Re: [asterisk-users] Early Media Issue

[Mark Farmer]

It's a good shout but sadly hasn't helped. Thanks anyway!

The issue seems to be that our provider expects to be able to send inband
early media.
There is an OpenSIPS box between the provider & Asterisk which essentially
just routes SIP traffic so the behaviour at our end is still controlled by
Asterisk which makes the call.

Using dtmfmode=auto it seems to be possible to switch to inband if RFC2833
is not advertised in SDP but the provider just honours what we set in the
call setup, which, since we only use RFC2833 is always advertised in SDP.

ATM I think it's a provider issue, according to another environment they
should never send us inband but it seems to not be working correctly in the
case.

Regards
Mark.


On Mon, 17 Jun 2019 at 10:11, Floimair Florian 
wrote:

> Just a guess, but I suspect that this might be related with strictrtp
> setting in rtp.conf, which learns the correct source in doing so drops a
> few packets.
>
> I would try to disable strictrtp for testing purposes and if this works
> add some delay before playing back the media.
>
>
>
>
>
>
>
> With best regards
>
>
>
> *Florian Floimair *Innovation - Software-Development
>
>
> *COMMEND INTERNATIONAL GMBH *A-5020 Salzburg, Saalachstraße 51
> http://www.commend.com
>
>
>
> *Security and Communication by Commend *FN 178618z | LG Salzburg
>
>
>
> *Von: *asterisk-users  im
> Auftrag von Mark Farmer 
> *Antworten an: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Datum: *Freitag, 14. Juni 2019 um 15:15
> *An: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Betreff: *[asterisk-users] Early Media Issue
>
>
>
> Hi all
>
>
>
> I've got an issue where when I call a number that just plays early media
> back to me.
>
> Instead of hearing the full sequence of tones I hear a short ringing then
> part of the sequence. What seems odd is that I can see
> the telephone-event/8000 being passed up the chain but when it gets to
> Asterisk, it is never sent back to the phone. Instead I just see the usual
> RTP flows.
>
>
>
> I've been trying to fix this for hours, does anyone have any ideas how to
> get this working correctly?
>
>
>
> Asterisk version is 13.25.0
>
>
>
> The settings I think are relevant (I'm using chan_sip):
>
>
>
> (sip.conf)
>
> ignoresdpversion=yes
>
> internal_timing=yes
>
> progressinband=never
>
> silencesuppression=no
>
> prematuremedia=no
>
>
>
> (Per peer)
>
> progressinband=yes
>
> directrtpsetup=no
>
> dtmfmode=rfc2833
>
> directmedia=no
>
> silencesuppression=no
>
> prematuremedia=no
>
>
>
>
>
> TIA
>
> Mark.
>
>
>
> --
>
> Mark Farmer
> farm...@gmail.com
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Re: [asterisk-users] Early Media Issue

[Floimair Florian]

Just a guess, but I suspect that this might be related with strictrtp setting 
in rtp.conf, which learns the correct source in doing so drops a few packets.
I would try to disable strictrtp for testing purposes and if this works add 
some delay before playing back the media.



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users  im Auftrag von 
Mark Farmer 
Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum: Freitag, 14. Juni 2019 um 15:15
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [asterisk-users] Early Media Issue

Hi all

I've got an issue where when I call a number that just plays early media back 
to me.
Instead of hearing the full sequence of tones I hear a short ringing then part 
of the sequence. What seems odd is that I can see the telephone-event/8000 
being passed up the chain but when it gets to Asterisk, it is never sent back 
to the phone. Instead I just see the usual RTP flows.

I've been trying to fix this for hours, does anyone have any ideas how to get 
this working correctly?

Asterisk version is 13.25.0

The settings I think are relevant (I'm using chan_sip):

(sip.conf)
ignoresdpversion=yes
internal_timing=yes
progressinband=never
silencesuppression=no
prematuremedia=no

(Per peer)
progressinband=yes
directrtpsetup=no
dtmfmode=rfc2833
directmedia=no
silencesuppression=no
prematuremedia=no


TIA
Mark.

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Re: [asterisk-users] Early media using ARI

[Jöran Vinzens]

Hi, thanks for the hint.

