The sipura 2100 does work good with a AS5300
Zoa wrote:
Does anybody know what devices really support t.38 ? I've seen a few
claiming they do on the box, but most do not seem to support it at all.
Zoa.
Kristian Kielhofner wrote:
Olle E Johansson wrote:
Friends in the Asterisk.org
Shawn, you ever get a fix for this problem?
samples are at
http://tumtum.no-ip.com/faxes/1128432831.3.tif
http://tumtum.no-ip.com/faxes/853107320051004-150908.tif
Both of these were faxed from a Brother intellifax 750 through a ring-it
single-line simulator into my asterisk box (through
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more error
routing in a real dial
See the message I post right before this one for a simple example.
Ray Yang wrote:
Apart from the dial plan issue, can anyone let Asterisk act like Cisco GW to
accept SIP call without registered in advance?
I've tried this for a long time but no answer yet.
, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really
struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need
Yes, asterisk makes a better voip to pstn gateway then Cisco.
Asterisk has more advanced call routing and restrictions then Cisco gear.
Nitesh Divecha wrote:
Hello,
Is it possible to implement Asterisk as a Gateway? For example like
Cisco 5300 or 5400 with 4 T1.
I was planning to buy
a clue as to the problem? Asterisk 1.0.9
sip.conf:
[bna-vonx-iad]
type=friend
context=trusted-out
host=192.168.7.250
canreinvite=no
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.250:5060;branch=z9hG4bK1A60
From: James Sizemore sip:[EMAIL PROTECTED];tag=19D8A640
the peer lookup. This is a * bug then. I have
documented it with both 1.0.9 and 1.2.1. Time to dig through the sip code.
James Sizemore wrote:
when my Cisco IAD send a call to my Asterisk gateway the gateway treats
it as if I don't have a peer statement in sip.conf, when I do. Here
I am trying to trouble shoot some problems with DTMF over PRI. I have a
digium wct1xxp card and these lines in extensions.conf:
exten = 5556000,1,Record(testtone:gsm)
exten = 5556000,2,Wait(2)
exten = 5556000,3,Playback(testtone)
I called in over the PSTN --to-- Asterisk. I did a pri debug,
Does the SPA-941 support stun?
Kerry Garrison wrote:
My two favorite phones (in order) are:
Linksys SPA-941
http://voipspeak.net/index.php?option=com_contenttask=viewid=41
Grandstream GXP-2000
http://voipspeak.net/index.php?option=com_contenttask=viewid=25
The problem is the change of
I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends
DTMF via sip info packets to another beta1 box. The peer is set to
receive info. What I get is a click sound and a very very short tone.
Sound like to me that I get the first part of the tone before it is
captured and put
192.168.1.36:5060
From: sip:101 at 192.168.1.36;tag=43
To: sip:201 at 192.168.1.38;tag=9753
Call-ID: 100450864100 at 192.168.1.36
CSeq: 3 INFO
Content-Length: 26
Content-Type: application/dtmf-relay
Signal= 2
Duration= 110
James Sizemore wrote:
I have a beta1 gateway with a 4 port card in PRI mode
I just upgraded to beta1 and everything does seem to be working, however
when reloading asterisk I see these error messages:
-- Reloading module 'chan_zap.so' (Zapata Telephony w/PRI)
== Parsing '/etc/asterisk/zapata.conf': Found
Oct 29 20:33:13 WARNING[10141]: chan_zap.c:10593 setup_zap:
Where can I change the Duration length of an INFO
packet?
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=5
Duration=250
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They are good phones they allow you to use speed-dial and hints to one
button park and intercom. The only bad thing I can say about them is
that there is no ground loop detection when using cinch headsets.
I only ever had one user complaint with the Snom phones, some users
don't like how soft
exten = s,1,SetVar(ALERT_INFO=Bellcore-dr4)
Bruce Komito wrote:
Greetings, I am trying to implement distinctive ringing on a Cisco 7960.
I have tried setting alert_info to chirp1 or chirp2 before dialing the
phone, but it has no affect. If you have successfully implemented
distinctive ringing
Fedora core 3 supports SATA on that model.
listas iPfone wrote:
Hi All,
I´m tryingo to install asterisk in an PROLIANT ML 150 G2 SATA and can´t
make it work because linux cant recognize the Hd (HP 160 mb).
