Patrick Friedel wrote:
Sorry if this is an obvious question, but I haven't seen an obvious
answer on the wiki that I remember. Has anyone managed to make the
record button on the snom 360 fire off the Monitor() application? I
don't see a bounty, and googling for snom 360 record button
Tim Karl wrote:
I don't know if this will help, but the checksum is part of the UDP
header which should be computed by the sender prior to the data being
sent. It is computed using the data that is to be sent. UDP at the
sender side performs the one's complement of the sum of all the 16-bit
In response to a large number of questions on the mailing list I've
decided to publish a presentation I have been running in the Asterisk
bootcamp - our one-week training class.
This presentation covers many, but does not claim to cover ALL, new
features of Asterisk version 1.2. I hope it will
Please file a bug report with a full SIP DEBUG output file. Set debug to
4, verbosity to 4 and turn on SIP debugging. Upload that file as an
attachment to the bug report and place the bug report in the SIP category.
Thanks!
/O
___
Asterisk-Users
Hall, Eric M. wrote:
Not sure what this is.
When I call my own ext the call will ring for 10 sec and goto the
voicemail. However the phone will keep ringing and I see this on the
asterisk CLI
Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host
'192.168.0.200' does not
When you write code for Asterisk, you are in the middle of a piece of
code and you add debug, log and console messages that you need yourself
to figure out whether the patch works or not. As a user, some of these
messages may be hard to understand, especially since a many of them look
like Ouch,
Kevin P. Fleming wrote:
Eric Wieling aka ManxPower wrote:
Thank you for the correction. Is this new to CVS-HEAD, or does it
apply to 1.0.x as well? 1.0.x sip.conf.sample only lists the host
option for a peer, not for a user.
It only applies to 'peer' entries, but the important point
TWV wrote:
Where can I find the documentation or an overview of everything that is new
in Asterisk 1.2 ?
There's no good documentation on that out there... yet.
Read all the sample configuration files, the READMEs and especially the
updating file.
Any documentation people with time to write
Dear Asterisk Community,
Asterisk 1.0 was released at Astricon 2004, in September last year. It's
been almost a year and we haven't been able to go ahead and release a
new version. Now is the time to try to move forward again.
As we've outlined before, the process is this:
[EMAIL PROTECTED] wrote:
Hello fellow asterisk people!
I have Asterisk listening on port 5061 and SER on port 5060.
Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.
My problems are with SIP. I can make incoming calls from SIP to asterisk
and to any of the other
Juraj Bednar wrote:
Hello,
There's some work on creating a multiprotocol solution for instant
messaging within Asterisk, but it will not be in the coming v1.2.
is the work somewhere as a patch to be tried or in some other form,
even if it's not coming to 1.2?
No, there needs to be
Ronald_Wiplinger wrote:
Is there a possibility to send text based messages from/to a sip phone
(text display) or to a video phone or from/to a messenger?
Yes, there is in SIP if the SIP user agents support it. But no, Asterisk
will not forward the SIP messages between the SIP user agents.
Andrew Kohlsmith wrote:
On Wednesday 20 July 2005 20:15, Eric Wieling aka ManxPower wrote:
As I understand it, adding VAD/Silence would require redesigning the
entire RTP stack of Asterisk.
My understanding is that with the new jitter buffer both of these things are
completely doable now
Matthew Boehm wrote:
Olle E. Johansson wrote:
...when the new jitterbuffer is included and if it's enabled...
Please help us test the SIP/RTP jitterbuffer!
It's available in the bug tracker!
/Olle
post 'da bug number
http://bugs.digium.com/view.php?id=3854
Download the patch
Chris A. Icide wrote:
Olle E. Johansson wrote:
Chris A. Icide wrote:
Note however, that in the unsolicited NOTIFY that Asterisk sends for
MWI, it includes a ;tag= as part of the NOTIFY. This will break some
devices as they will not accept the NOTIFY because the tag doesn't match
any
Brian West wrote:
Kristian Kielhofner, the lead developer of the AstLinux project, will
be speaking at ClueCon. His latest AstLinux Version 0.2.6 is a complete
Kristian will also be speaking at Astricon 2005 in California
http://www.astricon.net/2005/
/O
As we are getting closer to release of CVS head as version 1.2, we're
in need of your help.
