Jerry Geis wrote:
I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri
1.4.7
and I am getting the error:
-- Requested transfer capability: 0x00 - SPEECH
-- Called 23/317506
-- Channel 0/23, span 1 got hangup, cause 99
-- Hungup 'DAHDI/23-1'
sean darcy wrote:
I've have a simple caller id lookup on incoming:
[teliax-in]
..
exten =s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02135590993,1,Set(CALLERID(name)=Matthew )
Chances are you have a DNS problem. Asterisk is trying to look up a
hostname and it is not getting a response.
Todd Reese wrote:
New update: After the console being frozen for about 15 minutes, it
just responded to the last reload comand
___
Asterisk still does things like use DNS SRV records unless turned off,
it also tries to do a DNS lookup for all IP addresses on the system.
Update your hosts file and see if that helps.
Todd Reese wrote:
DNS server rebooted. Everything is back online now.
What threw me was that one of the
Elliot Murdock wrote:
Hello!
What kind of sms text messaging capabilities does Asterisk have?
Asterisk has the ability to use land lines to send SMS messages to a
remote device that supports landline SMS. In Europe and much of the
rest of the world SMS carriers provide a public PSTN
Sriram wrote:
Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed
that whenever during background(menu-filename) method - i try to
press any key for selection like 1 for some prompt, 2 for another
prompt etc...Asterisk takes a while before it takes me to the
respective
obitori junk wrote:
I am experiencing a 606 not Acceptable error trying to set up an
Asterisk server as an ekiga.net client. My server is behind a firewall
with NAT routing. I have googled this problem and read about Asterisk
feeding its local ip address to ekiga.net. That seems to be my
Andres wrote:
We are running 1.4.22 and have been experiencing problems with certain
IVRs and DTMF Tone duration. We would like to be able to increase DTMF
Tone duration by 50 to 100ms over what the user is pressing on his
phone. We have a PRI test circuit and an analyer in between to
Barry L. Kline wrote:
Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Not when you take the time to properly trim your reply
In the USA (maybe other T-1 countries) you can have a channelized T-1.
Each channel is assigned signaling just like an analog line, FXS, FXO,
EM, etc. I have worked with a carrier in the past that could put FXO
channels on a T-1 along with a PRI channels on the same T-1.
Andrew Thomas wrote:
Terry Wilson wrote:
On Dec 15, 2008, at 7:05 AM, Mike wrote:
Just so I'm clear: there is no way to do what I want short of
playing with the underlying code, correct?
Yes. I'm working on an issue right now related to parking and noticed
that Asterisk completely lies with the verbose
park-dial context.
Regards,
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
ManxPower Wieling
Sent: Monday, December 15, 2008 16:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
No, not on FXO ports. On FXO ports Asterisk considers the call answered
as soon as dialing is finished. Asterisk has no way to detect when the
far end answers when using FXO ports.
michel freiha wrote:
I would like to ask please if there is a way to play a ring back tone from
asterisk when
Philipp Kempgen wrote:
michel freiha schrieb:
I would like to ask please if there is a way to play a ring back tone from
asterisk when the customer try to make a call...I already added the ringing
function to the context in extensions .conf and it work perfectly...But the
issue that the
Remove the r option to Dial.
Bruno Castelo Branco wrote:
Hi all
When I call to any mobile and the device is power off the asterisk keep
ringing and I not able to hear the tradicional message saying this
mobile is power off.
When I call from a normal analogic line I got the
Use the docs, Luke.
dev-1*CLI core show application parkandannounce
dev-1*CLI
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
ParkAndAnnounce(announce:template,timeout,dial[,return_context]):
Park a call into the parkinglot and announce the call
Mike wrote:
Sure, that works too, but I needed access to context taken from the sip
entry because I needed to goto(${that_context}), and that context varies
depending on the phone used.
This is when setting __TRANSFER_CONTEXT in my cookie cutter really
big/complexe/database driven
There is a loopback somewhere on the line. Contact your telco and say
I see a loopback on the line. Please remove it.
