[asterisk-users] Connecting to multiple databases using res_config_pgsql
Hello, How do I use multiple postgresql databases using res_config_pgsql? I tried creating multiple contexts in res_pgsql.conf, but asterisk is only using the 'general' context. My res_pgsq.conf is [general] ;; Connect to mydb on localhost dbport=5432 dbname=mydb dbuser=pgdbuser requirements=warn [pgwritedb] ;; Connect to mydb2 on another host dbhost=IP of other db server dbport=5432 dbname=mydb2 dbuser=pgdbuser2 dbpass=x requirements=warn In extconfig.conf I configured sippeers = pgsql,general,sip_peers sipregs = pgsql,pgwritedb,sip_regs But asterisk is not using 'pgwritedb' to connect for sipregs, its trying to update 'sip_regs' on the local database. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables set by AGI lost in dialplan
On Thu, Feb 14, 2013 at 8:35 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 14 Feb 2013, Deepesh D wrote: The problem I am facing is that sometimes the variables are wrongly received as 0 (zero) in the dialplan even if the AGI has set it to a non-zero value. Are the variables actually set to '0' and not '' (like an undefined variable)? The variables are set to '0' This error does not happen for all calls and is not reproducible, it is random. Maybe you are violating the AGI protocol. Are you using an established library or did you roll your own? Yes, even I suspect that. I am not using any established library. I have written my own AGI. I get the error utils.c: write() returned error: Broken pipe when the AGI is called. In many forums and mailing-lists I read that this message can be safely ignored. Till now, I used to ignore this message as the AGI seemed to worked without any problem, it was only 2 days ago that I realized that there are a few calls being dropped. Now I have made some changes to the AGI and the error message has gone away. I will monitor it for a day to check if the problem happens again My asterisk server handles about 100 calls per minute, so its impossible for me to do an 'agi set debug' and observer the output That's less than 2 per second, so the output may be voluminous, but not impossible. Without some AGI debugging output it will be difficult to suggest anything relevant. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] What is the maximum number of meetme's allowed?
Hello, What is the maximum number of meetme's allowed by asterisk. On my server with an 8 GB memory, I start getting the following error after 150-160 meetme's are created WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo channel: Cannot allocate memory At this time the server still has about 6 GB of free memory. I even tried this on a server with higher memory, it gives the same result. I am using asterisk 1.4.44. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the maximum number of meetme's allowed?
I am using 64-bit Linux OS. Also before starting asterisk I have set the ulimit to a higher value. When this happened there was no calls in the system. There was only about 160 Meetme conferences, and in each Meetme there was only one channel. On Mon, Dec 24, 2012 at 11:36 PM, Johan Wilfer li...@jttech.se wrote: 2012-12-24 16:13, Deepesh D skrev: Hello, What is the maximum number of meetme's allowed by asterisk. On my server with an 8 GB memory, I start getting the following error after 150-160 meetme's are created WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo channel: Cannot allocate memory At this time the server still has about 6 GB of free memory. I even tried this on a server with higher memory, it gives the same result. I am using asterisk 1.4.44. You have probably run out of file descriptors. Try ulimit -n 8192 before starting asterisk (or in the safe_asterisk-script or the init.d-script). I think this is per default 1024 on debian, and if you use sip + meetme you will hit the limit with about 150 concurrent calls. -- Johan Wilfer JT Technologies Telecommunications AB Jabber: jo...@jttech.se | Web: www.jttech.se -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
I made these changes in dialplan and it worked. Thanks a lot.
In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those case. Is there some way I can get those also working.
On Fri, Oct 12, 2012 at 1:12 AM, Joshua Colp jc...@digium.com wrote:
Deepesh D wrote:
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
remains in the loop till the call is finished. What I wanted to do is
to reduce the number of calls on S1, so as soon as S1 receives a call
from C1 it redirects the call to S2 using 'Transfer' application and
exits from the loop, the call should now be handled by S2
With some crafty configuration you can achieve this using Transfer. With
promiscredir disabled Asterisk will not follow the SIP URI in the 302
response sent back as a result of calling Transfer. The call reenters the
dialplan at the user portion of the URI passed back and executes dialplan as
normal. By prefixing the user portion with a unique identifier you can write
dialplan that strips the prefix and then dials out to S2.
