You can try using the -p option to Asterisk.
Mark
On Wed, 2 Apr 2003, Jeff McClure wrote:
Good points. This system currently does not use any SIP or IAX channels (or
any other form of VoIP) and only deals with 1 call at a time (the single
FXO channel is the only link to the outside). At some
Does running it under valgrind produce more useful output?
Mark
On Wed, 2 Apr 2003, Alex Zarubin wrote:
OK, here it is. On a flow of shorter calls it lasted about an hour.
[EMAIL PROTECTED] asterisk]# gdb asterisk core.12348
GNU gdb Red Hat Linux 7.x (5.0rh-15) (MI_OUT)
Copyright 2001 Free
But if I try to call from one of them to the other, the remote end rings
just fine in both cases, but then as soon as asterisk bridges the two
channels, the remote end sends a Call/Leg Transaction Doesn't Exist
error and hangs up the line.
Apparently it doesn't like our reinvites for some
When I look at 'show dialplan' on the console the order of the various
entries has been shifted around in a different order to the way it was
entered into the extensions.conf file.. So the file must not processed
sequentially..
Generally entries are stored in increasing alphabetical order.
I have noticed when I add dtmfmode=inband under the [general] section
in sip.conf I get flooded with warnings on the console after asterisk
answers a sip call...
WARNING[16401]: File dsp.c, Line 1106 (ast_dsp_process): Unable to
detect process 2 frames
WARNING[16401]: File dsp.c, Line 1106
Not all interfaces support transmitting audio before the call is answered.
It may be necessary to answer the line first, if you haven't already.
mark
On Sun, 30 Mar 2003 [EMAIL PROTECTED] wrote:
it is possible to use musiconhold. i added
exten = s,5,SetMusicOnHold,default
exten =
I have seen a few posts that state that the hold and transfer features
on the SNOM and Cisco phones do not work in themselves and that,
instead, configuring the # key is the way to get transfers to work.
Can somebody please give me an indication as to why this is and what
needs to be done,
what is WMI ?
Something that now works on the SNOM 100/200 :) It's the message waiting
indicator which tells you if you have voicemail.
In order to use MWI, be sure to put mailbox=foo in your friend
declaration for it.
Mark
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Fixed, thanks :)
Mark
On Fri, 28 Mar 2003, Stephen Davies wrote:
On Fri, 28 Mar 2003, Mark Spencer wrote:
Last night I committed SIP retransmission support into Asterisk. Let me
know if this helps/hurts for anyone. Thanks!
You set a 15 second autokill timer but never cancel it. So
Turn off reinvites and that will likely fix this.
Notice how the first invite is totally ignored, and then for some reason
the second gives us the 481.
Mark
On Sat, 29 Mar 2003, Luke Howard wrote:
This seems to fix incoming calls but outgoing calls terminate
immediately, at least for me,
try turning off re-invite.
Mark
On Sat, 29 Mar 2003, Luke Howard wrote:
Latest CVS breaks outgoing SIP calls for me after a second or so
of audio (if that).
-- Executing Macro(SIP/515-Office-b922, iconnecthere|18006822878|60) in new
stack
-- Executing Dial(SIP/515-Office-b922,
Last night I committed SIP retransmission support into Asterisk. Let me
know if this helps/hurts for anyone. Thanks!
Mark
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you briefly tap the hook switch.
mark
On Fri, 28 Mar 2003, denon wrote:
How do I do a manual flash hook? And sorry, I didn't elaborate enough - I'm
using the Nortel Vista 390. I don't see a flash button anywhere, only the
link. And the link doesn't work when I load it with *'s adsiprog.
My extension definition for the conference room looks like this:
exten = 8600,1,Wait,1
exten = 8600,2,Playback(wstconfbeta)
exten = 8600,3,Meetmecount,8600
exten = 8600,4,Meetme,8600|p|1234
PIN is not yet implemented apparently. You could in the mean-time use
Authenticate like this:
exten
Quick question what happens if you go over
your channel licenses?
It cannot transcode.
Mark
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Using DISA.
Mark
On Thu, 27 Mar 2003, Eric Wieling wrote:
Is there any way to require a caller to enter their customer
number when they call in AND have the info put in the CDR info?
Also is there a way do to the same for outbound calls?
--Eric
as I understand it there are 2 h323 modules in the works (as well as others
for unrelated stuff.) It would be better for everyone if these were all in
one place (main cvs repository). The default modules.conf could simply be
not to load them, and notes should be provided as to their current
We've done 60 channels on a dual 1.8 Ghz Xeon. Trial channels are *not*
available because we have to purchase keys from Voiceage, and they are
unwilling to make any trial keys available.
