> > On Wed, 2003-07-30 at 22:33, Patrick wrote:
> > > Did it work after you left a new voice mail message?
> > >
> > > I was looking into the source code to fix it so that the euid was set to
> > > nobody, create the file and then change it back to uid 0, but that didn't
> > > work. Or, maybe c
Excellent idea mate,
Now I am able to do what I wanted with Great help from Jeremy McNamara.
Thanks alot
Foong
- Original Message -
From: "Andy Powell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:56 PM
Subject: Re: [Asterisk-Users] Call Transfer
> Foon
Inbound calls work pretty fine again,
Thanks for you help
Yves
|-+->
| | "Brenton D. Rothchild"|
| | <[EMAIL PROTECTED]> |
| | Sent by: |
| | [EMAIL PROTECTE
On Wed, 2003-07-30 at 21:59, Steven J. Sobol wrote:
> For anyone that cares...
>
> I am porting James Golovich's Manager.pm over to PHP. I plan on also
> doing some documentation which will cover both the Perl and PHP APIs,
> which will be almost identical (at least, to whatever extent is
> pract
On Wed, 2003-07-30 at 18:07, Ajit M Kallingal wrote:
> Hello
> Since I am getting a bit concerned about the SCO vs IBM issue, I was
> wondering if can I can setup Asterisk on FreeBSD is it supported ?
> Are drivers for Digium cards available on FreeBSD ?
If you are worried about it, you really sho
With shipping, I recall my 102 came to $97. I think it was $85 but
I'd need to look it up and don't have the papers nearby.
-reed
At 06:39 PM 7/30/2003 -0500, you wrote:
I was quoted $75 and $85 USD today.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: "Joe Cooke" <[EMA
using chan_h323 and g.711u
bkw
On Wed, 30 Jul 2003, Patrick wrote:
>
> Which codec are you using? and which H.323 channel driver? chan_h323 or
> chan_oh323 ?
>
>
> On 30 Jul 2003, Eric Wieling wrote:
>
> > That only works if you are using the G711 (ulaw/alaw) codecs. Other
> > codecs distort
http://store.yahoo.com/grandstream-networks-inc/products.html
I think that will clear it up.
On Wed, 30 Jul 2003, Ricardo Villa wrote:
> I was quoted $75 and $85 USD today.
>
> Ricardo Villa
> http://www.telesip.net
>
> - Original Message -
> From: "Joe Cooke" <[EMAIL PROTECTED]>
> To: <
Dear all,
Can some of you give us some suggestion how to configure the asterisk in
order to make a call to cisco5300 in g729a codec. And how to confiure the
cisco5300 part in order to receive a call from cisco5300 via h323 g729a.
Your advice /help will be highly appreciated.
Thanks,
George Lin
Hi Jeremy,
Ok, still learning to get the backtrace. will post a trace next.
When I issues a dial command on console, ex
dial H323/6031334000
I get seg fault also, this only happen if it involve dialing through H323
channels
Thank for your reply
Foong
- Original Message -
From: "Jerem
What's your concern with it? If any of SCO code made it into GNU stuff, it
will be removed and rewritten in a short time anyway...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ajit M Kallingal
Sent: Wednesday, July 30, 2003 7:08 PM
To: [EMAIL PROTECTED
Send me the backtrace and console output, off list.
That's a pretty crazy extension. I bet your trying to make some kind
of crazy callback system :)
Jeremy McNamara
Chee Foong wrote:
I dumped the following test.call file into /var/spool/asterisk/outgoing
gives me segmentation fault :(
Ch
For anyone that cares...
I am porting James Golovich's Manager.pm over to PHP. I plan on also
doing some documentation which will cover both the Perl and PHP APIs,
which will be almost identical (at least, to whatever extent is
practical).
Will let y'all know when I have some usable code to sho
I dumped the following test.call file into /var/spool/asterisk/outgoing
gives me segmentation fault :(
Channel: H323/0143126544
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: voip-test
Extension: 90324324433
Priority: 1
same thing happend if I execute dial command on console.
I figure out tha
But the mask of the file is set to 0700. I don't think the sgid bit will
make a difference if the file isn't written 0770. It's still on
readable/writable/executable by the owner.
Patrick
On 31 Jul 2003, Armand A. Verstappen wrote:
> On Wed, 2003-07-30 at 22:33, Patrick wrote:
> > Did it w
What are the results using X-lite to X-Lite to X-Lite, does transfer work ?