Flow being:
C1 executes Dial(SIP/${EXTEN}@S1) (matched using _1NXXNXX)
S1 executes Transfer(002${EXTEN}) (matched using _1NXXNXX)
C1 executes Dial(SIP/${EXTEN:3}@S2) (matched using _0021NXXNXX)
So simply:
C1 calls S1
S1 decides to send it to S2, but wants to tell C1 to do it directly
S1 sends back a SIP message saying hey call 00218005551212 instead
C1 matches the number to the dialplan which explicitly calls out through S2
S2 receives the call and life is good
* Note: This requires control over both C1 and S1, you can't just do it to
every random system calling.
If you don't get the finer details of this just experiment with the general
idea and configuration option I mentioned. It won't hurt.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
This doesn't work reliably well with all all clients. I tested it using a zoiper soft phone and it worked. But from an ATA device it failed. On the S2 server it failed to authenticate The console of S2 showed [Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth: username mismatch, have X, digest has [Oct 12 18:21:06] NOTICE[30483]: chan_sip.c:22046 handle_request_invite: Failed to authenticate device sip:X@192.168.1.1:14500;tag=3047 On Fri, Oct 12, 2012 at 5:00 PM, Joshua Colp jc...@digium.com wrote: Deepesh D wrote: I made these changes in dialplan and it worked. Thanks a lot. In most of the cases S1, S2 and C1 are in my control. But in some cases the dialplan of C1 is not in my control. Also in some cases C1 can be any SIP client like a softphone or SIP device, so it wont work in those case. Is there some way I can get those also working. You can certainly execute Transfer() and most clients will then send the call to where you have specified. If you have the same users on all possible servers you would Transfer to with the same username/password a challenge should occur and authentication happen. I haven't tested that though. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS})
exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I get
TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
S2). But the call does not get authenticated on S2 and goes into
default context instead of 'test_context'. How can I transfer the call
such that S2 authenticates the call and sends it to the required
context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
From C1 when I directly dial into S2, it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2, I define the context as
'test_context' and the default context is 'default'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS})
exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I get
TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
S2). But the call does not get authenticated on S2 and goes into
default context instead of 'test_context'. How can I transfer the call
such that S2 authenticates the call and sends it to the required
context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS})
exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I get
TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
S2). But the call does not get authenticated on S2 and goes into
default context instead of 'test_context'. How can I transfer the call
such that S2 authenticates the call and sends it to the required
context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
From C1 when I directly dial into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS})
exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I get
TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
S2). But the call does not get authenticated on S2 and goes into
default context instead of 'test_context'. How can I transfer the call
such that S2 authenticates the call and sends it to the required
context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.
When I directly dial from C1 into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the
call to S2 then the call goes into default context.
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS})
exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I get
TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the call to
S2). But the call does not get authenticated on S2 and goes into
default context instead of 'test_context'. How can I transfer the call
such that S2 authenticates the call and sends it to the required
context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
I am able to register S1 as a peer in S2 and dial from S1 to S2, but
this is not my requirement. I want to dial from C1 into S1 and S1
should redirect the call to S2.
I am trying to do a load balancing setup between S1 and S2. S1 will be
primary server which accepts all calls and then based on some
conditions redirect some calls to S2
On Thu, Oct 11, 2012 at 6:39 PM, Danny Nicholas da...@debsinc.com wrote:
So what happens when you dial directly from S1 to S2?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 5:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
In all my peer definitions on S1 and S2 I define the context as
'test_context' and the default context is 'default'.
When I directly dial from C1 into S2 it goes into the context
'test_context'. But when the call is made to S1 and S1 transfers the call to
S2 then the call goes into default context.
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a peer to S1.
From C1, I dial into S1 using 'Dial' command. What I want to do is,
use the Transfer command in S1 and transfer the call to S2.
Dialplan on S1
[test_context]
exten = _X.,1,Transfer(SIP/${EXTEN}@IP_of_S2)
exten = _X.,n,NoOp(${TRANSFERSTATUS}) exten = _X.,n,Hangup
Dialplan on S2
[default]
exten = _X.,1,Playback(somemsg)
exten = _X.,n,Hangup
[test_context]
exten = _X.,1,Answer
exten = _X.,n,Playback(msg)
exten = _X.,n,Hangup
The context for the SIP peer C1 is defined as 'test_context' in S1 and
S2.
