Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
On Sat, 18 Aug 2007, voiplist wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
Missing $ before the [
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback
In article [EMAIL PROTECTED], Mike Clark [EMAIL PROTECTED] wrote: Tony Mountifield wrote: Where can I find this paper? If you mean the one at voip-magazine, the links to it that I have found no longer seem to work. Tony: I believe this is it: http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf Mike, many thanks for that - I have it now! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Client NAT
Hi,
I am trying to run an Asterisk (1.4.10) server on Ubuntu Linux Fiesty Fawn. The
server is behind NAT. I am testing SIP with the X-Lite client from xten. The
client is also behind NAT.
I realize that this is amongst the worst configurations, but I have been made
to believe that it can work... eventually. However, currently SIP call set up
seems to go fine, but no media is transferred in either direction. For example,
the following is output on the asterisk CLI despite no voice being heard.
-- Executing [EMAIL PROTECTED]:1] Playback('SIP/john-081da978', 'hello-world')
in new stack
I have used tcpdump, ethereal and RTP debug to trace the problem. I have the
following data:
1. My SIP client is correctly receiving and processing SIP/SD packets. Ethereal
indicates that the client is told to send RTP traffic to server port 50296 and
all packets are sent to this port. However, this port IS NOT in the range
specified in rtp.conf. This is my first point of confusion.
2. Pushing onward, I forwarded ALL ports from my router (NAT) to the Asterisk
server to see if Asterisk would then pick up the voice stream. Tcp dump on the
Asterisk server indicates:
02:48:30.082693 IP pool-71-107-141-25.lsanca.dsl-w.verizon.net.50943
gaurav-desktop.local.50296: UDP, length 172
Each of these lines match a line in my ethereal capture on the client. However,
still no voice! This is my second point of confusion.
3. Finally, I tried using RTP debug to trace where audio packets from the
server are being sent. The output I got:
Sent RTP packet to 192.168.1.47:54626 (type 00, seq 058094, ts 012000, len
000160)
I have no idea where 192.168.1.47:54626 came from. There is no computer on my
server's LAN with that local ip, and it is NOT the local IP of my client (which
was 10.0.0.3).This is the third point of confusion.
My conf files are attached. Important details are below:
1. sip.conf
[global]
nat=yes
canreinvite=no
[john]
context=john
nat=yes
canreinvite=no
host=dynamic
2. extensions.conf
[john]
exten = 100,1,Dial(SIP/john) ; loopback
exten = 101,1,Playback(hello-world) ; the basics
3. rtp.conf
rtpstart=1
rtpend=2
Your expertise would be appreciated. My sincere thanks for your time and help
in advance.
--G
_
Explore the seven wonders of the world
http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE
sip.conf
Description: Binary data
extensions.conf
Description: Binary data
rtp.conf
Description: Binary data
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following is output on the asterisk CLI
despite no voice being heard. -- Executing [EMAIL PROTECTED]:1]
Playback('SIP/john-081da978', 'hello-world') in new stack
*sigh* The old NAT SIP issue - again... )-:
There is a lot of the VoIP WiKi on it. Eg:
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
However, assuming the asterisk and client are behind different NAT
firewalls, do this:
1. Tell the client to use a stun server and don't fiddle with the client's
firewall (other than to make sure it's not actually firewalling 5060 and
1-2)
If you're stuck for a stun server, use stun1.drogon.net:3478
2. Port forward 5060-5069 and 1-2 on the firewall that fronts the
asterisk box to the asterisk box.
3. Tell asterisk it's behind a NAT firewall.
1. sip.conf
[global]
nat=yes
canreinvite=no
This isn't enough. You also need to tell it the IP address of the external
firewall, and your local network address.
nat=yes
localnet=192.168.2.0/24
externip=1.2.3.4
Where 1.2.3.4 is the external IP address - the one the client is pointing
to. This needs to be a static IP address (or at least not change for the
duration of your use) the client can be behind a dynamic IP address.
you might need a bit more in the client definition - eg:
[100]
context=internal
type=friend
secret=very
qualify=yes
nat=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=100
callerid=Joe Bloggs 100
callgroup=1
pickupgroup=1
subscribecontext=BLF
And that's it.
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI for Asterisk realtime
MOR Billing comes in PRO and FREE versions. FREE version is OpenSource. It supports Realtime and GUI is written in RoR. It has own app_mor.so written in C and is very fast. You can check it: http://www.kolmisoft.com/mor Regards/Pagarbiai, Mindaugas Kezys http://www.kolmisoft.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Thursday, August 16, 2007 10:50 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] GUI for Asterisk realtime Thanks. I had googled as well and found basically the same links. We are building a DUNDi/Realtime cluster and need gui management. Mike.. http://www.bicomsystems.com/products/ Senad ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Client NAT
Hi Gordon,
I did everything that you suggested, however, the symptoms remain.