It works when using FWD for us ATA's to and from X-Lite work as well using
FWD and other SIP proxies.
What are the other endpoints that are being used in the transfer process?
Not sure what's going on with the SNOMs or th
On Wed, 2003-07-30 at 22:33, Patrick wrote:
> Did it work after you left a new voice mail message?
>
> I was looking into the source code to fix it so that the euid was set to
> nobody, create the file and then change it back to uid 0, but that didn't
> work. Or, maybe change the file mode was
On Wed, 2003-07-30 at 22:25, Carlos Crembil wrote:
> %unsuscribe
variable subsitution on the mailinglist contents of asterisk is not
implemented.
If i were, the correct syntax probably would have been:
exten => _asterisk,1,Agi(mailinglist,%{unsubscribe})
There's a link on the bottom of this mail
My trouble is that the MGCP devices lost the connection with the
asterisk
My gateways are ASKEY MGCP
Any comments?
Humberto
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Miércoles, 30 de Julio de 2003 05:07 p.m.
To: [EMAIL PR
I was quoted $75 and $85 USD today.
Ricardo Villa
http://www.telesip.net
- Original Message -
From: "Joe Cooke" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:31 PM
Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
> I was quoted the $75 and $8
I was quoted the $75 and $85 USD prices from Grandstream direct about 2
months ago. I'm not sure if it makes a difference, but I live in the US.
- Joe
- Original Message -
From: "marrandy" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 7:17 PM
Subject: [Asteri
I was quoted $85 per unit, and $80 shipping to New Zealand!
- Original Message -
From: "marrandy" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, July 31, 2003 11:17 AM
Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102
>
> Checking the earlier mails, it stated that
Hello
Since I am getting a bit concerned about the SCO vs IBM issue, I was
wondering if can I can setup Asterisk on FreeBSD is it supported ?
Are drivers for Digium cards available on FreeBSD ?
Thanks
Ajit
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wedne
Checking the earlier mails, it stated that the phones were $75 (100) & $85
(102) ref :-
http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html
Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person
said there was no price change.
Anyone on this list actu
Which codec are you using? and which H.323 channel driver? chan_h323 or
chan_oh323 ?
On 30 Jul 2003, Eric Wieling wrote:
> That only works if you are using the G711 (ulaw/alaw) codecs. Other
> codecs distort inband DTMF.
>
> On Wed, 2003-07-30 at 15:26, Patrick wrote:
> > I have the same s
thats all we use right now
On Wed, 30 Jul 2003, Eric Wieling wrote:
> That only works if you are using the G711 (ulaw/alaw) codecs. Other
> codecs distort inband DTMF.
>
> On Wed, 2003-07-30 at 15:26, Patrick wrote:
> > I have the same setup, and in the sip.conf file I set the dtmfmode=inband
>
Hello Wade:
There is MGCP firmware available too which appears to help in NAT mode. I have been
playing around with these CPG for quite a few months now.
I have been able to get them to work only with G711 codecs. I was unable to create a
coding profile to work successfully with either G723 an
This is getting too confusing for me ;-(
Could someone summarize what are the steps necessary to make vmail.cgi
work on a system? Something like this:
1) copy vmail.cgi to your cgi-bin directory
2) copy images/*.gif to your img directory
3) grant
4) grant
-Original Message-
F
I have the same with the transfer issue but also when I call between
X-Lite and a SNOM 200 there is no audio but if I call between X-Lite and
a Budgetone 102 all is OK.
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven J.
Sobol
Sent: 30 July 2
On Wed, 30 Jul 2003, Jeremy McNamara wrote:
> Because H.323 doesn't have a specific 'feature' of caller*id.
However, it does seem to have
- calling party number
- calling party name
- display string
and at least the last one seems to be set to whatever SetCallerID() tells
it to be if you're usin
%unsuscribe
___
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That only works if you are using the G711 (ulaw/alaw) codecs. Other
codecs distort inband DTMF.
On Wed, 2003-07-30 at 15:26, Patrick wrote:
> I have the same setup, and in the sip.conf file I set the dtmfmode=inband
> for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
>
I figured it out. I had a file called CVS in the
directory and it freaked out..
Kyle
- Original Message -
From:
Kyle Hagan
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 9:23
AM
Subject: [Asterisk-Users] CVS
Problem?