What we have done so far:

- get an incopming call
- create a new channel
- set stuff on outgoing channel
- dial outgoing channel
- get a Dial Evente State "PROGRESS"
- push both channels into the bridge

then nothing happens by default.

we will try your suggested way! (putting both Channels into bridge before
dialing the B channel)

BR
Jöran

On Thu, Jan 17, 2019 at 4:49 PM Joshua C. Colp  wrote:

> On Thu, Jan 17, 2019, at 11:40 AM, Jöran Vinzens wrote:
> > Hi all,
> >
> > we are working on a A to B basic Call scenario with early media.
> > On that scenario we get a call from a PJSIP endpoint and we place a new
> > call using ARI. On the created channel we receive a 183 Session
> > progress where we have an announcement regarding e.g. the cost of the
> > call (it's important for us to have this announcement to inform our
> > customers about the costs).
> > Using asterisk Dialplan this is done by App Dial automatically.
> > On ARI we receive a Dial Event "PROGRESS" where we thought we put both
> > channels into a bridge and the asterisk will then forward the RTP
> > towards the "A" Client using a 183 (since the channel is not answered,
> > yet). Unfortunately nothing happens.
> >
> > We searched the documentation and we have not figured it out. There is
> > no "/ari/channel/progress" command we can use and there is no
> > "early_media=true" in pjsip.conf which would enable the desired
> > behaviour.
> >
> > We would love to get a hint in the right direction and we very much
> > appreciate any help.
>
> There's a blog post which shows how it is supposed to work[1]. It expects
> the channel to be created, then both put into the bridge, and then dialed.
> This also requires Asterisk 14 or above to operate. What version are you
> using?
>
> [1]
> https://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] Early media using ARI

[Joshua C. Colp]

On Thu, Jan 17, 2019, at 11:40 AM, Jöran Vinzens wrote:
> Hi all,
> 
> we are working on a A to B basic Call scenario with early media.
> On that scenario we get a call from a PJSIP endpoint and we place a new 
> call using ARI. On the created channel we receive a 183 Session 
> progress where we have an announcement regarding e.g. the cost of the 
> call (it's important for us to have this announcement to inform our 
> customers about the costs).
> Using asterisk Dialplan this is done by App Dial automatically.
> On ARI we receive a Dial Event "PROGRESS" where we thought we put both 
> channels into a bridge and the asterisk will then forward the RTP 
> towards the "A" Client using a 183 (since the channel is not answered, 
> yet). Unfortunately nothing happens.
> 
> We searched the documentation and we have not figured it out. There is 
> no "/ari/channel/progress" command we can use and there is no 
> "early_media=true" in pjsip.conf which would enable the desired 
> behaviour.
> 
> We would love to get a hint in the right direction and we very much 
> appreciate any help.

There's a blog post which shows how it is supposed to work[1]. It expects the 
channel to be created, then both put into the bridge, and then dialed. This 
also requires Asterisk 14 or above to operate. What version are you using?

[1] https://blogs.asterisk.org/2016/08/24/asterisk-14-ari-create-bridge-dial/

-- 
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Re: [asterisk-users] Early Media Dialplan Issue

[Bobby Hakimi]

Replace vicidial with a better dialer :)
On May 9, 2016 10:39 AM, "Dan Adkins"  wrote:

> Hello all,
>
>
>
> Our company is working with a third party predictive dialer application
> that uses Asterisk 10.8.0 as its underlying telephony engine.  For several
> months, we have had issues with the execution of the dialplan due to early
> media packets being sent from our SIP provider.  My understanding of the
> problem is that an early media message arrives and the dial plan begins its
> execution, AMD hears silence and passes the connection to an agent.  The
> agent ends up with either a constant ringing or a connection without audio.
>
>
>
> Despite an active maintenance contract, the dialer company has given up
> trying to solve the issue.  Furthermore, they have no immediate plans to
> upgrade the underlying version of Asterisk.  I do have root access to the
> server but would be hesitant to make changes outside of the Asterisk
> configuration files.
>
>
>
> My question is twofold.  First, is there anything that I could adjust in
> one of the Asterisk configuration files that would take care of this
> problem?  Alternatively, is there a hardware\software  approach to
> filtering these packets?  As a side note, we do have a couple of Adtran
> 916e routers that could be used if they would be of any help.
>
>
>
>
>
>
>
> *Dan Adkins*
>
> IT Manager - TenPoint Complete
>
>
>
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Re: [asterisk-users] Early media recognition

[David Pinedo]

Finally I could do it using the AMI Originate
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate
command
and the parameter EarlyMedia=true. So, when you throw the call, when
detects the EarlyMedia (SIP 183) the channels is bridged to context and you
can do the recording.
This works from Asterisk 11.