I set the rtp.conf to use ports 1 to 2
I assured that my router was forwarding these ports. However, the Media
Description Section of the SIP/SD packet (captured with ethereal) reads:
Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101
50486 is the destination port of all RTP packets sent from the client. These
are filtered out by my server NAT's firewall. It seems that Asterisk is not
using rtp.conf
I did some searching and found the following link. This is right around the
time that I downloaded. Could this be the trouble?
http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html
Date: Sun, 19 Aug 2007 11:08:57 +0100
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following is output on the asterisk CLI
despite no voice being heard. -- Executing [EMAIL PROTECTED]:1]
Playback('SIP/john-081da978', 'hello-world') in new stack
*sigh* The old NAT SIP issue - again... )-:
There is a lot of the VoIP WiKi on it. Eg:
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
However, assuming the asterisk and client are behind different NAT
firewalls, do this:
1. Tell the client to use a stun server and don't fiddle with the client's
firewall (other than to make sure it's not actually firewalling 5060 and
1-2)
If you're stuck for a stun server, use stun1.drogon.net:3478
2. Port forward 5060-5069 and 1-2 on the firewall that fronts the
asterisk box to the asterisk box.
3. Tell asterisk it's behind a NAT firewall.
1. sip.conf
[global]
nat=yes
canreinvite=no
This isn't enough. You also need to tell it the IP address of the external
firewall, and your local network address.
nat=yes
localnet=192.168.2.0/24
externip=1.2.3.4
Where 1.2.3.4 is the external IP address - the one the client is pointing
to. This needs to be a static IP address (or at least not change for the
duration of your use) the client can be behind a dynamic IP address.
you might need a bit more in the client definition - eg:
[100]
context=internal
type=friend
secret=very
qualify=yes
nat=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=100
callerid=Joe Bloggs 100
callgroup=1
pickupgroup=1
subscribecontext=BLF
And that's it.
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi Gordon,
I did everything that you suggested, however, the symptoms remain.
I set the rtp.conf to use ports 1 to 2
I assured that my router was forwarding these ports. However, the Media
Description Section of the SIP/SD packet (captured with ethereal) reads:
Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101
50486 is the destination port of all RTP packets sent from the client.
These are filtered out by my server NAT's firewall. It seems that
Asterisk is not using rtp.conf
I did some searching and found the following link. This is right around
the time that I downloaded. Could this be the trouble?
http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html
I know what when I do that on my systems, it just works. Even with
xlite.
I've never fiddled with rtp.conf. Mine is as it came with the default
installation.
rtpstart=1
rtpend=2
However, I'm running asterisk version 1.2.X, so there might be some other
issues with 1.4.
This is the scenario that 99% of my installations work under for people
with phones not on the office LAN, and so-far so good (for me!)
Gordon
Date: Sun, 19 Aug 2007 11:08:57 +0100
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following is output on the asterisk CLI
despite no voice being heard. -- Executing [EMAIL PROTECTED]:1]
Playback('SIP/john-081da978', 'hello-world') in new stack
*sigh* The old NAT SIP issue - again... )-:
There is a lot of the VoIP WiKi on it. Eg:
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
However, assuming the asterisk and client are behind different NAT
firewalls, do this:
1. Tell the client to use a stun server and don't fiddle with the client's
firewall (other than to make sure it's not actually firewalling 5060 and
1-2)
If you're stuck for a stun server, use stun1.drogon.net:3478
2. Port forward 5060-5069 and 1-2 on the firewall that fronts the
asterisk box to the asterisk box.
3. Tell asterisk it's behind a NAT firewall.
1. sip.conf
[global]
nat=yes
canreinvite=no
This isn't enough. You also need to tell it the IP address of the external
firewall, and your local network address.
nat=yes
localnet=192.168.2.0/24
externip=1.2.3.4
Where 1.2.3.4 is the external IP address - the one the client is pointing
to. This needs to be a static IP address (or at least not change for the
duration of your use) the client can be behind a dynamic IP address.
you might need a bit more in the client definition - eg:
[100]
context=internal
type=friend
secret=very
qualify=yes
nat=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=100
callerid=Joe Bloggs 100
callgroup=1
pickupgroup=1
subscribecontext=BLF
And that's it.
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Change Packetization Time
Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Client NAT
Thank you very much for your prompt replies. Perhaps I will consider moving to
a 1.2 version of Asterisk.
