Re: [asterisk-users] Integrate Astreisk with SIP interface

2013-05-15 Thread luke devon 


Hi Longst, 

Sorry if I am leading my question in to meaningless. 

Let me explain my requirement . I don't think this something new to forum. 

Supposed I have setup an Asterisk box as a IVR.I want to get the traffic via 
and telecom / mobile operator. 
Meaning , for instance mobile user dial a IVR code it should come to my 
Asterisk IVR BOX.

Generally how do we capture traffic ? If operator uses singtran or some kind of 
IPbased system , can they directly route traffic to Asterisk box ?

Thanks in Advance
Luke


 From: longst 
To: luke devon ; Asterisk Users Mailing List - 
Non-Commercial Discussion  
Sent: Sunday, 12 May 2013, 14:18
Subject: Re: [asterisk-users] Integrate Astreisk with SIP interface
 


what SIP interface means? could you make an example 





On May 12, 2013, at 4:04 AM, luke devon  wrote:


Hi 
>
>
>
>Once I installed astrisk , how do we connect with SIP interface ? 
>
>Can somebody guide me how to integrate SIP interface with asterisk ? I want to 
>use Astrisk just for IVR purpose.
>
>
>Thank you
>Luke
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[asterisk-users] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan

2013-05-15 Thread Brian LaVallee 
Hi Matthew,

Thanks for the response.

> From: "Matthew J. Roth" 
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Wed, 15 May 2013 12:28:11 -0500 (CDT)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Initial REGISTER Request: Contains Credentials
> before 401
> 
> Brian LaVallee wrote:
>> 
>> My SIP provider is not happy that credentials (in the Authorization header
>> field) are provided in the initial REGISTER request.
>> 
>> The SIP provider ONLY wants the credentials AFTER rejecting the message with
>> a 401.
>> 
>> I know it's dumb, because the RFC says that the the initial REGISTER message
>> MAY include credentials.  If it fails, the proper authentication method is
>> included in the 401.  I know there is nothing wrong, it is how SIP is
>> supposed to work.
> 
> Who is your SIP provider?  They need to be called out so that other Asterisk
> users can avoid them.  This tendency to flip the customer/vendor relationship
> on
> its head must be discouraged.

The SIP provider is KDDI Japan.

>> However I would like to keep my SIP provider from complaining.
> 
> The only thing they should complain about is if you don't pay your bill on
> time.
> 
>> Asterisk is "NOT SUPPORTED" by the SIP provider.
> 
> The REGISTER request was successful so, at least from a practical standpoint,
> the provider does support Asterisk.  It would be ideal if all providers
> officially supported Asterisk, but this is just one example of how it's not
> worth trying to please everyone.

I know that the SIP provider is being overly diligent to prevent toll-fraud,
but some of their complaints about normal SIP communications have been
outrageous.  

I don't want to go into it, but KDDI has actually complained that the REPLY
to the 401 was too fast.
 
>> Does anyone in the Asterisk community know how to avoid sending the
>> credentials until AFTER receiving a 401?
> 
> Edit the source.  I'm sorry to be blunt, but I really can't see the developers
> adding another option to "sip.conf" just to satisfy such a pointless request.
> 
>> Any suggestions would be appreciated!
> 
> Ask the provider what platforms are "supported".  Pick one of them and use it
> to
> configure the "useragent" and "sdpsession" options in "sip.conf".  Or look for
> another provider that doesn't waste your time complaining about RFC-compliant
> behavior.  

KDDI does provide a list of supported equipment and vendors.  Specific
hardware or license based software products that quickly become cost
prohibitive.

I doubt that Asterisk will find it's way on the list any time soon.  Because
KDDI follows the traditional "big telco" method of interoperability, which
normally means licensing products for use on their network.

> Regards,
> 
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
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[asterisk-users] SetCallerPres questions

2013-05-15 Thread Adam Moffett 
Does SetCallerPres(Prohib) remove the ANI data from a SIP call or does 
it simply set a flag telling other devices not to display the data?


In other words, could another system override that and see the caller ID 
anyway?  The answer may affect how I handle 911 calls, so I'm very curious.




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Re: [asterisk-users] Polycom and forwarding.

2013-05-15 Thread Carlos Alvarez 
On Wed, May 15, 2013 at 12:10 PM, Ken D'Ambrosio  wrote:

> Hey, all.  I've got an office set up with Asterisk, and forwarding's got a
> bit of a glitch:
> When they forward, they listen for the remote phone to ring, then hang up.
>  If the remote phone doesn't connect, it goes to the original phone's VM.
>  Is this Polycom's "fault," or Asterisk's?  I've been reading up on
> blind/supervised forwards, and, honestly, have myself more confused than
> when I started.  If someone could give me a solid idea of how forwarding
> works, and a sample of how to send it to a remote extension, and have it
> *not* come back to the original extension, that'd be great.
>

You said "forwarding" but described a process that sounds like call
transfer.  I'm going to assume you mean the latter?

We just had a report of this from a customer on their own server.  I
haven't had time to investigate it.  We have confirmed it with Grandstream
and Cisco SPA phones, so it's not just Polycom.

As far as the atxferdropcall someone suggested, I did try that and then the
call is just dropped off into limbo.  The caller is left on hold, and the
nothing happens on the called extension or transfer-to extension.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Asghar Mohammad 
sip set debug peer 90102 and check in log why call drop or upload log
somewhere. configuration seems ok.


