Hi,
Cisco 7940/60 does P2P with FWD.
BR,
Dan
- Original Message -
From: Dave Packham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 29, 2003 5:30 AM
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server...
Check out this bug
Hi all ATA-Users,
after a lot of tests, i found the best (not complete working solution).
If you use an an MGCP-Image then
1. CLIP-CallerID works fine (with one Phone Callername-transmission
works too)
2. Blind transfer with # works fine
3. Attended transfer (Transfer with consultation?)
Is there any way to find out why this happens? why do I get complete
recived?
see my previous post
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
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Hi,
I have noticed that when I am calling from my Snom-phone to another
Snom-phone through Asterisk, the SIP-message's Contact -header could be
sometimes empty and for example other Snom get no BYE-message.
Here is example of that kind of message:
10 headers, 0 lines
Sending to 192.168.0.32 :
Ah, now that you mention it, I implemented this in my patch also and then
forgot about it: messages that are too short (less than 3 seconds) or all
silence (messages that ended with silence and are not longer than
maxsilence) are deleted. You could search for vm-tooshort.
Perhaps this should
Hi, I am testing a 7960 in this context:
[SIP] --- VPN --- [*] --- [ANY]
(ANY == any type of phone: isdn, SIP, IAX, etc.)
the call goes through and is dropped after 5 seconds with this message
in the log:
File chan_sip.c, Line 388 (retrans_pkt): Maximum retries exceeded on
call IP address
Thanks all,
I spent some time on this last night with packet sniffer in hand, the 'canreinvite'
option makes sense and seems to work well for me (running latest * CVS release) when
used between 79xx phones and the AS5300 gateway although I get some somewhat expected
problems with 79xx that are
1. what's the sequence to press on a SIP phone to transfer a
call to another
extension.
Which SIP phone? Soft/hard ? Phone specific ...
2. what's the same thing if you want to hold an incoming
call, speak to the
other extension, then pass the call?
Which SIP phone? Soft/hard ? Phone
Hi kapejod,
I tried the following firmware:
Driver 3.10-02 (from card's CDROM), Protocol DSS1 Line type
Point-to-Multipoint.
CAPI Channel driver is 0.2.4a.
My capi.conf looks like this:
[general]
nationalprefix=
internationalprefix=
rxgain=1
txgain=1
[interfaces]
msn=7810
incomingmsn=7810
Hey Dan,
On Mon, Jul 28, 2003 at 22:50:21 -0300, Dan Fernandez wrote:
Is there a way in iax to have to endpoints talk to each other directly (after the
call is setup by *) without going through *. In sip, with * you can do it by
configuring sip.conf with canreinvite = yes.
AFAIK, IAX
Sip phones on the system are Grandstream Budgettone 100's.
Was assuming it wouldn't be phone specific :)
they have flash key which is meant to send a DTMF.
thanks for the help with the dial string.
Dave
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent:
oh ok ;) just understood!!
call transfer is something the phone does
and asterisk picks up, not some sequence
you send directly to asterisk, hence from
the Grandstream manual :)
thanks very much for pointing it out!
cheers
Dave
- Original Message -
From: Dave Alan Caruana [EMAIL
You got it, I have cisco 7940 phones which have a transfer soft key which tells the
phones SIP UA to transfer the call via Asterisk to another SIP UA ...
-Original Message-
From: Dave Alan Caruana [mailto:[EMAIL PROTECTED]
Sent: 29 July 2003 13:26
To: [EMAIL PROTECTED]
Subject: Re:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I havn't used the h323 channel of Asterisk for a while, but today I needed to
test a few things only I found out that Asterisk/H323 crashes my Siemens
optipoint 400 phone. It seems to be the audio codecs that's causing it. Is
something broken
hi all
seems the problem with chan_capi and hanging channels now is solved, thanks to
kapejod and levon :)
roy
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Guys,
I have some answer about sample.call
1. Can we use sample.call to test (or simulated) asterisk (in a
predetermined scenario) to accept calls simultaneously?.
2. How many calls can be simulated?
3. Can we used the result as a basis on how many simultaneous calls can
handled by asterisk?
4.
What Linux distribution is best for use with Asterisk?
