Don't know if this helps, but my installed 4-port fxo card has the
rj11 jack closest to the pci edge connector as zap/4, and the rj11
away from the pci edge connector as zap/1. The board is installed
and working, so can't look at much more.
I have a TDM400P with 3 fxs
Phillip group,
I tried what you suggested and it did not work i included some more information
for you to take a look at...
i have got the MGCP working sort of for my asterisk server. My phone plugged
into the dlink gateway does not ring when i call it. My sip phone does ring
when i dial the
Really you should link /usr/src/linux-2.6 to /lib/modules/`uname -r`/build
then you don't have to do anything special and it'll build... That
directory and all the files in it are installed by the kernel rpm, you
don't even need kernel-source for it... Although I haven't tried compiling
without
this options remove first number
try
exten = _0.,1,Dial(h323/${EXTEN:[EMAIL PROTECTED]:1)
Hekuran Doli wrote:
Need to anounce that Im using sip to h323!
Is there any beter solution to do this ?
.
Can you tell us in details what the problem is (or I didnt understand)?
if the problem is on call
Hi,
new codec runs with snom 200 !
greetings
nicolas
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When I saw the update for Cisco Phone RTP issue I thought I would
try it.
Unfortunately chan_capi wont compile on this update.
Can anyone recommend a good * release for Capi, Bri ISDN and
Cisco 7940s SIP 6.3.
Or will CHAN_CAPI also be updated ?
Running Eicon Diva Bri Cards.
http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html
- Original Message -
From: Craig Waddington [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 12:24 PM
Subject: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
When I saw the update for
H - can anybody confirm this. I have generally had little luck with IAX
in any case so I must admit I assumed (due to info from www.voip-info.org)
that it was due to lack of timing device. I have actually not tried to do
any trunking - just normal calls.
-Original Message-
From:
Thanks.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chrétien Wetemans
Sent: 22 May 2004 12:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Chan CAPI and Latest CVS wont compile
Hi!
I'm tring to do some DB operations before and after a call. I see the 'g'
option in dial to continue in context if the destination hangs up, but
what if the originator hangs up?
You either need to run a CRON job for this clean up, or do that at the
beginning of the next call - whatever
Hi!
Below is my conf that i have now.Is there anything I need to configure in the
Dlink gateway for this to work with asterisk?
Here a few things you can try:
- upgrade to CVS-HEAD (not 0.9.0) and see if things are different
- issue a ngrep port 2727 to monitor what your dlink is sending
-
Hi,
The call waiting indicator do not work for me.
I am using a snom200 cwi is switched on in phone-config.
Have asked snom, but there are can not help me, because it is working for
them.
When it is coming in an call while the phone is busy.
The phone returns:
-- Got SIP response 486 Busy
I'm using Coloco now, which so far is working well.
Where companies like VoicePulse buy services from a patchwork of CLECs
in order to cover their markets, Coloco is a CLEC. The upside is that
you cut out the middleman. But if you need a number in an area they
don't serve you'll need to find a
Hi Phillip,
It needs to occur right after the call.
I'm tring to apply a sort of fromdomain call limit. So I need to keep
track of how many are currently active
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: Saturday, May 22,
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports:
handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx'
When I disable NOTIFY messages, * reports the device UNREACHABLE, followed
by REACHABLE every couple of minutes.
I think I want NOTIFY on, because the
At 7:18 AM -0700 on 5/22/04, Bruce Komito wrote:
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports:
handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx'
When I disable NOTIFY messages, * reports the device UNREACHABLE, followed
by REACHABLE every couple of
Brian Cuthie wrote:
I'm using Coloco now, which so far is working well.
Where companies like VoicePulse buy services from a patchwork of CLECs
in order to cover their markets, Coloco is a CLEC. The upside is that
you cut out the middleman. But if you need a number in an area they
don't
Welcome to Voicepulse and their lack of jitter buffer. This is the
cause of your horrible sound. Will be just as bad with SIP.
