Hi
Just an assumption. After packets reach Asterisk, it does the conversion
into the required format and forwards it to Zaptel driver, which in turn
combines and sends one RTP stream back to Asterisk.
How can a client check about number of participants etc.
hi List,
i've been trying to get festival work on my 1.4.4 *box for the last 3days,
i've used the tutorial on this page
http://www.voip-info.org/wiki-Asterisk+Festival+installation
with exactly the same line in my dialplan just to make a test
now when i try to call( dial 555 ) from my softphone
Hello,
I am using asterisk and a2billing. Can some one tell me how can I get
callingani2 field in a2billing. That way I will be able to identify if
the call is from a pay phone. My telco provider is sending me isup-oli
in the the from field.
Or if there is another way to get the information if
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me
Dear users,
I think I may found a bug in the voicemail module of Asterisk 1.4.2!
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so that
some mail servers won't reject the mails. That's why I've set the
Sven Jacobs wrote:
Dear users,
I think I may found a bug in the voicemail module of Asterisk 1.4.2!
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so
that some mail servers won't reject the mails.
On 7 May 2007, at 17:27, Florian Overkamp wrote:
Hi Everton,
Everton Goularth wrote:
I had success to do my asterisk to record CDR in a databese MYSQL...
Now, I need to do it to record CDR in Oracle...
Does Anybody knows how to do this??
Every hints are welcome
There is no native
Per Jessen wrote:
Outgoing email notifications should use a real existing domain (let's
call it domain.real) instead of the local domain (domain.local) so
that some mail servers won't reject the mails. That's why I've set the
serveremail option in voicemail.conf to [EMAIL PROTECTED]
On 7 May 2007, at 19:51, Gordon Henderson wrote:
On Mon, 7 May 2007, Tielin Xu wrote:
Hi list:
Has anyone done to set up two servers in different remote offices
through VPN
in order to get the VoIP communication?
Yes it will work, but depending on your hardware you might be
better off
Sven Jacobs wrote:
You fix that in your mail-server with aliasing and/or canonicalising.
I think the Asterisk behaviour is correct. It is similar to
receiving an email from cron or some other daemon. That is sent
from [EMAIL PROTECTED], which is fine for your internal purposes, but
if you
I will be out of the office until Monday, May 14. Please contact OWD at
800-337-3839 and ask for Client Services if you need immediate assistance.
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To UNSUBSCRIBE
As far as I can tell (but I'm on 1.4.1), the serveremail option only
sets the From-address, not the envelope-address. The envelope will
probably always be asterisk-user@hostname
The From-address ist set by the fromstring option - which works btw - so
you are wrong :) Unfortunately setting the
hi
vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c
this file and look for line that says 2.6.19 change it to 2.6.18 and save
and compile
arun
On 5/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, May 04, 2007 at 01:55:20PM -0400, mail-lists wrote:
I get the following error when
On Tue, May 08, 2007 at 01:27:03PM +0400, Arun Kumar wrote:
hi
vi /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c
this file and look for line that says 2.6.19 change it to 2.6.18 and save
and compile
I repeat again: please test the patch in
http://bugs.digium.com/view.php?id=9006 so
Sven Jacobs wrote:
As far as I can tell (but I'm on 1.4.1), the serveremail option only
sets the From-address, not the envelope-address. The envelope will
probably always be asterisk-user@hostname
The From-address ist set by the fromstring option - which works btw -
so you are wrong :)
Hi all, I have a problem with my beronet card with 2 isdn. I think
drivers and Asterisk are ok but the red led on the card always blinking.
The card is connected with PBX. I post some conf:
[EMAIL PROTECTED] ~]# misdnportinfo
Port 1: TE-mode BRI S/T interface line (for phone lines)
-
Hi Tim,
You will need an Oracle ODBC driver that Asterisk can use to connect to an
oracle instance (local/remote). As far as I am aware, Oracle don't have
unix/linux ODBC driver as of yet, but you can get one from EasySoft. They
have an eval version you can try out to see if it works, have a look
Hi Gavin,
I don't know if this will help, but can you check to see if you have libtool
installed?
I had a similar issue with unixodbc, and once I installed libtool, it
rectified the issue.
Once libtool is installed, re-run configure and it should hopefully work.
Thanks
Bruce
-Original
Hi,
Don't you have to configure the host option for each channel in iax.conf?
Look at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+iax.conf
Ronaldo.
chawki hammoud wrote:
Hi:
It's the first time I have this problem.