In C1, I have set 'promiscredir = yes' in sip.conf.
When I dial from C1, the call is successfully transferred to S1 (I
get TRANSFERSTATUS as SUCCESS and I can see C1 trying to send the
call to S2). But the call does not get authenticated on S2 and goes
into default context instead of 'test_context'. How can I transfer
the call such that S2 authenticates the call and sends it to the
required context?
Thanks
What happens when you dial into S2 from outside?
Did you set a context in sip.conf on S2?
sean
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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still remains in the loop till the call is finished. What I wanted to do is to reduce the number of calls on S1, so as soon as S1 receives a call from C1 it redirects the call to S2 using 'Transfer' application and exits from the loop, the call should now be handled by S2 On Thu, Oct 11, 2012 at 7:22 PM, Danny Nicholas da...@debsinc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D Sent: Thursday, October 11, 2012 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk? I am able to register S1 as a peer in S2 and dial from S1 to S2, but this is not my requirement. I want to dial from C1 into S1 and S1 should redirect the call to S2. I am trying to do a load balancing setup between S1 and S2. S1 will be primary server which accepts all calls and then based on some conditions redirect some calls to S2 -- My point in asking is that a transfer (in general, and specifically in this case) is a redial. If A dials B and B transfers to C, two dials occur; A-B to get to Asterisk, then B-C within Asterisk. So when you call from C1 to S1 and then S1 calls S2, it should look the same as if there were one call from S1 to S2. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Allowing transfer of only incoming calls
Hello, Is there a way by which a SIP peer can only transfer an incoming call? When I set 'allowtransfer=yes' the peer is able to transfer both outgoing calls and incoming calls. Is there any SIP setting or dialplan setting by which I can restrict transfer of outgoing calls. The asterisk version I am using is 1.8.8.2 and the peers are using the SIP REFER method for transferring calls. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Types of bridging
Hello all, What are the different type of bridging used by asterisk in a SIP call? What is the difference between Packet2Packet bridging, Remote bridging and Native bridging? Can someone please explain me the differences or point me to a good documentation of the same. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Types of bridging
Earlier I was using asterisk 1.4 and 1.6. In these version it used to do native bridging and the CPU load was not very high. Now after switching to asterisk 1.8 it has started to do remote bridging and the CPU load has often started to peak. Could this be a configuration issue. I have done the same SIP settings that was earlier there in 1.4 and 1.6. I have 'directmedia=yes' and 'directrtpsetup=yes' in sip.conf and both the peers use the same codecs and there are no nat issues as well Please help On Thu, Mar 29, 2012 at 7:29 PM, Phil Frost p...@macprofessionals.com wrote: On Mar 29, 2012, at 08:43 , Deepesh D wrote: What are the different type of bridging used by asterisk in a SIP call? What is the difference between Packet2Packet bridging, Remote bridging and Native bridging? Packet2Packet bridging is when RTP datagrams are forwarded by Asterisk without modification. This imposes little load on the CPU. Obviously this can only happen if both ends are using the same codec, and likely there are likely other less obvious conditions that must be met. Remote bridging happens when Asterisk can direct both ends to send media (RTP probably) to each other directly, by a SIP reINVITE, for example. Only works if both ends have a route to each other, Asterisk is configured to do it, each end shares a codec, and probably a dozen other more subtle conditions are true. In this case there is no load on Asterisk as it's not even in the media path. It also means it can't do things like intercept and act on DTMF or monitor the call. Native bridging is when media is forwarded with Asterisk, but for whatever reason (different codecs, maybe) Asterisk must inspect or modify the stream. Could mean a significant CPU load. -- Phil Frost Macprofessionals office 248-893-0738 direct 248-662-0809 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] default context overrides context of peer
This works when I change the host to non-dynamic and insecure=port,invite for the peer, but does not work when host=dynamic. Also my sip peers are realtime. If I remove the realtime peer and create a peer in sip.conf this works !! On Tue, May 3, 2011 at 11:15 AM, Justin Case nogoodnameswereavaila...@gmail.com wrote: On Mon, May 2, 2011 at 1:09 PM, Deepesh D deep.d2...@gmail.com wrote: Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to 'defcontext' and the calls fail. After the peer registers 'sip show peer peername' show the context as 'outcontext', but while making a call the default context in sip.conf overrides the peer context. Is there any other setting that I need to do in asterisk 1.6.2.17? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I am having the same issue and have not found a fix. Maybe we are both doing something wrong ;) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] default context overrides context of peer