No drivers for Centos ...Red Hat... i´t´s drivig me crazy..
Someone have a tip? if i
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
; notification. Inband indication, as used by Asterisk doesn't seem to ;work
; with all telcos.
; outofband: Signal Busy/Congestion out of band with
;RELEASE/DISCONNECT
; inband:
Install a smp kernel and you will use IO-APIC instead of XT-PIC you
typically will not share interrupts in APIC mode because it has twice
the numbers of interrupts to use.
Ronald Hartmann wrote:
Good Day list,
I am having some issues with my card in that it wants to
share IRQs
I have a PRI that if you dial a number that is busy, the channel does
not hang up, it then sends h|1to the phone company which will then
plays back to the end sip user You don't need to dial a one or zero
I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the
important bit from the
Already did.
Thanks for taking a stab though.
Brancaleoni Matteo wrote:
hi,
Il giorno mar, 01-02-2005 alle 13:30 -0600, James Sizemore ha scritto:
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = h,1,Hangup
try
extensions.conf:
[trunk]
exten = _X.,1,Dial(${TRUNK}/${EXTEN
I have a PRI that if you dial a number that is busy, the channel does
not hang up, it then sends h|1to the phone company which will then
plays back to the end sip user You don't need to dial a one or zero
I am running stable CVS-v1-0-01/20/05-02:45:17. I have pasted the
important bit from the
Try commenting out
;echocancel=yes
;echotraining=yes
I bet your faxs start working in both directions. But of course you will
now have
occasional echo problems.
Andrew Kohlsmith wrote:
On December 23, 2004 08:29 pm, Steve Underwood wrote:
This point is interesting. On most systems, if you
Do you have a wait(2) before your dial(SIP/) ?
You need to allow a full ring before you build your first sip
packet.
Jeremy Bogan wrote:
Yeah I have callerid=asreceived in my zapata.conf still nothing
unfortunately.
I get that when the calling party has caller id blocked on their end.
signalling=pri_cpe
callerid=asreceived
I see that I get the callerID CNAM in the cdr records, but the
same information does not show up on the display on my Cisco 7960
phone only the ANI. I do get Callerid from voip to voip calls .
Just not on the zap to voip calls.
My question is does
I have callerid setup on my PRI coming into my T100p and I know
this works because I can see the CNAM in my cdr records. But even
with callerid=asreceived set in zapata.conf I only get ANI to the
sip devices. Any ideals what I could have goofed up?
[channels]
rxgain=2.0
txgain=2.0
I have noticed a problem with the Cisco 7940/7960 phones where if
you put your voice-mail box on hold using soft keys and come back
you can no longer navigate. I am curious if anyone else can
duplicate this problem. Happens reliably for me with the 7940
phones.
I use rfc2833 for DTMF. I
Ex-girlfriend logic not working in latest CVS?
Incoming sip calls don't work. Anyone else seen this
problem?
Extension logic looks good:
exten = 6153248305/_931NXXX,1,Queue(queue1);
exten = 6153248305/_615NXXX,1,Queue(queue2);
;exten = 6153248305,1,Queue(queue3);
show dialplan looks good:
I would be interested.
Tenorio, Leandro wrote:
Seshu, I'm interested could u provide more info...
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu
Sent: Wednesday, July 14, 2004 11:02 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users]
I checked-out CVS Head today to get realm support, I have over hundred
Cisco phone on my servers and I have
not noticed any Qos problems. You may want to check the duplex of your
switches and Asterisk boxes. If you
don't have full duplex, that is more then likely your problem.
Brian Cuthie
Olle E. Johansson wrote:
James Sizemore wrote:
Has anyone else changed the Digest realm? Did you have any odd
problems?
In the chan_sip2 module, I've a setting called realm= in sip.conf
Time to port that over to chan_sip.
No, it doesn't cause any harm. On the contrary, the RFC states
Make sure you have canreinvite=yes in all peers in sip.conf that the
call goes through.
Also making sure that you don't have tT on any of your Dial
statements in extension.conf.