On Fri, Jun 27, 2014 at 11:00 AM, David Pinedo dpin...@presenceco.com
wrote:

 Hello,

 Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators
 sends an explaining audio, in situations as:
 The phone number does is not assigned
 The phone is powered off
 etc.
 The audio is sent before the call to be answered.
 So, in an automatic dialling application I'd like to recognize that audio
 to know what to do with those calls (queue them to a service, mark as wrong
 number, unavailable, etc.).

 Does Asterisk have any functionality to recognize the audio in early media
 (similar to the answer machine detection)?

 Is  there a way to record the audio in early media to implement my own
 early media detector?

 Thank you in advance

 ---
 David Pinedo




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Re: [asterisk-users] early media (video)

[Joshua Colp]

Fronc Hias wrote:

Hi!

sorry to poke in... but i haven't heard anything since posting my logs :(



No real additional thoughts. Everything looks as though it should work.

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Re: [asterisk-users] early media (video)

[Fronc Hias]

Hi!

sorry to poke in... but i haven't heard anything since posting my logs :(

any thougths on the issue Josuah? or somebody else?

thanks,
Hannes


On Thu, May 8, 2014 at 1:19 PM, Fronc Hias fronc.h...@gmail.com wrote:

 part #2

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Re: [asterisk-users] early media (video)

[Fronc Hias]

FYI: Joshua Colp already replied to my initial post of this message in
asterisk-app-dev.
he suggested to move it here (asterisk-users)

he so far stated, that early media/Video should theoretically work... but
probably no one tried this in recent times...

looking foreward to receive further information, thanks!


On Wed, May 7, 2014 at 9:09 AM, Fronc Hias fronc.h...@gmail.com wrote:

 Hi All,

 I've been looking for information on how to use asterisk and early media
 to allow for a video-preview of the caller at the callee's phone for
 days... but I haven't been too successful :(

 I found that there seems to be a company 2N Helios IP which claims
 (youtube-video) that their SIP server is able to provide early video
 (using a Grandstream 3157v2 with preview enabled), but I would like to
 have this with asterisk...

 I'm currently using asterisk 12.2.x.
 I tried with all kinds of combinations of prematuremedia and
 progressinband in sip.conf and many different
 dialplan-extension-scripts but to no avail...

 sniffing with wireshark shows me, that the caller (doorstation) is sending
 H.264 video but the RTP video stream is not passed on to the callee by
 asterisk. (establishing a direct-video - without preview - call does work
 of course)

 is it at all possible with a default asterisk installation? maybe using
 chan_pjsip instead of chan_sip?

 do the prematuremedia and progressinband properties only apply to
 audio and not to video?

 I think this feature is essential for a sip-device which is used as a
 door-station...

 any information / idea is highly appreciated!
 thanks,
 Fronc

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Re: [asterisk-users] early media (video)

[Joshua Colp]

Fronc Hias wrote:

FYI: Joshua Colp already replied to my initial post of this message in
asterisk-app-dev.
he suggested to move it here (asterisk-users)

he so far stated, that early media/Video should theoretically work...
but probably no one tried this in recent times...

looking foreward to receive further information, thanks!


If you can provide the console output with debug and dialplan I may be 
able to assist further. It would also be useful to try Asterisk 11 and 
see if this is a regression.


Cheers,

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Re: [asterisk-users] Early Media configuration doesn't seem to be working

[Maximilian Grobecker]

Hi,

on a similar setup I set in sip.conf:

prematuremedia=no
progressinband=never

in the peers configuration.
With this config you tell Asterisk not to handle inband information at
all. But: Maybe you won't get any inband error messages also.


Greetings from Wuppertal
Max Grobecker


Am 07.02.2012 11:53, schrieb Ishfaq Malik:
 Hi
 
 We are using asterisk 1.8.7.0
 
 Our Sip provider is passing us ringing via Early Media, i.e. using a SIP
 183 Session Progress, with session description message which is fine for
 the most part but some of our customers are terminating on an ISDN
 gateway which doesn't interpret this message and those customers get no
 ringing.
 
 After doing some reading on the subject I have tried the following
 set prematuremedia=yes in sip.conf
 set progressinband=never in sip.conf
 set progressinband=never in the peers configuration in question
 
 but the asterisk server still passes on the 183 message and RTP stream
 rather than converting it to a SIP 180 Ringing message.
 
 Is there a problem here or am I misunderstanding something?
 