Hello, I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17. I have context=defcontext set in sip.conf. For each peer I have context=outcontext in the peer definition since I want outgoing calls from registered SIP peers to go through context 'outcontext'. This used to work in the older version (1.6.2.7), but after upgrading this has stopped working. Now outgoing calls are going to 'defcontext' and the calls fail. After the peer registers 'sip show peer peername' show the context as 'outcontext', but while making a call the default context in sip.conf overrides the peer context. Is there any other setting that I need to do in asterisk 1.6.2.17? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents login
Hello Michael, You could try to achieve this functionality in dialplan by using the applications AddQueueMember/RemoveQueueMember which are used to dynamically add/remove queue members. An example dialplan flow for agent login will be 1. get the SIP interface from which the agent is logging in (like SIP/1234). You should be able to extract it from the CHANNEL variable 2. Ask for the agentid/code. Read the agentid/code and use an AGI to authenticate the agent 3. Add the SIP interface to the queue using AddQueueMember AddQueueMember(queuename,SIP/1234,) You could have a similar dialplan for agent logout, which removes SIP interface from the queue using RemoveQueueMember. For the DND you could then use PauseQueueMember/UnpauseQueueMember applications. Regards, Deepesh On Sat, Dec 25, 2010 at 7:01 PM, Michael voip.quest...@gmail.com wrote: Greetings and Merry Christmas, We're trying to implements a queue and agents login mechanism on our Asterisk. After going over the documentation, we're unsure if we got it right. We wish to setup a hotdesk mechanism, where an agent comes to a station with a PC IP phone (that is defined as a sip friend user in sip.conf), dials a certain number (agent login extension), enters his agent Id and code, and from now on this phone serves as his. Then, when a call comes into the queue he is associated to, if he is not on the line, the call would ring in his phone and he'll be able to pick it up. He should also be able to set a DND (if he needs a break). Is that possible?? From what we saw, the agents login works on a constantly open line. Any help (and examples) would be highly appreciated. Thank you in advance, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip peer becomes unreachable in Asterisk 1.6
Hello, I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP settings in sip.conf in this version also. I am facing a problem when a SIP client makes a call. When a SIP client registers to asterisk its status shows 'OK' and it is able to receive incoming calls. But as soon as this client make a call, its status becomes 'UNREACHABLE' and it cannot receive any incoming calls. Its status remains 'UNREACHABLE' until it re-registers again. I have faced this problem on various versions of 1.6 (1.6.2.0, 1.6.2.7, 1.6.1.1), but this never happened in the 1.4 (1.4.24, 1.4.26) versions. I have kept the SIP re-register time in the clients to a very small value to avoid becoming 'UNREACHABLE' for a long time, but his doesn't seem to be the solution. Is there any specific SIP settings which needs to be made in 1.6 to avoid this problem? I am using realtime sip. Some of my sip settings are rtcachefriends=yes rtupdate=no qualify=yes canreinvite=yes nat=yes Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get channel name of originated channel
Hello, I am using asterisk manager interface (http) for originating calls. How can I get the name of the channel which is created by originate? I want to use this channel for other manager commands like Atxfer, Monitor, Hangup etc. If I do action=originate, channel=SIP/200 then it creates a channel like 'SIP/200-0865ff80' which I can see in the asterisk console using core show channels verbose. Now if I want to transfer this call I have to use action=Atxfer, channel=SIP/200-0865ff80 for which I need the channel name. Is there any way to get this channel name or set the channel name during originate? On what basis does asterisk assign channel names, is it random? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate multiple channels
Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting hook flash in asterisk
Hello, Is it possible to detect a hook flash in asterisk. I want to be able to perform some functions an hook flash. I have the following entry in features.conf which executes a Macro on detecting key press '**'. [applicationmap] test = **,caller,Macro,testflash Is it possible to do this action on hook flash? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan for conference
Hello, I wanted to add the functionality of 3-way conference to my asterisk pbx using meetme or confbridge. During a call the user should be able to put the other party on hold and dial another number, then on dialing some key sequence all three of them enters into a conference. Is it possible to do this? Can someone please point me to any dialplan examples to do this. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Generate cdr on Hangup
Hello,
I have the following dialplan
exten = _X.,1,Set(CDR(userfield)=test)
exten = _X.,n,Do some checks and hangup if checks fail
exten = _X.,n,Dial(SIP/${EXTEN})
exten = _X.,n,Hangup
1. If the Dial fails with a busy, noanswer or congestion then a cdr is
generated.
