Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available (except on the mailing list and
the voip-info wiki (which is usually very old))?
I think this is not possible. If you take a look at main/rtp.c there is
no config option for an IP address.
regards
klaus
Jun Yin schrieb:
some vendors(like alcatel-lucent) developed a kind of sip proxy which
includes two parts: one sip signaling module and one or more voice
modules. voice
Klaus Darilion schrieb:
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Answer myself: I have found the documentation about chan_local's options
in
26 jun 2008 kl. 10.17 skrev Klaus Darilion:
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available (except on the mailing list and
Hi,
I tried this before I ask here on the list.
In 1.2 SetMusicOnHold did not work. The Moh class defined in queues.conf is
overwriting any SetMusicOnHold values of the caller channel.
You can see this if you use periodic announce, the Moh call is printed in the
CLI and is allways the class
Hi all,
I am trying to make an outbound video call to a mobile from asterisk.
however it keeps failing.
I can make inbound calls from a mobile and view video.
I am using x-lite to initiate the outbound call, however I have tried using
the management interface as well (action: etc...) and result
Finally did it but only one more problem, I want it to ring once before
going to the context or playing the background message.
[day_menu]
exten = s,1,Answer()
exten = s,2,Background(welcome-message)
exten = s,3,Dial(SIP/5960,200,rt) ; week day goes to Philadelphia
Office
[weekend__menu]
Johansson Olle E schrieb:
26 jun 2008 kl. 10.17 skrev Klaus Darilion:
Hi!
I am looking for authoritative documentation about channel driver
options, e.g. 'n' and 'j' option for chan_local or the SIP channel
option to set a specific To: header.
Is there such documentation available
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).
Another reason could be that the telco blocks video calls.
Grygoriy Dobrovolskyy schrieb:
You have 2 choices to pickup someone's phone with snom's
1: imagine yourself prefix of pickup, let's say 4
exten=4XX,1,Pickup([EMAIL PROTECTED])
so if u call 4 + phone number you will pickup that one.
Second you can add pickupgroup=number for each phone
Hi,
You could try to use libpri-1.4.7.1-llc-transmit-receive-patch.txt from
http://bugs.digium.com/view.php?id=11595 to signal H324M in LLC IE too.
Maybe the switch wants to have it in Bearer Capability and LCC (I once
had such a switch).
Just applied the patch, failed again. can you tell
Steve Kennedy a écrit :
[...]
Are the same rules and conditions that exist here in the States
mirrored elsewhere?
How does a person in Europe go fully VoIP and still keep the main
number?
In the UK numbers are portable, though the telco wanting the number must
have a
Well, I think I've solved the problem, just to let you know, I've just added
the Answer() app before the Call(Zap/N) app. Thanks a lot to Yannick Lam
Hang of Sangoma Technologies for suggesting that!!!
On Wed, Jun 25, 2008 at 9:04 PM, Raúl Gómez C. [EMAIL PROTECTED] wrote:
Well, I have new
On Thu, 2008-06-26 at 06:15 -0500,
[EMAIL PROTECTED] wrote:
Date: Wed, 25 Jun 2008 23:41:18 +0200
From: Michiel van Baak [EMAIL PROTECTED]
Subject: Re: [asterisk-users] SIP vs. SKINNY
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=us-ascii
Dear,
I am using ser + asterisk, for outgoing calls,
my problem is that the session was not closed if the caller says bye.
can u help me ?
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AstriCon 2008 - September 22
Anybody else get theses warning?
[Jun 26 10:08:55] WARNING[3158]: chan_zap.c:4747 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1
PB
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AstriCon 2008 - September 22
Hi
I followed this howto
http://www.voip-info.org/wiki/view/MeetMe-Web-Control
and
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
to install web meetme with asterisk, I know the meetme module is included
however I need to be able to ban and mute users
you also need (as stated in the bug report) the patch
10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch from
http://bugs.digium.com/view.php?id=10217
This enables LCC in chan_zap. Is this was done some time ago I do not
remember anymore who it is activated, I think you have to add the
Hello All,
This is my third freshly installed and updated CentOS 5.1 with installed
Digium 4-port Analog card and while compiling Zaptel I am getting this
error. If I run ./install_preq test and ./install_preq install it
says Install Successfully.
