Somesh S Shanbhag wrote:
Dear All,
Can I use Asterisk IP-PBX as Softswitch? If not, what
is lacking in asterisk
from not *becoming* softswitch?
What is your definition of a softswitch?
/O
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Matt Riddell wrote:
Marc Storck wrote:
Hello,
I would like to know if there is a way in IAX2 and SIP to tell a client
to register at a different server.
For example:
Client tries to register at server B but server B answers with some sort
of redirect to tell the client to register at server
BJ Weschke wrote:
On 11/19/05, Jean-Denis Girard [EMAIL PROTECTED] wrote:
Hi all,
Is sending text to a conference supported by asterisk-1.2, ie one member
of the conference sends text, it is received by all other members of the
conference (provided their channel supports text of course) ?
I
Anders Svensson wrote:
Someone who can recommend a good E1 gateway for terminating VoIP
traffic. H323 or Sip
Asterisk!
/O
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Jesus Bermudez Riquelme - Pcmur Soluciones Informaticas wrote:
Hi all,
i've a problem in my Asterisk system. We have around 30 SIP phones
connected to an asterisk system, and sometimes some SIP channel
(associated to an extension) gets busy all the time, even when that
extension isn't in use.
Pavel Siderov wrote:
Hi,
I'm experiencing some problems with my Asterisk 1.0.9. When a customer
tries to use transfer method sometimes Asterisk crashes. The following
message appears in /var/log/asterisk/messages
Nov 17 15:56:35 WARNING[759]: No path to translate from
Friends in the Asterisk community,
There have been a lot of questions about Asterisk version numbers on the
mailing lists. Here's a clarification:
* Executive summary
---
- Asterisk 1.2 = RELEASE version (previously called stable)
Asterisk 1.2.0 = First release of 1.2
Matt Florell wrote:
Hello,
Several of us were told that there would be a 1.0.10 release as the
final release of Asterisk 1.0 tree. There are several serious bugs in
the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to
have this packaged as a release before the tree stops
Pavel Siderov wrote:
Hi,
It's not possible to provide log due to the reason that system is in
production and there are many current calls. Crash happens on 1-2 weeks
once. I cannot simulate and get the same result with x-lite, cisco ata
and sipura 3000 when trying transfer. But some of the
Based on a discussion on the IRC a long time ago (several days) I've
created a patch for 1.2 in the bug tracker that allows you to see if a
parking lot is occupied or not - provided you use the Flash panel or SIP
subscriptions.
What you do:
* Patch the 1.2 source with the patch in
Matt Florell wrote:
Hello,
Several of us were told that there would be a 1.0.10 release as the
final release of Asterisk 1.0 tree. There are several serious bugs in
the 1.0 tree that have been fixed in v1-0 cvs and it would be nice to
have this packaged as a release before the tree stops being
Alexander Lopez wrote:
Does it hold state information for any channel? Even ZAP, IAX,
etc!!!
If it does, Olle, you have just placed us one step closer to being able
to emulate a Key system!!!
This fix is very focused on parking. Previous to this fix, we can check
device status
in
Pavel Siderov wrote:
Could you please advice me how to create log all calls or only for
those using Bye/Also. I've made some researche using google and found
that SJPhone use this method -
http://www.sjlabs.com/doc/SJphone%20Profiles.pdf .
If you can find out which peer uses SJphone by
Pedro wrote:
Yeah - tried that. Here are 2 lines I have in my modules.conf file:
noload = pbx_realtime.so
noload = app_realtime.so
For some reason, I still get the following in my logs even after a
restart of Asterisk.
Nov 21 13:17:08 ERROR[31192] res_config_mysql.c: MySQL RealTime:
Trond,
You need to tell us more. The SIP phones - what are they registering as?
(Show sip.conf peer configs)
If one register as a SIP peer trond you should be able to dial
SIP/trond and get a full URI. If not, something is really wrong.
/O
___
Andreas Sikkema wrote:
There should be other voices worth while...
Give other people the chance
The market is growing...
Be open :)
Asterisk as a product is in no way closed to Allison's voice prompts.
If that was the case, it would be a serious roadblock for international
use.