But your real problem is that you have some type of network problem use
mii-tool eth0
at a bash prompt, and make
Check the duplex on your ethernet conection on both the Cisco and the
Asterisk box. Make sure neither are half duplex.
Joseph wrote:
What codec should be used to connect a * box to
a cisco router which has a t1 with 24 trunks coming in?
My router voip dial plan looks like this:
dial-peer voice
?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore Posted At: Friday, March 12, 2004 2:58 PM Posted To: Asterisk
User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: Re: [Asterisk-Users] chan_sip.c
PROTECTED] On Behalf Of James
Sizemore
Posted At: Thursday, March 11, 2004 10:47 AM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
You
Andrew Gillham wrote:
Sounds good. I have not been that bothered with it when I make a
normal voice call.
It is mostly annoying when hitting the messages button on the phone.
My delay helped
that situation.
Perhaps on calls where asterisk is proxying the rtp stream we could
have an option
, but still no go... When the
phone registers with port 2842? Not the standard 5060? Any ideas? I
believe this is where my problem sits...
Thanks,
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore
Posted At: Friday, March 12, 2004 9:03 AM
69.133.182.77: icmp_seq=5 ttl=241 time=57.9 ms 64 bytes from
69.133.182.77: icmp_seq=6 ttl=241 time=56.0 ms 64 bytes from
69.133.182.77: icmp_seq=7 ttl=241 time=54.0 ms
Any thoughts there?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore
exten = 6500,1,Answer
exten = 6500,2,Wait,1
exten = 6500,3,VoicemailMain2
Or should I say, Me too!
Is this the bug for the case in question?
CSCed48311: Media takes 0.4 sec to be set up
Thanks.
-Andrew
Yes the problem is that when making outgoing calls, there is enough of a
delay in the
You do have :
nat_enable: 1
nat_received_processing: 1
On the Ciscos?
AstGrp wrote:
I am having a similar problem... I get the same message, but inbound
calls can go through This is only Cisco phones that are behind NAT.
I have tried your recommendations from below, but still no luck.. User
Thanks for the information. You have saved me a few hours on the phone
with TAC. smile
Low, Adam wrote:
We have a TAC case open on this issue (reference DDTS CSCed48311) as well, apparently it was going to be fixed in a 7.1 release (yes I know we're only at 6.2 but that's what Cisco stated)
When calling out on a Cisco 7960 there is a short delay before the call
gets setup and the other side can hear your voice.
Anyone know how to compensate for this effect?
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I can't believe you would add anymore digits to listen for.
I have thought about speeding up the digit play back.
It seems to take forever when waiting for 7.0.1 smile
Jim Sneeringer wrote:
Actually, it works fine as long as the parkpos values are numbers. If you
put in a * or #, it
You could always create a rule to match any-e-thing 3 or 4 digits, that
always forwards to the receptionist
[match_all_local]
exten = _NXXX,1,Goto(receptionist|s|1)
exten = _NXX,1,Goto(receptionist|s|1)
[trunk]
include = localnumbers
include = match_all_local
include = international
include =
You can only use the r option if you answer the call first
exten = 106,1,Answer
exten = 106,1,Dial(SIP/106,30,tr)
other wise remove the r
Erick Weber V. wrote:
Thank you very much
I just make the change and I'm up an running.
One more quick question, why I can not hear the ring in the phone
exten = 555,1,SetVar(Bellcore-dr1)
Will do what you want.
Andreas Anderson wrote:
Hiya,
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can trigger other
ringtones on the phone (in the section Support for SIP Alert-Info
This will give you what you want.I type a little to fast for
the brain buffer sometimes.
exten = 555,1,SetVar(ALERT_INFO=Bellcore-dr1)
Andreas Anderson wrote:
Hiya,
Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
straightforward. The release notes indicate that you can
A better option and one Asterisk desperately needs is some kind of
--lint option,
Which would check the config for errors and useless misspelled options.
smile
I personal find one or more typos or misspelling a month, On my PBXs.
Eric Wieling wrote:
Maybe someone will write a patch to print an
I'm interested.
TeleSIP wrote:
I'll try to hack a NAT friendly tftp server on monday.