One of the cool new features in CVS head is the ability to store the
actual voicemail messages in a database. This is not using ARA, the
Asterisk Realtime Architecture, but directly interfaces with ODBC
Paradise Dove wrote:
hi all,
is there any way to force all sip peers to re-authenticate themselves?
Turn off the power. Then turn on again :-)
No, there's not. I think I saw a SIP extension package that allowed the
SIP registrar to tell clients that the registration was forced to
unregister,
Juraj Bednar wrote:
Hello,
I found, that in CVS Head, in chan_sip.c, there's some support of
asterisk. I have one question -- how does it map extensions to sip
user names? When my client subscribes to other extensions' presence,
they see all users online, but it may be because of
Chris A. Icide wrote:
Note however, that in the unsolicited NOTIFY that Asterisk sends for
MWI, it includes a ;tag= as part of the NOTIFY. This will break some
devices as they will not accept the NOTIFY because the tag doesn't match
any transaction that is open. The correct way to send an
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.
The next community meeting is
This function is based on a non-standardized extension to SIP made
by Broadsoft. I have all the specs and are looking into this. Don't
expect anything to happen quickly though, I have to complete another
large SIP project first (SIP Transfers) and then start looking into
this. It requires quite a
Steve Blair wrote:
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Do you know which standard they base this on?
/O
___
Asterisk-Users mailing list
Astricon 2005 will take place in Anaheim, California October 12-14 2005.
Astricon is the Asterisk conference, arranged by IPsando LLC in
cooperation with Digium.
We are now looking for speakers. The conference will be bigger than last
year, so we are looking for more speakers in the conference
Joel Jn-Francois wrote:
Hi Guys,
I am using the latest stable version of asterisk which I updated
yesterday, but I keep getting this error message for some of my
accounts. Can anyone explain to me what does this mean?
WARNING[3032]: chan_sip.c:4832 check_auth: Stale nonce received from.
Dave Walker wrote:
Hi,
What could cause:
Got SIP response 406 Not Acceptable back from 10.0.0.10
10.0.0.10 = Hardware FXS
It means you have to reconfigure the device, I guess.
/O
---
Astricon 2005 - Anaheim, California, Oct 12-14 2005
http://www.astricon.net/2005/
Liu Peter wrote:
Can asterisk transfer a call with sdp passthourgh?
I don't want to asterisk to do rtp relay and only want it to process
the signal messages.
thanks.
Then do not use Asterisk. Install a SIP proxy instead!
/O
___
Asterisk-Users
Stojan Sljivic - GDS wrote:
Hi,
I have set autocreatepeer=no and it behaves just the same.
It seems that the default value is no, or Asterisk does not understand
this property.
In which version of Asterisk was this property introduced? I use 1.0.5.
autocreatepeer is off by default and
Stojan,
You have to check what extensions you have enabled in the context
specified in the sip.conf [general] section. All of those will be
reachable without authentication by anyone.
If there's no context= setting, Asterisk defaults to the context
named [default] in your dialplan. Any extensions
etc. However, I missed the initial part of this thread. Why is this an
issue? I went to voip-info.org today just to see what was going on
It is not really an issue at all. The thread started due to scheduled
maintenance of the server, which scared a lot of Asterisk users. The
wiki is safely
We're getting close to Astricon Europe 2005, the first Asterisk
Community gathering in Europe. Speakers are coming in from all over the
US and Europe, as well as far away as New Zealand, to talk, teach and
discuss Asterisk -the Open Source PBX.
At this time, we're still accepting registrations
James H. Thompson wrote:
Voip-info is back up -- in-spite of Murphy's law.
This was phase I (install latest version of O/S) of an upgrade to
improve performance and functionality.
Hopefully with Phase II we will see much better performance and new
functions.
For those that asked, the
sylvain garcia wrote:
Hi,
I have install asterisk and it works fine.
But when I use Sjphone and I use Ethereal a Client send Request:OPTIONS
sip:obelix.foo and Server answer Status: 404 Not found.
But i can talk with two client and asterisk.