Date: Sun, 19 Aug 2007 12:08:36 +0100
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi Gordon,
I did everything that you suggested, however, the symptoms remain.
I set the rtp.conf to use ports 1 to 2
I assured that my router was forwarding these ports. However, the Media
Description Section of the SIP/SD packet (captured with ethereal) reads:
Media Description, name and address (m): audio 50486 RTP/AVP 0 8 101
50486 is the destination port of all RTP packets sent from the client.
These are filtered out by my server NAT's firewall. It seems that
Asterisk is not using rtp.conf
I did some searching and found the following link. This is right around
the time that I downloaded. Could this be the trouble?
http://lists.digium.com/pipermail/asterisk-bugs/2007-July/001213.html
I know what when I do that on my systems, it just works. Even with
xlite.
I've never fiddled with rtp.conf. Mine is as it came with the default
installation.
rtpstart=1
rtpend=2
However, I'm running asterisk version 1.2.X, so there might be some other
issues with 1.4.
This is the scenario that 99% of my installations work under for people
with phones not on the office LAN, and so-far so good (for me!)
Gordon
Date: Sun, 19 Aug 2007 11:08:57 +0100
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk and Client NAT
On Sun, 19 Aug 2007, G B wrote:
Hi,
I realize that this is amongst the worst configurations, but I have been
made to believe that it can work... eventually. However, currently SIP
call set up seems to go fine, but no media is transferred in either
direction. For example, the following is output on the asterisk CLI
despite no voice being heard. -- Executing [EMAIL PROTECTED]:1]
Playback('SIP/john-081da978', 'hello-world') in new stack
*sigh* The old NAT SIP issue - again... )-:
There is a lot of the VoIP WiKi on it. Eg:
http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
However, assuming the asterisk and client are behind different NAT
firewalls, do this:
1. Tell the client to use a stun server and don't fiddle with the client's
firewall (other than to make sure it's not actually firewalling 5060 and
1-2)
If you're stuck for a stun server, use stun1.drogon.net:3478
2. Port forward 5060-5069 and 1-2 on the firewall that fronts the
asterisk box to the asterisk box.
3. Tell asterisk it's behind a NAT firewall.
1. sip.conf
[global]
nat=yes
canreinvite=no
This isn't enough. You also need to tell it the IP address of the external
firewall, and your local network address.
nat=yes
localnet=192.168.2.0/24
externip=1.2.3.4
Where 1.2.3.4 is the external IP address - the one the client is pointing
to. This needs to be a static IP address (or at least not change for the
duration of your use) the client can be behind a dynamic IP address.
you might need a bit more in the client definition - eg:
[100]
context=internal
type=friend
secret=very
qualify=yes
nat=yes
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=100
callerid=Joe Bloggs 100
callgroup=1
pickupgroup=1
subscribecontext=BLF
And that's it.
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Invite your mail contacts to join your friends list with Windows Live
Spaces. It's easy!
http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_
Connect to the next generation of MSN Messenger
http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
On 8/19/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Sat, 18 Aug 2007, voiplist wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
#
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
Missing $ before the [
Gordon
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Doh! Thanks, I guess I missed it when comparing my working examples to
this non working one.
Thanks all, I will give this a shot.
Regards,
Todd R.
--
Prestige Messaging
Live Answering Services
SIP or Toll-Free Connectivity
Light Accounts From $14.95/mo
http://www.PrestigeMessaging.com
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8
voiplist wrote:
I am using GotoIf all over the place in 1.4.8 but for some reason, the
following in my dial plan:
exten = _1NXXNXX,1,GotoIf([${EXTEN} = 15554441212]?100)
I would use something like
GotoIf($[${EXTEN} = 15554441212]?100)
because in Asterisk the quotes are part of the string.
But your subject implies that this expression works for
you with variables other than ${EXTEN} or Asterisk
before 1.4.8?