No matter how I configure my two IAX machines the
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
Hi,
you can use an AGI to connect to asterisk manager and retrieve the
info you need about the queue.
Hope it helps
Arun Kumar ha scritto:
Hi
I've few queues configured in * box is there any what that before
sending call to a particular queue can we get the status of the queue
that is
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
hello friends, I have a problem when I call to outside (9) from IPs
Telephones.
the incomning calls are OK.
in the console when I put sip debug peer 101 I have this lines:
*CLI sip debug peer 101
SIP Debugging Enabled for IP: 10.0.0.9:5060
*CLI
-- SIP read from 10.0.0.9:5060:
INVITE
Hello Josu,
In you're sip.conf you have the 2 phones configured that they are in the SOME
context.
Looking at the SOME contect in extensions.conf you only have the 2 phones
defined. If you want to call ouside from the SOME context as well, you need to
include the outgoing context there as
I understand that it is customary for SIP User Agents to send OPTIONS
packets every now and then to check that a peer is still alive and well.
Indeed I understand that Asterisk itself sends them if qualify is set to
yes in the peer configuration.
How is one supposed to configure the dialplan
Hi,
I hope this gets picked up by some bug marshall ...
I have downloaded (yesterday) the 1.2 branch from svn ...
When running: asterisk -c
loaded modules:
[modules]
autoload=no
load = pbx_functions.so
load = pbx_config.so
load = codec_a_mu.so
load = format_pcm_alaw.so
load = codec_ulaw.so
Maybe I'm misinterpreting things, but this is what I se:
fromstring = the From:-text, not the From:-address.
I'm just using the default fromstring, but I've set
serveremail = asterisk@realdomain
With this I get
From: Asterisk PBX [EMAIL PROTECTED]
Still, the envelope is [EMAIL
Hi
I already tried asterisk manager but Im not able to get status for each
queue member.
thanks
On 5/8/07, Edoardo Serra [EMAIL PROTECTED] wrote:
Hi,
you can use an AGI to connect to asterisk manager and retrieve the
info you need about the queue.
Hope it helps
Arun Kumar ha scritto:
thank you very much!
it works
- Original Message -
From: Dijkstra, Roelof
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, May 08, 2007 1:53 PM
Subject: RE: [asterisk-users] outgoing calls
Hello Josu,
In you're sip.conf you have the 2
On 08/05/07, Bruce McAlister [EMAIL PROTECTED] wrote:
Hi Gavin,
I don't know if this will help, but can you check to see if you have libtool
installed?
I had a similar issue with unixodbc, and once I installed libtool, it
rectified the issue.
Once libtool is installed, re-run configure and it
I have an issue that hopefully you can help me solve. I've got the sangoma a101
card and installed it on freebsd but I according to the manual I should be see
when running dmesg PCi0 vendor. something that tells me sangoma it's being
recognized by the system. Now this is the 2nd card I try,
Hello again,
I have a little problem, every time I switch on the Asterisk server I must load
two modules: modprobe zaptel and modprobe wctdm
Is there any way to load there automatically when the server start?
I have a Debian Etch.
One more cuestion, it's posible to start Asterisk (asterisk
Rohan Hathiwala wrote:
Hi,
I need asterisk to instruct the other side to send RTP to a conference
server running on a different machine. The conference server does not
understand SIP so I cannot use the SIP REFER method.
I have another question. Suppose when processing a SIP INVITE we want to
Sven Jacobs wrote:
Maybe I'm misinterpreting things, but this is what I se:
fromstring = the From:-text, not the From:-address.
I'm just using the default fromstring, but I've set
serveremail = asterisk@realdomain
With this I get
From: Asterisk PBX [EMAIL PROTECTED]
Still, the envelope is
Cesc wrote:
Hi,
I hope this gets picked up by some bug marshall ...
Eep! Filing a bug is best instead of email it here for future reference...
I attached gdb to the locked process:
0xb725af28 in std::_Rb_tree_increment () from /usr/lib/libstdc++.so.6
(gdb) bt
#0 0xb725af28 in
JK wrote:
Hello,
I am using asterisk and a2billing. Can some one tell me how can I get
callingani2 field in a2billing. That way I will be able to identify if
the call is from a pay phone. My telco provider is sending me isup-oli
in the the from field.
Or if there is another way to get the
chawki hammoud wrote:
Hi:
It's the first time I have this problem.