Are you still looking for it? I found one if you need it. Let me know and
I will post the info.
Andres.
--
Nicolas Bougues
Axialys Interactive
___
I have a need to make the outgoing volume on a meetme room
louder for an all sip setup. I could use txgain if I had
Zaptel devices in the loop. I do not think I have that
option with sip? Any Ideals?
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Do you type reload at the cli a few times a day?
If so try not reloading Asterisk and I'll bet Asterisk
stop blocking. If you don't normally reload the box
you will need to trouble shot normally.
mattf wrote:
Hello,
I have had several instances over the last month of Asterisk freezing,
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
So I could figure out the average time between crashs just by log level
and call volume! LOL
This is with out running into a single bug. smile Thankful I can
restart Asterisk from
time
What bug # should I look for you patch under? smile
Andrew Kohlsmith wrote:
You can not rotate logs with out dropping calls, and if logs get a
little over 2Gbs Asterisk will crashes...
Why not? Why are the logfiles kept open for the entire life of Asterisk?
Hell even my heavily loaded
I did not even know about it! But seeing as it is not in the change
log no wonder?
You have the bug number the notes are under for usage?
[EMAIL PROTECTED] asterisk]# grep log ChangeLog
-- Agent Callback-login support
-- Added Dialogic VOX file format support
Andrew Kohlsmith wrote:
You can
I would like to setup a hunt group, not a group ring, using sip phones.
Anyone done this with sip devices? Comments suggestions?
I have not had much luck with the outgoinglimit=1, incominglimit=1
stuff that I would need to get busy extinctions to work right, which is
why I'm asking on the list.
Vocal has redundancy.
Asterisk has features.
They both have bugs. smile
What a choice!
costas wrote:
I can't resist asking. What do you think of Vocal as compared to *? Anything Vocal has but missing in *?
-- Original Message --
From: Scott England
My guess is you need something like this:
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
Andrew Thompson wrote:
- Original Message -
From: jerk face [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, November 17, 2003 3:53 PM
Subject:
Amen
Lars Boegild Thomsen wrote:
I won't agree that RADIUS shouldn't ever have been deployed in a VoIP
environment. While it can be argued that RADIUS is not in any way an ideal
solution and it can also be argued if it is necessary in a PBX software such
as Asterisk, fact is that many IP
You will need to check with Cisco to see if the ATA188 has the same issues
with faxing as the ATA186.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Dave Weis wrote:
Should I expect a standard fax machine connected to an ata-188 connected
to an
I have been having a lot of problems with SIP calls and
gotos within contexts as well as between contexts.
They work some of the time and fail some of the time
but the console reads the same either way. Am I the
only one having this problem? A little sample config
below.
[macro-stdexten]
exten
Just a note, there is a bug with queue stats when you reload
Asterisk dynamic
users will not reset to 0 calls but as Troy suggested default queue
members will.
Meaning that if you stay in your set and take calls all day and the guy next
to you removes himself from the queue to go smoke, when
Does this have a bug number so I can track it?
Jan Janak wrote:
On 07-11 13:17, John Todd wrote:
From what I can understand of the issue you describe, it sounds like
the problem resides on the remote side, and not Asterisk's side.
You are sending an invalid request in your first query, and
Asterisk would need scalability and redundancy on the voip side to
play in the soft-switch area. The biggest issue stopping Asterisk having
redundancy and scalability using sip is the inability to work with just
about any sip device without canreinvite turn off. If Asterisk could
handled
Besides you got list four times since May!smile
http://slashdot.org/search.pl?query=asterisk
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Thanks a bunch you were on the money. Do you know about when that changed?
John Todd wrote:
Grandstreams phones can't call out with the latest CVS, anyone know
what the
last good CVS date was?
You may be experiencing difficulty due to bad codec permissions, since
the latest CVS updated
The ATA186 is only rated to 9600 baud, it is not usable for faxing as
most faxs
are 19200. See Cisco sight for details.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Eric Wieling wrote:
If you have ANY chance of sending a fax over VoIP your
Grandstreams phones can't call out with the latest CVS, anyone know what the
last good CVS date was?
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To be a true ip tel softswitch, Asterisk would need SS7 support.
