Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf,
sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to
call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using
Chris Bagnall schrieb:
- No shared adress book (especially it should be shared between phone on
different base stations). I can access an online adress book, but only
the built in, and you cannot set up your own online book.
You can send address books to the phone in standard vcard format
- Original Message
From: Olivier [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 6 February, 2008 7:35:10 AM
Subject: Re: [asterisk-users] External MWI question for Asterisk
Do you send those
Femi wrote:
Does anyone have data on the switching capacity of Asterisk based on the
hardware?
I need to know what type of hardware would be required to switch 100
simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP
VoIP calls
That largely depends on whether you
Hi,
Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk?
Please let me know how it performed and what issues you faced
Thanks
Femi
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Hi,
Does anyone have data on the switching capacity of Asterisk based on the
hardware?
I need to know what type of hardware would be required to switch 100
simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP
VoIP calls
Thanks
Femi
Hi,
I have a queue with one agent added using AddQueueMember
(FAO|Local/[EMAIL PROTECTED]|0||Agent/602). My extensions.conf is
[general]
static=yes
writeprotect=yes
autofallthrough=no
clearglobalvars=no
priorityjumping=no
[from-sip]
exten = 11000,1,Dial(SIP/11000,,t)
exten =
On 8 Feb 2008, at 00:39, David Hogan wrote:
Alternatively you could fix the client :-)
Heh :) Although it's a situation that won't happen in (our)
production,
for the sake of completeness I'll probably upgrade the client.
Actually, it does (assuming you guys still run Tesco's UK
This will be closed service provider network with own VoIP phones and
gateways so we can assume that there is no transcoding
Femi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
Kempgen
Sent: 08 February 2008 12:15
To: Asterisk Users
Subject:
Cosini SS7 work for 99% of all cases.
Femi ha scritto:
Hi,
Has anyone on this list used the Cosini iAN7s SS7 gateway with Asterisk?
Please let me know how it performed and what issues you faced
Thanks
Femi
___
-- Bandwidth and Colocation
That's very unfortunate.
I use now a workaround : I'm just switching (with gotos) between
extensions and use some macros but always within the same context.
I'll try to remeber it for next time :)
Cheers,
Yves.
On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote:
On 2/8/08, Yves Räber
Hi list,
can anyone tell me how problematic it is setting canreinvite=yes ? I know if
its to avoid issues with bad implementatins of
SIP on other devices then maybe you cant give a black and white answer, but any
constructive comments welcome!
Reason being I think I have to set this to yes to
On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote:
That's very unfortunate.
I use now a workaround : I'm just switching (with gotos) between
extensions and use some macros but always within the same context.
Well, you should create contexts for your main features, and you can
write few of them in
Vytenis,
As far as I understand it 1.2 zaptel should be used with 1.2 Asterisk,
1.4 with 1.4.
As for 1.2 vs 1.4, it depends on if you want new features and any
bug-fixes. 1.2 is a closed project (I think).
Just compile from source if its not available as an RPM in 1.4 for
Debian.
Adrian
Hello,
I wrote a small AJAX phonebook targeted at Grandstream phones although
the basic functionality doesn't require a SIP-phone.
Asterisk integration (call history, incoming call info, click-to-dial)
is not yet implemented, but on my ToDo list.
Features
* Licenced under the GPL v2
*
We are building a Ruby on Rails interface that we will probably put out
as open source later this spring. I worked around this problem with
#exec statements. This is what is in my extensions.conf file:
#exec /path/to/scripts/load_extensions.rb
This runs a Ruby script that rips through my
Hi,
I am about to purchase an Eurotech PRI E1 GSM CDMA Gateway to operate
with my Asterisk's TE412P interface.
Anyone here has any experience of having this combination? Any success
or failure stories would be greatly appreciated.
Thanks in advance.