Mark
On 27 Mar 2003, Jared Smith wrote:
That's my question exactly... How many concurrent calls can I run
I have gotten relatively far with support from Sayson and Aastra, but
the vibe I'm getting at this point is that ADSI is a standard, we
implement it but we're not responsible for helping you develop your
implementation. The specs are available from Telcordia (and perhaps
belcor?) for a
The separator on app_record was ':' instead of '|'. I've modified it to
accept either one. Please try a cvs update.
Mark
On Thu, 27 Mar 2003, Michael K. Rodriguez wrote:
This the error I receive when I try to record a GSM file
-- Executing Record(SIP/67.98.37.220:5060, intro|gsm)
Will to ports on this card be able to act as FXO as well, or just as FXS?
If the answer is yes, can we control which ports do which in any
combination? Finally, can this card coexist with the X100P FXO card in the
same PC and will Asterisk support them all at the same time?
They are FXS
That is an annoying, arguably misfeature, of the Aastra. The idea is that
the use of the programmed buttons should eliminate the need for the Link
button since manual flash hooks can get your phone out of sync. Don't
worry you can use manual flash hooks in the mean time.
Mark
On Thu, 27 Mar
On every call using the X101P I get
Unable to set audio mode on channel 1 but I got no problem
This error is fixed in CVS. While the error itself is harmless, some
other code that was included in the same patch could potentially cause
problems with echo and even possibly a segfault.
Mark
Seems that there're some problems with dtmf detection chan_zap.
(today CVS)
Should be fixed. Contact me off-list if it isn't so I can debug it with
you.
Mark
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Be sure to use the linux version and not the windows version.
Mark
On Wed, 26 Mar 2003, Warren Bird wrote:
Hi Roy,
I have taken a look at Gastman, but unfortunately it always crashes as soon
as a call comes through. So not too much help.
I see that there is an Asterisk Management System
How about we say yes only if there are new messages
cvs update and let me know if that worked.
Mark
On Wed, 26 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
Looking at the cvs rdiff from 1.5 to 1.6, apparently the behavior changed?
The old way it just checked inbox for new
Update your libpri
On Tue, 25 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
gcc -shared -Xlinker -x -o chan_phone.so chan_phone.o
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations
-g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6
Not sure how I can look inside to see what asterisk could be doing. .
While it's running, you can do ps auxww | grep asterisk. Look for the
PID of the process taking the CPU. For example:
root 1206 0.0 0.9 118576 4904 pts/2 SMar21 0:00 ./asterisk -vvvgc
.
.
.
root 1216
Digium has provided some cards to the David Sugar of the Bayonne project,
as well as made ourselves available for support, but as far as I know,
they have not added support for the boards yet. I would suggest
contacting David Sugar for more information.
Mark
On Sat, 22 Mar 2003, d hinton wrote:
Is there a Record-Route header in the response that comes back from
iconnect?
Mark
On Sun, 23 Mar 2003, Luke Howard wrote:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
Tried that, doesn't work.
I should add that in my config I'm totally behind NAT, both
It *shouldn't* be necessary to answer the line before executing dial. At
the point the SIP phone answers, it should *then* answer the X100P.
As usual, find me on IRC and I can spend some time debugging it. See
http://www.digium.com for IRC info.
Mark
On Sun, 23 Mar 2003, Steven Critchfield
Whoa, this is pretty weird. Somebody find me on IRC and lets get this
fixed!
Mark
On 22 Mar 2003, Matteo Brancaleoni wrote:
Ciao lele.
Yes, this is known.
also, if you use inband dtmf you can't log into the voicemail
(doesn't recon. the dtmf), but voicemail works with rfc2833
matteo
In case I typed it wrong:
http://www.digium.com/handbook-draft.pdf
Mark
On Fri, 21 Mar 2003, Brian Capouch wrote:
But the link in Mark's mail to the pdf of the rev II manual comes up
Cannot find link target or somesuch.
Is there something wrong with the server, or is it on my end?
B.
You mean Greg? He will be on vacation this coming week but Call me (x
6275) and I'll try to find them for you.
Mark
On Fri, 21 Mar 2003, d hinton wrote:
hi i sent gary an email about those fcc #'s. no response yet.
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have you tried nat=1 in your friend declaration? I notice in your dump it
says non-NAT
Mark
On Fri, 21 Mar 2003, denon wrote:
Oh, and yes, the * is current as of a few days ago .. so it should have
that new SIP code mark was working on a while back.