There are
Avi Miller wrote:
Olle E. Johansson wrote:
be seen as a sample of a full prompt set and something that is extremely
This actually leads to a question I've had for a while: Is there a list
somewhere of all the prompts (by filename) and what is said? I've
searched the Wiki but haven't
Matt Riddell wrote:
Is it just me or have the monthly tips from Olle stopped. I just opened my
mail client and the last few posts were about 80% HTML.
Please Olle if you already posted it this month, can you step it up to once
every couple of weeks!
Well, the monthly tip of this month is:
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
These are exciting times for
Eric ManxPower Wieling wrote:
Neil Cherry wrote:
Does someone explain the 's' extension? In the Wiki it says it's
the catch all extension. In the Asterisk 1.2-rc1 it say it isn't
but doesn't say anything more. Needless to say I'm confused.
When a call comes into Asterisk (PSTN, VoIP,
Peter Dean wrote:
I have now been successful in getting the notification lights working.
Then asterisk extensions hint required a reference to the extension
being monitored and the extension monitoring the call status.
i.e. _226,hint,SIP/226SIP/101
So with this change the asterisk hint
[EMAIL PROTECTED] wrote:
Hi List
I’m getting this notification from my one and only SNOM 360 every time a
number button is pushed.
I know that it’s only a notification, but it really irritates me. Is it
anything I can/should do anything about ??
Not really. We do not support
Harry,
RFC 3265 is a generic framework for subscriptions and notification. No
one can't answer a question whether we support it or not, because you
also need to specify which event-package you are interested in to make
it a meaningful question. I'll give you a general answer:
We do support
Jason Pyeron wrote:
take for example a phantom SIP/400b from a previos phone config, without
restarting * how can I purge only 400b?
testserver*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
400c/400c (Unspecified)D 0
[EMAIL PROTECTED] wrote:
After some experimentation and posting, I have concluded
that in the file sip.conf, the line:
username = irrelevant
Please read sip.conf.sample in your distribution for updates on
configuration parameters.
The username parameter has nothing at all to do with a
Adam Moffett wrote:
does anyone know when 1.2 will no longer be beta?
The quick answer is: When it's ready for release.
Open Source software doesn't really follow a set agenda. We have been in
code freeze for quite a while, fixing bugs. A lot of people are testing
the 1.2 beta and reporting
Mark Hulber wrote:
In recent CVS Head build when I run: sip show peers my dynamic peers
show:
Name/username HostDyn Nat ACL Port Status
sipura2_2/sipura2_2(Unspecified)D N 0UNKNOWN
sipura2_1/sipura2_1(Unspecified)D N
Juan Salas wrote:
yes,
I tested too and it's works.
The Problem is when we want to add (or delete)
registers without reload the asterisk.
We are using it like a border server wich
is registered like many users in a SER machine
and the real endpoints are registered in the
asterisk.
I
Luki wrote:
Hi all,
this is probably a asterisk-devel question but I'll try it here first.
Is there a way to delay a ReInvite on SIP? I have an issue where my
provider's server is only ~1 ms RTT away and for about 1/3 of the
incoming calls I get a 482 Loop Detected error because the
marek cervenka wrote:
hi,
will be somewhere materials (videos, presentations) from astricon?
Registered attendees will get information about the material soon.
No videos where recorded this year.
The 1.2 presentation I made together with Kevin has been available
for a while at
The Norwegian Asterisk user's group is meeting on Tuesday next week. A
full one-day seminar in several tracks covering Asterisk is arranged in
Oslo.
See http://www.asterisk.no for the agenda.
I will attend the meeting and enjoy listening to people's experience of
Asterisk and various
Juan Salas wrote:
Hello!
As I know, the register is a variable of [general] section in sip.conf.
You can't use it in database, ie you can't add new registers without reload
the asterisk..
You can have a static config in a database, but you will still have to
reload.
/O
Ronald Wiplinger wrote:
I was looking for the text in the /etc/asterisk directory, but it must
be somewhere else. Can anybody tell me where? And can it include Chinese
as well?
Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
/configs directory of your source code tree.
I
Guido Hecken wrote:
I was looking for the text in the /etc/asterisk directory, but it must
be somewhere else. Can anybody tell me where? And can it include Chinese
as well?
Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the
/configs directory of your source code tree.