Ash
Greetings,
I've set up the AMI and am able to authenticate, however I am unable to
execute Action: Status. I get a permission denied error:
asterisk:/etc/asterisk# telnet localhost 5038
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call
Steve Shepherd wrote:
Greetings,
I've set up the AMI and am able to authenticate, however I am unable to
execute Action: Status. I get a permission denied error:
*snipped
read = system,call,log,verbose,command,agent,user
write = none
without the ability to 'write' a command, you
Don't forget to 1000,1,Answer the call
Moj
John Von Essen wrote:
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last
On Fri, Feb 08, 2008 at 05:12:33PM +0200, Vytenis Sabaliauskas wrote:
Hello,
I would like to consulate with you guys. I'm setting up an Asterisk
server on Debian. The problem is that Rhino drivers are only compatible
with Zaptel 1.2.
Thats seems odd to me. Are you sure? I
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
Soumya Kat wrote:
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rhino Drivers are agnostic to the version of zaptel you are using with
1small exception. You can build any of our drivers against zaptel up
to 1.4.7 without any patching or fancy foot work. You can guild our
2.2.3beta2 and when released the 2.2.3
Hi,
I have installed Asterisk 1.4 along with net-SNMP 5.4.1 in my Fedora 8
system. Asterisk works fine for me and I can log into Asterisk-GUI and
monitor asterisk.
What I would like to know is how to get information such as SIP users,
number of SIP connections and traffic associated with those
After Andrew's suggestion, if that isn't the problem, spend some more
time on OSLEC to be darn sure it's operating properly -- that thing
works like a champ for my crappy lines!
Moj
Brent Davidson wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an
Thanks!
Femi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: 08 February 2008 15:20
To: asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Scalability
Hi!
Does anyone have data on
Hello,
I would like to consulate with you guys. I'm setting up an Asterisk
server on Debian. The problem is that Rhino drivers are only compatible
with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel
1.2, and that suits our needs. Is there a bleeding need to use
Lol - so I read about this website today - www.dealipedia.com
http://www.dealipedia.com/
And I thought cool, lets start typing in a few names of companies I know
who have taken funding recently.
Check out the username of the person who submitted the Fonality deal -
Hi,
I use outgoing URI-dialing for my sip-phones as suggested in
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
The relevant extensions look like this:
[dial-uri]
exten = _[a-z].,1,Macro(uridial,[EMAIL PROTECTED])
exten = _[A-Z].,1,Macro(uridial,[EMAIL PROTECTED])
exten =
I'm using FreePBX/Trixbox with Asterisk 1.4.17-1 trying to register
against a Sylantro server in front of a Metaswitch. I'm able to register
and receive inbound calls but outbound calls are rejected by the far
end. The username and password have been checked repeatedly. Putting the
same
I believe trixbox can fulfill your requierements.
regards
- Mensaje original
De: Paul Hales [EMAIL PROTECTED]
Para: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Enviado: jueves 7 de febrero de 2008, 22:45:00
Asunto: Re: [asterisk-users] Asking
Don't worry - I paste this leink becaus eyou should have e good
understanding about what the queue() cmd does to be safe in implementation
phase: http://www.voip-info.org/wiki-Asterisk+cmd+Queue
See also: http://astrecipes.net/index.php?n=118
Thanks
l.
On Tue, 05 Feb 2008 06:31:16 +0100,
On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote:
* Version: Asterisk 1.4.14
* Commas instead of pipes = already tried, this is not working at all
* Realtime switch for script_13_0 = No, should I ? This would be really
a shame, I want to use realtime BECAUSE I don't want to play with my
This is standard stuff.
I have switch over 200 simultaneous with g711 on a 1-U, Xeon-DualCore @ 3.0
using RH versions of Linux. Even higher with pass thru (no-transcoding)
on g729.
..mike..
At 07:54 AM 2/8/2008, Femi wrote:
This will be closed service provider network with own VoIP phones and
2008/2/8, Tobias Wolf [EMAIL PROTECTED]:
Chris Bagnall schrieb:
- No shared adress book (especially it should be shared between phone
on
different base stations). I can access an online adress book, but only
the built in, and you cannot set up your own online book.
You can send
Hi!
Does anyone have data on the switching capacity of Asterisk based on the
hardware?
I need to know what type of hardware would be required to switch 100
simultaneous calls as opposed to 1000 or 1 calls, no TDM just SIP to SIP
VoIP calls
Use the Wiki, Luke!
Hi All,
I'm going to be upgrading our 1.2 Asterisk system. At the moment we use
the Enicomms SLN files. Are there major differences in the 1.4 default
voicefile packs, or will I be able to re-use Enicomms??