 Thanks
 in Advance
 
 Ish 

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Re: [asterisk-users] Early media and IAX2

[Russ Dill]

On Tue, Aug 31, 2010 at 8:11 PM, Matt Riddell li...@venturevoip.com wrote:
 On 28/08/10 10:18 AM, Russ Dill wrote:
 My IAX2 trunk provider, Teliax, seems to be forcing early media. Early
 media is cool and all, but my Asterisk install doesn't seem to be
 fully supporting it. My initial setting was using Dial() to call all
 of my dahdi (TDM400P) extensions. The results were that incoming calls
 would not hear any ringing tones and the call would be ended by Teliax
 after 21 seconds.

 You could just answer the call before dialling your internal extensions.

It was problem on Teliax's end. They were very responsive and took
care of the issue quickly. I'm still confused as to why I couldn't get
Asterisk to send ringing as early media.

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Re: [asterisk-users] Early media and IAX2

[Matt Riddell]

On 28/08/10 10:18 AM, Russ Dill wrote:
 My IAX2 trunk provider, Teliax, seems to be forcing early media. Early
 media is cool and all, but my Asterisk install doesn't seem to be
 fully supporting it. My initial setting was using Dial() to call all
 of my dahdi (TDM400P) extensions. The results were that incoming calls
 would not hear any ringing tones and the call would be ended by Teliax
 after 21 seconds.

You could just answer the call before dialling your internal extensions.

-- 
Cheers,

Matt Riddell
___

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Re: [asterisk-users] early media issue from phone co.

[Trevor Hammonds]

Edwin, 
In your outbound context, you need to have the dialplan evaluate the
hangupcause variable and send an appropriate message to your callers. 

Check out the following URL for some samples that you may adapt for your
circumstance.

http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause


If you need more specific assistance, let me know.

Sincerely,
Trevor Hammonds 

-Original Message-
From: Edwin Lam
Sent: Tuesday, June 08, 2010 4:11 PM
Subject: [asterisk-users] early media issue from phone co.

hi folks. i have the following puzzle:

when i call certain cell phone# using a regular phone  POTS.
the called cell phone co. usually return a message such as
phone travel out of range or phone is busy etc. if the phone is
unreachable. now when i have the following setup:

sip phone - asterisk - PRI - phone co.

i call the same cell# and if it's unavailable. the PRI return
cause code 31 and hangup, asterisk will then send a SIP BYE to
the sip phone and the channel will simply hangup. how do i
get the message on the sip phone?


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Re: [asterisk-users] Early Media

[D Tucny]

I can't say it's always been like this, as I don't recall, but, Background
in 1.0 behaved like this, answering the channel if it wasn't already
answered and playing the sound file/s until they finished or an exten was
dialed...

in 1.0 the 'skip' option would cause playback to be skipped if the channel
was not 'up', the 'noanswer' option would cause the channel to not be
answered
in 1.2 the options became 's' for skip and 'n' for noanswer though the
original 'skip' and 'noanswer' options are still valid even in 1.6

That said, in this example, you'd never leave background as it would sit
there playing the background_song file waiting for digits to be dialled...
using the dial option would be the way...

d

2009/3/27 Danny Nicholas da...@debsinc.com

 Is this correct for all versions, or does it start at 1.4 or 1.6?  I did
 put
 a YMMV on the comment, so my answer was not to be taken as fact.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
 Sent: Thursday, March 26, 2009 1:29 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Early Media

 On Wed, 2009-03-25 at 08:34 -0500, Danny Nicholas wrote:
  YMMV, but you might try this
 
  Exten = s,1,background(background_song)
 
  Exten = s,n,Answer() ;start billing

 This is not correct.  Background() automatically answers the call if it
 hasn't been answered already.

 The way to accomplish the task the original poster asked is to use the
 m option to the Dial() application.

 --
 Jared Smith
 Training Manager
 Digium, Inc.


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Re: [asterisk-users] Early Media

[Jared Smith]

On Wed, 2009-03-25 at 08:34 -0500, Danny Nicholas wrote:
 YMMV, but you might try this
 
 Exten = s,1,background(background_song)
 
 Exten = s,n,Answer() ;start billing

This is not correct.  Background() automatically answers the call if it
hasn't been answered already.

The way to accomplish the task the original poster asked is to use the
m option to the Dial() application.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Early Media

[Danny Nicholas]

Is this correct for all versions, or does it start at 1.4 or 1.6?  I did put
a YMMV on the comment, so my answer was not to be taken as fact.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Thursday, March 26, 2009 1:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Early Media

On Wed, 2009-03-25 at 08:34 -0500, Danny Nicholas wrote:
 YMMV, but you might try this
 
 Exten = s,1,background(background_song)
 
 Exten = s,n,Answer() ;start billing

This is not correct.  Background() automatically answers the call if it
hasn't been answered already.