I have
Cirelle Enterprises wrote:
is the cvs head version considered 1.0 or 1.1 with
regard to spandsp
CVS head would be considered 1.1 at this time.
/O
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Steve Davies wrote:
On 10/21/05, Steve Davies [EMAIL PROTECTED] wrote:
I have noticed that when a SIP redirect is sent back to Asterisk by a
SIP peer, that Asterisk will (quite appropriately) do a
Dial(LOCAL/redirect-number) in the context of the original callee.
It also forks the CDR, which
Someone wrote me off list:
I would like to be able to help, but I'm not a C programmer - is there
any other way I can assist the project?
There are many ways! Testing new patches, making sure they are
documented properly, that they work as expected. Making sure the Wiki is
up to date with
Jay Milk wrote:
I'm having the following recurring problem with asterisk:
When for any reason one of my SIP providers fails to register (i.e.
internet connection dropped), ALL SIP channels fail. This means that,
for example, when my internet connection is out, none of my internal sip
Please move all discussions about this service provider to the
asterisk-biz list.
Thank you!
/O
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Ted Cabeen wrote:
In August and September of last year, there was some discussion of
changing the Voicemail and Record applications to send back CNG RTP
packets during recording to prevent inbound calls from dropping when
they assumed a disconnect after 30 seconds of no RTP frames.
Was
Sherwood McGowan wrote:
I've been poring over the sample configs for the latest CVS-HEAD as well
as the readmes from the source's docs directory. I'm finding a lot of
options that weren't previously available, and would like to know if
anyone's gone so far as to play with these various new
Sergey Okhapkin wrote:
http://bugs.digium.com/view.php?id=5472
The users will not learn about undocumented AEL features. Sure I'm not
going to reopen the problem.
Sergey,
I am sorry if you took our comments that badly. I proposed a worthing
and you did not accept that and refused to update
Doug Meredith wrote:
Olle E. Johansson [EMAIL PROTECTED] wrote:
Sergey,
I am sorry if you took our comments that badly. I proposed a worthing
and you did not accept that and refused to update according to our
suggestions. Tilghman therefor decided to close the bug.
I suggest you try again
Jason Becker wrote:
Vahan Yerkanian wrote:
I'd recommend using native mp3 support that is available in CVS HEAD,
as madplayer mp3 decoder gives a lower quality sound (audibly more
cranky/noisy).
I don't follow CVS commits but if that's the case the mpg123 target
should be removed from
Richard Cook wrote:
A great stance. Another contributor most likely lost. Nice job.
Welcome to be a bug marshal and help keep the contributors still around!
Contact [EMAIL PROTECTED] to sign up today. We do need your help
so this does not happen again.
/Olle
Matt Hess wrote:
Attached is a pcap of sip packets that pertain to another call similar
to the history shown.. it's hard to nail these down as it takes a lot of
time, patience and sifting through dumps.
Well, a pcap does not tell me how Asterisk reacts, sorry. That was what
I wanted to see -
Matt Hess wrote:
I should have mentioned that I can't do a full sip log.. with several
calls a second whipping through this system it's almost impossible to
weed out the info for the proper call.. and usually I don't see the dead
channel until well after the fact.
Looked at this with
Matt Hess wrote:
I should have mentioned that I can't do a full sip log.. with several
calls a second whipping through this system it's almost impossible to
weed out the info for the proper call.. and usually I don't see the dead
channel until well after the fact.
kotesh m wrote:
Hi,
I am new Asterisk. I configured asterisk1.5 and be able to communicate
That is amazing. You are new and already at version 1.5. I have been
around for a while and only reached 1.1dev, working on 1.2 :-)
Guess I have some catching up to do...
/O ;-)
(Sorry, could not
Federico Alves wrote:
My company can supply inbound numbers in 49 States, via IAX, SIP or H323. I
noticed that nobody offers this much needed service to the Asterisk
This kind of commercial notices should *not* be sent to this mailling
list. Please use the asterisk-biz mailing list for all
Matt Hess wrote:
I have this in sip show history for a particular channel marked as dead
(should be removed) in sip show channels:
1. TxReqRelINVITE / 102 INVITE
2. Rx SIP/2.0 / 102 INVITE
3. CancelDestroy
4. Rx SIP/2.0 / 102 INVITE
5. CancelDestroy
6.