(easiest compile, least problems, etc)
Thanks,
Sean Rodger
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Personally, I've compiled Asterisk on Redhat and Debian without any problems on
either, I think generally Asterisk compiles very easily no matter what the distro but
I would recommend that you use the one you are most comfortable/experienced with.
-Original Message-
From: Sean Rodger
There are some SIP errors that appeared in CVS in the last couple of
days. I checked out some CVS source from last week and everything
works properly.
Maybe that's part of the problem.
-bill
On Tuesday, July 29, 2003, at 04:59 AM, Louis-David Mitterrand wrote:
Hi, I am testing a 7960 in
In my opinion, Debian is the best for compiling programs, because you can
'apt-get' any dependencies and its respective dependencies in a quick and clean way.
You can also use auto-apt.
And if you don't want to compile, you can 'apt-get install asterisk' and get
asterisk running
Hi,
I try to compil the nex cvs version of asterisk cvs and i have this error
gcc -shared -Xlinker -x -o cdr_mysql.so
r_mysql.o -lmysqlclient -lz -L/usr/local/mysql/lib
/usr/bin/ld: cannot find -lmysqlclient
collect2: ld returned 1 exit status
make[1]: *** [cdr_mysql.so] Error 1
make[1]:
Sure, nothing special though:
[4840]
type=friend
username=4840
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband
[4842]
type=friend
username=4842
host=dynamic
canreinvite=yes
nat=no
qualify=200
mailbox=4840
dtmfmode=inband
-Original Message-
From: Dave
Great.
I will ask him ASAP.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Spencer
Sent: 28 July 2003 22:10
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Offering an Asterisk Documentation and FAQ
Portal
I'm interested!
Mark
On Mon,
Try increasing busycount (a hidden parameter) at Zapata.conf
Mine works like a charm with
busydetect=yes
busycount=6
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerk Face
Sent: July 29, 2003 9:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Tuesday, July 29, 2003 9:15 AM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] Linux flavor?
Personally, I've compiled Asterisk on Redhat and Debian
without any problems
Install mysql
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rattana BIV
Sent: Tuesday, July 29, 2003 9:07 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Compilations errors
Hi,
I try to compil the nex cvs version of asterisk cvs and i
can you do the stutter tone on Multiple SIP voicemail extensions? or only one
extension listed in the zapata.conf?
Dave
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At 01:58 PM 7/28/2003 -0400, you wrote:
Jeremy,
While I see your point, I don't think it's reasonable to ask an end user (as
opposed to a system admin) to hang out on IRC to learn how to use his/her
phone while dealing with live calls and trying to do their job (sales,
marketing, support,
On Tuesday 29 July 2003 09:41, Troy Settle wrote:
-Original Message-
From: Low, Adam
Sent: Tuesday, July 29, 2003 9:15 AM
Personally, I've compiled Asterisk on Redhat and Debian
without any problems on either, I think generally Asterisk
compiles very easily no matter what the
I would love to see FreeBSD support. Any links on the OpenBSD port?
bkw
On Tue, 29 Jul 2003, Tilghman Lesher wrote:
On Tuesday 29 July 2003 09:41, Troy Settle wrote:
-Original Message-
From: Low, Adam
Sent: Tuesday, July 29, 2003 9:15 AM
Personally, I've compiled
made those changes and still no P2P
[70900]
type=friend
insecure=yes
username=70900
secret=youwish
host=dynamic
context = campus
mailbox=70900
canreinvite=yes
nat=no
qualify=200
dtmfmode=inband
is what I have for my Cisco 7960's
Dave
[EMAIL PROTECTED] 7/29/2003 8:01:41 AM
Sure, nothing
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tilghman Lesher
Sent: Tuesday, July 29, 2003 12:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Linux flavor?
Actually, it's been ported over to OpenBSD. It shouldn't be
too much
For the development team to get * (and the zaptel cards) running on BSD
shouldn't take too much effort. Perhaps it's just a matter of finding the
right incentive? My only request would be that it be installed to match
BSD
filesytem standards (everything in /usr/local).
One of my next
But when I run the safe_asterisk, I got the Asterisk died with code 127
error.
run asterisk -gvvvc to get full error outputs :)
roy
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Hi,
One of my pbx's seems to be having some new issues. crackling interference
on the zap channels running through the channel bank, and I noticed that
these happen when I hit an irq miss in zttool:
Current Alarms: No alarms.