No matter how I configure my two IAX machines the
connection is rejected.
chan_iax2.c:5550 socket_read: Call rejected by :
No authority found
Without seeing your dialplan it is a little hard to determine why but
I'll
Hi Zvonimir
I have an issue that hopefully you can help me solve. I've got the sangoma
a101 card and installed it on freebsd but I according to the manual I should
be see when running dmesg PCi0 vendor. something that tells me sangoma
it's being recognized by the system. Now this is the 2nd
On 5/7/07, nik600 [EMAIL PROTECTED] wrote:
i am experiencing problem with asterisk 1.2.18
I've downloaded and installed pwlib and openh323 with the following commands:
cd /path/to/pwlib
./configure
make clean opt
cd /path/to/openh323
./configure
make clean opt
then 'ive set the corresponding
Alex Lake wrote:
I understand that it is customary for SIP User Agents to send OPTIONS
packets every now and then to check that a peer is still alive and well.
Indeed I understand that Asterisk itself sends them if qualify is set to
yes in the peer configuration.
How is one supposed to
Zvonimir Mileta wrote:
I have an issue that hopefully you can help me solve. I've got the
sangoma a101 card and installed it on freebsd but I according to the
manual I should be see when running dmesg PCi0 vendor. something
that tells me sangoma it's being recognized by the system. Now
Hi Nitesh -
Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2
using H.263 Video Coder.
I had to update both phones firmware with new one...
Out of curiosity - do you like the phone? I've looked for reviews,
but I haven't found any that rate the phone's functionality.
I have an asterisk Server (2.1.17) behind NAT with a static IP and port
forwarding enabled. The remote SIP phone is also behind NAT.
I've gotten them to work except when I specify a secret.When there is a
secret configured the phone can not authorize. Has anyone gotten this
Stewart, till what time on Monday will you be out?
On 8 May 2007 04:10:44 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I will be out of the office until Monday, May 14. Please contact OWD at
800-337-3839 and ask for Client Services if you need immediate assistance.
I have two asterisk servers. One is at location 1 and the other is at
location 2.
What I am trying to do seems straightforward. I want the Asterisk
server at location 2 to
send all it outbound calls to the Asterisk Server at location 1.
Both asterisk servers can dial each other using
- Noah Miller [EMAIL PROTECTED] wrote:
1. A patch allowing capture audio streams in a way that will allow
[us] to
debug (and presumably fix) the problem was mentioned by Kevin. -
Anything new about it ?
I couldn't find it in Zaptel 1.2.17.1 nor 1.4.2.1 changelog.
I believe it was
Joshua Colp wrote:
Alex Lake wrote:
I understand that it is customary for SIP User Agents to send OPTIONS
packets every now and then to check that a peer is still alive and
well. Indeed I understand that Asterisk itself sends them if qualify
is set to yes in the peer configuration.
How is
Chris Shipman wrote:
I have an asterisk Server (2.1.17) behind NAT with a static IP and port
forwarding enabled. The remote SIP phone is also behind NAT.
I've gotten them to work except when I specify a secret.When there
is a secret configured the phone can not authorize. Has
Hi all,
We are an ISP in Switzerland and we propose VoIP with Asterisk.
Everything works perfectly for all clients but one. In a conversation,
they have no sound during 2 to 8 seconds using the G729 codec (I didn't
make the test with G711).
The Client configuration is perfect (QoS and bandwidth
Hello,
So far yes... The Video phones are behaving good and all the
functionality working.
I have 5 phone on the network and planning to put more by next week.
Cheers,
Nitesh
Noah Miller wrote:
Hi Nitesh -
Thanks everyone... The GXV-3000 IP Video Phone works with Asterisk 1.2
using
Joshua Colp wrote:
The voicemail email gets handed off to sendmail for actual sending.
It's adding on the envelope above.
Yes, but asterisk is writing the From: header.
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user -
SIP wrote:
Joshua Colp wrote:
Handling of OPTIONS in Asterisk has changed a little bit through
chan_sip versions... but for the most part the other side usually just
wants you to respond with something/anything. Is the other side
unhappy with the 404 Not Found?
Joshua Colp
Software
Turn off VAD (Voice Activation Detection) on the client software or your
carrier is using VAD.
Asterisk does not like VAD
Best regards
Erick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas Deillon
Sent: Tuesday, May 08, 2007 10:07 AM
To:
Using a quad core 1.86GHz Xeon CPU here, running Asterisk 1.4.2. Noticed
the following:
Cpu(s): 4.3% us, 95.4% sy, 0.0% ni, 0.2% id, 0.0% wa, 0.0% hi, 0.0% si
30908 asterisk 18 0 188m 10m 5152 S 400 0.3 51051:13 asterisk
Asterisk is eating up all the cores running the CPU at
Hi guys,
I had the same problem ... and then remembered that my asterisk
1.2.9.1 compiled just fine ...