Uros Djokic wrote:
Hi I have problem with TE121 Digium card. I connected it to modem keymile
Music 200 (provided by telco) but I can see 2 red lights on modem (both
bellow
go to Asterisk2
But RTP go to Asterisk1 and no more.
Where have I to insert canreinvite ?
Thank you
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Eric
ManxPower Wieling
Envoyé : mercredi 3 décembre 2008 19:25
À : Asterisk Users Mailing
Next time I'll be sure to finish my morning coffee before posting. 8-)
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote:
There is a loopback somewhere on the line. Contact your telco and say
I see a loopback on the line. Please remove
Welcome to the world of FreePBX. It would save me quite a bit of time
if you could list what ports (port number and signaling) you have on the
card and what context you want each port to go into. When I manually
merge the two files (after stripping out 37 billion comment lines) I see
that
ICMP is used to determine maximim packet size. If you or the other end
are blocking all ICMP then MTU Path Discovery will not work. It's a
classic newbie network admin mistake. Symptoms of this problem would be
exactly like you describe.
Typically I see this on PPPoE connections.
More
canreinvite=yes should work as long as 1) there is no NAT involved
anywhere in the call path, 2) All legs of the call are using the same
codec, 3) you do not have the t/T/w/W (and maybe a few other) options to
the Dial line.
Remember the only way you can really tell if a reinvite happens is by
John covici wrote:
OK, I can park the calls OK, but I don't get the announcement -- I am
using freepbx if that makes any difference.
If you park a call and do not hear the announcement then you are doing a
BLIND transfer, not an ATTENDED transfer. You should be doing attended
transfers for
transfer?
on Wednesday 12/03/2008 Eric \ManxPower\ Wieling([EMAIL PROTECTED]) wrote
John covici wrote:
OK, I can park the calls OK, but I don't get the announcement -- I am
using freepbx if that makes any difference.
If you park a call and do not hear the announcement then you
It is not a parking solution.
Sebastian wrote:
Any idea? Please I need advice.
Thanks!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sebastian
Sent: lunes, 01 de diciembre de 2008 11:58 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
You can either add that feature to chan_iax2.c or pay someone to add
that feature to chan_iax2.c.
Bruno Castelo Branco wrote:
Somebody know some work around for it?
I still trying to find a solution but nothing seems to work
thanks
Eric ManxPower Wieling wrote:
The problem is that IAX2
The problem is that IAX2 does not seem to support call pickup.
Bruno Castelo Branco wrote:
hi
I'm using only IAX extensions and inserted callgroup=1 and callpickup=1
for all IAX extensions in iax.conf. Didn't works for while.
thanks
Tim Panton wrote:
I think it doesn't work across
Bruno Castelo Branco wrote:
Somebody knows if pickup call works with IAX2?
I enable *8 in features.conf, but doesn't works with IAX2 extensions.
As I understand it, IAX2 does not support callgroup= and pickupgroup=
and *8.
This link might be helpful:
Maybe because there is no such thing as a SIP trunk, at least in the
Asterisk world. Most of us call them peer or friend.
The term you are looking for is reinvite. Reinvites allow two devices
to send audio directly between the two end points of the call. the
SIGNALING stays on the servers,
Philipp Kempgen wrote:
Olivier schrieb:
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address are using
CLI as it seems
Olivier wrote:
2008/11/20 Eric ManxPower Wieling [EMAIL PROTECTED]
Personally I use the MAC-x wherex=the line appearance number. MAC-a for
first line appearance, MAC-b for 2nd, etc.
Is it easy to use (CLI, logs...) ?
Would you step back to an extension-based identification scheme ?
I
From IRC:
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP, etc),
you
Alex Balashov wrote:
Steve Totaro wrote:
I have done some large installs where people are going to be in the
office, sometimes out, work from home, it always changes sorta thing..
