that the analog PBX
port that my Voicetronix card is plugged into does NOT support callerid,
hence the lack of detection in asterisk.
My kingdom for a PSTN line...
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: James
(failing to hang-up).
P.S. It's NOT a wireless problem. I'm sitting right on top of the access
point, and have a strong, clean signal.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600
You want him to plug a loopback plug into the front of his Dell server? I
think he means the system status LED on the server, not the port LED on the
Zaptel...
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 04, 2005
on this?
--
==
Rod Bacon - VOIP Systems Engineer
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
-bits...
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 23, 2005 2:55 PM
Subject: [Asterisk-Users] SpanDSP - Still can't send
I have googled until blie in the face, WiKi'd until physically exhausted
and searched
Now that I've finally got faxing working, I'm wondering if there's any
way to retrieve specific fax status messages from SpanDSP when sending
faxes through *. E.g. completed/error status, etc. If so, is there any
simple way to have this status dumped into * CDR?
Do broadvoice limit the number of concurrent calls that any given sip
registrant can make? What about other similar providers?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
Did you ever work this out?
- Original Message -
From: Niksa Baldun [EMAIL PROTECTED]
Sent: Monday, January 03, 2005 1:07 PM
Subject: [Asterisk-Users] Line-in as MOH source
Hello,
Most traditional PBX-es have the ability to use external audio source
(e.g. radio tuner) for music on hold.
I'm sorry if this has been answered before. I seem to remember reading a
similar thread a while back, but for the life of me I can't find it
(after 30 minutes of intensive googling).
I have a voicetronix openline 4 card in an * server running CVS HEAD as
of 1st March. Everything is working
I've posted this to SNOM, but was wondering wheter anyone here has issues with
SNOM 190 phones not showing the correct DST adjusted time (using the latest
firmware).
--
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
This is probably a very simple question, but I can't for the life of me work it
out. I'm trying to use Asterisk as a PTSN gateway to OCS (and believe I have
all the SIP issues sorted), but OCS wants to dial in e164 format
(+613blahblah). Because Asterisk sees the + in the SIP URI, it doesn't
individual
config is not an option?
Rod Bacon
Technical Manager
JASCO Consulting Pty. Ltd.
http://www.jasco.net.au http://www.jasco.net.au/
Ph. 03 9432 6376
Fax: 03 9432 6378
systems now run close to 100% in zttest, never miss an irq and don't seem to
generate PCI parity errors any more.
I don't know if I've fixed it, but you should really go through the whole
process anyway.
==
Rod Bacon
Empowered Communications
Ground Floor
If you do a make install samples in the asterisk src dir, it will put them
into /var/lib/asterisk/sounds
Chadwick E. Labno wrote:
where should the sound (.gsm) files be located?
Currently the are in /usr/src/asterisk/sounds.
I feel they should be located else ware, like in
2 - Check your span line in your zaptel.conf. You should be receiving
timing, not giving it, when using a PRI (generally). Change the second
number from 1 to 0. Save and restart asterisk. (span=1,0,0,esf,b8zs)
I think you've got this cocked-up. A 0 in the second position tells zaptel
1 - 100 of 213 matches
Mail list logo