The way to accomplish the task the original poster asked is to use the
m option to the Dial() application.

-- 
Jared Smith
Training Manager
Digium, Inc.


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Re: [asterisk-users] Early Media

[Danny Nicholas]

YMMV, but you might try this

Exten = s,1,background(background_song)

Exten = s,n,Answer() ;start billing

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Early Media

 

Dears,

 

 

-  Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.

 

Regards

 

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
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Re: [asterisk-users] Early Media

[Khaled W. Chehab]

What I am meaning is .

 

I want to start a music on hold and dial the number  (009713045212) In the
same time and when the call is connected the music will stop  and I will
talk to the called number 

 

Exten = 444,1,--

exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|)

 

is it feasible 

 

regards

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, March 25, 2009 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Early Media

 

YMMV, but you might try this

Exten = s,1,background(background_song)

Exten = s,n,Answer() ;start billing

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Early Media

 

Dears,

 

 

-  Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.

 

Regards

 

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
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Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*



*
No employee or agent is authorized to conclude any binding agreement on behalf 
of Xplorium with another party by e-mail without express written confirmation 
by an officer of Xplorium. Any views expressed by an individual in this 
electronic message do not necessarily reflect views of Xplorium or its 
subsidiaries and associates.

This electronic message and its attachments are solely addressed to the 
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If you are not the intended addressee of this electronic message and its 
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Re: [asterisk-users] Early Media

[Danny Nicholas]

Change line 2 to this:

 

exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m)

 

this will play moh for 300 seconds or until the other end answers.  The only
issue you may have is that some carriers don't generate a proper response
when answering to the music would continue over your conversation. (ATT
conferences in particular).

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 9:36 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Early Media

 

What I am meaning is .

 

I want to start a music on hold and dial the number  (009713045212) In the
same time and when the call is connected the music will stop  and I will
talk to the called number 

 

Exten = 444,1,--

exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|)

 

is it feasible 

 

regards

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, March 25, 2009 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Early Media

 

YMMV, but you might try this

Exten = s,1,background(background_song)

Exten = s,n,Answer() ;start billing

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Early Media

 

Dears,

 

 

-  Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.

 

Regards

 

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*

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Re: [asterisk-users] Early Media

[Kinjal Dixit]

am i right in understanding that this feature is called color ring back
tone?

On Wed, Mar 25, 2009 at 8:16 PM, Danny Nicholas da...@debsinc.com wrote:

  Change line 2 to this:



 exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m)



 this will play moh for 300 seconds or until the other end answers.  The
 only issue you may have is that some carriers don’t generate a proper
 response when “answering” to the music would continue over your
 conversation. (ATT conferences in particular).
  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Khaled W. Chehab
 *Sent:* Wednesday, March 25, 2009 9:36 AM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Early Media



 What I am meaning is .



 I want to start a music on hold and dial the number  (009713045212) In the
 same time and when the call is connected the music will stop  and I will
 talk to the called number



 Exten = 444,1,--

 exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|)



 is it feasible



 regards





 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas
 *Sent:* Wednesday, March 25, 2009 3:34 PM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* Re: [asterisk-users] Early Media



 YMMV, but you might try this

 Exten = s,1,background(background_song)

 Exten = s,n,Answer() ;start billing




  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Khaled W. Chehab
 *Sent:* Wednesday, March 25, 2009 8:27 AM
 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
 *Subject:* [asterisk-users] Early Media



 Dears,





 -  Anyone know how to play an early media as (background song)
 with no billing and when the call is connected the song will stop and the
 billing starts.



 Regards




  --

 *

 No employee or agent is authorized to conclude any binding agreement on 
 behalf of Xplorium with another party by e-mail without express written 
 confirmation by an officer of Xplorium. Any views expressed by an individual 
 in this electronic message do not necessarily reflect views of Xplorium or 
 its subsidiaries and associates.


 This electronic message and its attachments are solely addressed to the 
 addressee(s), and contain confidential information protected from disclosure 
 belonging to Xplorium.


 If you are not the intended addressee of this electronic message and its 
 attachments, kindly delete it immediately from your system and notify the 
 sender by electronic mail. You must not copy this message or attachment or 
 disclose its content to any other person.


 Xplorium does not guarantee the integrity of this electronic message and any 
 of its attachments, or that they are free from computer viruses or other 
 defects.
 *


  --

 *

 No employee or agent is authorized to conclude any binding agreement on 
 behalf of Xplorium with another party by e-mail without express written 
 confirmation by an officer of Xplorium. Any views expressed by an individual 
 in this electronic message do not necessarily reflect views of Xplorium or 
 its subsidiaries and associates.