I have found that setting all device profiles to Nat=yes Just Works
whether they are on the LAN or not
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check the doc directory in the Asterisk source
code for channelvariables.txt, but for some reason that I
Tilghman Lesher wrote:
On Wednesday 12 November 2008 23:15:01 Eric ManxPower Wieling wrote:
Use ${HANGUPCAUSE} which provides a Q.931 hangup cause. chan_sip and
chan_iax (maybe other channels) translate the protocol specific causes
to a Q.931 hangup cause.
I would recommend you check
You can sometimes find the older Cisco Aironet boxes that run at 900Mhz.
That frequency is AWESOME in rural areas. Mountains will still block
it, but trees and water does not.
Drew Gibson wrote:
Wilton Helm wrote:
Good points. I got an access point instead of a router specifically
so I
Thomas Winter wrote:
On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote:
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Really, thats not the way it should work.
How I can find out the codec of an incomming call
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Thomas Winter wrote:
Hi,
I have a problem with codecs.
I have an provider with allowed codec alaw, ulaw, g.723
I have SIP clients with codec allowed alaw, ulaw, g.723
If a SIP clients wants
It sounds like you have analog lines. If that is the case, the silence
you experience is Asterisk sending the DTMF down the line. Asterisk
collects the DTMF and when you are done dialing it retransmits those
digits down the analog line. I think each digit is by default 300ms.
If you are
Most IVRs want longer DTMF tone lengths. If you shorten the
toneduration= then many IVRs won't work.
Wilton Helm wrote:
If it is 300 ms, that is way to long. I don't know any CO grade receiver
that can't decode in 80 ms and some can do 40. There is also a similar size
gap between digits.
Historically Asterisk's config file parser ignored unknown keywords.
This is useful for exactly the things you are trying to do. I hope 1.6
did not remove this feature.
Rob Hillis wrote:
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Then you should read the READMEs right now. See the 3 upgrade info
files as well as any other READMEs.
Christian wrote:
Hello,
Many thanks for the info.
OK, I didn't know that. I just installed it. Usually I read the included read
me files and so on but not at this time.
But I will be
core show application dial (this is the official application doc)
Pay special attention to the D() option.
Rodolfo Alcazar Portillo wrote:
Hi. In creating a custom extension, and dialing
SIP/222/333#444, seems the party receives only 333
What should I do to send the # symbol? or
From zapata.conf.sample:
; 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
ast_request: No channel type registered for ''SIP'
Notice the extra ' in the message.
That is either an error in the error message or you have a an extra ' in
your Dial line. Something like Dial('SIP/
I'm surprised nobody else noticed this.
Stephen Reese wrote:
On Sun, Oct 19, 2008
exten = _+X.,1,Goto(${EXTEN:1},1)
michel freiha wrote:
Dear All,
i have the following context defines in etensions.conf:
[a2billing]
exten = _X.,1,Gotoif($[${EXTEN} = 111] ? 21)
exten = _X.,2,DeadAGI,a2billing.php
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten =
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Brian J. Murrell wrote:
On Fri, 2008-10-17 at 10:32 -0500, Andres wrote:
Because Asterisk does not support
Brian J. Murrell wrote:
On Fri, 2008-10-17 at 11:18 -0500, Eric ManxPower Wieling wrote:
It should be fairly easy to write an AGI script that does the SRV query,
do whatever you want with the response, set a channel variable with
the results and use that in your dialplan.
Maybe. If I
exten = +13129842314,1,Noop(Happy match!)
or
exten = _+1NXXNXX,1,Noop(Happier match!)
Karl Fife wrote:
Steve Murphy [EMAIL PROTECTED] wrote:
People have voiced this before; but the cut-down version of RE's that
the matching algorithms allow are fairly fast, both in the new and
the old
Olivier wrote:
2008/10/10 Eric ManxPower Wieling [EMAIL PROTECTED]
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant extra latency while causes you you to perceive the
echo
Try setting canreinvite=no in each of the device sections on a couple of
phones, reload chan_sip.so and see if that fixes things. It has fixed
the issue when I've tried it.
[EMAIL PROTECTED] wrote:
Hello,
We are using asterisk 1.6, sangoma pri card, and Cisco 7960 phones. When we
make or
by normal echo canceling systems.