2. If the call fails before Dial (if the checks fail) then no cdr is
generated.
I would like to generate a cdr in the second case also. Is there a way to do
this?
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Re: [asterisk-users] Getting 'username' of sip peer
Yes, setting a fullname=xxx in peer definition sets the CALLERID(name) But if the peer sets it own callerid then will it override this value? On Wed, May 26, 2010 at 10:52 PM, Danny Nicholas da...@debsinc.com wrote: I might be wrong, but I think that adding fullname=xxx to the context will populate CALLERID(name) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D Sent: Wednesday, May 26, 2010 12:18 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Getting 'username' of sip peer Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting 'username' of sip peer
Thanks. Got this working by using setvar=variable=value in the peer
definition
SIPCHANINFO(peername) is giving me the 'name' of the peer i.e.
'TestSIPUser' and not the 'username'.
On Wed, May 26, 2010 at 11:20 PM, Jared Smith jsm...@digium.com wrote:
On Wed, 2010-05-26 at 22:48 +0530, Deepesh D wrote:
When a call is made from any of these peers I want to get the username
of the peer.
for eg:- If a call is being made from 'TestSIPUser' then I want to be
able to get the value 'testuser'
I can think of two ways of doing this. The first is to use the
SIPCHANINFO() dialplan function, like this:
exten=123,1,Verbose(0,The call came from ${SIPCHANINFO(peername)})
The other option is to use the setvar=variable=value setting in the
peer definition in sip.conf. For example, if you add
setvar=USERID=jsmith in a user/peer/friend definition, Asterisk would
automagically create a channel variable named USERID with a value of
jsmith every time this device made a call into Asterisk.
--
Jared Smith
Sr. Trainer
Digium, Inc.
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[asterisk-users] Getting 'username' of sip peer
Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from 'TestSIPUser' then I want to be able to get the value 'testuser' Is it possible to get the value of 'username' of the peer in the dialplan using some application/function ? Thanks, Deepesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using unix socket to connect with database
Hello, I tried removing the dbhost and dbport entries and restarting asterisk. During startup the following warnings are shown and it gets stuck up at this point for a few seconds. WARNING[1819]: res_config_pgsql.c:1367 parse_config: PostgreSQL RealTime: No database host found, using localhost via socket. WARNING[1819]: res_config_pgsql.c:1383 parse_config: PostgreSQL RealTime: No database port found, using 5432 as default. But there is no connection being made to the database. On Sat, May 22, 2010 at 3:25 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/21/2010 02:48 AM, Deepesh D wrote: I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to database, it is making a tcp connection with the db. Is there anyway I can force asterisk to use the unix socket for db connection? You've specified *both* a socket to be used and a hostname/port number. The way the code is written, if both are supplied, the host/port combination is used and the socket path is ignored. If you don't want the host/port to be used, don't specify them. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using unix socket to connect with database
I am using Asterisk 1.6.2.7 On Sat, May 22, 2010 at 7:20 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 05/22/2010 02:07 AM, Deepesh D wrote: I tried removing the dbhost and dbport entries and restarting asterisk. During startup the following warnings are shown and it gets stuck up at this point for a few seconds. WARNING[1819]: res_config_pgsql.c:1367 parse_config: PostgreSQL RealTime: No database host found, using localhost via socket. WARNING[1819]: res_config_pgsql.c:1383 parse_config: PostgreSQL RealTime: No database port found, using 5432 as default. But there is no connection being made to the database. What version of Asterisk are you using? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using unix socket to connect with database