On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk wrote:

> Current configuration follows:
>
> [general]
> context=default
> allowguest=no
> alwaysauthreject=yes
> allowoverlap=yes
> allowtransfer=yes
> tcpenable=no
> tlsenable=no
> srvlookup=yes
> vmexten=vm
> rtcachefriends=yes
> nat=no
> directmedia=nonat
> directrtpsetup=no
> videosupport=yes
> maxcallbitrate=384
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> allow=ilbc
> allow=speex
> allow=g726
> allow=g723
> mohinterpret=default
> mohsuggest=default
> dtmfmode=rfc2833
> timer1b=6
> transport=udp
>
> [carrier-1]
> host=a.b.c.d
> type=friend
> context=from-pstn
> disallow=all
> allow=ulaw,alaw
> qualify=yes
> trunk=yes
>
> [90102]
> secret=xx
> mailbox=90102@default
> cid_number=NX
> accountcode=401
> type=friend
> host=dynamic
> port=5060
> qualify=yes
> nat=yes
> transport=udp
> context=users
> disallow=all
> allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263
> directmedia=no
> canreinvite=no
> videosupport=no
>
>
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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Daniel - Asterisk 
Current configuration follows:

[general]
context=default
allowguest=no
alwaysauthreject=yes
allowoverlap=yes
allowtransfer=yes
tcpenable=no
tlsenable=no
srvlookup=yes
vmexten=vm
rtcachefriends=yes
nat=no
directmedia=nonat
directrtpsetup=no
videosupport=yes
maxcallbitrate=384
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=ilbc
allow=speex
allow=g726
allow=g723
mohinterpret=default
mohsuggest=default
dtmfmode=rfc2833
timer1b=6
transport=udp

[carrier-1]
host=a.b.c.d
type=friend
context=from-pstn
disallow=all
allow=ulaw,alaw
qualify=yes
trunk=yes

[90102]
secret=xx
mailbox=90102@default
cid_number=NX
accountcode=401
type=friend
host=dynamic
port=5060
qualify=yes
nat=yes
transport=udp
context=users
disallow=all
allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263
directmedia=no
canreinvite=no
videosupport=no




On Wed, May 15, 2013 at 2:47 PM, Asghar Mohammad wrote:

> please show us peer configuration.
>
>
> On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk 
> wrote:
>
>> Users (softphones) are behind a NAT, Asterisk has its own public ip
>> address
>>
>>
>> On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:
>>
>>> asterisk is behind nat?
>>>
>>>
>>> On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk >> > wrote:
>>>
 Hello everyone,

 I've suffering cut offs after 6 or 7 seconds a call is answered,
 incoming calls are working fine, but outgoing ones show the gollowing
 messages when are being dropped:

 [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
 Retransmission timeout reached on transmission
 ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
 Response) -- See
 https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 6399ms with no response
 [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt:
 Hanging up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to
 our critical packet (see
 https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
 This is happening with my PBX hosted on an external network and peers
 on my local network.

 It seems the SIP ACK is not being received properly.

 I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9

 Elder D. Arohuanca
 Lima - Peru

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>>>
>>>
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>>
>>
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>
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Re: [asterisk-users] Polycom and forwarding.

2013-05-15 Thread Richard Mudgett 
> Hey, all.  I've got an office set up with Asterisk, and forwarding's
> got
> a bit of a glitch:
> When they forward, they listen for the remote phone to ring, then
> hang
> up.  If the remote phone doesn't connect, it goes to the original
> phone's VM.  Is this Polycom's "fault," or Asterisk's?  I've been
> reading up on blind/supervised forwards, and, honestly, have myself
> more
> confused than when I started.  If someone could give me a solid idea
> of
> how forwarding works, and a sample of how to send it to a remote
> extension, and have it *not* come back to the original extension,
> that'd
> be great.

What you are describing is an attended call transfer not
call forwarding.  Call forwarding is a different feature.
>From the behavior you describe, you are using DTMF to initiate the
attended transfer.  There is an option in features.conf called
atxferdropcall that you need to set to yes to have the call not come
back to the transferrer.

Richard

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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Asghar Mohammad 
please show us peer configuration.


On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk wrote:

> Users (softphones) are behind a NAT, Asterisk has its own public ip address
>
>
> On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:
>
>> asterisk is behind nat?
>>
>>
>> On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk 
>> wrote:
>>
>>> Hello everyone,
>>>
>>> I've suffering cut offs after 6 or 7 seconds a call is answered,
>>> incoming calls are working fine, but outgoing ones show the gollowing
>>> messages when are being dropped:
>>>
>>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
>>> Retransmission timeout reached on transmission
>>> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
>>> Response) -- See
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>>> Packet timed out after 6399ms with no response
>>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt:
>>> Hanging up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to
>>> our critical packet (see
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>>> This is happening with my PBX hosted on an external network and peers on
>>> my local network.
>>>
>>> It seems the SIP ACK is not being received properly.
>>>
>>> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>>>
>>> Elder D. Arohuanca
>>> Lima - Peru
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>http://www.asterisk.org/hello
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
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>
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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Daniel - Asterisk 
There was 2-way audio and suddenly, the calls when down.