Most echo canceling systems I've seen (mostly tellabs) only cancel echo
in one direction. I suspect all of Digium's EC systems only do echo
canceling in one direction as well.
Olivier wrote:
2008/10/11 Eric ManxPower Wieling [EMAIL PROTECTED]
Olivier wrote
You should not get that message on analog lines in the USA or Canada. I
suspect your line has a provisioning issue or is using different
signaling than you think it is using.
Jim Duda wrote:
Can anyone tell me what this message means?
Got event 17 (Polarity Reversal)...
I'm running
All calls with a 2-wire analog piece have echo. You cannot perceive the
echo because it happens so fast on non-VoIP connections. On VoIP calls
you have significant extra latency while causes you you to perceive the
echo. Echo must be removed before the call is converted to VoIP -- in
your
Olivier wrote:
I don't have any spare zaptel enabled system I could try this on, but I
was not aware of this CHANNEL variable.
Now, I can see it here http://www.voip-info.org/wiki/view/Asterisk+variables
Maybe, I will add a line in www.voip-info.org http://www.voip-info.org
to keep
The most reliable ATA is a channel bank.
Vieri wrote:
--- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote:
Why not swap it all with just IP phone?
That's because we have almost 400 analog phones already wired in our building
(which is very large). So we need to take advantage of the
and imports it into voip-info.org when the the
docs are changed?
I'd be glad to write and host such a script if the community desires the
feature.
-josiah
SIP wrote:
Eric ManxPower Wieling wrote:
Olivier wrote:
I don't have any spare zaptel enabled system I could try
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am
This is done in sip.conf, iax.conf, etc, not in extensions.conf. By the
time a call gets to extensions.conf it must already be authenticated.
Assume the username is robertdobbs and the ip is 209.17.71.61
In sip.conf you would have something like this:
[robertdobbs]
deny=0.0.0.0/0
Tilghman Lesher wrote:
On Wednesday 17 September 2008 21:42:20 Steve Totaro wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL PROTECTED] wrote:
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming
Where did you hear this?
Shaun Wingrin wrote:
I have heard it said that, Asterisk falls over at 100 simultaneous
calls. Is this true?
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL.
SCCP (aka Skinny), H323, MGCP, and SIP all use the RTP protocol for
audio. For all signalling protocols (except maybe H323) use rtp.conf
for the RTP ports.
OCG Technical Support wrote:
I have a 7921 wireless phone working with Asterisk, and I want to
tighten the wide open port range of
ManxPower Wieling [EMAIL PROTECTED] wrote:
From: Eric ManxPower Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] FAX over T1 Question
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date: Friday, September 5, 2008, 10:04 PM
You're joking, right?
exten = _1xx,1,Dial(SIP/200SIP/201SIP/202SIP/203,30,tr)
exten = _1xx,n,Voicemail([EMAIL PROTECTED])
Use whatever voice mailbox and voicemail context you want.
Joseph L. Casale wrote:
I have a setup with a SIP DID inbound, and several SIP phones inside.
If I am not mistaken every single echo canceler out there will disable
itself if it detects a fax tone.
Echo Cancelers do not screw up faxes, people screw up faxes. 8-)
Bob Pierce wrote:
On Fri, 2008-09-05 at 09:19 -0700, Amaru Netapshaak wrote:
I have a Sangoma A104d T1 card, a Rhino 24-port
The thing is, you are doing FAX over PRI, not FAX over T-1 (which to me
implies Channelized T-1). Seems like most people gave you advice that
might apply to a Channelized T-1, but would not apply or be practical
for a PRI.
Amaru Netapshaak wrote:
Bob,
I should have added that I have
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for inband tones like after an
Answer())
eng. Anatoli Marinov wrote:
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
The problem appears when the call comes
ManxPower Wieling [EMAIL PROTECTED]:
It will do so by default if you have a valid
/etc/asterisk/indications.conf (only used for inband tones like after an
Answer())
eng. Anatoli Marinov wrote:
Hi guys,
I am trying to configure an asterisk server for our office.
