with integratin CCM and *,
but without any details, especially around MTP configuration.
Any help would be greatly appreciated. BR, - Patrick -
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the Sangoma card with the hardware echo can on board.
Regards,
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On Wed, 2005-10-19 at 14:06 +0200, [EMAIL PROTECTED] wrote:
On Wed, 19 Oct 2005, Patrick wrote:
Unless you have the Sangoma card with the hardware echo can on board.
So am I right in saying that the normal Sangoma uses the standard Zaptel
software echo canceller - the same one
but when the dial command hangs up normally, line 2 won't get
executed.
Try with h (for hangup):
exten = 1234,1,Dial...
exten = 1234,h,...
From the command line in asterisk show application dial gives a lot of
info what it can do.
Regards,
Patrick
Title: Patrick Briefpapier
What large number of
answers?
If I scroll through
the lists no answers are present..and previous posts do not seem to help as
well..
Please
clarify?
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Title: Patrick Briefpapier
All,
Currently I've got
my Asterisk machine running smoothly on IP bases. Meaning I can reach all phones
or softphones within my LAN or remote LAN's via VPN. The next step for me is
connecting it to the PSTN network.
After some tweaking
with the modem.conf I got
Title: Patrick Briefpapier
Some additional
information:
mchan_modem.so[0;37;40m] =
([33;40mGeneric Voice Modem Driver[0;37;40m)Parsing
'/etc/asterisk/modem.conf': FoundLoading modem driver chan_modem_i4l.so
= ([33;40mISDN4Linux Emulated Modem Driver[0;37;40m)Configured modem
/dev
Title: Patrick Briefpapier
I would prefer to
get it working with i4l at the moment, and migrating later on to CAPI if
needed.
Thanks for any help
you can give me..
-
Patrick
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Title: Patrick Briefpapier
Hi
Martin,
I saw your problem
listing on the Asterisk mail archives. I seem to have the same problem with the
ISDN 'lacking dialtone' message
I still have not
been able to get it working, could you share your modem / extension / sip conf
files?
Thanks
hotswap PS)?
Regards,
Patrick
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one. Anyone know a vendor that does
(barebones or just a 1U case with a dual hotswap PS)?
Regards,
Patrick
My Sun server has hotswap power supplies. I have not tried to install
asterisk on it since a couple years ago but I am pretty sure that I
On Sat, 2005-10-08 at 18:01 -0400, Cory Andrews wrote:
The F3000 is not anticipated to be available for distribution until late
December/January, FYI.
I came across this one. Haven't seen one in real life though.
http://www.gemtek.com.tw/pro_whsg103g.htm
Regards,
Patrick
Cirelle Enterprises wrote:
we also experienced this with asterisk 1.0.9 and rev H of the tdm with
4 fxo modules
we were restarting asterisk every night via cron and this still happened
in our case, 3 out of 4 fxo modules (2,3,4) crapped out and stopped
ack'ing incoming
calls (outgoing
My office has been running Asterisk 1.0.8 and a TDM04B for a few months
now without too much trouble. After a while we discovered that after a
certain period (about a month), asterisk stopped acknowledging inbound
calls. Since I was out of the office the first time it happened,
another admin
Patrick Friedel wrote:
I couldn't find a changelog for 1.0.9 to see if it's worth the
off-hours maintenance window, and we're too dependant on the phones to
try 1.2. Should I try the next step up in the probably unnecessary
preventative maintenance and unload/reload the wctdm module during
Rich Adamson wrote:
My office has been running Asterisk 1.0.8 and a TDM04B for a few months
now without too much trouble. After a while we discovered that after a
certain period (about a month), asterisk stopped acknowledging inbound
calls. Since I was out of the office the first time it
decisions like that that contribute to 3Com's fading
away.
Regards,
Patrick
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. Never
laughed so hard when I saw the incredulous faces of the M$ drones. We
brought in a Stratus based solution and won the project.
Regards,
Patrick
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-class box.
Regards,
Patrick
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forgotten to ALSO plug power into the board from the power supply.