No one is working on SS7 signaling for Asterisk.
WipeOut wrote:
Victor Medrano wrote:
Asterisk will become a real ip tel softswitch or is going to other way ?
like vovida
regards
It is already an IP softswitch.. Or may
Yes
Ing. Angel Gomez Garcia wrote:
Hello.
I have this issue, when I pickup a call that is ringing in a SIP
Phone, it keeps ringing.
There is bug #116 that mention something about these, but it does
not seem to be resolved , at least, not yet.
Anybody else has seen it behavior ?
10 Fix call waiting tone.
9Fix the tftp configs so that I can host my own provisioning server.
Or make a command prompt based tool kit, so that I can use
Gaps with out writing a http screen scraper.
4 Having the Conference button do something would be cool.
John Brown (CV) wrote:
Agreed, don't drive up my shipping cost. light is good.
Tilghman Lesher wrote:
I'd have to respectfully disagree. If this is really a problem I'd
suggest taking advantage of the mounting bracket on the bottom
and either attach the phone to the desk or attach a sheet of lead.
-Tilghman
learn.to can kiss my *** smile I'll top quote till death, And so will
almost
every other person on this planet. grin
Tilghman Lesher wrote:
On Tuesday 14 October 2003 18:15, Uriel Carrasquilla wrote:
I have to tell you, at the expense of offending you, that I use
MS-Outlook and the responses
I wish you would take this stuff to personal email, I am tired of
wasting my time
reading this crap. If you idiots want to give lesions on how YOU
would like
people to post on list servers _DO_IT_VIA_PERSONAL_EMAIL_!!! None
of the rest of us care. This is a personal messages from you to
You will need :
extensions.conf
indications.conf
logger.conf
manager.conf
rtp.conf
sip.conf
modules.conf ; with a crap load of stuff turned off:
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
noload = chan_phone.so
noload =
Most PBX do park the way your old KSU system did.
As a matter of fact Asterisk is the only PBX I have ever seen
that parks the way it does.
If given a choice my uses would use the normal way. And I would
be happy not to here the question can you speed up her talking? LOL
Andrew Kohlsmith wrote:
Can a call be kicked out of a queue if it reaches a specific timeout?
I don't see an obvious way to do this in either queues.conf or
extensions.conf any pointers or patches to do this? smile
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Red Carpet will give you some serious dependency problems later down the
road.
Bisker, Scott (7805) wrote:
I found the best way to upgrade is install Red Carpet from www.ximian.com.
Subscribe to the RH 9.0 channel. And do a complete update. The only
drawback is that this method doesn't update
I have been testing Gastman and Astman with SIP calls. As I have no Zap
phones, so I have a few question on what is normal behavior? When a call
comes in and I have created extensions for all phones (example: Channel
= SIP\3846) Should the little lines connect between the pre-made
extension or
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user documentation hiding
in Internet land some place? If not there well
be soon. Ho hum.
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[EMAIL
Does any-e-one know if the 4 port FX0 cards will
be shipping anytime soon?
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there
were some keys for the users to skip the greetings or a key to
goto the VoiceMailMain from VoiceMail. But have not seen
any yet.
Olle E. Johansson wrote:
James Sizemore wrote:
Has anyone browsed through the source code and
made a list of menu option for VoiceMailMain2?
Or know of some user
Here are a few outgoing gateway configs that work for me.
[vocal]
type=friend
host=1.1.1.7
insecure=1
port=5065
accountcode=memrtr
;dtmfmode=info
[cisco]
type=friend
host=1.1.1.3
insecure=1
canreinvite=no
port=5060
[EMAIL PROTECTED] tftpboot]# cat OS79XX.TXT
P0S30100
Get this image as well.
Shaun Ewing wrote:
Hello all,
I know this isn't strictly Asterisk, but I'm sure that there are more people
here using the Cisco 7960 w/ SIP, so I thought I'd post here.
I've just bought a Cisco 7960 phone to use with
I use both Ciscos and Asterisk as Sip gateways to pstn.
I can say a lot of good things about both, and a few bad
things as well.