When I use Xlite i don't have this request
Chris Coulthurst wrote:
If you activate (via sip.cfg) the feature Group Call Pickup, its no
surprise that asterisk doesn't know what to do with this feature
request. But it is being sent as a SIP SUBSCRIBE request, and I'm
wondering if, as asterisk stands, there is a way to take advantage of
Joshua Colp wrote:
You're actually confusing me when you say this due to the fact you're not
giving much information, probably why nobody has responded yet. If the SIP
server on the Nortel does an INVITE for the phone number, then asterisk will
act accordingly and go to the phone number in the
Joshua Colp wrote:
Okay lemme give you something that should work some magic!
Stuff for sip.conf:
[nortel]
type=peer
host=IP ADDRESS OF NORTEL
disallow=all
allow=ulaw
context=inbound_nortel
insecure=very
Stuff for extensions.conf:
[inbound_nortel]
exten =
Nabeel Jafferali wrote:
I have a sip.conf entry for a customer's PBX (IP based authentication) that
reads:
[customer]
type=friend
context=customer
host=x.x.x.x
accountcode=1
disallow=all
allow=g729
When the customer makes a call to my * server, * recognizes the peer
In the development tree of Asterisk we've changed two things lately that
may affect you:
* Asterisk can fail an outbound registration
If you enter a register= statement with an incorrect password, wrong
hostname or anything else that is wrong, Asterisk will give up
registration after 10
Nir Simionovich wrote:
Hi All,
I'm trying to connect to a SIP carrier who never connected with Asterisk.
I managed to connect with a sipura phone or a grandstream, no problem.
When I configure asterisk, I'm able to send out calls to the carrier
no problems,
however, receiving calls
I believe there is a patch for this problem in the bug tracker, it was
recently fixed in CVS head.
The problem is that the registering device is sending proper auth, but
with an old challenge (nonce). Asterisk incorrectly answers wrong
password where the situation is more wrong challenge but
Ed Greenberg wrote:
--On Saturday, June 04, 2005 10:41 PM -0700 Chris Coulthurst
[EMAIL PROTECTED] wrote:
I have a Polycom 500 and would love one of the line-appearance keys to
show me if a certain person/people are on the phone upstairs. This
'hint' priority seems to have little-to-no
Mirko Marghitola wrote:
Asterisk don't send the 180 Ringing SIP message to the calling phone
when the called party is ringing. How can I force asterisk to send the
ringing messages? The option 'r' in the Dial() command or the Ringing()
command didn't solve the problem.
I think you have to
Ross Kevlin wrote:
the subscription is sent a reply and the reply has content that indicates
its state is terminated
From: sip:117 at 192.168.2.252;user=phone;tag=as77c7b911
To: sip:83 at 192.168.2.252;tag=z6kvtd67bu
Contact: sip:117 at 192.168.2.252
Call-ID: 3c2670ad35b6-68nuemr6pg58
Bjorn wrote:
I guess the simple solution for the problem below would be if there was a
way through the management interface to establish a call between two
extensions defined in the dialplan, and not an extension and a specified
channel. If anyone knows how, I'd appreciate the feedback.
[EMAIL PROTECTED] wrote:
Hi all,
I've programmed on a SNOM function key the destination **77
which in my extension plan reprograms/toggles an asterisk DB variable
which I use in the extension plan for some call routing decisions.
I would like the SNOM extension light to permanently
[EMAIL PROTECTED] wrote:
Hi all!
So far I've always used plaintext passwords for SIP, but now I've decided
to use MD5 encryption.
For each client I edited its section as follows, then:
auth=md5
md5secret=hashed_passwd
;secret=plaintext_passwd
where hashed_passwd is the output of
Joseph wrote:
I notice these notices running cvs head:
NOTICE[6154] chan_sip.c: No field 'From' present to copy
Any ideal what make this notice?
You need to turn on SIP debugging for us to be able to help you.
I suspect you have a buggy device.
/Olle
Astricon - the Asterisk
Joseph wrote:
On Fri, 2005-06-03 at 19:35 +0200, Olle E. Johansson wrote:
Joseph wrote:
I notice these notices running cvs head:
NOTICE[6154] chan_sip.c: No field 'From' present to copy
Any ideal what make this notice?