On Wed, May 15, 2013 at 1:30 PM, Gertjan Baarda wrote:

> When the call is snswered, is there 2-way audio? Seems a natting issue.
>
>
> On Wednesday, May 15, 2013, Daniel - Asterisk wrote:
>
>> Hello everyone,
>>
>> I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
>> calls are working fine, but outgoing ones show the gollowing messages when
>> are being dropped:
>>
>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
>> Retransmission timeout reached on transmission
>> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
>> Response) -- See
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>> Packet timed out after 6399ms with no response
>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
>> up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
>> critical packet (see
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>> This is happening with my PBX hosted on an external network and peers on
>> my local network.
>>
>> It seems the SIP ACK is not being received properly.
>>
>> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>>
>> Elder D. Arohuanca
>> Lima - Peru
>>
>
>
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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Daniel - Asterisk 
Users (softphones) are behind a NAT, Asterisk has its own public ip address


On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad wrote:

> asterisk is behind nat?
>
>
> On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk 
> wrote:
>
>> Hello everyone,
>>
>> I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
>> calls are working fine, but outgoing ones show the gollowing messages when
>> are being dropped:
>>
>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
>> Retransmission timeout reached on transmission
>> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
>> Response) -- See
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>> Packet timed out after 6399ms with no response
>> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
>> up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
>> critical packet (see
>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>> This is happening with my PBX hosted on an external network and peers on
>> my local network.
>>
>> It seems the SIP ACK is not being received properly.
>>
>> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>>
>> Elder D. Arohuanca
>> Lima - Peru
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
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>>
>
>
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[asterisk-users] Polycom and forwarding.

2013-05-15 Thread Ken D'Ambrosio 
Hey, all.  I've got an office set up with Asterisk, and forwarding's got 
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then hang 
up.  If the remote phone doesn't connect, it goes to the original 
phone's VM.  Is this Polycom's "fault," or Asterisk's?  I've been 
reading up on blind/supervised forwards, and, honestly, have myself more 
confused than when I started.  If someone could give me a solid idea of 
how forwarding works, and a sample of how to send it to a remote 
extension, and have it *not* come back to the original extension, that'd 
be great.


Thanks,

-Ken

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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Gertjan Baarda 
When the call is snswered, is there 2-way audio? Seems a natting issue.

On Wednesday, May 15, 2013, Daniel - Asterisk wrote:

> Hello everyone,
>
> I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
> calls are working fine, but outgoing ones show the gollowing messages when
> are being dropped:
>
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
> Retransmission timeout reached on transmission
> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
> Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6399ms with no response
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
> up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
> critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> This is happening with my PBX hosted on an external network and peers on
> my local network.
>
> It seems the SIP ACK is not being received properly.
>
> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>
> Elder D. Arohuanca
> Lima - Peru
>


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Re: [asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Asghar Mohammad 
asterisk is behind nat?


On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk wrote:

> Hello everyone,
>
> I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
> calls are working fine, but outgoing ones show the gollowing messages when
> are being dropped:
>
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
> Retransmission timeout reached on transmission
> ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
> Response) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6399ms with no response
> [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
> up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
> critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> This is happening with my PBX hosted on an external network and peers on
> my local network.
>
> It seems the SIP ACK is not being received properly.
>
> I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9
>
> Elder D. Arohuanca
> Lima - Peru
>
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Cut offs on outgoing SIP calls

2013-05-15 Thread Daniel - Asterisk 
Hello everyone,

I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:

[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3670 retrans_pkt: Hanging
up call ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
This is happening with my PBX hosted on an external network and peers on my
local network.

It seems the SIP ACK is not being received properly.

I'm using asterisk 1.8.19.0 on Debian 6.0.6 and 1.8.11.1 on Centos 5.9

Elder D. Arohuanca
Lima - Peru
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Re: [asterisk-users] Initial REGISTER Request: Contains Credentials before 401

2013-05-15 Thread Matthew J. Roth 
Brian LaVallee wrote:
> 
> My SIP provider is not happy that credentials (in the Authorization header
> field) are provided in the initial REGISTER request.
> 
> The SIP provider ONLY wants the credentials AFTER rejecting the message with
> a 401.
> 
> I know it's dumb, because the RFC says that the the initial REGISTER message
> MAY include credentials.  If it fails, the proper authentication method is
> included in the 401.  I know there is nothing wrong, it is how SIP is
> supposed to work.

Who is your SIP provider?  They need to be called out so that other Asterisk
users can avoid them.  This tendency to flip the customer/vendor relationship on
its head must be discouraged.

> However I would like to keep my SIP provider from complaining.

The only thing they should complain about is if you don't pay your bill on time.

> Asterisk is "NOT SUPPORTED" by the SIP provider.

The REGISTER request was successful so, at least from a practical standpoint,
the provider does support Asterisk.  It would be ideal if all providers
officially supported Asterisk, but this is just one example of how it's not
worth trying to please everyone.

> Does anyone in the Asterisk community know how to avoid sending the
> credentials until AFTER receiving a 401?

Edit the source.  I'm sorry to be blunt, but I really can't see the developers
adding another option to "sip.conf" just to satisfy such a pointless request.

> Any suggestions would be appreciated!