The Ciscos are a very solid product with very good very fast tech
support. It also has some really nifty fax detection with
redirection via email options (AS5300 and
If one is using SIP the CVS-current can be extremely unstable.
I would say about half the time I have tried a new CVS checkout
on a test box. (about once a week) I have had lockups or missing
features. I like Asterisk and CVS but with out testing in a semi
large environment the cvs -current is
Know bug
http://bugs.digium.com/bug_view_page.php?bug_id=116
Pertti Pikkarainen wrote:
I have problems with this as well ( similar config ). My CVS is 10
days old.
I can get the call picked up with *8 ( *8# does not work ) but
the phone B never stops ringing.
B rings forever. I'm
know bug http://bugs.digium.com/bug_view_page.php?bug_id=116
WipeOut . wrote:
I have just started to play with callgroups and pickupgroups..
I updates my * from CVS this morning (about 15 mins ago)..
I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in
I know *8 kind of works with SIP
but what about the rest should they work
do they work with a zap device?
*0# sends flash
*8# remote call pickup (pickup phone in your group)
*67# disable caller id
*70# no call waiting
*78# do not disturb on
*79# do not disturb off
*72# enable call forwarding
I agree the * functions should be selectable and doing this
at the dial plan level seems a good place to put them.
Having not look at the code. I don't know how hard
this will be.
Digium wants $150 an hour for contract work, I wonder
if Mark could give me a quote on adding these features either
How hard do you think it would be?
Tilghman Lesher wrote:
On Monday 08 September 2003 01:56 pm, Brian West wrote:
I agree they should stay at the dialplan level.
It's not a matter of staying; it's a matter of moving. Those
features are already present within the chan_zap channel
driver.
If you put Tt in your dial statement you can type # some number to
transfer to.
Of if you can send flash hooks that will work as well.
Dave Alan Caruana wrote:
what i'm asking is what is the key sequence
you have to dial for the transfer ..
it was something like *7# if I remember
well, I know
Yes its a known issue.
http://bugs.digium.com/bug_view_page.php?bug_id=116
Louis-David Mitterrand wrote:
Hello,
As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco
7960 with *8 but the extension then keeps ringing indefinitely, even
though I picked up the call.
Is this
The client device has to support stun.
Bugetones do, ATA do, 7960 don't..etc
Dave Cotton wrote:
On Wed, 2003-09-03 at 09:01, WipeOut . wrote:
Sorry to answer a question with a question..
Can stund and * be loaded on the same server and run at the same time?
I've also never been
Someone told you wrong. Works fine, volume is a little low however
without powered headsets.
Erik Anderson wrote:
I converted my Cisco 7960 to SIP. I did not try the headset port because I
was told that Cisco did not enable the headset port for SIP.
Erik
-Original Message-
From:
screechy.
If this is something anybody wants to see more about,
I'll be happy to provide more info.
Thanks,
Brenton Rothchild
- Original Message -
From: James Sizemore [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 25, 2003 1:42 PM
Subject: Re: [Asterisk-Users] Is the DTMF bug
DTMF tones are not long enough on out going calls, when I'm using either
info or rfc2833. Does anyone know if the tone length value is in rtp.c
or chan_sip.c ?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
I have not had any problem at all with the 10 I have.
They sound good and work well. The only problem I
ever had was a problem with remote ntp servers.
Andres wrote:
Hi,
I would like to know if others have experienced a high percentage of Budgetone
defective units. We purchased 4 to test
Blind and assisted transfer work with Cisco 7960 phones.
Blind transfer works fine with Budgetones.
As long as you register to Asterisk.
Jamie Carl wrote:
Ok, just been thinking about this and thought I would ask before
trying it out again.
What is the state of SIP transfers? By this I mean
Do you have transfer turn on in zapata.conf?
transfer=yes
Hi,
I cannot use '#' to initiate transfers.
I have tried on different phones (7960, ATA, X-Lite).
When I press '#' during a call, nothing happen.
I have both T and t switches in Dial application.
The transfer function works with Flash key
If you add t to you out-going trunk Dial lines:
exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t)
exten = _NXX,2,Congestion
so that you can still use park to park a call or transfer
the phones, You have a problem of not being able to use
# on external IVR systems. Is there any solution
to
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