You need to turn on SIP debugging for us to be able to help you.
I
Ross Kevlin wrote:
I have a snom 360 that im trying to get the extension lights working i can
see the subscription being sent and a reply but the reply is a terminate.
Terminate being?
/O
___
Asterisk-Users mailing list
Jeff Heath wrote:
I checked out CVS HEAD today and tried to compile it with no luck, so
then I checked out the stable version and compiled it successfully. I'm
99% sure that I'm not missing anything and that I'm following the
instructions correctly (I'm no guru, but I've compiled lots of
Sandeep A.S wrote:
For bridging VOIP with PSTN Lines
Which one is giving better performance SIP or IAX ?
I am looking at a result without NAT in picture ?
Can some body give details from experiance ?
I can't say anything based on experience, but guessing that IAX2
trunking will
Betl Gzlkolu wrote:
Hi ;
Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions) to
handytone line port directly and
vice versa ?...
The answer to the same question a few days ago was,
At this point we have close to 200 registrations for Astricon Madrid and
new attendees are registering all the time. Make sure you register now
in order to get a seat at this first community meeting in Europe,
arranged by IPsando and Digium - the company behind Asterisk.
The detailed tutorial
Lance Grover wrote:
I have been working with this for a wile and I have been watching
the list for about a month on this subject, to no avail.
I am wondering if anyone has successfully configured asterisk for
clients to connect to it when the clients are behind nat. I mean
successfully
Lance Grover wrote:
On 6/2/05, Wiley Siler [EMAIL PROTECTED] wrote:
Have you configured your sip.conf to use these aprameters under General?
;externip=66.213.227.66
;localnet=192.168.1.0
;localmask=255.255.255.0
If Asterisk is on Internet and the phones is on nat, these do not
help at all.
Ross Kevlin wrote:
I recently got a SNOM 360 and have been trying to get the extension
lights to work. I can see the subscriptions with sip show subscriptions
but I don't see any notifies when a call is made. I must be missing
something because I've tried looking to see if anyone else has had
Ronald Wiplinger wrote:
One of our remote user's phone reports frequently:
Got SIP response 481 Call Leg/Transaction Does Not Exist back from IP
What can I do ???
Turn on SIP debug, set verbose to 4, debug level to 4 and trace what
happens. If we can't see that, an error message out of
Tim P wrote:
I have multiple Sipura ATA 2100s attached to normal analog phones that
are all configured as extensions in *
When I call an extension it rings and will go to voicemail if no one answers
it.
When I call the same extension a second time after no answer (went to
VM) the phone
Manjit Riat wrote:
Hi,
I prevoiusly has asterisk on a public static ip and had a phone from
a different location registering to the asterisk box. But now we have
dropped the previous connection and the current connection has a
dynamic ip. Is there any way for the phone to register to
Betl Gzlkolu wrote:
Hi ;
Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions) to
handytone line port directly and
vice versa ?...
On my 486 I can't dial out on the FXO port, it's just a
asterisk asterisk wrote:
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these
extesion can call each other, and ext 300 can call outside to pstn, and
ext 301 to call internatonal.
How can I do that ?
You read the samples and the guides
I think good practise is
* Not confuse [peer] names in sip.conf with extension numbers
(as you have learned). Extensions will match several peers
some times, and when using trunks, it will cause problems.
It is also much easier if the peer is a name and an extension
is a number. (But I
Mohamed A. Gombolaty wrote:
Dear All,
I am totally new in this arena and I am still waiting for my
installation process on freebsd to finish, but I wanted to make sure of
the following:
- Call routing between IP telephones can be done regardless of who made
the phones?
As long as you
Sean Kennedy wrote:
Hi all,
I'm trying to see if I can get the hint priority working with my polycom
500.
So far I have 2 /reg entries with the same sip registration, one is
labeled as private, the other as shared. I have set the hint priority
before anything else in my dialplan for my
Waldo Rubinstein wrote:
Hi all,
Assuming 1.0.7 is the latest stable version, how/where can I find out
the different CVS revisions available and a description of what has
been patched/updated in each CVS revision so I can decide whether to
leave my 1.0.7 installation as is, or if I need
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.