Ask the provider what platforms are "supported".  Pick one of them and use it to
configure the "useragent" and "sdpsession" options in "sip.conf".  Or look for
another provider that doesn't waste your time complaining about RFC-compliant
behavior.  

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[asterisk-users] How to allow AMI access to Originate yet deny Application: System

2013-05-15 Thread Alex Villací­s Lasso 
While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, 
combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug in our system. However, there are third parties that plug into our system, and that reconfigure the 
manager.conf file to allow remote access to AMI logins that allow Originate (by default, the manager.conf remains configured to deny login to any system except localhost). I want to have a guideline on how to proceed in order to make these applications 
work, without allowing malicious users to compromise the system. I know that one way to proceed is to deny remote access to AMI, and build an application-specific proxy that will perform the Originate on behalf of the remote requester, after filtering the 
values. However, I want to know if there is a simpler way to remove the danger of code execution while allowing applications to use AMI to place calls.


The intended scenario is that a remote desktop application (for Windows) is configured with the AMI credentials, and connects over the LAN to Asterisk in order to place calls and otherwise monitor the system. The attack I want to protect against is that of 
a malicious user that collects the credentials from the desktop application and proceeds to use the Application: System trick. I know of the SSL support for AMI, but it will not protect against a malicious end user.


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Re: [asterisk-users] 3. mfcr2 channel state IDLE 0x00 and call trace log file not ended ?? (Leonardo Rivanera)

2013-05-15 Thread Mc GRATH Ricardo 
Hi Leonardo

At first should be useful to post your message at 
asterisk-r2-requ...@lists.digium.com group.
By the other way let me advice, to make an explained detail of your problem as;
Asterisk version
Openr2 version
Configurations files
Dialplan dahdi pattern detail 
Detail of the call process (inbound or outbound call failed?)
In case of outbound call failed
extension dial 
wait time  of the call process (waiting for ringback tone, busy, reorder, 
silence etc)
call trace log (for better collect information, you could move or delete old 
data log on /var/log asterisk/.)
In case of inbound call failed, it could be possible with call log and 
monitoring bit CAS signalling

Based on your explanation and information could let you know the possible 
reason of the weird issue.

By the other way, let me explain when  dahdi restart, it make hardware reset, 
therefore will stop card control software and restart again, these will cause 
lost frame connection between you box and PSTN, and call process, etc.
When dahdi restart again it recover frame between both side and set CAS control 
channel according to dahdi parameters setting on system.conf
These is for preventing to let to PSTN start call handler process during 
asterisk starting process or stop.
Another point it seems on your log information a call have been answered,  so 
it seems is working.

Good luck   
PD: 0x00 it doesn't mean idle state (it mean force release)
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
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Re: [asterisk-users] dial and bridge

2013-05-15 Thread Dan Cropp 
You could use AsyncAGI to achieve this.

 

Originate the first call (passing in some unique identifier as a variable), 
then using AMI you will see the channel data.  When you see an Event: AysncAGI 
for that channel (with that id, you have control of the call).  Send a Dial 
Action telling it to dial the call and bridge them together if the person 
answers.  If they don’t answer, you will be notified and can do something with 
the original call (play a message, hangup, etc).  If they are bridged, you can 
see how long, etc.

 

Setup an extension, naming it something like patching

 

exten => patching,1,AGI(agi:async)

 

Action: Originate
Channel: Local/300@from-internal

Async: 1 
Exten: 1

Context: patching
Data: 1973

Variable: YourUniquePatchID=1234

 

 

Using AsyncAGI and AMI, you can have full control of the call.  You do have to 
setup a very simple dial plan so Asterisk knows you are using AsyncAGI to 
control the call.

 

Have a great day!

Dan

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Tuesday, May 14, 2013 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] dial and bridge

 




Hi all,

I need some advice - I have been working on originating multiple calls using 
AMI and then joining them. 

What I want to do is:

- dial call 1 (where the caller is in a "channel" format, like SIp/1234 or 
Local/1234@ext) and "park" it somehow

- dial call 2 (where again the caller is in channel format) and join it to the 
previous call.

 

As a requirement, I cannot use the dialplan as an end-point (as I cannot change 
it) but need to use the AMI only.

 

I tried doing something like:

 

Action: Originate
Channel: Local/300@from-internal

Async: 1 
Application: Wait
Data: 1973

 

So that the call goes to 300 and then basically stays there forever, and then I 
dial again:

 

Action: Originate
Channel: Local/500@from-internal

Async: 1 
Application: Wait
Data: 1973

  

And then try to bridge the results, but it does not seem to work.

What I would like to do would be more on the lines of:

 

Originate call 1 and park it (using a park or waiting)

Originate call 2 and bridge it immediately to call1 (using the Application part)

 

But maybe I am missing something? is there anybody who has better suggestions?

 

Thanks

l.

 

 

 

 

 

 

-- 

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Test-drive WombatDialer beta @ http://wombatdialer.com 

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Re: [asterisk-users] dial and bridge

2013-05-15 Thread Ioan Indreias 
BTW - what was exactly the problem when trying to bridge the two channels
that you have sent to the wait application?