Which providers give you a jitter buffer?
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I recently upgraded to the latest CVS and when a caller presses 'o' in
voicemail, I get listed below. I have searched the archive for a
suggestion and pared the sip.conf and extensions.conf to bare minimum to
duplicate this scenario. Any suggestions?
== Parsing '/etc/asterisk/enum.conf':
Hello
I am trying to setup Asterisk on 2 servers PBX300 and PBX200.
PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device.
Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from
PBX200.
I can call from PBX300 outside but I am unable to
Hi !
Running asterisk (cvs 20/05/04) with config
intel 2.4GHz(no SMP); 1GB RAM; 1*E100P; IAX1 slinear; RH 9-2.4.20-8(no
patches)
When running load (30 simultaneous calls), the server utilizes approx
10% CPU, but every 20-30 seconds it's a short peek where the
asterisk-process takes 99% CPU.
Have
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to make as
much of the configuration dynamic as I possibly can. The only part that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by
Hi,
I am setting up a dispatch center where will have 4 call takers, all
with Polycom IP 600 Sip phones. Each phone will be setup with 6
extensions each. When a new call comes in, the first extension on all
the phones will ring. This works fine, the problem is when one of the
dispatchers is
Hi,
I am trying to receive a fax with the spandsp library.
The sending fax says success but there is no tiff file generated.
I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my extensions.conf.
The connection is via SIP/G.711 as I have read on the list that this
can sometimes work (I know Fax over
So I've been kind of struggling with the notion of making my Asterisk
implementations dynamic, too. While I'd like to make everything directly
database driven, I'm not sure Asterisk is quite there yet.
I've been thinking of writing something that creates appropriate
configuration files from
[EMAIL PROTECTED] wrote:
Welcome to Voicepulse and their lack of jitter buffer. This is the
cause of your horrible sound. Will be just as bad with SIP.
Which providers give you a jitter buffer?
In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure
there are more.
Lars Boegild Thomsen wrote:
H - can anybody confirm this. I have generally had little luck with IAX
in any case so I must admit I assumed (due to info from www.voip-info.org)
that it was due to lack of timing device. I have actually not tried to do
any trunking - just normal calls.
That
You might consider using the Cisco SIP phones. They're smart enough to
accept incoming calls for as many call appearances you have with the
same SIP registration.
-brian
Tor Roberts wrote:
Hi,
I am setting up a dispatch center where will have 4 call takers, all
with Polycom IP 600 Sip phones.
Call the PBX300 using IAX2 from PBX200, make sure that the call goes into
the context that allows dial out.
Example.
exten = _543219XX,1,StripMSD,5
exten = _9XX,2,Dial/[EMAIL PROTECTED]/BYEXTENSION
The first line looks for an access code '54321' followed by the access code
for an
John Todd wrote:
At 7:18 AM -0700 on 5/22/04, Bruce Komito wrote:
When I enable NOTIFY messages in my SIP device (Sipura), Asterisk
reports:
handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx'
When I disable NOTIFY messages, * reports the device UNREACHABLE,
followed
by
Another problem with the SIPURA is the lack of a working STUN
solution. Even Grandstream works better with NAT today. /O
I second that!!!
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To
What I do is have a cron with a perl script that recreates a sip-db.conf
(which has an #include in the sip.conf) then do a sip reload.
I's rather simple, all you have to write a temp and diff, if its
changed replace and reload, if not don't do a thing.