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
My pick of the month: rfc 2822 3.6.5
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Client NAT
Thank you very much for your prompt replies. Perhaps I will consider moving to a 1.2 version of Asterisk. There is one setting in X-lite (under Network) that can cause one-way audio. It used to be called Transmit silence and should be on. It is now called Preserve bandwidth during silence and should be UNchecked. If this is a small installation you don't need 10,000 RTP ports, they could be something like 1-10020. This matters if the router doesn't allow a range in the forward parameters. As a last resort, go to the Free World Dialup forums and look for one way audio or firewall or nat and xlite. There are a large number of solutions there. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pre-recorded first and last names audio database
On 8/4/07, John Vogel [EMAIL PROTECTED] wrote: My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. Hi, John Brown. Your book is due back at the library. So I need John and Brown in audio files along with LOTS of other names - The only good solution to this problem is getting the customers to record their names when their records are created. This could be done at the library itself, if that's who is doing this, during a signup process. NO TTS will be able to do this. The reason is that names can be spelled the same and pronounced differently depending on the person. Even some first names like Stephen have at least two ways of being pronounced. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Someone please explain FXO/FXS module/channel relationships in Zaptel configuration to me
Please explain the relationship between modules from the driver (wctdm), the /etc/zaptel.conf file and zapata.conf. Specifically, if I have a FXS module 0 and FXO module 1, what should be used in zaptel.conf and what should be used in zapata.conf? Then finally, in extensions.conf, what is the Zap channel for dialing out? Zap/? % dmesg Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) % cat /etc/zaptel.conf fxoks=1 fxsks=2 % cat zapata.conf ... signalling=fxo_ks context=outgoing-analog echocancel=yes callerid=asreceived channel = 1 signalling=fxs_ks context=incoming-analog echocancel=yes callerid=asreceived channel = 2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] flash zap FXO port from SIP device (SPA-2002) using RFC2833 or SIP INFO
Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box and an SPA-2002, the Asterisk box has a TDM400P populated with FXO ports. The Asterisk box is attached to the station (FXS) ports of a traditional analog PBX, as you might imagine it uses flash to access features like transfer. I want to be able to flash the FXS port of the SPA-2002 and have that flash propagate to the corresponding FXO port (on the TDM-400P). After you disable the various features of the SPA-2002 that require flash, it can then send flash as either an RFC2833 or SIP INFO event rather than handling it internally. How do I get Asterisk to act on one of those two events and flash the FXO end? This seems to be exactly what I'm trying to do: http://bugs.digium.com/view.php?id=4283nbn=8 setting Flash TX method to SIP INFO. --- SIP read from 192.168.4.98:5060 --- INFO sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.4.98:5060;branch=z9hG4bK-34d6f914 From: spa1l1 sip:[EMAIL PROTECTED];tag=27e0074d58f7a49ao0 To: sip:[EMAIL PROTECTED];tag=as7da5d155 Call-ID: [EMAIL PROTECTED] CSeq: 103 INFO Max-Forwards: 70 Proxy-Authorization: Digest username=spa1l1,realm=asterisk,nonce=7a6afc99,uri=sip:[EMAIL PROTECTED],algorithm=MD5,response=5488f11b8ec0bf6298dc70ffd79559c7 User-Agent: Linksys/SPA2002-3.1.20 Content-Length: 11 Content-Type: application/hook-flash Signal=hf - --- (11 headers 1 lines) --- Receiving INFO! [Aug 18 21:28:48] WARNING[13157]: chan_sip.c:10977 handle_request_info: Unable to parse INFO message from [EMAIL PROTECTED] Content The above seems strange to me, that there is nothing after the word Content in the warning. Seems like there should be hf there. using RFC2833 (AVT) and application/hook-flash shows nothing on console using sip debug and doesn't work. using RFC2833 (AVT) and application/dtmf-relay does the same as above. Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Increase Volume on AGI
Hello All, I have my AGI scripts streaming wav files and I would like to increase the volume on it. Is there any way to increase the volume on outbound SIP trunks, instead of me changing the wav file one by one? Please help? Cheers, Nitesh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Snom 300 Hints and LIne Buttons
Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing the hold music (or making any otehr call). As soon as I put the first call on hold though (by pressing the L2 button), Asterisk pops up the message xtension Changed 116 new state Hold for Notify User Russell. If I drop the first 'line', there's no message from Asterisk. When I flip back to the second line Asterisk says Extension Changed 116 new state Idle for Notify User Russell - even though it's patently not! This obviously makes the BLF lamp on my Snom 370 pretty useless as it only lights up when the Snom 300's got two lines going :-( Can anyone point me in the right direction to getting this fixed? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Packetization Time
Dovid wrote: Does anyone know if it is possible to change the packetization time in Asterisk ? I was told by a client of mine that adjusting this with using G729 can greatly lower the amount of bandwidth used. Your client is correct. Configurable packetization was added introduced with the release of 1.4.0. For details look at the rtp-packetization.txt file in the doc directory for full details. The short answer is to append :size to any codec on your allow directive that you want to change from the default of 20ms. Ex. Allow=g729:40 Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Snom 300 Hints and LIne Buttons