On Wed, May 15, 2013 at 4:29 PM, Ioan Indreias  wrote:

> I think you could use twice the Park action to park the channels ->
> https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Park
>
> In the end you will have to bridge the parked channels.
>
> HTH,
> Ioan
>
>
> On Wed, May 15, 2013 at 1:03 PM, Lenz Emilitri wrote:
>
>> I never actually used parking, but should it work if I call the Park
>> application as the second leg of the Originate (w/o going through the
>> dialplan)? I dont seem to be able to make it work.
>>  l.
>>
>>
>> 2013/5/15 Mitul Limbani 
>>
>>> The dial n bridge might work, but there ain't indefinite wait in that
>>> scenario.
>>> Direct calls to parking you might try Local(70X@from-internal) but I m
>>> not sure if this method works reliably.
>>>
>>>  The method I mentioned is used by vicidial and it works flawlessly, yes
>>> it comes with some computing load, however you can try the newer ConfBridge
>>> app to see if its cheaper.
>>>
>>> Mitul
>>>
>>> On Wednesday, May 15, 2013, Lenz Emilitri wrote:
>>>
 Hi Mitul,
 I agree that the dialplan way is easier, but it's a client requirement
 to avoid using it. I was wondering if there was a way to send a call
 directly to a parking slot right from the originate, because that is
 cheaper than running conferences, and then joining the second call right to
 the parked call, so that all we have to do is two originates.
 l.


 2013/5/14 Mitul Limbani 

> Dial first call and put it into a conference, then dial second call
> and put him into same conference to bridge both.
>
> However dial plan way is much more simpler.
>
> Mitul
>
>
> On Tuesday, May 14, 2013, Lenz Emilitri wrote:
>
>>
>> Hi all,
>> I need some advice - I have been working on originating multiple
>> calls using AMI and then joining them.
>> What I want to do is:
>> - dial call 1 (where the caller is in a "channel" format, like
>> SIp/1234 or Local/1234@ext) and "park" it somehow
>> - dial call 2 (where again the caller is in channel format) and join
>> it to the previous call.
>>
>> As a requirement, I cannot use the dialplan as an end-point (as I
>> cannot change it) but need to use the AMI only.
>>
>> I tried doing something like:
>>
>> Action: Originate
>> Channel: Local/300@from-internal
>> Async: 1
>> Application: Wait
>> Data: 1973
>>
>> So that the call goes to 300 and then basically stays there forever,
>> and then I dial again:
>>
>> Action: Originate
>> Channel: Local/500@from-internal
>> Async: 1
>> Application: Wait
>> Data: 1973
>>
>> And then try to bridge the results, but it does not seem to work.
>> What I would like to do would be more on the lines of:
>>
>> Originate call 1 and park it (using a park or waiting)
>> Originate call 2 and bridge it immediately to call1 (using the
>> Application part)
>>
>> But maybe I am missing something? is there anybody who has better
>> suggestions?
>>
>> Thanks
>> l.
>>
>>
>>
>>
>>
>>
>> --
>> Loway - home of QueueMetrics - http://queuemetrics.com
>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>
>
>
> --
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mi...@enterux.in
> DID: +91-22-71967121
> Cell: +91-9820332422
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com

>>>
>>>
>>> --
>>> Regards,
>>> Mitul Limbani,
>>> Chief Architech & Founder,
>>> Enterux Solutions Pvt. Ltd.
>>> 110 Reena Complex, Opp. Nathani Steel,
>>> Vidyavihar (W), Mumbai - 400 086. India
>>> http://www.enterux.com/
>>> http://www.entvoice.com/
>>> email: mi...@enterux.in
>>> DID: +91-22-71967121
>>> Cell: +91-9820332422
>>>
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs

Re: [asterisk-users] dial and bridge

2013-05-15 Thread Ioan Indreias 
I think you could use twice the Park action to park the channels ->
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Park

In the end you will have to bridge the parked channels.

HTH,
Ioan


On Wed, May 15, 2013 at 1:03 PM, Lenz Emilitri  wrote:

> I never actually used parking, but should it work if I call the Park
> application as the second leg of the Originate (w/o going through the
> dialplan)? I dont seem to be able to make it work.
> l.
>
>
> 2013/5/15 Mitul Limbani 
>
>> The dial n bridge might work, but there ain't indefinite wait in that
>> scenario.
>> Direct calls to parking you might try Local(70X@from-internal) but I m
>> not sure if this method works reliably.
>>
>> The method I mentioned is used by vicidial and it works flawlessly, yes
>> it comes with some computing load, however you can try the newer ConfBridge
>> app to see if its cheaper.
>>
>> Mitul
>>
>> On Wednesday, May 15, 2013, Lenz Emilitri wrote:
>>
>>> Hi Mitul,
>>> I agree that the dialplan way is easier, but it's a client requirement
>>> to avoid using it. I was wondering if there was a way to send a call
>>> directly to a parking slot right from the originate, because that is
>>> cheaper than running conferences, and then joining the second call right to
>>> the parked call, so that all we have to do is two originates.
>>> l.
>>>
>>>
>>> 2013/5/14 Mitul Limbani 
>>>
 Dial first call and put it into a conference, then dial second call and
 put him into same conference to bridge both.

 However dial plan way is much more simpler.