The next community meeting is
Personally I would rather see realtime load everything into memory and
not go to the database unless something has changed or you reload.
Then maybe add a new manager command like the following:
Then don't use realtime extensions, use the static database functions in
realtime to load your
, Digium: Extending Asterisk: The AMI and the AGI
* Serge Kruppa: Building a carrier class hosted contact
center platform with Asterisk
* James Jones, Signate: Res_Perl: Perl embedded in your Asterisk!
* Kindy Conley: The basics of telephony billing
* Olle E. Johansson
Personally, I'd like to see this changed so there are two 'general'
sections--one for default parameters to use unless overridden when there
*is* a peer section below, and a different one to describe parameters to
use when the remote peer is not previously known. I know there are ways
to
Irakli Natsvlishvili wrote:
100k question - does asterisk correctly handle following situations:
There are plenty of good documents on Asterisk, SIP and NAT on the
voip-info.org wiki. Please look them up. There are also information
within the configs/sip.conf.sample file within Asterisk.
1.
Elmar Haneke wrote:
Hi,
how can I determine the status (busy, offline, ringing, duration of
current call) of an SIP phone?
Remember that the SIP phone is a kingdom of it's own. Right now,
Asterisk does not really now anything about what is happening out there
in the SIP woods. We know about our
Ronald Wiplinger wrote:
I would like to get a notice by email, if we run out of gateways!
exten = _9011Z.,410,Busy
exten = _9011Z.,411,EMAIL = How to?
-= Info about application 'System' =-
[Synopsis]:
Execute a system command
[Description]:
System(command): Executes a command
administrator tootai wrote:
Nabeel Jafferali a écrit :
Does anyone else have this problem? Is there a workaround?
Yeah, I had this problem when I added a lot of SIP register statements
and SIP peers. Changing the hostnames (FQDNs) to IP addresses solved the
problem. It seem * was getting stuck
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.
Our community is also growing
Rich Adamson wrote:
Cross posted on purpose
FYI, just upgraded from cvs-head from March 23 to this morning (March 31).
All compiles and installs completed normal.
Loading asterisk via safe_asterisk (or asterisk -cdvvv) fails with the
standard oche... message. Piped the output to a text file
During the developer's conference call yesterday evening,
it was decided that we finally should release the much-awaited
Asterisk 2.0 Stable release, also called codename AAFJ.
This relaese is based on the hidden cvs that has been in
operation for six months by a group of core development members
These OPTIONs packets are what we send for qualification, a scheme that
can also be used for NAT keepalives. You turn them on by adding
qualify=yes in the [peer] section of sip.conf
With qualification on, we regurlarly measure the latency between
Asterisk and the client and decide whether the
[EMAIL PROTECTED] wrote:
hi,
Is asterisk a registrar server.
It all depends. If you mean registrar for Inter-Galaxy Travel
Permissions, no. If you mean SIP registrar, yes.
But we are not a SIP proxy ;-)
/O
___
Asterisk-Users mailing list
Klaus Peras wrote:
Hey there,
does anybody know a CLI SIP Client für Linux?
I think you may find one in Vovida.org
/O
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE
Atif Rasheed wrote:
HI all,
I have the following setup running:
EP---Calling Asterisk---Relaying Asterisk---Softswitch--- PSTN
The Endpoint EP is registered with the Calling Asterisk. Calls are
forwarded from this machine to
Relaying Asterisk which in turn forwards it to the Softswitch. In
In the Grandstream setup, turn off subscribe to message waiting
indication.
...or upgrade to CVS head, where I've fixed this problem with SUBSCRIBE.
Best regards,
/Olle
___
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Steve Kann has developed a new jitterbuffer for IAX2, that hopefully
will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable
relase.
Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs
support in the form of funding in order to take the time to test this
Steve Underwood wrote:
re here: http://www.astertest.com/forum/viewtopic.php?t=13
Thank you for your contribution!
The hard work of building the thing was done for free, and now someone
brings out the begging bowl for the relatively minor activity or porting
into to another home. Frankly, that
I've added an introduction article about the ARA on my web site
http://www.voip-forum.com/
The same text is now also added to CVS head as README.realtime.