Since yesterday i get the
Hi folks
I have a X100P in my home asterisk box and I have it setup as a default context of
'incoming-pstn'
in my extensions.conf i have
[incoming-pstn]
exten => s,1,Goto(incoming,01225,1)
then:
[incoming]
exten => 01225,1,Answer
exten => 01225,2,Dial(SIP/data|m)
etc etc
Anywho back to the p
Try setting dtmf-relay h245-alphanumeric in the MC3810 dial-peer.
On Wed, 30 Jul 2003, Brian West wrote:
> I have done that but I think its the Asterisk => MC3810 via h323 thats
> causing that. Does anyone have an example on how i can dump sip to and
> from the MC3810 to my asterisk server?
>
I have done that but I think its the Asterisk => MC3810 via h323 thats
causing that. Does anyone have an example on how i can dump sip to and
from the MC3810 to my asterisk server?
bkw
On Wed, 30 Jul 2003, Patrick wrote:
>
> I have the same setup, and in the sip.conf file I set the dtmfmode=inb
Did it work after you left a new voice mail message?
I was looking into the source code to fix it so that the euid was set to
nobody, create the file and then change it back to uid 0, but that didn't
work. Or, maybe change the file mode was 770 with the group set so that
the webserver could m
I have the same setup, and in the sip.conf file I set the dtmfmode=inband
for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
On Wed, 30 Jul 2003, Brian West wrote:
> I have this setup:
>
> Sip Phones -> Asterisk -> h323 gateway -> ptsn
>
> Sip phones are setup for ou
On Wednesday 30 July 2003 02:49 pm, Todd Lieberman wrote:
> I did the chown and now I get
>
> [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45]
> Setuid/gid script is writable by world., referer:
> http://asterisk.weichertrents.com/cgi-bin/vmail.cgi
chmod o-w vmail.cgi
btw, 'man chmod' h
I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked.
you still need to make sure nobody has read/write permission on
/var/spool/asterisk/vm/$MBOX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Todd
Lieberman
Sent: Wednesday, July 30,
I have this setup:
Sip Phones -> Asterisk -> h323 gateway -> ptsn
Sip phones are setup for out of band dtmf
but the h323 gateway is inband. Is their a way to pass the digits from
the sip phones to the ptsn via the h323 gateway?
bkw
___
Asterisk-Users
I did the chown and now I get
[Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gid script
is writable by world., referer:
http://asterisk.weichertrents.com/cgi-bin/vmail.cgi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paulo
Mannheimer
Se
I did that and now I get
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paulo
Mannheimer
Sent: Wednesday, July 30, 2003 3:23 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] voicemail file access problems
Thanks!
-Original Message-
From: [E
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: July 30, 2003 4:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] voicemail file access problems
On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote:
> Hi folks,
On Wed, 30 Jul 2003, Brian West wrote:
> Same here. Same build.
--
JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity]
22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve Sobol, Proprietor
888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED]
___
Hi,
After spending some time trying to get a DG-104S working behind NAT,
I finally found the problem.
I made the incorrect assumption that nat=yes in mgcp.conf works just
like sip.conf. The channels within a gateway are treated more closely to
zap channels than sip channels (from
On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote:
> Hi folks,
>
> I'm having problems accessing my voicemail files through the web
> interface.
>
> I remember that this was discussed on the list, and it seems to be
> a permission problem, but I couldn't find any answer by searching
> the
Hi folks,
I’m having problems accessing my voicemail files
through the web interface.
I remember that this was discussed on the list, and it seems
to be a permission problem, but I couldn’t find any answer by searching
the archives.
Any hint?
PauloHM
On Wed, 30 Jul 2003, Benjamin Miller wrote:
> I have not had time to complete an "Unified Messaging" component to
> voicemail, but I would see this as an admiral goal. Most modern
> voicemail systems have some kind of way to delete or mark the voicemail
> as read when the message is deleted or re
Same here. Same build.
On Wed, 30 Jul 2003, Dan wrote:
> Hi Erik,
>
> I have the version "X-Lite 2.0 private build 1050".
> When I click on transfer then extension then transfer, the call is closed,
> but the final destination does not ring.
>
> Thanks,
> Dan
>
>
> - Original Message -
>
I believe this is fixed. Sorry.
Mark
On Wed, 30 Jul 2003 [EMAIL PROTECTED] wrote:
>
> Hi,
>
> I am using the latest cvs release of asterisk, and the behaviour is in fact
> the same,
>
> outbound calls work fine,
> but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked"
> by as
Hi Erik,
I have the version "X-Lite 2.0 private build 1050".