Error
=
CC [M]
Yes, it's happening to me too
[Jun 26 07:54:56] WARNING[24659] chan_zap.c: CallerID returned with error on
channel 'Zap/3-1'
[Jun 26 07:54:57] WARNING[24659] chan_zap.c: Ring/Off-hook in strange state
6 on channel 3
Mostly of the time this two messages comes together. The other situation in
Ali wrote:
I followed this howto
http://www.voip-info.org/wiki/view/MeetMe-Web-Control
and
http://asteriskpr.blogspot.com/2005/09/guide-to-install-web-meetm_112614171575673316.html
to install web meetme with asterisk, I know the meetme
module is included however I need to be able to ban
This Friday June 27th at Noon EDT, JR Richardson is joining us to talk
about scaling asterisk by clustering and server specialization. JR has
authored multiple documents on the subject but I'm unclear as to
whether he intended these to be published, so I'll wait to hear about
that.
Many
Hello,
If am connecting a digium E1 card to a PSTN Switch in the same equipment
room would I need an echo canceller? wouldnt the Switch handle echo
cancellation for dial-in users?
Responses would be appreciated.
BR,
Robor
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Hello,
If am connecting a digium E1 card to a PSTN Switch in the same equipment
room would I need an echo canceller? wouldnt the Switch handle echo
cancellation for dial-in users?
Responses would be appreciated.
BR,
Robor
___
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On Thu, Jun 26, 2008 at 12:17 PM, Robor Oghene [EMAIL PROTECTED] wrote:
Hello,
If am connecting a digium E1 card to a PSTN Switch in the same equipment
room would I need an echo canceller? wouldnt the Switch handle echo
cancellation for dial-in users?
Responses would be appreciated.
BR,
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia: PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and
PSTN to Cell are NOT OK.Dean Collins
Same here.
Some of our clients upgraded from 1.4.18.1 to 1.4.21.
After some time CLI stops responding and no calls are possible.
Killall -9 is the only way to solve (get out) of this situation till next
time it hangs.
Example CLI screenshot:
On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote:
I think it would be a good idea to start an item in the Wiki about this.
Can anyone else chime in for their countries??
Others in the EU, Eastern, Far East?
So Far I have:
Australia:PSTN to PSTN and Cell to Cell are OK
Hi all,
I am getting a weird error here. When i send a call to a sip peer on one of our
servers i get a 'Nobody picked up in -1 ms' immediately following the SIP
INVITE then the call hangs up.
I do not have a timeout in the Dial, if i send the call to a different peer the
call works fine.
When using console/dsp is that play only?
Is it play/record mode? If so how can I make it play only?
When I play wave files on a machine with aplay everything is fine. (no
record)
When I use asterisk and console/dsp I am getting seg faults in alsa-lib.
I want to make sure there is NO record
On 6/26/08, Grey Man [EMAIL PROTECTED] wrote:
On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote:
This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
are getting closer to being merged into 1.4, trunk, and 1.6.x.
If CDR's are important to you, and
Astricon 2008 is less than three months away - the Early Bird
discounts will expire on the last day of the month, which is next
Tuesday - please get your registrations in by then to get up to $100
off the normal rates. Making hotel reservations now is also a good
idea, since while there is a
On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote:
On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote:
This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
are getting closer to being merged into 1.4, trunk, and 1.6.x.
If CDR's are important to you,
Nice, i will search for that.
2008/6/26 Klaus Darilion [EMAIL PROTECTED]:
Grygoriy Dobrovolskyy schrieb:
You have 2 choices to pickup someone's phone with snom's
1: imagine yourself prefix of pickup, let's say 4
exten=4XX,1,Pickup([EMAIL PROTECTED])
so if u call 4 + phone number
If I remember correctly there was a security patch released after
1.4.19, I think that's shwy.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Totaro
Sent: Thursday, June 26, 2008 12:42 AM
To: Asterisk Users Mailing List - Non-Commercial
Well, here in Venezuela there is no way to port out numbers between Telcos
(as far as i know)
On Fri, Jun 27, 2008 at 12:28 PM, Steve Kennedy [EMAIL PROTECTED]
wrote:
On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote:
I think it would be a good idea to start an item in the Wiki
Hi,
iam using and queue and starting an AGI script after caller connected to
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to
be not work, either as variable in the queue command and also not in the AGI
script.