Everything worked fine after that (yep, I was a noob). :-)
Patrick
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($[${GROUP_COUNT()} 1],?100) ;Full group
exten = isdn,2,Ringing()
exten = isdn,3,Dial(SIP/302,120,tT)
exten = isdn,5,Congestion
exten = isdn,6,Bussy
Shouldn't that read Busy instead of Bussy?
Regards,
Patrick
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appreciated.
Patrick
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was
not successful or was the workaround an easier solution?
Regards,
Patrick
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extensions from any phone. The solutions I've come up
with so far (individual contexts for each extension or customised
dial strings for each extensions) are pretty gruesome. Is there a
neat way of achieving this functionality?
Thanks
Patrick
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On Tue, 2005-09-20 at 21:09 +0200, Armin Schindler wrote:
Hi all,
it took a while, but on sourceforge.net I added the new release 0.6 of
chan_capi-cm driver.
[snip]
Thanks for the new release Armin. I will test it tomorrow with cvs HEAD.
Regards,
Patrick
load? Just wondering if
that copy action wouldn't also create an I/O bottleneck and cause call
quality issues under load. Did you consider using remote storage e.g.
via nfs, a fibre channel or iSCSI link to a SAN?
Regards,
Patrick
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card as an
Asterisk server and it works fine.
Regards,
Patrick
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are running this on top of vmware then forget it.
Given the time sensitivity of Asterisk it is not suited
to run in a virtual OS environment.
Regards,
Patrick
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Just one line. Do you think you could point me to the SPA3K. A google
search doesn't yield any results. Is that a discontinued product? Would
I not need something on the other end where the POTS phone line is
located? Thanks!
--
Patrick Campbell
*?
--
Patrick Campbell
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,
Patrick
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running * at location B, but what type of hardware will let me connect the
POTS line to the * server?
Thanks!
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. If you get your calls via PRI's I suggest
exploring 2 MaxTNT's each with DS3/GbE interfaces and a lot of
redundancy.
Regards,
Patrick
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to
be transferred to the new extesion. Once the call is terminated by the
person to whom it is transferred the agent returns to their appropriate
context and extension as shown in agent status. The same result is had for
sip or PBX transfers. I am running asterisk stable.
thanks,
Patrick
this inside the context in
extension
exten = _.,1,DeadAGI(testbefore.agi)
exten = h,1,DeadAGI(testafter.agi)
Thanks.
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FreeBSD Brasil LTDA.
(31) 3281-9633 / 3281-3547
[EMAIL PROTECTED]
http://www.freebsdbrasil.com.br
Long live Hanin Elias, Kim Deal
I have a small phone system built around Asterisk stable utilizing a PRI
trunk and approximately 25 Uniden UIP 200 sip phones. I have two call
queues, nothing exotic, serviced by up to three call agents. Whenever the
agents transfer a call, the queues do not register a call transfer or
app_voicemail depends on this adsi stuff. Is there a way
to disable adsi in app_voicemail? I looked through app_voicemail.c but
don't have enough knowledge of C how I would go about it.
Regards,
Patrick
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, there is no noise problem.
Digium has no explanation for now and have asked for a RMA of one of the cards.
I will keep you informed. If someone else has seen this behaviour, tell me
if there is an explanation.
Patrick
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Patrick Tracanelli wrote:
Hello List,
This is my first message herein. I was playing around with System() and
AGI() and found out something I cound not determine my configuration
error. I added before.agi and after.agi to the agi-bin dir. Tried to
make before.agi get run before the dial
suggestions?
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(31) 3281-9633 / 3281-3547
Long live Hanin Elias, Kim Deal!
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On Tue, 2005-08-23 at 19:57 +0100, razza wrote:
[snap]
pbx_dundi.c:31:18: zlib.h: No such file or directory
Seems you are missing zlib-devel
sh: line 1: flex: command not found
Seems you are missing the flex app
Regards,
Patrick
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And the static noise is gone !!
Anobody have an idea why this happened
Of course it doesn't solve my problem because we have one old card and
several new card but it may give digium an idea of where is my problem
Patrick
Hi
We have static noise problem on our asterisk server. latest
Hi
We have static noise problem on our asterisk server. latest stable release.