Do the same with extensions
On Sat,
Afternoon all,
I'm trying to load Asterisk, however I am getting the following error:
[skipping res_musiconhold.so]
[chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol:
ast_moh_stop
May 22 18:42:24 WARNING[16384]:
Leif Madsen wrote:
Afternoon all,
I'm trying to load Asterisk, however I am getting the following error:
[skipping res_musiconhold.so]
[chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined
symbol: ast_moh_stop May
Please reply with sip.conf extension.conf for both servers. Hard to tell
what the problem is without see config info
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 11:39 AM
Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk
Dunno about not being able to generate a tiff, I got rxfax to do that, but
they're badly malformed.
http://roanoke-voip01.psknet.com/fax/
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Senad Jordanovic
Sent: Saturday, May 22, 2004 2:07 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] loader.c:240 ast_load_resource:
/usr/lib/asterisk/modules/chan_sip.so:
Brian Cuthie wrote:
So I've been kind of struggling with the notion of making my Asterisk
implementations dynamic, too. While I'd like to make everything directly
database driven, I'm not sure Asterisk is quite there yet.
I've been thinking of writing something that creates appropriate
Leif Madsen wrote:
I'm trying to load Asterisk, however I am getting the following error:
[skipping res_musiconhold.so]
[chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240
ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol:
ast_moh_stop
May 22 18:42:24
Am 22.05.2004 um 20:09 schrieb Troy Settle:
Dunno about not being able to generate a tiff, I got rxfax to do that,
but
they're badly malformed.
This is more than I get ;-)
Does the fax on the other side get a success message?
I get fax-rx-audio and fax-tx-audio files in /tmp but no tiff output
Hi Philipp
On Sat, May 22, 2004 at 02:29:18PM +0200, Philipp von Klitzing wrote:
Note: The h extension is not reliable enough to solve your problem.
What is the problem with the hangup extension?
Thanks in advance
--
Stefan Tichy [EMAIL PROTECTED]
Hi!
On Sat, May 22, 2004 at 02:29:18PM +0200, Philipp von Klitzing wrote:
Note: The h extension is not reliable enough to solve your problem.
What is the problem with the hangup extension?
Not reliable - ask bkw for details, he can elaborate.
P.
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote:
[snip]
Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk
can handled the NAT traversal all by itself with Qualify (as John points
out) disabling the NOTIFY will not change anything.
The NOTIFY will in no way affect the
IMHO, the Polycom IP600 is a superior phone to the cisco 79XX phones. I
base this off of having had both an IP600 and a 7960. The two advantages
the 7960 had over the IP600 was appearance and ease of
configuration. Outside of that, the IP600 (IMHO) beat the cisco hands down.
Now, you MAY
Sipura does include STUN as an option. It has for quite some time. We are
using it with all of our Sipuras behind NAT'd gateways and it works great!
Try upgrading to the latest Sipura firmware rev.
Darren Nay
-Original Message-
From: John Todd [mailto:[EMAIL PROTECTED]
Sent:
On Sat, May 22, 2004 at 05:39:48PM +0100, Fran Boon wrote:
Only by extending the functionality of sip friends to include this extra
field...
In chan_sip.c the configuration data from sip.conf is used to build
a list of sip friends. Checks for waiting voice mail are done for the
members of this
Beyond this, you can still just use the NAT keepalive in the Sipura.
While It only provides for either a NOTIFY or REGISTER (which both
generate errors in asterisk) if you change it to something else (I just
have it send blank, but a few ... or anything will do) asterisk won't
complain and
David,
Not sure if you already got a reply or not - but it looks to me like
your FXO module is on port 3 - not 2 (see the dmesg output). Give that
a try.
HTH- Ben
On Wednesday, May 19, 2004, at 12:51 PM, David Creemer wrote:
Hi-
I'm totally stumped configuring my TDM400P with one FXS and one
Another problem with the SIPURA is the lack of a working STUN solution.
Even Grandstream works better with NAT today.
/O
I disagree. We have hundreds of Sipura customers using STUN with our
SER Solution. The are the most stable SIP UA we have ever tested. We
had to dump loads of Grandstream
Hi guys!