In the Make menuselect, I noticed theres no .SLN voicefile selection for
the basic
Jaap Winius wrote:
* Why can't I delete any voicemail messages?
(Response: Message undeleted.)
* Why can't I listen to the messages in the Old folder?
* Why can't I use the advanced options?
(Response: I'm sorry, I did not understand your response.)
* How
As far back as I can remember in 1.4, the option of sending a VM from
voicemailmain (3-5 or 3-5-1), depending if you could use the directory has
been broken. In the ChangeLog for 1.4.18 a bug (11735) was mentioned. I
do seem to remember that in 1.2, it wasn't possible to send a message to
Hello Everyone,
I am currently operating a VoIP business in Canada
and joined the list. I have seen some very useful ideas and information
posted daily in this forum. I have also noticed that there are user who
barter, sell, trade services, products, etc. Wonderful
On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW +1-303-722-7209
wrote:
As far back as I can remember in 1.4, the option of sending a VM from
voicemailmain (3-5 or 3-5-1), depending if you could use the directory has
been broken.
So does this work if you use the directory, if you
Hi,
I have the Cisco PIX 506 firewall right in front of the asterisk and I am
getting a one-way audio. I need your help/guidance to resolve this problem.
I have the fixups disabled for SIP in the Cisco PIX 506. Any help
rendered by you in this subject is greatly appreciated. I have been
Ravi,
Open up the RTP (UDP) ports on your pix. (EX. conduit permit udp host
x.x.x.x eq 10049 any). Also set your asterisk rtp config span to
something you can configure (1 to 10200) unless you write a script
to just copy and paste about 1 to 2 ports in your config on the
pix.
Otis,
I am new to Cisco PIX 506 and I am learning this. If you can help me with
how to do this change on Cisco PIX it would be greatly appreciated.
Thx
Ravi
-Original Message-
From: ListAcct [mailto:[EMAIL PROTECTED]
Sent: Friday, February 08, 2008 11:11 PM
To: [EMAIL PROTECTED];
On Fri, 8 Feb 2008, Tilghman Lesher wrote:
On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW +1-303-722-7209
wrote:
As far back as I can remember in 1.4, the option of sending a VM from
voicemailmain (3-5 or 3-5-1), depending if you could use the directory has
been broken.
So
Ravi,
I will explain changing the config in asterisk and the pix:
Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to
1 to 10050 (to start, you will need to increase later as ports fill up)
(use insert to make a change in a file)
to save:
1. esc
2. shift + colon
LOL I guess all I was asking for the changes to be made in the Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks once again
for this help. Let me work on these changes and test the one-way audio
problem and go from there.
Thx
Ravi
-Original Message-
From: ListAcct
No problem. :-P I thought it might wise to include everything you
needed just in case!! LOL! You are welcome!!!
--Otis
Ravichandran Rajagopal wrote:
LOL I guess all I was asking for the changes to be made in the Cisco PIX
506. I think you gave me a short tutorial on VI as well. Thanks once
On Friday 08 February 2008 23:28:01 William F. Acker WB2FLW +1-303-722-7209
wrote:
On Fri, 8 Feb 2008, Tilghman Lesher wrote:
On Friday 08 February 2008 18:21:20 William F. Acker WB2FLW
+1-303-722-7209
wrote:
As far back as I can remember in 1.4, the option of sending a VM from
Otis,
I wanted to clarify what you said and what I comprehended.
the SIP protocols are disabled in fixup.
Having said that I guess all I have to do is just the following.
the inside IP of asterisk server is 192.168.5.0
On the cisco PIX
2008/2/8, Grey Man [EMAIL PROTECTED]:
- Original Message
From: Olivier [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, 6 February, 2008 7:35:10 AM
Subject: Re: [asterisk-users] External MWI
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:[EMAIL PROTECTED]);
User on sip server (192.168.2.81):
[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
Note also that if you point to the DNS name rather than the IP address of the
asterisk server on the phones trying to register, you can set NAT=NO on the
asterisk side and the sip FIXUP command on the PIX will handle everything
correctly making this workaround unnecessary
- Original
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