 Mitul


 On Tuesday, May 14, 2013, Lenz Emilitri wrote:

>
> Hi all,
> I need some advice - I have been working on originating multiple calls
> using AMI and then joining them.
> What I want to do is:
> - dial call 1 (where the caller is in a "channel" format, like
> SIp/1234 or Local/1234@ext) and "park" it somehow
> - dial call 2 (where again the caller is in channel format) and join
> it to the previous call.
>
> As a requirement, I cannot use the dialplan as an end-point (as I
> cannot change it) but need to use the AMI only.
>
> I tried doing something like:
>
> Action: Originate
> Channel: Local/300@from-internal
> Async: 1
> Application: Wait
> Data: 1973
>
> So that the call goes to 300 and then basically stays there forever,
> and then I dial again:
>
> Action: Originate
> Channel: Local/500@from-internal
> Async: 1
> Application: Wait
> Data: 1973
>
> And then try to bridge the results, but it does not seem to work.
> What I would like to do would be more on the lines of:
>
> Originate call 1 and park it (using a park or waiting)
> Originate call 2 and bridge it immediately to call1 (using the
> Application part)
>
> But maybe I am missing something? is there anybody who has better
> suggestions?
>
> Thanks
> l.
>
>
>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>


 --
 Regards,
 Mitul Limbani,
 Chief Architech & Founder,
 Enterux Solutions Pvt. Ltd.
 110 Reena Complex, Opp. Nathani Steel,
 Vidyavihar (W), Mumbai - 400 086. India
 http://www.enterux.com/
 http://www.entvoice.com/
 email: mi...@enterux.in
 DID: +91-22-71967121
 Cell: +91-9820332422



 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

>>>
>>>
>>>
>>> --
>>> Loway - home of QueueMetrics - http://queuemetrics.com
>>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>>
>>
>>
>> --
>> Regards,
>> Mitul Limbani,
>> Chief Architech & Founder,
>> Enterux Solutions Pvt. Ltd.
>> 110 Reena Complex, Opp. Nathani Steel,
>> Vidyavihar (W), Mumbai - 400 086. India
>> http://www.enterux.com/
>> http://www.entvoice.com/
>> email: mi...@enterux.in
>> DID: +91-22-71967121
>> Cell: +91-9820332422
>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>
> --
> _

Re: [asterisk-users] dial and bridge

2013-05-15 Thread Lenz Emilitri 
I never actually used parking, but should it work if I call the Park
application as the second leg of the Originate (w/o going through the
dialplan)? I dont seem to be able to make it work.
l.


2013/5/15 Mitul Limbani 

> The dial n bridge might work, but there ain't indefinite wait in that
> scenario.
> Direct calls to parking you might try Local(70X@from-internal) but I m
> not sure if this method works reliably.
>
> The method I mentioned is used by vicidial and it works flawlessly, yes it
> comes with some computing load, however you can try the newer ConfBridge
> app to see if its cheaper.
>
> Mitul
>
> On Wednesday, May 15, 2013, Lenz Emilitri wrote:
>
>> Hi Mitul,
>> I agree that the dialplan way is easier, but it's a client requirement to
>> avoid using it. I was wondering if there was a way to send a call directly
>> to a parking slot right from the originate, because that is cheaper than
>> running conferences, and then joining the second call right to the parked
>> call, so that all we have to do is two originates.
>> l.
>>
>>
>> 2013/5/14 Mitul Limbani 
>>
>>> Dial first call and put it into a conference, then dial second call and
>>> put him into same conference to bridge both.
>>>
>>> However dial plan way is much more simpler.
>>>
>>> Mitul
>>>
>>>
>>> On Tuesday, May 14, 2013, Lenz Emilitri wrote:
>>>

 Hi all,
 I need some advice - I have been working on originating multiple calls
 using AMI and then joining them.
 What I want to do is:
 - dial call 1 (where the caller is in a "channel" format, like SIp/1234
 or Local/1234@ext) and "park" it somehow
 - dial call 2 (where again the caller is in channel format) and join it
 to the previous call.

 As a requirement, I cannot use the dialplan as an end-point (as I
 cannot change it) but need to use the AMI only.

 I tried doing something like:

 Action: Originate
 Channel: Local/300@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 So that the call goes to 300 and then basically stays there forever,
 and then I dial again:

 Action: Originate
 Channel: Local/500@from-internal
 Async: 1
 Application: Wait
 Data: 1973

 And then try to bridge the results, but it does not seem to work.
 What I would like to do would be more on the lines of:

 Originate call 1 and park it (using a park or waiting)
 Originate call 2 and bridge it immediately to call1 (using the
 Application part)

 But maybe I am missing something? is there anybody who has better
 suggestions?

 Thanks
 l.






 --
 Loway - home of QueueMetrics - http://queuemetrics.com
 Test-drive WombatDialer beta @ http://wombatdialer.com

>>>
>>>
>>> --
>>> Regards,
>>> Mitul Limbani,
>>> Chief Architech & Founder,
>>> Enterux Solutions Pvt. Ltd.
>>> 110 Reena Complex, Opp. Nathani Steel,
>>> Vidyavihar (W), Mumbai - 400 086. India
>>> http://www.enterux.com/
>>> http://www.entvoice.com/
>>> email: mi...@enterux.in
>>> DID: +91-22-71967121
>>> Cell: +91-9820332422
>>>
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>
>>
>>
>> --
>> Loway - home of QueueMetrics - http://queuemetrics.com
>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>
>
>
> --
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mi...@enterux.in
> DID: +91-22-71967121
> Cell: +91-9820332422
>
>
>
> --
> _
> -- 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
>



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[asterisk-users] Initial REGISTER Request: Contains Credentials before 401

2013-05-15 Thread Brian LaVallee 
My SIP provider is not happy that credentials (in the Authorization header
field) are provided in the initial REGISTER request.