I just try to compile Asterisk with make all and get the following
lines multible times:
cli.c:31:19: build.h: No such file or directory
dlfcn.c:40:25: mach-o/dyld.h: No such file or directory
dlfcn.c:41:26: mach-o/nlist.h: No such file or directory
dlfcn.c:42:28:
Nothing, it's normal to get those errors - I get them all the times I
compile asterisk on Linux, FreeBSD, and Windows. Your failure to compile is
being caused elsewhere.
- Joshua Colp.
- Original Message -
From: Julian Pawlowski [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May
Nothing, it's normal to get those errors - I get them all the times I
compile asterisk on Linux, FreeBSD, and Windows. Your failure to compile is
being caused elsewhere.
Ah okay, thanks. Although make all is successfully, I say these
messages and tought that it could result in some incorrect
John Todd wrote:
At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote:
[snip]
Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk
can handled the NAT traversal all by itself with Qualify (as John points
out) disabling the NOTIFY will not change anything.
The NOTIFY will in no
Andres wrote:
Another problem with the SIPURA is the lack of a working STUN solution.
Even Grandstream works better with NAT today.
/O
I disagree. We have hundreds of Sipura customers using STUN with our
SER Solution. The are the most stable SIP UA we have ever tested. We
had to dump loads
I see that * refers to the channels this way on the console output, but
I get warnings when I try to use the new naming in the extensions.conf
dial plan - anyone else notice this? How do you refer to the channels
in extensions?
On Tuesday, May 18, 2004, at 07:50 PM, Leo Ann Boon wrote:
f. Be
Tor Roberts wrote:
Hi,
I am setting up a dispatch center where will have 4 call takers, all
with Polycom IP 600 Sip phones. Each phone will be setup with 6
extensions each. When a new call comes in, the first extension on all
the phones will ring. This works fine, the problem is when one of the
I am trying to recreate an * server from backups.
I copied /var/spool/asterisk/voicemail/context/109/INBOX/* from backups.
The voicemail files got restored
msg.gsm
msg.txt
msg.wav
but when the user goes into voicemail, * says there is no voicemail.
Thanks!
Paul
Andres wrote:
[EMAIL PROTECTED] wrote:
Which providers give you a jitter buffer?
In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure
there are more.
Clearpath gives jitter buffer as well. http://www.clearpath1.com/
John
___
The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the
world to work. Is this necessarily true, or does it only need some of these
outgoing?
I'm concerned as anyone that could guess an extension numberpassword could
use my server to make outgoing calls. It would help if
Hi All,
Having searched the archives, I can see there has been much discussion
at various points regarding capture of caller id information from good
old BT.
If I understand correctly, it seems that not only do the drivers not
currently support it, but my X101P possibly/probably can't do it
I personally only allow IAX2 in and out from my asterisk box, due to the
simplicity of one (udp) port. I do not relish the thought of trying to
open the port ranges for SIP securely!
As long as your inbound stuff in iax.conf lands in a sensible context,
inbound connections would only be able to
Hi,
i'd like to know more about this issue, i'm always getting this message while in call
with anyone from sip to zap or zap to sip.
ast_rtp_read: Unknown RTP codec 72 received
here is my current setup:
client side, x-lite, with the transmit silence to yes, using ulaw,alaw
on asterisk server
Hi
On Tuesday, May 18, 2004, at 07:50 PM, Leo Ann Boon wrote:
f. Be careful about the zap channel naming. With the old XP101, the
first channel (card) is Zap/1 and the second Zap/2. With the TDM, it's
Zap/1-1, Zap/2-1 ... Zap/4-1 for the 4 ports on the first card and
Zap/1-2 ...
The new app_groupcount looks great for most applications but it a is a
step back for call queueing...
since app_queue calls physical interfaces and not extensions,
app_groupcont can't be used to limit the calls passed to a dynamically
added agent.
I presently use the broken sip incominglimit
Chris,
As far as the Cisco phones, they are not an option as I already have the
Polycoms. The Ciscos are overpriced anyway.
It was my understanding that asterisk would not let you register the
same extension more than once. If that is not the case, I will try to
register the same extension to
The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the
world to work. Is this necessarily true, or does it only need some of these
outgoing?