On the same site, you will also find the news item about how we used
Asterisk for a call from an airline jet above Greenland to Stockholm,
Sarat Vemuri wrote:
While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the
following. I limited the RTP ports from 8000-8050 to limit holes in
firewall. Pretty soon Asterisk ran out of RTP ports. Traced the
problem back to how * is handling SUBSCRIBE. A sip structure is
Anton Krall wrote:
I think it would be your last suggestion.. When I pickup the phone I hear a
tone, the sip phone box tone... Then I hit 9, no tones :) and enter the
whole phone number and it starts to ring on the other side.. So no outside
dialtone get heard ever.. I was wondering if it could be
Stig Andersson wrote:
So, I try
-
SetVar(cid=${CALLERIDNUM:-5:5})
The result is a empty string if CALLERIDNUM is less than 5 digits long,
which is NOT the case of SubString. SubString command returns what remains of
the variable,
that is - if CALLERIDNUM is 4 digits in length, it
During the last week, we have had several support issues being reported
as bugs on the bug tracker. Since we are going into a final development
stage on version 1.1dev (CVS HEAD) in order to complete the 1.2 release
we are under pressure to fix bugs and handle a lot of reports in a short
time
Asterisk wrote:
I've got a test * server (hppbx) where I install CVS-HEAD as often as
possible, with my extension registered to this, talking through IAX to
our production server which then channels out to the PSTN.
After completing a call just now, the following appeared on the CLI of
hppbx
Peter Svensson wrote:
On Wed, 16 Feb 2005, Rob Scott wrote:
Why is it that Asterisk can't cope with silence suppression?
All the clients seem to be able to but not Asterisk.
What would be needed to get it to work with silence suppression?
What is the problem?
Asterisk clocks outgoing rtp data to
Chris St Denis wrote:
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
In the 1.0 stable release, you can not send MWI for database peers.
In CVS head, the base for the future
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.
Our community is also growing
Marcello Lupo wrote:
Hi,
notice that i have Grandstream phones and i have the problem if i activate the
Send Anonymous function on them.
If i do not activate that option the ACCOUNTCODE is correctly populated. SO i
think it may be a bug of asterisk.
I'm using Asterisk CVS-HEAD-10/07/04-18:07:25
Kevin P. Fleming wrote:
I have a patch in my local system that allows the canreinvite setting
(which I renamed) to actually be based on IP address masking, so that
Asterisk can make a more intelligent decision, but even that has
problems, because we don't actually _know_ that any given IP route
Spencer Nassar wrote:
Does anyone know if the tutorial materials from Atricon 2004 are
available for download anywhere? I'm particularly interested in Joachim
Vanheuverzwijn's Performance and Scalability tutorial slides (Asterisk -
building your system for performance and scalability).
I'm sitting in a hotel close to the Madrid airport... Any Asterisk users
in the neighbourhood that wants to meet me for a beer and some Asterisk
hacking this evening?
Send e-mail to me *off list*, thank you.
/O
___
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Vlasis Hatzistavrou wrote:
Hello,
We hve been trying to make Asterisk work with SIP proxies with no success.
Is there support for SIP proxies in Asterisk in the latest versions?
A lot of people use Asterisk with SIP proxys.
What is your problem, give us a bit more information.
/Olle
Robert Spielmann wrote:
Hi everyone,
I have a question concerning DNS SRV lookups. The situation is like this:
- one central Asterisk server
- many domains with SRV records, let's say we have bar.com and doe.com
Now the question is: if the SRV lookup is done for [EMAIL PROTECTED] the call is
Andrew Thompson wrote:
I am about to start a program that will be generaging sip device
configurations for sip.conf. My current sip.conf contains friend entries
for each SIP device connected to asterisk.
Should I even be attempting to split these in to seperate user/peer
devices?
Can two
Geoff Speicher wrote:
Sipura has implemented auto-answer in version 0.9.5 of the SPA-841
firmware. However, it is implemented via the Call-Info header, which
Asterisk stable doesn't currently support.
The attached patch implments a quick hack to support the Call-Info
header from the Dial()
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