The card is a new TDM04B
We have it installed on the following hardware
Motherboard Intel SE7520BD2SCSI
2x POWER SUPPLY 730W INTEL
I will not mention the other hardware because we have desactivated/changed
all the
declaration of
'uniquelock' was here
make: *** [channel.o] Error 1
error: Bad exit status from /var/tmp/rpm-tmp.94064 (%install)
When I force the use of gcc32 all is well and the asterisk srpm compiles
fine. Any ideas how I can make asterisk compile with gcc4 too?
Thanks and regards,
Patrick
Hi,
I recently encountered a weired situation where my budgetone stopped
working. My network is like this:
Asterisk on Public IP --- ADSL NAT Router - GS01, GS02,
GS03 on Internal IP
We have an Asterisk server running with a public IP address, which
serves as the master PBX. On a
issues please let me know so I can fix them.
Regards,
Patrick
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Olle E. Johansson wrote:
Patrick Friedel wrote:
Sorry if this is an obvious question, but I haven't seen an obvious
answer on the wiki that I remember. Has anyone managed to make the
record button on the snom 360 fire off the Monitor() application? I
don't see a bounty, and googling
set verbose 255
Verbosity was 9 and is now 255
voip*CLI
[boring build up that isn't new to anyone]
Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih;rport
From: Patrick sip:[EMAIL PROTECTED];tag=4rbnk4yyxd
To: sip:[EMAIL PROTECTED];user=phone
Nils Ohlmeier wrote:
On Thursday 28 July 2005 17:12, Patrick Friedel wrote:
Yeah, that was in the middle of a call - the only other SIP debug
information is the normal call build up and tear down. I can generate
it if you want, but it's nothing exciting, just the usual handshaking
as the unique entry to your local
network (DMZ).
Try with ordinary NAPT rules first and if that doesn't work perhaps try
to assign the ADSL IP address to the phone (I don't know about the
security implications though). Good luck!
Regards,
Patrick
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this is just because I don't have anything set up for it:
SIP Debugging Enabled
voip*CLI
*[I hit the record button at this point]*
Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-q6cqmwneki97;rport
From: Patrick sip:[EMAIL PROTECTED];tag=1pvw4rlq7s
not go for FC2 is that Adria needs kernelcapi
support. There have been many bugfixes in the capi modules in more
recent kernels that are part of (updated) FC3, FC4, CentOS 4.1 and afaik
not FC2. For that reason I would always use a recent FC distro like FC4.
Regards,
Patrick
[EMAIL PROTECTED] wrote:
That's exactly my opinion: isn't ironic that the only function joe
sixpack will use in a pbx is the worst implemented ?
Maybe because most asterisk PBX's are implemented using business class
softphones rather than analogue phones? Most business class SIP
/_760899,3,Goto(menu,s,1)
^^^
Regards,
Patrick
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when I repeatedly press # while a call is established
between two local sip phones. Nothing happens except for these messages
(including no Parking stuff).
Regards,
Patrick
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PROTECTED],30)
exten = _1NXXNXX, 2, congestion()
exten = _1NXXNXX, 102, busy()
Remove the spaces
Regards,
Patrick
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Pavel Jezek wrote:
according to this debate, I would like to try snom 360 still more
(features, opensource support, linux based) ;-)
any good or bad experience with support from snom? or reliability of
snom phones?
PJ
I've been fiddling with a set of Snom 360's for a while now and
So I decided, for the formal asterisk rollout, to change over to less
commercially-infringing MOH than the prior admin had thrown on the
server. (plus: it was blown out and nasty sounding over the phones.
Ew.) I changed the files in /var/lib/asterisk/mohmp3 to something else
(can't dig up
Bob Goddard wrote:
There are 2 problems here, the first is if you click on memory and
the connection count is not 0, then you will be unable to reboot the
phone, all you can do then is power cycle it.
Secondly, to update the phone, you have to create 2 files, the first
is entered into the
you kill the mpg123 processes?
-Bryce
[EMAIL PROTECTED]
NOTICE: The views expressed in this e-mail do not neccesarily reflect
those of my employer. This is a personal e-mail and as such, the
opinions expressed are my own.