I'm concerned as anyone that could guess an extension numberpassword could
use my server to make outgoing calls. It would help
Hi
Il dom, 2004-05-23 alle 00:11, Tony Hoyle ha scritto:
The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the
world to work. Is this necessarily true, or does it only need some of these
outgoing?
all depends on what you need to do.
if you use only zap channels and no
Karl Dyson wrote:
Hi All,
Having searched the archives, I can see there has been much discussion
at various points regarding capture of caller id information from good
old BT.
If I understand correctly, it seems that not only do the drivers not
currently support it, but my X101P
i try to place a call
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
where sip.conf has an entry
[foo]
secret=torture
callerid=local ext 103 1914666
type=friend
fromuser=asterisk
auth=both
host=dynamic
canreinvite=yes
context=in-914
i try to place a call
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
^^^
That foo name needs to be changed to an IP address of whatever it
is that is suppose to handle the call. Asterisk is doing a DNS name
lookup and can't resolve it,
Tor Roberts wrote:
It was my understanding that asterisk would not let you register the
same extension more than once. If that is not the case, I will try to
register the same extension to all 6 lines.
On the 7960's, * does not get upset with having multiple appearances of
the same line on a
Randy Bush wrote:
i try to place a call
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
where sip.conf has an entry
[foo]
secret=torture
callerid=local ext 103 1914666
type=friend
fromuser=asterisk
auth=both
host=dynamic
canreinvite=yes
Brancaleoni Matteo wrote:
if you plan to do only IAX, only port 4569 UDP needs to be opened.
but if you plan to do only sip you need only port 5060 UDP
and 1 to 2 UDP for sip rtp stream (configurable
into rtp.conf)
so... all depends :)
Surely it depends on who's calling me - if they're
Congradulations to the Asterisk gang on getting slashdotted!
http://slashdot.org/article.pl?sid=04/05/22/1840220
Cheers,
Rich
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Randy == Randy Bush [EMAIL PROTECTED] writes:
Randy i try to place a call
Randy exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
Randy where sip.conf has an entry
Randy [foo]
Randy type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user
So I just saw this VoIP-centric article at slashdot
(http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions
e164.org. It's a non-profit public DNS root designed to map phone numbers
to Internet protocols. Is anyone on this list actually using this?
They have asterisk config
Simon Dorfman wrote:
I wonder if someone can help me understand this. Let's say I configure my
asterisk box to use e164 and then I try to call a phone number in Germany.
I'm in the U.S.A. So if the number I'm calling in Germany is registered in
e164's dns, would my call be routed directly via
On Sat, 2004-05-22 at 18:08, Tony Hoyle wrote:
Simon Dorfman wrote:
I wonder if someone can help me understand this. Let's say I configure my
asterisk box to use e164 and then I try to call a phone number in Germany.
I'm in the U.S.A. So if the number I'm calling in Germany is
Hi, i'm having another problem I can't work out -
make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper'
Dean Collins wrote:
Tony, as per you inference that e164 are up to something shady, you
should talk to one of the founders Duane, he currently has about 5 open
If it's the same duane who runs cacert he probably means well... however
having read the site I'm still not sure whether i'd use it
Check your README file again.
In order to compile 0.6.1 you need newer versions of pwlib and
openh323 (1.6.6 and 1.13.5)
Then it should work just fine
Pablo
--
Pablo Endres [EMAIL PROTECTED]
ComVoz Comunications
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Matthew Asham wrote:
; north america enum
exten = _1NX,1,Playback(doing-enum-lookup)
exten = _1NX,2,EnumLookup(${EXTEN})
exten = _1NX,3,BackGround(enum-lookup-successful)
exten = _1NX,4,Dial(${ENUM},30,tr)
exten = _1NX,5,Hangup
exten =
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily pasted and renumbered,
exten = _1NX,6,Playback(enum-lookup-failed)
exten = _1NX,7,Hangup
are actually:
exten = _1NX,103,Playback(enum-lookup-failed)
exten =
Thank you, Michael
I tried to switch to FR mode... but it did not help. I tied DLCI as 16
and 99... the same result.