 This electronic message and its attachments are solely addressed to the 
 addressee(s), and contain confidential information protected from disclosure 
 belonging to Xplorium.


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 sender by electronic mail. You must not copy this message or attachment or 
 disclose its content to any other person.


 Xplorium does not guarantee the integrity of this electronic message and any 
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 defects.
 *

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Re: [asterisk-users] Early Media

[Danny Nicholas]

It's possible.  My understanding of it is just as another feature of Dial.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit
Sent: Wednesday, March 25, 2009 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Early Media

 

am i right in understanding that this feature is called color ring back
tone?

On Wed, Mar 25, 2009 at 8:16 PM, Danny Nicholas da...@debsinc.com wrote:

Change line 2 to this:

 

exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m)

 

this will play moh for 300 seconds or until the other end answers.  The only
issue you may have is that some carriers don't generate a proper response
when answering to the music would continue over your conversation. (ATT
conferences in particular).

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 9:36 AM


To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Subject: Re: [asterisk-users] Early Media

 

What I am meaning is .

 

I want to start a music on hold and dial the number  (009713045212) In the
same time and when the call is connected the music will stop  and I will
talk to the called number 

 

Exten = 444,1,--

exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|)

 

is it feasible 

 

regards

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, March 25, 2009 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Early Media

 

YMMV, but you might try this

Exten = s,1,background(background_song)

Exten = s,n,Answer() ;start billing

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Wednesday, March 25, 2009 8:27 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Early Media

 

Dears,

 

 

-  Anyone know how to play an early media as (background song) with
no billing and when the call is connected the song will stop and the billing
starts.

 

Regards

 

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*

 

  _  

*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*


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Re: [asterisk-users] Early media support for Asterisk behind NAT

[Johansson Olle E]

8 jan 2008 kl. 07.41 skrev Mayur:

 Hi,
I have asterisk 1.4.16 behind a NAT-FW which is using a hosted  
 SIP trunk for PSTN calling. Asterisk is configured to support nat  
 with nat=yes in sip.conf. Now the hosted PSTN Gateway supports  
 symmetric RTP and early media using 183 Session Progress. So If I  
 call a PSTN number which has IVR message played before the call is  
 connected (via 183), those media RTP packets do not reach the  
 asterisk inside till asterisk sends out media packet to the PSTN  
 gateway. I have used rtpkeepalive option and set it to 1 sec. But it  
 seems that I drop rtp voice packets in the initial instructions  
 played by the IVR.

 How do I make sure that asterisk sends RTP packets (null rtp) to the  
 PSTN gateway just after receiving the media details in 183 SDP to  
 open the firewall port from inside?

That's a very interesting question. We are able to receive media as  
soon as we send the INVITE, but I am unsure on when we actually start  
sending media. Turn on RTP debugging in your asterisk to check. I  
would assume that if you have rtpkeepalive, we should start sending as  
soon as we get somewhere to send to, which in this case is when we get  
the SDP in the 183. There might be issues with some packets being sent  
at the same time as the gateway sends 183. With QoS priority for  
media, these may arrive to the NAT before the 183 SIP reply, which  
will be a problem in this NATted situation.

There's no way we can actually send anything before the 183, so there  
will always be time between SDP exchange and first functional media  
packet in NAT situations. I always consider this when playing prompts  
and wait at least a second before important audio begins.

/O

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Re: [asterisk-users] Early Media Handling

[Noah Miller]

Hi Arun -

 using php script and Asterisk manager I'm dialing numbers and once gets
 connected send to an exten in my dial plan that plays an automated message
 but some time without answering even it goes to my exten. How can I handle
 early media in Asterisk that is I want only when user answer the call it
 should goto my specified extension.

You could put in an explicit Answer() and then a Wait(1).  That
will force asterisk to answer the call first, wait 1 second, and then
move on through the extension's priorities.


- Noah

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RE: [Asterisk-Users] Early media after a dial command

[Benjamin Lawetz]

Actually Harry, there is no setup needed. You can send early audio with the
Playback command by adding the noanswer parameter (see example). But the
other end of must support/offer it.

But back to my problem, I thought maybe if asterisk generated the ring tone
it might accept the early audio afterwards, but nope, doesn't work either.

Anybody have any ideas/explanations?

Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: April 27, 2006 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Early media after a dial command

Hi Benjamin,

How do you setup early media in asterisk ?