Sync Source:Digium Wildcard T100P T1/PRI C
IRQ Misses:
OK calls thru the * server are looped and calls with the same phones thru Free WOrld
Dialup are P2P. same configs...
Anyone have any ideas? I know its a bug but we need to fix this one I think its
pretty big one. it would HAMMER the scalability of * servers
Dave
[EMAIL PROTECTED]
I have an AVM Fritz card using CAPI which seems to work quite well apart
from the CLID is not being captured correctly.
On my SNOM 200 the CLID displays odd characters and in CAPI debug the
CLID reported is £.
Anyone any ideas ?
Rgds,
Stuart
___
Hi Stuart,
you have to order CLIP for your BRI from BT and pay
them for it.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
Thanks. I will chase them tomorrow. Might as well ask for MSN numbers
whilst I am at it.
Stuart
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaus-Peter
Junghanns
Sent: 29 July 2003 21:57
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] CAPI CLID
Where do I get a Dialogic driver for Asterisk from?
The handbook mentions it in passing as a paid-for option.
How much does this cost, and how does one go about obtaining it?
--
Alastair Maw
MX Telecom http://www.mxtelecom.com
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Asterisk-Users mailing
OK I have revised this a bit it might help someone out there:
[dnd]
exten = _74,1,DBput(DND/${CALLERIDNUM}=YES})
exten = _74,2,Playback(dnd-on)
exten = _74,3,SoftHangup
exten = _73,1,DBdel(DND/${CALLERIDNUM})
exten = _73,2,Playback(dnd-off)
exten = _73,3,SoftHangup
[callforward]
exten =
Hi Roy,
When I run asterisk -gvvvc, it complains about some *.conf files not found
(see the output
in the end of the mail):
logger.conf, manager.conf, rtp.conf, modules.conf, adsi.conf,
musiconhold.conf,
indications.conf, and modem.conf.
My linux box does not have sound card and modem
I'm having trouble getting the cards configured. I
have followed the instructions but get errors.
when I modprobe zaptel
/lib/modules/2.4.21-0.13mdk/misc/zaptel.o:
unresolved symbol
proc_mkdir_Re6122de7/lib/modules/2.4.21-0.13mdk/misc/zaptel.o: unresolved
symbol
On Wednesday 30 July 2003 00:29, Wen Wen wrote:
ERROR[1024]: File chan_modem.c, Line 852 (load_module): Unable to load
config modem.conf
add this line
noload = chan_modem.so
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On Wednesday 30 July 2003 00:29, Wen Wen wrote:
ERROR[1024]: File chan_modem.c, Line 852 (load_module): Unable to load
config modem.conf
er
add this line to modules.conf
noload = chan_modem.so
potentially also these
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload =
Hi,
Can I do variable substitution in the [globals] section of extensions.conf?
For example something like this:
[globals]
EXT_BOB=4206
PHONE_BOB=SIP/${EXT_BOB}
Thanks,
Justin
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Hello,
Is anybody else out there using pingtel phones? If so, I like to hear your
experiences...
Sincerely,
Andy Hester
Consero
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No, not at this time. If you think that's valuable, you could request it
as a feature in the bug tracker.
Mark
On Tue, 29 Jul 2003, Justin Eckhouse wrote:
Hi,
Can I do variable substitution in the [globals] section of extensions.conf?
For example something like this:
[globals]
I need to make a little IVR app and get/send the data
into a MS-SQL database.
As far as I know, it doesn't have driver for Linux.
Anybody here already found here any workaround for this situation?
Maybe, I can use an AGI interface to do that, maybe perl+ODBC?
Isamar
On Wed, 2003-07-30 at 07:14, [EMAIL PROTECTED] wrote:
I need to make a little IVR app and get/send the data
into a MS-SQL database.
As far as I know, it doesn't have driver for Linux.
Anybody here already found here any workaround for this situation?
Maybe, I can use an AGI interface to do
Like all the other M$ Databases:-( They should realy think about to make
them DBs able to be reached from other platfroms then their own...
Well, once I needed to reach some Access Database from Linux. What I use is
a service on 2K Server that can be found at http://odbcsock.sourceforge.net
. Be
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