When I click on transfer then extension then transfer, the call is closed,
but the final destination does not ring.
Thanks,
Dan
- Original Message -
From: "Erik Lagerway" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wedn
Hi Dan,
We had problems with that during a transitional build but the new v2.0 build
1050 should have fixed that, can you confirm that you are using the most
recent build?
-Erik
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Dan
Sent: Wednesday, July 30, 2
We have built this in-house and it works well but we had to overcome some
challenges along the way. For example,
1) Your voicemail server may be off-net (behind NAT or using private ip
addresses) while your email users are on-net. The way we approached this was
to abstract the functionality such t
On Wed, 2003-07-30 at 09:40, Rattana BIV wrote:
> Hi,
>
> I try to make some statistics about call on Asterisk. Is there
> something who makes it ?
> I will be interesting to have the time of a call and a list of current
> calls.
Current calls can be found either from the CLI, or from the manage
I'm not running a high security solution so an link in w/a MD5 encrypted
path name or guid would be sufficient. I don't want to enter a password
every time. Can Voicemail2 save the files by unique file name? I'd be happy
to write the cgi that deletes the message or marks it as read. TL
-Ori
Actually, this is a much bigger task than you would imagine.
I have not had time to complete an "Unified Messaging" component to
voicemail, but I would see this as an admiral goal. Most modern
voicemail systems have some kind of way to delete or mark the voicemail
as read when the message is dele
Hi,
Anyone succeed using call transfer function in X-Lite?
It is stated that this feature is available in the Lite version too, but for
me it doesn't work.
Clicking on Transfer button, then entering the number and then clicking
again on transfer doesn't work.
I miss something?
Thanks,
Dan
Since yesterday i get the following message when
downloading anything from the CVS.
cvs [checkout aborted]: reading CVS/Tag: Not a
directory
Is it a problem on my end or digium? I havnt
changed anything on my end.
Kyle
We have talked about a cdr_tds.c how many times now on this list?
I recently rewrote cdr_mysql.c to use a new table structure
CREATE TABLE cdr (
accountcode varchar(45) NOT NULL default '',
src varchar(45) NOT NULL default '',
dst varchar(45) NOT NULL default '',
dcontext varchar(45) NOT
Will there be a way to delete messages from email. I love getting voicemail
in wav to my email, but I hate having to delete them when I call in to get
my messages. If we could add a link and have a cgi delete the messages that
would be a nice time saver. TL
-Original Message-
From: [EMAI
I have Panasonic TD1232 pbx, few cisco ata186 and linux box with asterisk.
Can I dial from asterisk into ata, then indicate phone number playing
tone (use DISA feature at panasonic) and connect to any analog phone
connected to panasonic ?
I think some of Playtones application within Dial applic
Try ftp://angelgomez.homelinux.com/pub/audiocodes
Anton Tinchev wrote:
Kelvin Chua wrote:
hi guys,
have anybody tried using audiocodes sip fxs against asterisk? how's the device fairing?
~kelvin
Can someone send me SIP firmwire for audiocodes 104.
I has h.323 only and it sucks
__
Hi,
Do you have some experience with the "best" values for those parameters in
youyr particular case?
I mean the best raport between sound level in both direction and echo
cancellation.
For me, the best result I can get is with:
rxgain=10
txgain=15
... the sound level is good, but the echo is a l
Hi,
It was one of the possible way to get a workaround with my ATA and attended
call transfer. It was solved for now.. check one of my previous mails.
Thanks,
Dan
- Original Message -
From: "Armand A. Verstappen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003
Thanks Ben.
I have entered an item in the bug track on Digium site.
This is not a real issue for me, more an observation.
It can be an issue in some circumstances, but I'm sure that it will be
solved in the near future.
Thank you again for your info.
Dan
- Original Message -
From: "Benj
On Wednesday 30 July 2003 08:06 am, Wen Wen wrote:
> great! it works. I am now able to get the cli prompt with "asterisk
> -vvvcg"
>
> when i try to use "/usr/sbin/safe_asterisk" script to start, i
> still got the error of
> "Asterisk ended with exit status 127". I guess I can use the
> "asterisk -
Check the source on this. When I send Mark a patch way back when to
allow the e-mail to be triggered to the person who gets the forwarded
message, I selfishly only coded it to handle a wav file. Mark may have
fixed or changed this at a later point, but if not, there is probably a
to-do comment in
On Wed, 2003-07-30 at 16:44, Dan wrote:
> Thanks for the suggestion.