On Jul 12, 2005, at 11:52, Patrick Friedel wrote:
So I decided
OK, last showstopper that I just can't puzzle my way through - parking
calls with the snom phones. I get the two phones connected, I hit
transfer on one, the other phone goes to MOH and the first phone gives
me DT, so I dial 700 and hit the OK button. Call transferred, the SNOM
hangs up
users but it
would be nice to know how you are doing it.
If the gsm owner has activated the email2sms service than you can send
an sms message to gsm number@gin.nl. This will cause the gsm owner to
be charged so many have it turned off.
Regards,
Patrick
.
My configuration looks something like this:
sip.conf:
[mjg]
type=friend
username=mjg
context=sip
callerid=Masuo 6001
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6001
subscribecontext=sip
[pjf]
type=friend
username=pjf
context=sip
callerid=Patrick 6003
secret
base station with an EuroISDN interface?
Nope.
Regards,
Patrick
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On Thu, 2005-07-07 at 17:04 +0200, Frank Sautter wrote:
hi patrick,
Patrick schrieb:
Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
ftp
/app_capiHOLD.so: undefined
symbol: get_ast_capi_MessageNumber
Jul 4 22:56:58 WARNING[1013]: loader.c:523 load_modules: Loading module
app_capiHOLD.so failed!
Ouch ... error while writing audio data: : Broken pipe
I appreciate any suggestion how I can fix this.
Thanks and regards,
Patrick
built in
hands free operation
display size
codecs
communication protocol (SIP, h.323)
price
availability
reliability
know bugs / limitations
asterisk compatibility
If someone has done this recently that would save me some time
Patrick
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correctly.
Regards,
Patrick
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...
Set signalling to qsig in zapata.conf:
http://lists.digium.com/pipermail/asterisk-users/2005-February/091109.html
Other info:
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk+legacy+integration
Regards,
Patrick
to hook up my ISDN/BRI line through ETSI
and capi.conf to asterisk.
Regards,
Patrick
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Hi
We are about to buy several Snom phones.
Does anyone have warnings or advices against these phones ?
Our finalists were Cisco, Polycom and Snom.
We will be using only the SIP protocol.
Thanks
Patrick
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Asterisk
, updated to latest HEAD and all is
well again.
Regards,
Patrick
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: No translator path from
unknown to unknown
translate.c:134 ast_translator_build_path: No translator path from
unknown to alaw
Did you try using a codec that is supported by asterisk and each phone.
alaw or ulaw would be a good start.
Regards,
Patrick
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you could try using ulaw and see if that improves it. Or try using
app_conference and see if that works for you:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_conference
Regards,
Patrick
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had quite a lot of money in their accounts:
Level 3: $2,500
Palavon: $10,000
VoiceConduits: $10,000
Telesthethics: $24,500
Surprising to see a company like Level 3 do business with them.
Regards,
Patrick
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all the cards?
Regards,
Patrick
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is
suffering from. Wouldn't surprise me if they both shared a dark cold
slippery cave with Gollum hidden deep inside the code :)
Regards,
Patrick
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.
Regards,
Patrick
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Hi all,
I was reading http://bugs.digium.com/view.php?id=2639 and it seems that
anthm's great native MoH patch only works on HEAD. Does anyone have a
version of the native MoH patch that works on 1.0.8? If so please point
me to its location or email it off-list.
Thanks and regards,
Patrick
Upgrading
Software on the screen. It then continues to re-request the same image from
the tftp server at 10s intervals indefinitely. What am I doing wrong?
Thanks
Patrick
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that is valid for my network, but
then never attempts to connect to any TFTP server. I then set up a dummy
network (DHCP, TFTP server) which matches the network parameters of the
legacy config, and I still don't see any TFTP requests. Any suggestions on
what to do next? I'm out of ideas...
Patrick
something with DHCP, but
never generates a TFTP request - apparently?
Patrick
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. The phone automatically
connects to the dock'n'talk when it comes into bluetooth range.