Asterisk 1.4.17 SIP only
It would be clearer if it said Hookstate (FXS ports only): Offhook
i.e. the state information is not valid for FXO ports.
Jay Ray wrote:
Any pointers on this one?
--- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
From: Jay Ray [EMAIL PROTECTED]
Subject: [asterisk-users] X100P Card in
+ is not a valid Caller*ID character. Asterisk allows you to use + in
Caller*ID, but many carriers will reject the call if you do that.
Benny Amorsen wrote:
ronald [EMAIL PROTECTED] writes:
Is it possible to assign a plus sign on the callerid(num) ?
Yes.
currently this is what i do
Steve Totaro wrote:
On Thu, Aug 21, 2008 at 12:50 PM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
Hello all!
my last month's phone bill sky rocketed after i setup asterisk with
softphones all over the house!
could someone help me set up a limitation for my wife and kids not to be
able to talk
Your script is not catching SIGHUP, which is what Asterisk uses to tell
the AGI the channel went away.
Ruddy Gbaguidi wrote:
Hi all.
I'm using asterisk 1.4.21.2 and when I run
ps -ef |grep defunct,
I can see a lot of my perl agi still pending there.
The channel has been cleaned up in
What would on-board NIC be?
Jay R. Ashworth wrote:
On Wed, Aug 13, 2008 at 11:54:23PM -0400, Steve Totaro wrote:
NIC card is redundant ;-)
And you can take that to the ATM machine.
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux,
But what would you call it? It's not a card, so it can't be a NIC, right?
Steve Totaro wrote:
Er, it would be one integrated with the MoBo, on the board if you will...
Thanks,
Steve Totaro
On Thu, Aug 14, 2008 at 11:50 AM, Eric ManxPower Wieling
[EMAIL PROTECTED] wrote:
What would
Steve Totaro wrote:
On Wed, Aug 13, 2008 at 7:40 PM, Jonathan Miller [EMAIL PROTECTED] wrote:
From what I can determine while troubleshooting a voice-dropping
issue, the Asterisk server in my organization has been dropping RTP
packets between the asterisk server process and the network
Yes you should be able to do that with an E-1 (it's called DACS).
HOWEVER, you can't do DACS on a PRI, as you would need the D-Channel
replicated and you can't do that.
If you just want Asterisk to provide PRIs to your users, then that's a
different story.
Hans Witvliet wrote:
Before trying
The something is generated by Asterisk at the time the call is
created. You should never add it, since you don't control that call
instance info. In fact, you should almost never care about the call
instance string. The -1 means first instance of a call on this
channel, a -2 would be seen
bilal ghayyad wrote:
The reason that I need to do this is:
I will have two Asterisk PBX's, and I need both of them to use same Internet
(so both of them will be behind NAT under same DSL router), in that case, how
I will distinguish on the router the calls that need to be send for box A
Jerry Geis wrote:
I dont see any errors in the dialplan while loading.
I tried to past the whole log but it was rejected.
Again 1.2 works, 1.4 works, no on 1.6 I made no changes to the files.
I cant even dialplan show default at this time.
It looks like you did not read the UPGRADE
I sit corrected. He should still be reading the upgrade files.
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
It looks like you did not read the UPGRADE files for 1.2, 1.4, 1.6 that
should have been included in the source code. If you read that you'll
realize that dialplan show
How about:
exten = _9X,n,Goto(not-parked,s,1)
Doug Lytle wrote:
Everybody,
I have a fall though context that, among other things, tests to see if
someone it trying to pick up a non-existent parked call (Defined from 90
to 99). I have the following:
[not-in-service]
exten =
Tell your box to not expect Caller*ID information. You set that with
usercallerid=no in /etc/asterisk/zapata.conf
Since you are using the Asterisk Appliance you would have to contact
Digium for support.
Sydney Web Hosting wrote:
Hi All,
I have just setup an asterisk box (AA50) and all is
chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8).