Harry
--- Benjamin Lawetz [EMAIL PROTECTED] a écrit :

 Hello all,
 
 I've been playing around with early audio, and I'm able to get some 
 things working
 
 We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. 
 If I do the following:
 
 Exten = i,1,Playback(ss-noservice,noanswer)
 Exten = i,2,Congestion(15)
 Exten = i,3,Hangup()
 
 The PSTN caller does not get an answered call (doesn't get billed) but 
 hears the ss-noservice message.
 
 But the early media fails when I try to do the
 following:
 
 Exten = 100,1,Dial(SIP/100,15)
 Exten = 100,2,Playback(standby,noanswer) Exten = 
 100,3,Dial(SIP/[EMAIL PROTECTED],20)
 
 The PSTN caller hears the ringing for the time of the 3 priorities 
 (20s+15s+ time of standby sound file)
 
 My guess is the cisco is receiving a 183 Ringing
 and generates (or the
 remote PSTN side generates) a ring tone until the call is answered.
 Is there any way to get to have early media passed once a ringing is 
 generated?
 Would there be a way to have asterisk generate the ring tone as early 
 media to the switch to the standby message in early media?
 
 Thanks for your help
 Benjamin
 
 
 
 
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RE: [Asterisk-Users] Early media after a dial command

[hgaillac-sip]

Hi Benjamin,

How do you setup early media in asterisk ?

Harry
--- Benjamin Lawetz [EMAIL PROTECTED] a écrit :

 Hello all,
 
 I've been playing around with early audio, and I'm
 able to get some things
 working
 
 We have PSTN calls coming in to asterisk in SIP from
 a Cisco AS5300. If I do
 the following:
 
 Exten = i,1,Playback(ss-noservice,noanswer)
 Exten = i,2,Congestion(15)
 Exten = i,3,Hangup()
 
 The PSTN caller does not get an answered call
 (doesn't get billed) but hears
 the ss-noservice message.
 
 But the early media fails when I try to do the
 following:
 
 Exten = 100,1,Dial(SIP/100,15)
 Exten = 100,2,Playback(standby,noanswer)
 Exten = 100,3,Dial(SIP/[EMAIL PROTECTED],20)
 
 The PSTN caller hears the ringing for the time of
 the 3 priorities (20s+15s+
 time of standby sound file)
 
 My guess is the cisco is receiving a 183 Ringing
 and generates (or the
 remote PSTN side generates) a ring tone until the
 call is answered.
 Is there any way to get to have early media passed
 once a ringing is
 generated?
 Would there be a way to have asterisk generate the
 ring tone as early media
 to the switch to the standby message in early
 media?
 
 Thanks for your help
 Benjamin
 
 
 
 
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RE: [Asterisk-Users] Early Media Enable?

[Nabeel Jafferali]

Early audio is played, as long as you do not have a r in your Dial
statement.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Mohammed Salim
 Sent: April 13, 2006 2:17 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Early Media Enable?
 
 Hi,
 
 I've searched almost everywhere but have not come across a solution so I
 was
 hoping one of your fine folks can help me out.
 
 The problem is that a carrier is passing me early media on calls that
 sometimes have problems connecting. For example, calls to India mobile
 might
 play an early media message saying the phone is out of reach if mobile
 is
 out of area of coverage.  Problem is that asterisk does not play this
 early
 media message and simply continues to ring indefinitely.
 
 Now I know asterisk will not open the audio streams till it gets acks from
 both sides but is there a way around it?  To open one way audio right
 away?
 Any solution for this problem?  Thanks for any help in advance.
 
 Regards,
 Mohammed Salim
 
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RE: [Asterisk-Users] Early Media in 100 Ringing

[Joshua Colp - Asterlink]

Hello Ronald,

A 180 Ringing is something that should not have SDP because it's out of band
signaling of the exact status of the call, ringing. The PSTN Gateway should
return a 183 Session Progress if it wants to deliver inband audio progress.
Their SIP implementation doesn't look the best either... so to get it to
work you'd either have to hack Asterisk, or get the manufacturer of the PSTN
gateway to fix their stuff.

Joshua Colp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Voermans
Sent: Monday, September 26, 2005 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Early Media in 100 Ringing

Hello,

I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:

As you can see below, the SIP message from 10.254.254.1 (the PSTN
Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content.

How can this be solved?