Steve Gladden wrote:
Sorry this is a bit of a newbie question, I've been at this for a few
months and still have not quite figured this one out.
I've been able to setup one itsp (incoming calls) (sip account) with a
register line like this:
register = nnn:[EMAIL PROTECTED]
-or-
Luca wrote:
Hi Matthew,
It’s really possible to allow register = to work in Realtime mode?
Only in static mode. Since it's realtime, we do not load the peers at
load time and unless we force Asterisk to check *every* peer in the
database and register, I fail to see how this would work.
Dan Journo wrote:
Hi Guys,
Basically, i'd like to be able to force a call to drop by using one of
the following methods. Does anyone know if it is possible? It has to be
more or less realtime.
a) Issuing a command to asterisk via telnet.
b) Altering a field in the realtime database
c)
Matt Florell wrote:
Hello,
We've released another update to our Asterisk GUI Client suite: 1.1.7
http://astguiclient.sf.net/
Matt will show this version in the Asterisk Solutions Showcase at
Astricon next week. It's a part of the Astricon exhibition where Open
Source projects can show
Dinesh Nair wrote:
hey all,
am wondering if anyone has successfuly done a SIP attended transfer
using the REFER method (after an INVITE obviously) and the Replaces:
header.
That is not supported today. However, I have working code that will be
submitted to the bug tracker after Astricon.
This work will belong to a future version of Asterisk, not 1.2 release.
/Olle
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According to
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+incominglimit
incominglimit and outgoinglimit are deprecated.
incominglimit is replaced by call-limit. Please read sip.conf.sample.
Outgoinglimit has not worked for ages, so we removed it. One limit works
for both
Arne Morten Johansen wrote:
Does the incomming call context in extensions.conf always have to be
[default]?
Can't i define different context for incomming like i can for Outgoing
in the sip.conf? My default conf is getting very large.
In sip.conf, you can set context in a few places:
In
Benjamin Lawetz wrote:
I'm looking for a way to transmit a user specific variable to my dialplan
If we use the example of the hair color, I was thinking of having something
like:
[bob]
context=users
host=dynamic
secret=password
type=friend
username=bob
hair=brown
Use
Doug Lytle wrote:
[EMAIL PROTECTED] wrote:
On Mon, 3 Oct 2005, Corey S. McFadden wrote:
Am I just using the Set() command wrong? It seems pretty
counter-intuitive not to enclose multi-word strings in quotes but if
that's the problem let me know.
Yeah, that's the problem.
René Enskat [Teamware GmbH] wrote:
Hey guys.
I have to put my * behind a Firewall through nat on the firewall.
The asterisk is running, but for example a register to an outside PSTN
provider won't work.
I enabled nat for the register but i only get Code 120 Send request.
The other problem
Anders Svensson wrote:
I have a problem. Incoming calls work without problem but I cant call
out. Using AAH.Gets a busy tone
Anyone who can see a mistake in Outgoing settings
context=from-pstn
host=ipkund1.rixtelecom.se
insecure=very
nat=yes
secret=xxx
type=peer
stevanus wrote:
Hi,
Outgoing setting is in zapata.conf. I think you should read the wiki
more ;).
If what you mean by outgoing is another sip extension then you should
look for extension.conf.
Links:
http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
Alex Lake wrote:
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally
forward the Remote-Party-ID from inbound SIP calls (where
trustedrpid=yes) to outbound SIP calls. I guess this is going to be
something we have to use SER for, unless we make our own custom build
(which
Anders Svensson wrote:
Hi!
Is there a simple way for an * newbie to force * to use different
sip-trunks for different calls. I have 2 siptrunks, one for inland calls
and one for international calls. All in country numbers starts with 0
and all international starts with 00. This I have
Balaji NJL wrote:
Announcing – Voice over IP Directory Services
(http://www.voipDS.org)
To make this global, where any VOIP user could make
peer-to-peer call to any other VOIP user, we need the
following
a central repository which stores peer connection
information of all users
an
BJ Weschke wrote:
Upgrade Asterisk. Versions of HEAD post 8-29-05 have this functionality
built in.