Thanks
] /usr/src#
On Wednesday 19 March 2003 19:35, Mark Spencer wrote:
CVS is now back online, but here are a few things you all need to know:
1) cvs.digium.com is now on a new IP address. It may take some time for
this IP address (216.207.245.20) to get propagated. If your machine still
please provide the fcc cert # and any euro cert #
we need them thanks
dwayne
We have FCC certification numbers on file. We have not obtained Euro
approval on our cards. What's your need for fcc numbers, just curious?
Mark
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wait--- to connect these cards to us phone lines i believe you also need fcc
cert, is that true??
Yes, in principle you need Part 68 for the phone lines. Part 68 also
verifies high-pot, as does UL (i.e. it shouldn't catch on fire, even after
a lightening strike).
Mark
So how would a more complex definition like:
exten=1234,1,Dial,Zap/1|30|Tt
Either as
exten = 1234,1,Dial(Zap/1|30|Tt)
or
exten = 1234,1,Dial(Zap/1,30,Tt)
Asterisk automatically converts your , to a | if it's in the (). This
change was made to make it easier to read dialplans as though
sounds like you need to update your kernel-source RPM as well.
Mark
On Thu, 20 Mar 2003, Frank Hoonhout wrote:
I am in the process of trying out interesting software.
I setup Redhat 8.0 and updated everything. (including the kernel)
Now when I compile the code I get this error.
If you run asterisk with -vvvgc you can force Asterisk to dump core when
it crashes. then you can run:
gdb ./asterisk core.foo
.
.
.
(gdb) bt
And that will give you a backtrace of where the crash occured. Even just
the backtrace will give some useful information, but if you have even a
vague
Upsides:
- would allow me to use multiple analog lines with consistent
outbound appearance
- would allow me to overflow CID buffers to remove my original CID
with junk (marginally useful)
- ? others
Only some callerid boxes are susceptible to this, but you can use Asterisk
to
Call it a RDBMS, LDAP server or even an NIS map.
Asterisk is designed to support multiple switch backends, so you can put a
statement like this in your dialplan:
switch = IAX2/host/context
Which pulls in extensions from another remote host. But just as easily
one could create a switch for
Matteo, your fix looks like a good temporary solution, but I don't want to
merge it with CVS because the *right* thing that I *need* to do is to
implement retransmissions on SIP. Once retransmissions are in place, then
we can keep the channel around until we receive the 487 back that we
expect.
ms
I heard quite shortly moh sound at starting Dial sequance.
and playing RBT normally.
and already checking this problem following SIP terminals are
snom100, BCM HP.323, WinRTC v4.6.
---
Masakazu Nakano as [EMAIL PROTECTED]
On Mon, 17 Mar 2003 23:50:48 -0600 (CST)
Mark Spencer [EMAIL
If XML is important to your needs, why not write a translation script to
parse XML and write the asterisk configs? Scripting languages abound and are
appropriate to the task. Obviously, the transaltion script could grab your
XML and write fresh asterisk configs every time you started asterisk.
His 481 issue is not the same as your 487 issue.
Mark
On 18 Mar 2003, Brancaleoni Matteo wrote:
hi. read my mail 'bout 487 response.
I wrote a patch to fix that in chan_sip .
It's good for a occasional fix, until
mark updates chan_sip to handle retransmissions.
matteo.
Il mar,
There appears to be an Asterisk WebMin module on the Digium FTP site, in
some state of development. Is this being officially developed?
No, it hasn't been developed in at least 12 months. If anyone wants to
play with it they're welcome to.
Mark
We could implement a ringdown on MGCP or IAX, but SIP would not support
such a system.
Mark
On 18 Mar 2003, Stephen Webb wrote:
Does anyone know if this can be done by any VoIP Technology (SIP, IAX,
IAX2 or MGCP) I don't know the protocols!
On Tue, 2003-03-18 at 09:56, Steven Critchfield
modversions.h is actually created when you configure your kernel. You
need a kernel source tree which matches the kernel you're running.
I don't mean to start a distribution flame war on here, but if you don't
know how to compile a kernel (and don't really want to learn) install
RedHat and just
SIP Debugging Enabled
*CLI DEBUG[2051]: File chan_sip.c, Line 401 (create_addr): Setting NAT on RTP to 0
Interface is eth0
IP Address is 172.16.17.7
11 headers, 1 lines
XXX Need to handle Retransmitting XXX:
NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
Just update your CVS this bug was fixed earlier today.