The SIP provider ONLY wants the credentials AFTER rejecting the message with
a 401.

I know it's dumb, because the RFC says that the the initial REGISTER message
MAY include credentials.  If it fails, the proper authentication method is
included in the 401.  I know there is nothing wrong, it is how SIP is
supposed to work.

However I would like to keep my SIP provider from complaining.  Asterisk is
"NOT SUPPORTED" by the SIP provider.

Does anyone in the Asterisk community know how to avoid sending the
credentials until AFTER receiving a 401?

Any suggestions would be appreciated!


Sincerely,
Brian LaVallee

  
# ===
# sip.conf
# Asterisk 1.8.15-cert1
# ---
; 
[general]
;
; - trucated
; 
register=>accountnum...@server.carrier.tld:secret:acco...@proxy.carrier.tld/
DID
;
; - end

# ===
# SIP REGISTER Dialog
# ---

IP 4.4.4.4.sip > 8.8.8.8.sip: UDP, length 602
REGISTER sip:server.carrier.tld SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa
Max-Forwards: 70
From: ;tag=as6c2d23d4
To: 
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 190 REGISTER
Authorization: Digest username="account", realm="carrier.tld",
algorithm=MD5, uri="sip:sip:8.8.8.8", nonce="1368595443265327",
response="0b833bff6d83337f9f88f6fb53bbcef6"
Expires: 1800
Contact: 
Content-Length: 0


IP 8.8.8.8.sip > 4.4.4.4.sip: UDP, length 469
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=aAaAaAaAaAaAaAa
From: ;tag=as6c2d23d4
To: ;tag=3577586129
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 190 REGISTER
Content-Length: 0
Date: Wed, 15 May 2013 05:55:29 GMT
WWW-Authenticate: Digest realm="carrier.tld", domain="sip:8.8.8.8",
nonce="1368597329273572", opaque="", stale=TRUE, algorithm=MD5


IP 4.4.4.4.sip > 8.8.8.8.sip: UDP, length 602
REGISTER sip:server.carrier.tld SIP/2.0
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB
Max-Forwards: 70
From: ;tag=as333ffab1
To: 
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 191 REGISTER
Authorization: Digest username="account", realm="carrier.tld",
algorithm=MD5, uri="sip:sip:8.8.8.8", nonce="1368597329273572",
response="097ee5b915cd39c1407c785fb3c06caf"
Expires: 1800
Contact: 
Content-Length: 0


IP 8.8.8.8.sip > 4.4.4.4.sip: UDP, length 373
SIP/2.0 200 OK
Via: SIP/2.0/UDP 4.4.4.4:5060;branch=BbBbBbBbBbBbBbB
From: ;tag=as333ffab1
To: 
Call-ID: 3e47b75000b0924b6c9ba5759a7cf15d@4.4.4.4
CSeq: 191 REGISTER
Contact: ;q=0;expires=1901
Content-Length: 0
Date: Wed, 15 May 2013 05:55:29 GMT

# ===





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Re: [asterisk-users] dial and bridge

2013-05-15 Thread Mitul Limbani 
The dial n bridge might work, but there ain't indefinite wait in that
scenario.
Direct calls to parking you might try Local(70X@from-internal) but I m not
sure if this method works reliably.

The method I mentioned is used by vicidial and it works flawlessly, yes it
comes with some computing load, however you can try the newer ConfBridge
app to see if its cheaper.

Mitul

On Wednesday, May 15, 2013, Lenz Emilitri wrote:

> Hi Mitul,
> I agree that the dialplan way is easier, but it's a client requirement to
> avoid using it. I was wondering if there was a way to send a call directly
> to a parking slot right from the originate, because that is cheaper than
> running conferences, and then joining the second call right to the parked
> call, so that all we have to do is two originates.
> l.
>
>
> 2013/5/14 Mitul Limbani  'mi...@enterux.in');>>
>
>> Dial first call and put it into a conference, then dial second call and
>> put him into same conference to bridge both.
>>
>> However dial plan way is much more simpler.
>>
>> Mitul
>>
>>
>> On Tuesday, May 14, 2013, Lenz Emilitri wrote:
>>
>>>
>>> Hi all,
>>> I need some advice - I have been working on originating multiple calls
>>> using AMI and then joining them.
>>> What I want to do is:
>>> - dial call 1 (where the caller is in a "channel" format, like SIp/1234
>>> or Local/1234@ext) and "park" it somehow
>>> - dial call 2 (where again the caller is in channel format) and join it
>>> to the previous call.
>>>
>>> As a requirement, I cannot use the dialplan as an end-point (as I cannot
>>> change it) but need to use the AMI only.
>>>
>>> I tried doing something like:
>>>
>>> Action: Originate
>>> Channel: Local/300@from-internal
>>> Async: 1
>>> Application: Wait
>>> Data: 1973
>>>
>>> So that the call goes to 300 and then basically stays there forever, and
>>> then I dial again:
>>>
>>> Action: Originate
>>> Channel: Local/500@from-internal
>>> Async: 1
>>> Application: Wait
>>> Data: 1973
>>>
>>> And then try to bridge the results, but it does not seem to work.
>>> What I would like to do would be more on the lines of:
>>>
>>> Originate call 1 and park it (using a park or waiting)
>>> Originate call 2 and bridge it immediately to call1 (using the
>>> Application part)
>>>
>>> But maybe I am missing something? is there anybody who has better
>>> suggestions?
>>>
>>> Thanks
>>> l.
>>>
>>>
>>>
>>>
>>>
>>>
>>> --
>>> Loway - home of QueueMetrics - http://queuemetrics.com
>>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>>
>>
>>
>> --
>> Regards,
>> Mitul Limbani,
>> Chief Architech & Founder,
>> Enterux Solutions Pvt. Ltd.
>> 110 Reena Complex, Opp. Nathani Steel,
>> Vidyavihar (W), Mumbai - 400 086. India
>> http://www.enterux.com/
>> http://www.entvoice.com/
>> email: mi...@enterux.in 
>> DID: +91-22-71967121
>> Cell: +91-9820332422
>>
>>
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
> Test-drive WombatDialer beta @ http://wombatdialer.com
>