Some of it is currently broken, but there is a patch in the bug tracker
that fixes status notification for Eye-beam. haven't tried with Snom.
/O
___
Ray Van Dolson wrote:
On Thu, Sep 29, 2005 at 08:54:42PM -0500, Kevin P. Fleming wrote:
Ray Van Dolson wrote:
Our SIP/PSTN gateway provider seems to think that Asterisk should initiate
a
renegotiation to G711 when it sends the 488 message rejecting T38.
This is not correct. The 488 response
Damian Funnell wrote:
Hi,
Does anyone have remote call pick-up working on * (either via SIP or
otherwise)? If so then can you post your features.conf, sip.conf and/or
zapata.conf?
We can't seem to get this (seemingly simple) function to work.
Check callgroups and pickupgroups in the
Asterisk buddies!
If you have open issues in the bug tracker, please help us with
providing fast responses. All developers are working real hard to close
bugs pending the new release, so we kindly ask you for fast responses on
our questions in the bug tracker. The quicker the better and we'll get
Script Head wrote:
I am upgrading to Asterisk-Realtime and stumbled upon a problem
converting my existing sip.conf register command to the RealTime format.
It seems that sip_friends table setup doesn't allow for such thing to
happen. So far the only way I see to do this is dumping the
Paul Conn wrote:
I’m receiving the following error over and over, adnauseam:
Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce
received from ‘CNAME-CID sip:[EMAIL PROTECTED]’
Does anyone know what “stale nonce” is?
I've answered this question many times, so
Corey S. McFadden wrote:
We've been experiencing an odd issue lately. I'm not sure when it started
because it's not happening on most calls--it seems confined to a couple of
our queues. It's consistent though.
Here's the CLI output:
-- Got SIP response 400 Bad Request back from
Doug Lytle wrote:
Olle E. Johansson wrote:
Corey S. McFadden wrote:
Here's the CLI output:
-- Got SIP response 400 Bad Request back from 192.168.249.94
-- SIP/502-9a58 is circuit-busy
I've tried a few different Asterisk versions CVS-HEAD, stable, even
1.2 beta. I've also
Friends,
I have updated my Asterisk 1.2 presentation with the latest information.
It is still available in the same place as before:
http://www.astricon.net/asterisk1-2/
Please continue to test the beta of Asterisk 1.2, available at
ftp.digium.com. We need all the feedback we can get. If you are
Ronald Voermans wrote:
Hello,
I have asterisk connected to SER/RTPProxy which is again connected to a
IP-PSTN gateway. When calling with a UA, registered at * to a SIP phone
connected to the IP-PSTN gateway, I get 'empty ACKs':
U 192.168.0.173:5060 - 10.254.254.1:5060 ACK SIP/2.0.
Via:
Sherwood McGowan wrote:
Bumping, just in case it got lost in the shuffle today... I think this is an
important thing to be able to do.
Subject: [Asterisk-Users] Asterisk and RTP streams
Guys, I've been poking around trying to find a good answer for this via
voip-info, google, etc...
Tony Hoyle wrote:
Olle E. Johansson wrote:
SIP phones need to re-register every once in a while to tell the server
where it can be reached. If you have a soft phone on a laptop that you
move from network to network - home, office, airport, Barnes Noble etc
- you want to be reached
Chris Stenton wrote:
With todays CVS head I am getting the following being sent after a call
has been terminated
on my Cisco 7960. It eventually gives up with a critical error.
chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission
[EMAIL PROTECTED] for seqno 102
(Critical
Andres wrote:
8.1.50;tag=as12a1c927
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
As you can see. This is just a NOTIFY message. Probably a Keep Alive.
User-Agent: Asterisk PBX
Event: message-summary
Content-Type:
Zeeshan Zakaria wrote:
Hi,
When a SIP client registers on Asterisk server, why it expires after
certain amount of time?
Because it is the way SIP registrations work. For more information, find
a SIP book or read the SIP RFC 3261.
SIP phones need to re-register every once in a while to tell
Brian Capouch wrote:
Olle E. Johansson wrote:
Try setting _ALERT_INFO
The reason for this is that if you set *any* variable with one
underscore prefixing the name, that variable will be copied to the new
channel created by dial() - without the underscore. If you create a
variable called
taf taffey wrote:
Is there a way of increasing the delay before asterisk
picks up the incoming PSTN call?