So, i tried that Makefile ... and voila! :)
See attached patch ...
Cesc
On 5/8/07, nik600 [EMAIL PROTECTED] wrote:
On 5/7/07, nik600 [EMAIL PROTECTED] wrote:
i am experiencing problem with
ax.
The downside of rx_fax is that you need to compile it into asterisk.
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail
system. The channel must be dedicated to faxing, and that's that. This
may or may
The downside of rx_fax is that you need to compile it into asterisk.
The downside of iaxmodem is that (to my knowledge) you can't easilly
implement an auto-answer/detect fax/voice/ auto attendant/voicemail
system. The channel must be dedicated to faxing, and that's that. This
may or may
Hi,
I will add the report ... though I find the system a bit cumbersome
for sporadic users like me.
Oh, and you are right ... without chan_h323 asterisk shuts down just fine.
Regards,
Cesc
On 5/8/07, Joshua Colp [EMAIL PROTECTED] wrote:
Cesc wrote:
Hi,
I hope this gets picked up by some
We have several Sangoma cards that we used during a transition time in
our the replacement of our legacy voice system that we no longer need.
Each of them saw about a month of service and are in good working order.
We'd be happy to get 70% of retail for them. They are as follows:
qty 2 A104D
I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf
being selected. And when reading rtp if 'f' character shows up vector to
fax extension
Kevin Collins
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of mail-lists
Sent:
NM == Noah Miller [EMAIL PROTECTED] writes:
NM If it helps at all, I read a study that said that SSL VPN's can
NM actually help with jitter problems. So it might be preferable to
NM implement something with OpenVPN (uses SSL) rather than an
NM IPSec-based VPN. I found the link:
Only if you use
many thanks for your help!
i have used a makefile of a release 1.2.13 and now i've correctly compiled it.
On 5/8/07, Cesc [EMAIL PROTECTED] wrote:
Hi guys,
I had the same problem ... and then remembered that my asterisk
1.2.9.1 compiled just fine ...
So, i tried that Makefile ... and voila!
Hello,
Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
Extension xxx is unavailable
The goal is
We are using Record to monitor calls. We use this because it has the
option of a max time in it's call.
the problem is, and I'm not at all sure it is happening in record, the
recordings have sections of the conversation missing, sometimes. there
is not significant pattern as to the types of
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
mysql Select extension from
I will be out of the office until Monday, May 14. Please contact OWD at
800-337-3839 and ask for Client Services if you need immediate assistance.
___
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To UNSUBSCRIBE
Is this for asterisk 1.2 or asterisk 1.4?
2007/4/26, bkruse [EMAIL PROTECTED]:
Hey guys,
In my spare time(off of work, not digium related whatsoever) I finished
the cacti php script.
I need someone to help me do some finishing touches and make a basic
layout and pretty colors for the
On 5/8/07, Pedro Silva [EMAIL PROTECTED] wrote:
Can i identify the sound files that are played in the asterisk
console? I defined the verbose to 100 but i can not see the sound
files that are played in some situations... :(
For example, I need to know what files are played for the message:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every
new PC seems to be coming with Vista these days. I expect it'll only be a
matter of time for all of us before clients start needing Vista-compatible
softphones (if it's not already happened).
So,
Forrest Beck wrote:
I have all my SIP users in a realtime database. I would like to use
MySQL command to query the database and use the results from the query
to page all the phones found in the query.
The results from the MySQL query will be multiple rows of extension:
Something like:
- Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly,
nearly every new PC seems to be coming with Vista these days. I expect
it'll only be a matter of time for all of us before clients start
needing Vista-compatible
Hi
Can you send me output from 'pciconf -l'?
Thanks
Alex Feldman
Software Project Leader
905.474.1990 x104
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zvonimir
Mileta
Sent: Tuesday, May 08, 2007 8:53 AM
To: asterisk-users@lists.digium.com
Subject:
Hi all,
I have an Asterisk server connected in a PABX (TELEDATA) by channel
Unicall..
I`m having problem when somebody call from PABX to Asterisk..
Eg: When somebody dial 1234, I received 113344 in the
Asterisk CLI...
If somebody can help me... or already saw this...
Well This seems to work.
[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to
Hi:
It's been a few weeks since the great voip-info.org crash.
Around that time there was some lofty talk about a set of mirrors being
set up for it.
Has anything happened with that, or are we just going back to business
as usual?