How to found out which agent is connected to
On Thu, Jun 26, 2008 at 8:21 PM, Steve Murphy [EMAIL PROTECTED] wrote:
On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote:
Hi murf,
CDR start answer end
112 4
245 6
Well, time 3 does get lost, but I thought it might be nice to
be able to
Yes I do remember now, I believe that there was a security vunerability
in 1.4.19 and below that was addressed, that is why I updated. Do you
ask because you want to know if you should upgrade yours or to give me
one of those you shouldn't upgrade a production server if its not
needed and working
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage. This could be useful when you do
not have a backup T1
On 17:36, Thu 26 Jun 08, Steve Finkelstein wrote:
Hi all,
I was curious if anyone can recommend a company that would work with
small businesses, and capable of using a fallback number (mobile
phone, home number etc) in the event SIP or IAX2 peering was to
terminate because of some outage.
We're personally located in a small office based in Manhattan. Would
need DIDs for the greater Manhattan area. But it sounds like Speakup
is the type of service we're looking for that would cater to us
domestically.
On Thu, Jun 26, 2008 at 5:50 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On
I use Vitelity strictly for fall back to my cell (and testing).
Thanks,
Steve T
On Thu, Jun 26, 2008 at 5:56 PM, Steve Finkelstein [EMAIL PROTECTED] wrote:
We're personally located in a small office based in Manhattan. Would
need DIDs for the greater Manhattan area. But it sounds like
I think Voicepulse is out of NYC... not sure if they have failover
though... but they have iax2 and sip.
http://connect.voicepulse.com/ is their asterisk page.
Fred Posner
Tel: +1 (212) 937-7844 x501
Fax: +1 (954) 252-4187
www.teamforrest.com
FWD#: 902963
On Jun 26, 2008, at 5:56 PM,
VoicePulse looks awesome, but they do not have the feature I need ...
which is to be able to dial my mobile phone in the event my asterisk
box or the Internet goes kablunk.
On Thu, Jun 26, 2008 at 6:07 PM, Fred Posner [EMAIL PROTECTED] wrote:
I think Voicepulse is out of NYC... not sure if they
List,
Anybody have a script around that will do this? We have to run
valgrind and asterisk to help troubleshoot a bug in the tracker.
Since we do not know how to reproduce the error, we'd like to run them
from an init.d script (simalar to safe_asterisk), email on crash, and
restart asterisk.
Asterisk 1.2, and Cepstral 5, Allison voice.
I execute:
swift Please enter your pin. -o please-enter-your-pin.ulaw -p
audio/channels=1,audio/encoding=ulaw,audio/sampling-rate=8000
then copy it up to /var/lib/asterisk/sounds, and Play() the file.
The sound file seems corrupted. All I hear is
The scenario:
This is all done SIP with a VOIP provider (have to register to single IP)
We have two inbound DID numbers / Accounts.
We have to register each individually with the VOIP provider.
I'd like inbound from each registered account (DID)
to be able to come into a unique PEER or dialplan
Thomas Winter wrote:
Hi,
iam using and queue and starting an AGI script after caller connected to
agent.
How to find out in the script the connected agent, MEMBERINTERFACE seemed to
be not work, either as variable in the queue command and also not in the AGI
script.
How to found out
Hello,
I've spent a couple days searching and posted into the forum with no luck,
apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B connected
to a POTS
line. Also connected to the POTS line are plain
On Thu, Jun 26, 2008 at 9:11 PM, Steve [EMAIL PROTECTED] wrote:
Hello,
I've spent a couple days searching and posted into the forum with no luck,
apologies
to anyone who reads the Digium forums for the cross-post.
I'm having a problem with an asterisk set up where I have a TDM402B
I'm finding that my Asterisk server is bombarding my DNS servers with
lookups like the following:
Queries
5060-b7bfce38: type A, class IN
Name: 5060-b7bfce38
Type: A (Host address)
Class: IN (0x0001)
One call alone has a handful of requests
On Fri, Jun 27, 2008 at 2:20 AM, Steve Gladden
[EMAIL PROTECTED] wrote:
In other words how to match a registration to a peer or inbound context
other that the single defined default.
I've also been told back in the asterisk 1.2 days that it was not possible.
Not true. When you register the
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