Hello, I am using asterisk realtime with a postgresql database on the same server. In res_pgsql.conf I have specified [general] dbhost=localhost dbport=5432 dbname=asteriskdb dbuser=psql dbsock=/tmp/.s.PGSQL.5432 Since both asterisk and db are on same server, I would like asterisk to connect to db using the local unix socket. However asterisk is not using the local unix socket to connect to database, it is making a tcp connection with the db. Is there anyway I can force asterisk to use the unix socket for db connection? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unexpected message received when receiving Fax
Hello, I have been trying to setup asterisk 1.6.2.0 to receive fax. I have two SIP trunks connected to asterisk. One of them is a VoIP service provider and the other is an audiocodes gateway connected with pstn and fax lines. I am able to receive faxes on the DID numbers provided by the VoIP service provider, but I am not able to receive fax through the fax lines connected through audiocodes. udptl debug shows T.38 packets coming from the audiocodes, but the line gets disconnected with an error phase_e_handler: Error transmitting fax. result=13: Unexpected message received. What could be the reason for this error? Is this a problem with misconfiguration of audiocodes or asterisk. I have the following t.38 settings in sip.conf 't8pt_udptl=yes,redundancy,maxdatagram=400' and the dialplan I use is exten = s,1,Answer exten = s,2,Wait(3) exten = s,3,ReceiveFAX(filename) Thank you -- _ -- 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] T.38 with reinvite
Hello, Is it possible to use asterisk in T.38 pass through mode with reinvite? My fax calls are getting disconnected if canreinvite=yes. It works only if I make canreinvite=no. Normal calls work in both cases. Thanks -- _ -- 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] Not able to receive fax
Hello, I have been trying to setup asterisk (1.6.2.0) to receive fax. I am able to receive faxes sent from a zoiper softphone connected to asterisk. I have some DID numbers (with T.38 support) forwarded to my asterisk pbx. I am not able to receive faxes from these numbers. The error I get on the asterisk console is phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely Can someone please help. Thanks. -- _ -- 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] Not able to receive fax
Thanks. It's working now. In my sip.conf I had 't8pt_udptl=yes'. I changed it to 't8pt_udptl=yes,redundancy,maxdatagram=400' and it started working. On Tue, Feb 9, 2010 at 6:22 PM, Tommy Botten Jensen tommy.jen...@freecode.no wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA512 I have been trying to setup asterisk (1.6.2.0) to receive fax. I am able to receive faxes sent from a zoiper softphone connected to asterisk. I have some DID numbers (with T.38 support) forwarded to my asterisk pbx. I am not able to receive faxes from these numbers. The error I get on the asterisk console is phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely Have you set the vendor connection point to enable T.38?: t38pt_udptl=yes ; redundancy,maxdatagram=400 You can also use the CLI command 'udptl set debug' to retrieve debug information about the T.38 data stream. Hope this helps. - - Tommy Botten Jensen -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEAREKAAYFAktxWm8ACgkQ573V05EH/palCgCgiuAebIOxERfIG50+LNTUfBXp HNcAn0Kuu355yfftodPeeP5eDMZl+5+G =ogQX -END PGP SIGNATURE- -- _ -- 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
[asterisk-users] Asterisk status 488 Not acceptable here on receiving fax
Hello, I have been trying to setup asterisk 1.6.1.1 to receive fax. Whenever a SIP peer (zoiper soft phones) tries to send a fax message asterisk responds by sending a 488 Not acceptable here and the sending fails. I tried changing a few sip settings like canreinvite and codec preferences, but it did not help. The same sip peer is able to make normal calls. The same settings works on on asterisk 1.6.2.0 and I am able to receive fax successfully in asterisk. I would like to get this working in 1.6.1.1 as It is not possible for me to upgrade asterisk on my production servers. Can someone please help. Thanks, Deepesh -- _ -- 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] Adminpin for conference room
Hello,
Can someone please explain me how the adminpin for conference rooms is used?
In the following dialplan
exten = 1122,1,MeetMe(${conf-room-no})
If users join this conference by dialing adminpin or pin will it make
any difference to the user?
Does dialling the adminpin give the user any kind of control of the
conference room?
Thanks
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