I'm using a tdm400p with fxo card. It seems to pick up
the inbound call immediately. I want to delay
detecting the call by about 10 secs if poss.
Done some searching but couldn't
Irakli Natsvlishvili wrote:
If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between
them will ALWAYS go via Asterisk.
Dial plan contexts has nothing to do with how we set up RTP traffic.
I.e. Asterisk WILL NOT issue Re-INVITE even if:
1. Both UAs have canreinvite=yes in their
Vahan Yerkanian wrote:
What is the proper way of adding hints to multiple extensions?
In my case extensions are the same as the sip usernames, while as per
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
exten = 1234,hint,SIP/1234 works,
exten =
Brian Capouch wrote:
I am wondering if there are any tricks getting the Cisco ATAs to do
distinctive rings via the ALERT_INFO variable?
I have seen some contradictory information in the Wiki, and I tried the
example there. I then sniffed the connection between the server and the
ATA and
Steve Gladden wrote:
You also want to look at the registertimeout and registerattempts
Yes!!!, thank you VERY much this is what I needed.
Where are these options documented at?
I'm guessing the source code?
Or is there a better place to find this stuff?
A search on the wiki for
Matt wrote:
Am I correct in thinking that at this time the CVS-HEAD supports
Jitter Buffer for SIP on Asterisk?
No, you are incorrect.
/o
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Tim Connolly wrote:
Asterisk cvs-head (up to date) keeps core dumping on me. I finally
tracked it down to my register command for Vonage in the sip.conf file. If I
remove the username and password from the register command, it won't core
dump, but of course won't register either... This
Mike Hansford wrote:
If Asterisk is not able to function as a SIP proxy, how do I re-write
and/or route messages? Can Asterisk fake these processes or will I
require a proxy like SER to do it for me?
Please read
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20not-proxy
/Olle
harry gaillac wrote:
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
THat is a response to the polycom's PUBLISH request, a
harry gaillac wrote:
Hello,
I patched asterisk cvs head sources with
http://juraj.bednar.sk/work/software/asterisk/messaging/
and presnce patch without success.
asterisk send 405 method not allowed to sender.
I use polycom ip300.
THat is a response to the polycom's PUBLISH request, a
Timur V. Elzhov wrote:
So I definitely misunderstand something in Asterisk SIP channel
engine :-/ Where I'm wrong?
You are wrong in not reading the available sample configurations and
configuration files. Read the sip.conf that is installed when you
install with make samples and check
Damon Estep wrote:
When executing: Dial (SIP/[EMAIL PROTECTED],60
mailto:SIP/[EMAIL PROTECTED],60) I get about 15 seconds of
ringing, the called party rings, but if not answered in the ~15 seconds
I get back SIP 480 temporarily unavailable.
If the call is answered everything is fine
Gulzar Hussain wrote:
Hi
how to disable call waiting on SIP User agents
Configure it on the SIP user agent!
(incominglimit=1 is Deprecated , End of life already
announced no idea how to use setgroup to achieve same
functionality)
We will have to change that. Incominglimit has an
Nicolas Schmerber wrote:
[EMAIL PROTECTED] a écrit :
On Thu, 11 Aug 2005, Nicolas Schmerber wrote:
All the features I need work just not one : the supervised call
transfers. I know there are a lot of posts about that, but none gave
me the correct answer (unless I missed it).
You'll
Tomáš Komárek wrote:
Hello,
I've got such a problem. I'm configuring Asterisk as a backup server, if
call to the first one fails.
My problem is, that the redirection from the sending machine work so,
that in the INVITE line of the INVITE message is the presentation number
of the Asterisk
Tomáš Komárek wrote:
Well,
that is great, but I'm not a good programmer, so I would need some
furher details. Probably I will need to edit the file chan_sip.c and
then recompile Asterisk.
Is it true??
No, it's a dialplan function in CVS head. You do not need to program
anything. CVS head
Christian Stredicke wrote:
It would be nice if the PBX can acknowlegdge the Record header - then it
would have the chance to paint a record icon on the screen.
In the next release.-)
Right.
Is there another header for turning off recording?
Anyway, we should not send unsupported media
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