-- 
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967121
Cell: +91-9820332422
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_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] dial and bridge

2013-05-15 Thread Lenz Emilitri 
Hi Warren,
the problem is that all I have is two channels, so the specs might be "join
SIP/123 and SIP/345" not "join SIP/123 to 456@from-internal". They might be
Local channels, but this should be able handle the general case. The reason
why I have channels and not ext@ctxt is that I read them live from the AMI
itself. any idea on how to do this?
Thanks
l.



2013/5/14 Warren Selby 

> On Tue, May 14, 2013 at 11:16 AM, Lenz Emilitri wrote:
>
>>
>> Hi all,
>> I need some advice - I have been working on originating multiple calls
>> using AMI and then joining them.
>> What I want to do is:
>> - dial call 1 (where the caller is in a "channel" format, like SIp/1234
>> or Local/1234@ext) and "park" it somehow
>> - dial call 2 (where again the caller is in channel format) and join it
>> to the previous call.
>>
>>
>>
> Why not just originate from one extension to the other?  Something like
> this (not tested):
>
> Action: Originate
> Channel: Local/300@from-internal
> Context: from-internal
> Exten: 500
> Timeout: 30
>
> Should dial extension 500 in the from-internal context after the call to
> 300@from-internal is answered.  Meaning, the person at 300@from-internalwould 
> have their phone ring, they'd pick it up, and then they'd hear
> ringing on the line as asterisk then dialed extension 500@from-internal.
>
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
> --
> _
> -- 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
>



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Test-drive WombatDialer beta @ http://wombatdialer.com
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_
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Re: [asterisk-users] dial and bridge

2013-05-15 Thread Lenz Emilitri 
Hi Mitul,
I agree that the dialplan way is easier, but it's a client requirement to
avoid using it. I was wondering if there was a way to send a call directly
to a parking slot right from the originate, because that is cheaper than
running conferences, and then joining the second call right to the parked
call, so that all we have to do is two originates.
l.


2013/5/14 Mitul Limbani 

> Dial first call and put it into a conference, then dial second call and
> put him into same conference to bridge both.
>
> However dial plan way is much more simpler.
>
> Mitul
>
>
> On Tuesday, May 14, 2013, Lenz Emilitri wrote:
>
>>
>> Hi all,
>> I need some advice - I have been working on originating multiple calls
>> using AMI and then joining them.
>> What I want to do is:
>> - dial call 1 (where the caller is in a "channel" format, like SIp/1234
>> or Local/1234@ext) and "park" it somehow
>> - dial call 2 (where again the caller is in channel format) and join it
>> to the previous call.
>>
>> As a requirement, I cannot use the dialplan as an end-point (as I cannot
>> change it) but need to use the AMI only.
>>
>> I tried doing something like:
>>
>> Action: Originate
>> Channel: Local/300@from-internal
>> Async: 1
>> Application: Wait
>> Data: 1973
>>
>> So that the call goes to 300 and then basically stays there forever, and
>> then I dial again:
>>
>> Action: Originate
>> Channel: Local/500@from-internal
>> Async: 1
>> Application: Wait
>> Data: 1973
>>
>> And then try to bridge the results, but it does not seem to work.
>> What I would like to do would be more on the lines of:
>>
>> Originate call 1 and park it (using a park or waiting)
>> Originate call 2 and bridge it immediately to call1 (using the
>> Application part)
>>
>> But maybe I am missing something? is there anybody who has better
>> suggestions?
>>
>> Thanks
>> l.
>>
>>
>>
>>
>>
>>
>> --
>> Loway - home of QueueMetrics - http://queuemetrics.com
>> Test-drive WombatDialer beta @ http://wombatdialer.com
>>
>
>
> --
> Regards,
> Mitul Limbani,
> Chief Architech & Founder,
> Enterux Solutions Pvt. Ltd.
> 110 Reena Complex, Opp. Nathani Steel,
> Vidyavihar (W), Mumbai - 400 086. India
> http://www.enterux.com/
> http://www.entvoice.com/
> email: mi...@enterux.in
> DID: +91-22-71967121
> Cell: +91-9820332422
>
>
>
> --
> _
> -- 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
>



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