Re: [asterisk-users] Early Media Issue

r > *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:

Re: [asterisk-users] Early Media Issue

terisk 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.

[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 Asteri

Re: [asterisk-users] Early media using ARI

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! (p

Re: [asterisk-users] Early media using ARI

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 h

[asterisk-users] Early media using ARI

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 importa

[asterisk-users] Early Media Dialplan Issue

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 understan

Re: [asterisk-users] Early Media Dialplan Issue

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 exe

[asterisk-users] Early Media Dialplan Issue

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 understand

Re: [asterisk-users] Early media recognition

Finally I could do it using the AMI 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 recordin

[asterisk-users] Early media recognition

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 re

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

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. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:

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

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 wrote: > part #2 > -- _ -- Bandwidt

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

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 re

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

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 inform

[asterisk-users] early media (video)

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

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

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.

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

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 th

Re: [asterisk-users] Early media and IAX2

On Tue, Aug 31, 2010 at 8:11 PM, Matt Riddell 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() t

Re: [asterisk-users] Early media and IAX2

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

[asterisk-users] Early media and IAX2

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 rin

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

+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

[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 -

Re: [asterisk-users] Early Media

art 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,

Re: [asterisk-users] Early Media

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) &g

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 accompli

Re: [asterisk-users] Early Media

- 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 wrote: Change line 2 to this: exten => 444,n,Dial(SIP/OutGoingGateway/009713045212|300|m) t

Re: [asterisk-users] Early Media

om:* 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

Re: [asterisk-users] Early Media

ial 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,-- exte

Re: [asterisk-users] Early Media

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-us

Re: [asterisk-users] Early Media

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 sta

[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 a

[asterisk-users] Early Media before 200 ok

Hi , Can anyone suggest me how to start Early Media in asterisk . -- Thanks & Regards, Rajkiran Reddy, 09825698439 , Ahmedabad , India . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBS

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

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

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

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

Re: [asterisk-users] Early Media Handling

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 answ

[asterisk-users] Early Media Handling

Hi 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 i

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

t: 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 worki

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

[Asterisk-Users] Early media after a dial command

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 calle

RE: [Asterisk-Users] Early Media Enable?

.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

[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 me

[Asterisk-Users] Early media and custom SIP return codes

Is it possible to make Asterisk: - answer a SIP call with a "183 Session Progress" code - play an audio message - quit the call with a given SIP error code (402, 403, etc...) Thanks. -- ___ __ |- [EMAIL PROTECTED] |

[Asterisk-Users] early media

Hi,all   Does asterisk support sip early media?   I have a setup asterisk for sip ATA boxs and a SIP trunk (SIP GATEWAY) for PSTN access. The ATA can call PSTN phone, cell phone, BUT it can’t receive early media. I am sure the SIP GATEWAY support early media.  If use the ATA connect to

RE: [Asterisk-Users] Early Media in 100 Ringing

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 Gatewa

[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 SI

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

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?

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

ED] 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, t

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

[Asterisk-Users] Early Media in 180 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 180 Ringing WITH SDP, but Asterisk forwards the 100 Ringing WITHOUT SDP: As you can see belo

Re: [Asterisk-Users] Early Media with Asterisk

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 answe

RE: [Asterisk-Users] Early Media with Asterisk

[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

[Asterisk-Users] Early Media with Asterisk

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=-

Re: [Asterisk-Users] Early media dectection problem

It may be a problem with your sip phone, as some doesn't support early media connect, and you just hear local ringback until the call is answered. I had exactly this kind of problem until Swissvoice (IP10s) released last firmware. Snom has no problems neither. Julian J. M. On 7/5/05, kurt x <[EMA

[Asterisk-Users] Early media dectection problem

I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt ___ Asterisk-Us

[Asterisk-Users] Early media problems...

The problem is * not supporting or handling early media.  I have looked through the sniffer traces and I see the RTP stream being setup between * and the gateway during the invite and or 183 message, but * does not setup a corresponding stream to the client until it sees an OK (200) message