If you are happy with a fixed solution, where you leave the SIM permanently
installed, you might want to look for a Nokia Premicell or equivalent on
e-bay. This would also connect to a standard FXO port.
HTH
Patrick
/remove PRI cards without
shutting down the asterisk system
Is there a solution that exist ?
Someone told me to look at the C-PCI technology, it seems that telecom
companies use this.
Thanks
Patrick
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,
Patrick
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/remove PRI cards without
shutting down the system
Is there such a thing as an asterisk compatible hot-swappable PRI card and
board ?
Someone told me to look at the C-PCI technology, it seems that telecom
company use this.
Thanks
Patrick
Thx Scott,
However, and from what I can read, chan_sccp is not really meant for
server-to-server trunks, yet...
Am I wrong ?
BR, - Patrick -
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Herrick
Sent: jeudi, 26. mai 2005 04:24
To: Asterisk
Hi,
I have the following config:
[7960] --skinny-- [Cisco CCM] --SIP_trunk-- [asterisk] --SIP--
[X-lite]
Is there a chance to avoid the RTP stream from passing through the Cisco
CCM ? I would like to have all RTP handled by the *.
This is just a testbed, for a larger project. What I want to
I use * @Home for my home and home office lines. Nothing to exciting except
that it is full VoIP, no POTS lines allowed! This gives me access to
near-worldwide $.02 calling through teliax, and allows us to use exising
cordless phones via an ATA device, and a Cisco 7960 for the business lines,
In the interest of factual correctness, Warren Buffet, much to his chagin, does
not get to set interest rates!
Quoting Colin Anderson [EMAIL PROTECTED]:
they come out with ANOTHER box that they will loose even MORE money on. It
will be as good as, or better than Cell, run on broadband, and
This is a little off topic, but I cant seem to hit
the right google keywords to get an answer to what should be a simple question.
My [EMAIL PROTECTED] box sits behind a firewall and needs to use an
internal host to relay all email (voicemail notifications). I cant for
the life of me
Did you dial the 800 number correctly? You need to dial *1800XXX. I
had this problem for a while and then checked out the docs on FWD's website.
Any toll-free number seems to require a * before dialing. You can setup
your dialing prefixes to add it automatically so it becomes transparent to
are there in the extensions.conf file.
- Original Message -
From: Patrick M. Gray, Jr. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 04, 2005 1:02 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
I found where
: Patrick M. Gray, Jr. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, May 04, 2005 1:02 PM
Subject: Re: [Asterisk-Users] 7960 'multi-line' configuration
I found where it's getting this flag in the agi script, but I'm
Yes.
Quoting Henry Devito [EMAIL PROTECTED]:
Are you using asterisk @ home?
- Original Message -
From: Patrick M. Gray, Jr. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 03, 2005 9:22 PM
Subject: Re
?
Thanks!
Pat
Quoting Henry Devito [EMAIL PROTECTED]:
It has something to do with the AGI script. Scroll down!
- Original Message -
From: Patrick M. Gray, Jr. [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday
My 7960 doesn't behave this way. With all usernames/display names/extensions
the same, a second incoming call goes directly to voicemail. I'm on SIP
firmware 7.1... could that be part of the problem or is it likely something in
*?
Thanks!
Pat
Quoting Henry Devito [EMAIL PROTECTED]:
]:
On Tue, 2005-05-03 at 08:25 -0400, Patrick M. Gray, Jr. wrote:
My 7960 doesn't behave this way. With all usernames/display
names/extensions
the same, a second incoming call goes directly to voicemail. I'm on SIP
firmware 7.1... could that be part of the problem or is it likely something
Great info! The only question I would have is on the call waiting setting.
What should it be set to, and is the setting the one in the SIPX.conf file?
Pat
Quoting Corey S. McFadden [EMAIL PROTECTED]:
Guys,
I added some content to the Wiki on this feature. I don't think it's well
I still can't get the multi-line magic to happen. When I get the second call,
this is what appears on the CLI.
Any ideas?
Thanks!
Pat
dialparties.agi: Caller ID is not set
-- dialparties.agi: Added extension 200 to extension map
-- dialparties.agi: Extension 200 cf is disabled
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