> I have change it like that:
>
> ;dummy extension
> exten => 199,1,Ringing
> exten => 199,2,Wait(60) ; give illusion we might pick up
> exten => 199,3,Hangup
>
> in order to hear the ring too.
>
> ..but now... how can I do t
I also had the same problem with sip, I also moved back a couple of
weeks in cvs.
I also use a AS5300 Cisco in my call chain.
I got a bunch of "Ignoring this request" in debug. I have not had time
to trace the call path on this problem yet.
Low, Adam wrote:
All,
I've found problems in my setup
On Wed, 2003-07-30 at 16:40, John Congdon wrote:
> Has anyone solved the problem on the ADSI phones
> that when you hit one of the soft keys, the Number Pad
> stops working?
No, I haven't. Just confirming that I have the same problem here, using
the VoiceMail2 app. Do you experience this outside V
Hi,
On Wed, 2003-07-30 at 16:15, [EMAIL PROTECTED] wrote:
> is it possible to use a Teles16.3 via isdn4linux for the external phone
> connections (phone provider net)?
Yes, it is. I tested using an old card I had lying around. I quickly
switched to a Fritz card and chan_capi however. This solved
I do part time consulting work. I need to setup an asterisk system to
allow me to record both inbound and outbound calls to clients. I have one
client that is just a PITA. The client has changed their mind three times
so far and we are at step one.
I have a spare slackware box and a seperate phone
Hi again,
I think I have now a workaround for call transfer on ATA 186.
This is the extension corresponding to the phone connected to an ATA186
exten => 103,1,Dial(SIP/103,20),Tt
exten => 103,2,Voicemail2(us101)
exten => 103,3,Hangup
exten => 103,102,Ringing
exten => 103,103,Wait(1)
exten => 103
On Wed, 30 Jul 2003, Steven J. Sobol wrote:
> I've successfully used the FreeTDS libraries on a Linux box to connect to
> a MySQL server
s/MySQL/MS SQL/g
--
JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity]
22674 Motnocab Road * Apple Valley, CA 92307-1950
Steve So
On Wed, 30 Jul 2003, Florian Overkamp wrote:
> As suggested by another poster: MS-SQL is mostly based on Sybase, so any
> Sybase driver (there is one for PHP for instance) can probably be used,
> from AGI or otherwise...
I've successfully used the FreeTDS libraries on a Linux box to connect to
Hi,
Thanks for the suggestion.
I have change it like that:
;dummy extension
exten => 199,1,Ringing
exten => 199,2,Wait(60) ; give illusion we might pick up
exten => 199,3,Hangup
in order to hear the ring too.
..but now... how can I do to call this extension from a Dial command?
What I want in
Has anyone solved the problem on the ADSI phones
that when you hit one of the soft keys, the Number Pad
stops working?
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Hi,
I try to make some statistics about call on
Asterisk. Is there something who makes it ?
I will be interesting to have the time of a call
and a list of current calls.
Regards
Rattana
Hi,
is it possible to use a Teles16.3 via isdn4linux for the external phone
connections (phone provider net)?
Internally I want to use the TDM30B card to connect my analogue interfaces.
Lars
___
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[EMAIL PROTECTED]
http://l
That also worked for me. My AudioCodes MP-104 FXO has no problem
making inbound calls now.
Thanks Patrick and Adam.
-Brenton
- Original Message -
From: "Low, Adam" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 8:45 AM
Subject: RE: [Asterisk-Users] chan_sip.
On Wed, 2003-07-30 at 15:55, Dan wrote:
> It is possible to create a dummy account (SIP or IAX type) in order to be
> used in a "dummy" extension?
> I want to be able to use it as a normal extension (as an IP phone connected
> to it), but without the need to answer or call from that extension.
> I
Hi,
It is possible to create a dummy account (SIP or IAX type) in order to be
used in a "dummy" extension?
I want to be able to use it as a normal extension (as an IP phone connected
to it), but without the need to answer or call from that extension.
I want that when I call that extension to hear
///shameless advert///
I have just listed my two 4 channel voicetronix cards on ebay if anyone is
interested. The cards work great but we have now moved to E1 so can no
longer use analogue cards so have moved over to asterisk running on digium
hardware.