I attached one more full config from Netopia and from my Linux+Zaptel
T100P systems.
DEVICE=hdlc0
# MODE=hdlc
# MODE=cisco
MODE=fr
NETMASK=255.255.255.252
ok done, but now i'm getting different errors -
/usr/src/pwlib/include/ptlib/args.h:389: virtual outside class declaration
/usr/src/pwlib/include/ptlib/args.h:389: non-member function
`UnknownOption (...)' cannot have `const'
method qualifier
/usr/src/pwlib/include/ptlib/args.h:397: parse error
You forgot to allow for tel: N+51
bkw
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 7:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was
Hi Mike,
Your log seems to be incomplete. It stops in the middle of the call.
Regards,
Steve
Mike Heininger wrote:
Hi,
I am trying to receive a fax with the spandsp library.
The sending fax says success but there is no tiff file generated.
I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my
'local' target? What's that?
- Original Message -
From: Matthew Asham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, May 22, 2004 9:55 PM
Subject: Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example
I pasted was hastily
Hi Troy,
People had a lot of problems like this with earlier versions of spandsp.
However, the latest version is pretty solid, and people are using it in
high volume production applications. If you are getting these bad
results with the latest version I would be interested to see the audio
log
[foo]
type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user blocks in sip.conf it may work. Expecially
if you also specify a port in the Dial().
Else, use the hostname (or a const).
hmmm. then, how do i let it be dynamic if it has
Hi Tony, it is the same duane - lol you are hardly allowing it to
perform least cost routing, it just does one check for ip to ip call
then drops back to whatever you have written on your asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony
Hello,
I am trying to get
an Adtran 750 w/ 1 Quad FXO and 1 Quad FXS to work with a TE405Pand I am
having a few problems. I have the FXO on channels 1-4 and the FXS on
channels 5-8. I have a single analog phone set connected to the first port
on the FXS (channel 5) and an analog line
Hello All,
Does anyone tried to use CallerID in Eastern Europe (Russia/Ukraine)?
Our teleco provides CallerID, as well as AON, then can send _callerid_, as well as AON
signals non of those 2 works on TDM400P card with FXS ports.
They are using Siemence systems.
How can I debug this and
Billy Huddleston wrote:
'local' target? What's that?
http://www.voip-info.org/wiki-Asterisk+local+channels
It's like a subroutine, so you can use it to call bits of the dial plan
that get repeated a lot, like dialing FWD after first setting the caller ID.
(AFAIK anyway... not tried to get them
Just would like to add, of course if it is going to help:
I am using Linux 2.4.26 on Linux, compiled from sources and latest zaptel sources.
We have T1 Internet from Verizon.
... any help appreciated.
= Original message ===
From: Vasyl Rublyov [EMAIL PROTECTED]
Dean Collins wrote:
Hi Tony, it is the same duane - lol you are hardly allowing it to
perform least cost routing, it just does one check for ip to ip call
then drops back to whatever you have written on your asterisk.
So eg. if I've registered 3 different sip providers and an IAX provider,
plus a
Hi Mike,
How do you run rxfax? You problem is probably something to do with that.
Your's is the first report I have had of no TIFF file whatsoever.
Regards,
Steve
Mike Heininger wrote:
Am 22.05.2004 um 20:09 schrieb Troy Settle:
Dunno about not being able to generate a tiff, I got rxfax to do
At 7:31 PM -0700 on 5/22/04, Randy Bush wrote:
[foo]
type=friend
I do not beleive that will work for type=friend. If you use separate
type=peer and type=user blocks in sip.conf it may work. Expecially
if you also specify a port in the Dial().
Else, use the hostname (or a const).
hmmm.
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