U 10.254.254.1:5060 - 192.168.0.173:5060
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.0.173:5060;rport=5060;branch=z9hG4bK454e2d35.
Record-Route: sip:[EMAIL PROTECTED]:5060.
Record-Route: sip:[EMAIL PROTECTED]:5060;lr;nat=yes.
From: 0161801019 sip:[EMAIL PROTECTED];tag=as02de1b95.
To: sip:[EMAIL PROTECTED];tag=00-04094-52dbe3bc-6cf68a723.
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE.
Contact: sip:212.241.48.70:5060.
server: Cirpack/v4.38f (gw_sip).
Allow: UPDATE, REFER.
Content-Type: application/sdp.
Content-Length: 253.
.
v=0.
o=cp10 112775383044 112775383045 IN IP4 10.166.38.109.
s=SIP Call.
c=IN IP4 10.254.254.1.
t=0 0.
m=audio 35058 RTP/AVP 18 101.
b=AS:64.
a=rtpmap:18 G729/8000/1.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000/1.
a=fmtp:101 0-15.
a=ptime:20.

#
U 192.168.0.173:5060 - 192.168.1.103:5062
SIP/2.0 180 Ringing.
Via: SIP/2.0/UDP 192.168.1.103:5062;branch=z9hG4bKff31d98edbf2b265.
From: 411 sip:[EMAIL PROTECTED];tag=f93ee2f65c6906cb.
To: sip:[EMAIL PROTECTED];tag=as675f246d.
Call-ID: [EMAIL PROTECTED]
CSeq: 60590 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
Contact: sip:[EMAIL PROTECTED].
Content-Length: 0.
.
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Re: [Asterisk-Users] Early Media in 180 Ringing

[Hauke Zuehl]

Hi :)

Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans:
 Hello,

 As you can see below, the SIP message from 10.254.254.1 (the PSTN
 Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content.

 How can this be solved?


Well, I am not that expert but AFAIK your PSTN gateway should send a 183 
(Session progress) than a simple 180.
Do you use Dial(SIP/blah|30|m(moh_class)) to start early media?

Regards,
Hauke
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RE: [Asterisk-Users] Early Media in 180 Ringing

[Ronald Voermans]

If guess I figured it out already.

I made some changes in chan_sip.c (when ringing was received, it didn't
check for SDP), and recompiled.

It's working now! 

Ronald
-

-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Hauke Zuehl
Verzonden: dinsdag 27 september 2005 10:02
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Early Media in 180 Ringing

Hi :)

Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans:
 Hello,

 As you can see below, the SIP message from 10.254.254.1 (the PSTN
 Gateway) has SDP, while * (with 192.168.0.173) removes the SDP
content.

 How can this be solved?


Well, I am not that expert but AFAIK your PSTN gateway should send a 183
(Session progress) than a simple 180.
Do you use Dial(SIP/blah|30|m(moh_class)) to start early media?

Regards,
Hauke
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Re: [Asterisk-Users] Early Media in 180 Ringing

[Kevin Bockman]

Ronald Voermans wrote:

If guess I figured it out already.

I made some changes in chan_sip.c (when ringing was received, it didn't
check for SDP), and recompiled.


I don't know what all of this means, but I'm sure it could be of value 
to others.  Can you submit your patch to bugs.digium.com?



Kevin
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RE: [Asterisk-Users] Early Media with Asterisk

[Andreas Sikkema]

[EMAIL PROTECTED] wrote:

 Now, I traced RTP packets and see how sip2.provider1.de sends
 packets to my Asterisk but the port seems closed on my server so the
 inquiring server of
 provider1 will never get an answer and sends a port unreachable.

Did provider1 send the exact same SIP message types to you 
as provider2? It looks to me like provider1 is not sending 
a 183 Session Progress message. Which is usually used for 
this kind of functionality I think.


-- 
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544  f: +31 (0)10 413 65 45
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Re: [Asterisk-Users] Early Media with Asterisk

[Hauke Zuehl]

Hi :)

Am Donnerstag, 22. September 2005 12:48 schrieb Andreas Sikkema:
 [EMAIL PROTECTED] wrote:
  Now, I traced RTP packets and see how sip2.provider1.de sends
  packets to my Asterisk but the port seems closed on my server so the
  inquiring server of
  provider1 will never get an answer and sends a port unreachable.

 Did provider1 send the exact same SIP message types to you
 as provider2? It looks to me like provider1 is not sending
 a 183 Session Progress message. Which is usually used for
 this kind of functionality I think.

Oops!
Mistake by myself:
They start with INVITE (sure!) with all the SDP stuff. I answer 100 followed 
by 183.
So I do send the 183 message to provider1 and provider2.

Oh, I've forgotten to tell: My problem are incoming calls with early media  
from PSTN to my server.

Sorry for that :)

Regards,
Hauke
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