Items no 2745081372 and 2745081930
I'm not
Well found Patrick, that did the trick for me as well !
I had been trying to debug 1.135 where this portion of code wasn't added yet ... thats
a lesson learnt ...
-Original Message-
From: Patrick
To: '[EMAIL PROTECTED] '
Sent: 30/07/03 15:04
Subject: RE: [Asterisk-Users] chan_sip.c probl
How the X100P card detect the arrival of a call? I
have a bunch of cards and they perform perfectly when connected to a PBX, but
when I connect to another brand of PBX, they don't unhook the line, or in other
words, they don't answer the line. If I connect a normal analog phone on the
same l
I was told in #asterisk that you just hit transfer, dial the extension,
speak to caller and press transfer once your done talking and it should do
it. In addition you can do transfer+extension+transfer+hangup...
Thats how I was told it would work.
bkw
On Wed, 30 Jul 2003, denon wrote:
> Last
great! it works. I am now able to get the cli prompt with "asterisk -vvvcg"
when i try to use "/usr/sbin/safe_asterisk" script to start, i still got the
error of
"Asterisk ended with exit status 127". I guess I can use the "asterisk
-vvvcg" to
start the server. but I would like to know why the "
It is in the find_user() routine. If it is not an extension on the PBX,
it should return a zero
if ( isfound ) {
ast_log(LOG_DEBUG, "%s is not a local user\n", name);
ast_pthread_mutex_unlock(&userl.lock);
return 1; <--- this is the problem - change it to a 0.
}
It isn't an error,
Brenton, Yves, ...
I've located the cause of the problem in chan_sip.c but am still trying to find the
exact cause being completely new to the asterisk code. It seems that there was an
added function in 1.135 called 'find_user' that is supposed to lookup the users
incoming call limit but the ro
Hi,
I am using the latest cvs release of asterisk, and the behaviour is in fact
the same,
outbound calls work fine,
but for inbound calls (from C2651 over PSTN) , SIP messages get "blocked"
by asterisk, and never reach the phone.
The setup is the same : 7960 <--> asterisk <--> C2651<---
Hello,
Can anyone spare a configuration file for using asterisk as a pure
Voip-PSTN gateway running H323 and Sip only ?
Many thanks,
Abdul Hakeem
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Dan,
The time to call could be stored into database with party A and party B
phone number.
Asterisk or perhaps a script (mentions by Andy Powel in another reply) just
keep checking the database and make calls if time is < current time and the
call has not been processed yet.
In this manner, the c
Thanks Andy
Will try that
Thanks again.
Foong
- Original Message -
From: "Andy Powell" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 6:56 PM
Subject: Re: [Asterisk-Users] Call Transfer
> Foong
>
> Take a look at the sample.call file, modifying the settings
There is no need to create a Meeting Room... just to initiate a conference
in three...
- Original Message -
From: "Chee Foong" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, July 30, 2003 1:02 PM
Subject: Re: [Asterisk-Users] Call Transfer
> Hello
>
> But If i do that I ha
Foong,
> Actually, we have a client that is too lazy to do all the dialing, he want
a
> system that will call him and also the person he wanted to call, just like
> some receptionists do theese days. The different is that asterisk is
taking
> over the receptionist's job
... then... who decide when
Foong
Take a look at the sample.call file, modifying the settings in there and copying the
file to /var/spool/asterisk/outgoing will cause asterisk to dial the call.. an example
config is below
Channel: SIP/[EMAIL PROTECTED]
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: mysipcontext2
Extens
Hello,
This morning I just started to have this problem calling from a SIP phone to
a regular phone, using one of the 4 BRI cards (passive) I've in my * box.
It calls regularly, but somewhere after 8-10 secs, it random hangups, or it
hangups immediately after a hold, and so.
I've looked into /v
> As suggested by another poster: MS-SQL is mostly based on Sybase, so any
> Sybase driver (there is one for PHP for instance) can probably be used,
> from AGI or otherwise...
I wrote a check_mssql perl script for nagios (dot org). Take a look at this
for reference - it really isn't hard :)
http:
chan_h323 passes Caller*id, if there is one.
Then you can specify a type=h323 for a specific H.323ID, if you wish.
Jeremy McNamara
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 30 July 2003 11:58, Jeremy McNamara wrote:
Because H.323 doesn't have a specif
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