Enrico Pasqualotto wrote:
Hi all, One question
I have set in the extensions.conf of my asterisk that all incoming call
go in the wait application because I need to not connect the caller
but remain in the ringing
Make the card stop sharing it's IRQ with your IDE controller. Try
moving the card to another slot.
Asterisk has to send an audio packet every 20ms for VoIP calls. I
believe Zaptel expects no more than a few ms of latency. If something
is causing a delay, like the IDE controller locking
You should contact Digium for support for the Asterisk Appliance. It
works totally differently from other Digium products.
Fidel Garcia wrote:
I just found it at : /ramfs/etc/asterisk/http.conf
How do I restart the http service without affecting the phone service?
Fidel Garcia
If any docs were the cause of this (very important) misconception, maybe
the docs could be reworded. Do you remember what caused you to think
that context was created automatically?
broadband Voice wrote:
fc7234153*CLI dialplan show open
There is no existence of 'open' context
I was under
This will happen if the other side is configured the same as the
Asterisk side. i.e. PRI CPU mode on both ends or PRI NET mode on both
ends. This can also happen if the line is in loopback mode at the far end.
Eve-Ellen Cole wrote:
The underscore helped, but didn't resolve the real
Asterisk allows you to add custom SIP headers. SER is a *very* powerful
SIP proxy. I imagine you should be able to make SER translate those
headers into the URI as it routes the SIP packet.
Tom Browning wrote:
To send calls into a custom SER implementation, I need to be able to add
some
Joseph L. Casale wrote:
So my SIP Provider states they do not offer the service to list my numbers w/
the Whitepages.
We phoned the Whitepages and they said we can't do it, the SIP Provider must?
Either one/both of them is/are useless or I must switch SIP providers to one
that can get
This should do it, but I've not actually tested it. It is based on a
line from my own dialplan.
_X.,n(entrada),Set(CALLERID(num)=${IF($[${LEN(${CALLERID(num)})} =
0]?00:${CALLERID(num):0:11})})
Venefax wrote:
I have two lines in my dialplan that I wish to make it into only one, and I
Oddly core show function SPRINTF works on my 1.6. SPRINTF function
does not seem to be in 1.2 and I don't have any 1.4 systems.
Venefax wrote:
Believe it or not, I cannot find online a single piece of documentation for
the Asterisk function SPRINTF. This example does not work, for it changes
Answer() is seldom the solution.
Rob Hillis wrote:
Steve Totaro wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Answer is the /cause/? Or do you mean it's the solution?
--
Consulting for Asterisk, Polycom, Sangoma, Digium,
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to do this, but
since your Internet connection was down it could not do that. You
should look at /etc/hosts on the Asterisk machine and make sure that
each IP address of
:
On June 9, 2008 01:34:31 pm Eric ManxPower Wieling wrote:
You should not expect FaxOverVoiceOverIPOverInternet to work well. If
you stick to ulaw codec for the entire call, it might work well enough
for your use, but it might not.
Just as an FYI - you have too many Over's in your description
The G729 codec is neither open source, nor is it free, and the
license/patent does not make an exception for educational use.
The Intel LIBRARIES are free for educational/personal use, but the
license for that software says that you still need a license from the
G729 patent holder before use.
The external DNS server would immediately return with a not found
message. Without internet access you'll have to wait for the timeouts, etc.
Joseph L. Casale wrote:
Asterisk gets very upset if it can't lookup the host name associated
with every IP on the system, normally it would use DNS to
You should not expect FaxOverVoiceOverIPOverInternet to work well. If
you stick to ulaw codec for the entire call, it might work well enough
for your use, but it might not.
John Morey wrote:
I've been thinking about something around these lines that I'd like feedback
on. What I'd like to
Correct. The previous poster was wrong.
Drew Gibson wrote:
Nope, didn't help.
Doesn't the context declaration come *before* the channel assignment in
zapata.conf?
It's working that way in our main Asterisk server.
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN,
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as screeching,
feedback, static, or other useless terms. If users report static
on a system where there cannot be static (all digital, PRI, SIP, etc),
you might be
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