-Stephen-
___
Hi,
Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog
4187?
I was told some functionalities like CLID will only work with an Ericsson
PABX but other than that I would like to hear from anybody using this phone
on a FXS port.
Thanks,
Jose Limeres
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me
How about required MTU and jitter? I think openvpn will add some latency and
frames will be charged with supplementary encapsulation bits.
On 08 May 2007 19:03:09 +0200, Benny Amorsen [EMAIL PROTECTED]
wrote:
NM == Noah Miller [EMAIL PROTECTED] writes:
NM If it helps at all, I read a study
Cohen,
Thanks for your help, but I solved this problem removing the ACPI and
APIC from the boot in /boot/grub/menu.lst.
Thanks,
MCelo.
2007/5/8, Tzafrir Cohen [EMAIL PROTECTED]:
On Mon, May 07, 2007 at 05:15:26PM -0300, MCelo wrote:
Cohen,
On different boots you get the modules loaded
Hi,
Does anyone know how to adjust the volume of the ringing application? I have
done a lot of internet searching and have not found much.
Thanks.
Jad
Network Blitz Bkgrd.gif___
--Bandwidth and Colocation provided by Easynews.com --
Jadrien Wauthier wrote:
Does anyone know how to adjust the volume of the ringing application? I
have done a lot of internet searching and have not found much.
You cannot do this in Asterisk.
Some SIP phones might allow you to do so by setting an option on the
phone, but you would have to
Hi All,
We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that
there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there
was something better. Are people using LDAPget or something else?
___
--Bandwidth and
http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4
jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP
RTP packets renumbered on transmit, or is the original sequence number
preserved in the UDP
Forrest Beck wrote:
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
I don't use Realtime in Asterisk personally so I'm not sure if it
implements it or not, but I agree that being able to iterate over a
ResultSet is a pretty basic need.
Lee Jenkins wrote:
Forrest Beck wrote:
How does asterisk handle the multiple results. Is there a way to loop
until there are no more rows?
Sorry, I forgot the last link for the AEL2 scripts:
http://sourceforge.net/projects/aelscriptlib/
--
Warm Regards,
Lee
(repost - can anyone confirm whether they've seen this before, or have
any tipes in debugging it?)
Hi Everyone,
I was hoping someone might know why I am experiencing a problem with
Asterisk logging the event:
[May 3 12:07:41] WARNING[30371] chan_sip.c: Maximum retries exceeded on
transmission
Damon Estep wrote:
http://www.asterisk.org/node/48317 does a nice job of explaining the
1.4 jitter buffer, however it raised a question in my mind.
In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the
UDP RTP packets renumbered on transmit, or is the original sequence
Sorry for being a little off topic, but I'mconsidering a few new phones
for my Asterisk installation. I have a mix of Polycom 500/600s and an
Aastra 480i CT. I'm considering adding a couple of Aastra 57i or 57i
CT.
Does anyone here have experience with the 480i CT and the newer 57i CT?
I'm
Does anyone know how to adjust the volume of the ringing application? I
have done a lot of internet searching and have not found much.
You cannot do this in Asterisk.
Some SIP phones might allow you to do so by setting an option on the
phone, but you would have to ask the company that makes
I have Vista on my new HP laptop X-lite soft phone works like charm with it,
I tried sjphone, I couldnt get that working, its gets hung.
--
Deepak
Chris Bagnall [EMAIL PROTECTED] wrote:
Greetings list,
I've noticed over the last couple of weeks that, unsurprisingly, nearly every
new
Hello,
A few days ago I've asked about the ability to play a stuttered ringing
tone when the called party is already on the phone. I've found a partial
solution for it.
To describe again the problem: When a user is on a call and someone else
calls him, the caller does not know that the
Hi Alex,
How is one supposed to configure the dialplan so that Asterisk responds
correctly to these requests?
At the moment, I'm seeing Looking for s in default and then a 404 Not
Found being returned - which can't be right.
Not specific to an OPTIONS packet, but I know that I previously
I can confirm the same error message... i haven't done nearly the amount of
debuggin you have but it's the exact same error message i receive when i use
a software based SIP phone connecting to another internal software SIP
phone... some times it's twinkle to xlite some times xlite to xlite and
hi List,
i've been trying to get festival work on my 1.4.4 *box for the last 3days,
i've used the tutorial on this page
http://www.voip-info.org/wiki-Asterisk+Festival+installation
with exactly the same line in my dialplan just to make a test
now when i try to call( dial 555 ) from my softphone
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