Mark
On Mon, 17 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
-- Registered SIP '' at 192.70.239.2 port 5060 expires 3600
-- Executing Playback(SIP/lenny-4ee2, transfer|skip) in new stack
-- Executing
Will Asterisk support GSM with SIP? I'd like to deploy a snom phone at home
(which will likely only have 64k ISDN to the office) without necessarily
needing an Asterisk server there too (although doing so would, relative to
bandwidth costs, still be cheaper :-)).
It does, but for some reason
Fixed a big bug in the generator. This should fix musiconhold with SIP,
using Ringing before SIP, use of r flag with all sorts of channels, and
more.
Mark
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IAX2 now has support for a trunk mode (trunk=yes in the appropriate
friend section). Trunk mode allows IAX2 to use bandwidth extremely
effectively. The original impetice (and strategy) was a result of a
mistake in which it was claimed that Asterisk with a T100P could send 96
simultaneous calls
If you have parkedcalls included in the context your phone is in, that
should be sufficient. show dialplan should show you what Asterisk
actually believes your dialplan to be.
Mark
On Sun, 16 Mar 2003, James Sizemore wrote:
I got some time this week end to play with
this. By add the pickup
I don't understand how #define AST_FORMAT_ADPCM(1 5) becomes
a format = 32 in the * console display.
1 shift left 5 == 2 to the power of 5 = 32.
Mark
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This is what I have. In the MGCP box you have a few settings to deal with
as well, of course. Currently MGCP is not functioning well. The guys are
working on it. Right now, * will seg fault when you place station to
station calls.
Already fixed in CVS as of this morninga ctually.
Mark
Perhaps we should have BYEXTENSION print a warning that says the option is
deprecated, what do you think?
Mark
On 14 Mar 2003, Steven Critchfield wrote:
On Fri, 2003-03-14 at 10:22, Don Pobanz wrote:
We have a group of lines (FXO/FXS) between our Rolm PBX and our
Asterisk server. From the
Mark - I beg to differ. Generally callerid works with Deltathree; but
sometimes they seem to reject it / mess it up.
Nevermind, I was thinking this was incoming Caller*ID.
Mark
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The more logical breakup is asterisk-users and asterisk-dev which are both
there, and everyone who subscribed to [EMAIL PROTECTED] is now subscribed
to both the users and devel list.
Mark
On Thu, 13 Mar 2003, Roy Sigurd Karlsbakk wrote:
join both
On Thursday 13 March 2003 12:57, Michael
LIghtweight
Voice over IP
Exchange
Or:
Lightweight
Internet
Voice
Exchange
Mark
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Who is generating this ringback? The ATA or asterisk? What if I call
a non-suping number with a the number has been changed recording?
Will I never hear it because audio will never be cut through without
answer supervision?
Find out by doing a trace. If you're using callprogress, then you
Please check latest CVS. This issue has been fixed and was related to the
dynamic payload merger.
Mark
On Wed, 12 Mar 2003, Lubomir Christov wrote:
Hi Gregg,
I'm using iconnect with LineJACK/PhoneCARD and G723.1 codec from about 1
mount without any problems. The quality is perfect and
how does your cisco send DTMF?
Mark
On Wed, 12 Mar 2003, Lenny Tropiano / asterisk.org Mailing list wrote:
I had updated CVS this morning and it broke me being able
to call the voicemail extension from my SIP/Cisco 7960 phone
it won't receive DTMF digits...
Restored back to Mar 10 2003 and
there is a relaxed dtmf mode that may help.
Mark
On 12 Mar 2003, James Hines wrote:
On Wed, 2003-03-12 at 13:42, Brian J. Schrock wrote:
I am using background, the pbx-invalid stuff should (if DTMF
recognition is working correctly) not get played.
My users here have complained about
Are you running latest CVS? I think we addressed this issue a few days
ago with the hit='f' thing.
Mark
On Wed, 12 Mar 2003, Darrell Eldridge wrote:
I doubt it's a signal loss problem. It's a simple
circuit, connected through our local Meridian, so it's
fax-to-Meridian [A]
Meridian
Probably you should do dtmfmode=inband in the general section.
Mark
On 12 Mar 2003, Matthew Farley wrote:
Finally, I have NATted ATA-186s working with Asterisk (thanks to
all who made this happen)! My final troubles were with the firmware
version in the 186 -- if you have troubles with
it does need to service multiple lines at the same time.
Also as a background music (on loudspeaker type phones) would be nice.
We are currently playing with festival, so the caller can select which
area they want details for and we hourly download and massage the
hourly updates avail from
The friend would only happen if the From: was iconnect. Unfortuantely
SIP does not differentiate a user from Caller*ID. The only way to make the
peer match would be if we matched the peer based on IP address.