Russell Brown wrote: I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing the hold music (or making any otehr call). As soon as I put the first call on hold though (by pressing the L2 button), Asterisk pops up the message xtension Changed 116 new state Hold for Notify User Russell. If I drop the first 'line', there's no message from Asterisk. When I flip back to the second line Asterisk says Extension Changed 116 new state Idle for Notify User Russell - even though it's patently not! This obviously makes the BLF lamp on my Snom 370 pretty useless as it only lights up when the Snom 300's got two lines going :-( Can anyone point me in the right direction to getting this fixed? Do your peers in sip.conf have call-limit=something, e.g. call-limit=10 sip.conf [general] settings: allowsubscribe=yes subscribecontext=default notifyringing=yes notifyhold=yes limitonpeers=yes Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 2 Speechphone/Mandi
Has anyone that has the Speechphone/Mandi service been able to set up a SIP connection directly with their servers? If so, would you want to share any information on how to do this? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rewriting the From and Subject from voicemail for a MMS Message to a Cell Phone - like visual voicemail
I would like to send Multimedia Messaging (MMS) email (gateway) to my cell phone and have the from and subject be the callerid/calleridnam information from the voice mail message. I know there is a way to call another perl script or program up when an email message is written, but I am not a programmer. I know there could be a perl script or program that could run every minute and check the /var/spool/asterisk/voicemail/default//INBOX and read the msg.txt file and get the information and then attach the msgxxx.wav file and email it but again I am no programmer. Does anyone know if this has been done or is willing to do it? This would be similar to the iPhone visual voicemail using MMS on cell phones. Just a thought. Any ideas or thoughts? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How many calls can use the same username
Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal Luggage? GPS? Comic books? Check out fitting gifts for grads at Yahoo! Search http://search.yahoo.com/search?fr=oni_on_mailp=graduation+giftscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many calls can use the same username
Hi List; If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? If yes, how can I control the number of calls per account? Regards Bilal Hi It can be done with call-limit into sip.conf And I'm not pretty sure but in iax.conf it must be incominglimit/outgoinglimit jat ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How many calls can use the same username
bilal ghayyad wrote: If I configured one SIP account or one IAX account [sipuser1] or [iaxuser1] then how many calls can be originate/terminate using the same account [sipuser1] or [iaxuser1]? The number of calls per account is not really limited (for SIP at least). In other words, can 10 IP Phones (users) do a calls via Asterisk using the same account (SIP or IAX2)? Unless things have changed: No. (not sure about IAX) The number of registrations to an account *is* limited (to 1). If yes, how can I control the number of calls per account? sip.conf: call-limit Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nokia cell connected to Asterisk
Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF winmail.dat___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk multiport
thank you. On 8/17/07, Steven [EMAIL PROTECTED] wrote: Ahh, I see. Good point. -- -- Steven http://www.glimasoutheast.org Steve Totaro [EMAIL PROTECTED] wrote in message news: [EMAIL PROTECTED] Steven wrote: I am curious. Why would one need to do this? If a phone connect to 5060 from another port number, asterisk happily works, so why use multiple port on asterisk? I cannot see the thread history but from the context, I would say because many ISPs block 5060, 25, and others. Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quick DUNDi Poll Questions, For All Asterisk, Users, Please Give Feedback
Questions: 1. Is the wiki DUNDi example and the dundi.conf file too difficult to follow for new users? I wouldn't exactly say that it is too difficult but that the target audience for the default examples is not the average person/entity that could make use of the power inherent with DUNDi. When an average * user/admin wants to use DUNDi they will want to start out small and local rather than worry about all of the intricacies of the e164 standard. It is much easier, in my opinion, to learn the power of DUNDi on a simple level and scale that up to a more globally connected platform. 2. Does the complexity of the DUNDi setup discourage you from using it or even attempting to configure it? I don't see this as the case. Most people who use * are comfortable with the level of complexity that is present in DUNDi, they just don't know where to start. 3. If there was a simple tutorial, step by step guide with easy to setup and test examples, would this encourage more users to investigate and use DUNDi? Absolutely. If you need any help in putting this together or if you simply need people to review a tutorial, I would be glad to assist. I'm interested in putting together a new-user tutorial about DUNDi configuration and setup. There is a lot of great information, setup guides already but the feedback I get is that the current examples are a bit complicated to follow for new users. Thank you for being a part of the conference last Friday. Your participation is greatly appreciated. Matthew Brothers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Application for Home Delivery Restaurants
Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and order detail. - Invoice generation and print functionality of Invoice. - Black list a customer if he placed fake order and next time its black list status would show based on his CLI. - Call recording - Sales Analysis Regards, -- Kashif Naeem Director Soft Hand www.softhand.com.pk Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk1.2.24 or asterisk1.4.10.1
Hi: You offer me use asterisk1.2.24 or asterisk1.4.10.1.How's it if I want to use astbill? Best Regards. - Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