Mark
On Wed, 12 Mar 2003, Jim Archer wrote:
Hi All...
We have found that the
turn off the secret then, or tell the phone to call.
Mark
On Tue, 11 Mar 2003, Eric Wieling wrote:
Let us know if you ever get it working. I also have an ArrayVox
phone that I've never gotten working.
On Tue, Mar 11, 2003 at 05:16:27PM -0500, Raymond McKay wrote:
For some reason, this
Actually I think it was an issue with incrementing the sequence number on
the bye andshould be fixed now. RTCP is irrelevant in SIP signalling.
Mark
On Tue, 11 Mar 2003 [EMAIL PROTECTED] wrote:
1 - From watching the udp fly by, it seems that iconnect does not know
when we hang up. For
only for # transfer, not for flash-hook transfer.
Mark
On Sun, 9 Mar 2003, Mike Reiling wrote:
Didn't know you needed the t. Is that a new thing?
On Sunday, March 9, 2003, at 11:38 AM, TC wrote:
Anyone having trouble parking calls? I haven't tried it in a while,
but it seems to have
Check your zapata.conf too to be sure stripmsd=0 or is undefined.
Mark
On Sun, 9 Mar 2003, Raymond McKay wrote:
Are you stripping the digit you use to specify it is goin to be an
external call?
I don't use a digit for that purpose. I have always used
Try turning off the jitter buffer and/or using SIP and see if things
change. The IAX jitter buffer needs some work.
Mark
On Sat, 8 Mar 2003, Martin Pycko wrote:
but he was asking about iax too:
1. Is it normal that I get such a crappy quality with iax, some drops and
clicks?
Could anyone
GSM should now work with Microsoft. They understood our GSM and now we
understand theirs too.
Mark
On Mon, 24 Feb 2003, Dan Fernandez wrote:
Folks,
I cannot seem to be able to place a call from a dialup connection (this is the first
time I try to do this)
From my notebook, connected to
It appears that the pharsing for the wav49 extension which is .WAV isn't
correct in app_voicemail.c. I can't attach the wav49 format although gsm and
wav work fine.
What about just putting WAV instead of wav49?
Mark
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It's the From: line.
Mark
On Thu, 6 Mar 2003, Eric Wieling wrote:
I have debugging on in Asterisk and sip debug.
How do I tell what username a SIP client is trying to use to
register with Asterisk as?
--Eric
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You could turn off reinvite.
Mark
On Thu, 6 Mar 2003, Eric Wieling wrote:
I'm getting the following message:
-- Executing Dial(SIP/2111-b825, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/2111-0bd5 answered SIP/2111-b825
-- Attempting native bridge
Is this GSM or ulaw? You can force ulaw by doing:
disallow=all
allow=ulaw
in your /etc/asterisk/sip.conf in the general section.
Mark
On Thu, 6 Mar 2003, David Davis wrote:
I have a problem with the snom (200) phones that causes all sound to be
'raspy' or stuttered
BUT when I use the same
exten = 2111,1,Dial(SIP/[EMAIL PROTECTED])
exten = 2111,2,Voicemail(u2111)
exten = 2111,3,Hangup
exten = 2111,100,Voicemail(b2111)
exten = 2111,101,Hangup
Needs to be 102 and 103... If that doesn't work, find me on IRC.
Mark
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We just added support for the ;received= method of NAT translation.
This works only if your NAT does not translate the port number. Otherwise
your call will come up w/out RTP.
Mark
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Find me on IRC and I'll try to (yet again) work on this.
Mark
On Wed, 5 Mar 2003, Wade Weppler wrote:
There is now (thanks, Mark!) an addition in sip.conf called nat=1
that can flag a sip user/peer/friend as being behind a NAT address
translator. The good news is that the REGISTER and
Picking up the phone now results in a brief dialtone followed by bursts
of random static and crackling noises and then a second or two of
fast-busy followed by silence and more crackling noises.
I've tried the hardware attached to a couple of different machines as
well as a couple of
Yes, it does. Note also that you can specify mailbox=XXX,YYY and MWI will
display if there are messages in either mailboxe XXX or YYY.
Mark
On Fri, 28 Feb 2003, Raymond McKay wrote:
Will that work in the zapata.conf file also for phones that support MWI?
- Original Message -
From:
It is in-band. My original SIP patch contained code to do inband DTMF
detection in a very ugly way, which apparently got dropped when it was
integrated.
We have a much cleaner interface now, using the dsp functions in Asterisk
(see how it's done in chan_zap.c.
Mark
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