Re: [asterisk-users] 180 Ringing missing

On Tue, Dec 1, 2020 at 11:24 AM marek wrote: > > Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a): > > On Tue, Dec 1, 2020 at 7:20 AM marek wrote: > >> hi, >> >> after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, >> its old. customer is very conservative...) >> >> i have

Re: [asterisk-users] 180 Ringing missing

Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a): On Tue, Dec 1, 2020 at 7:20 AM marek > wrote: hi, after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, its old. customer is very conservative...) i have problem with missing 180

Re: [asterisk-users] 180 Ringing missing

On Tue, Dec 1, 2020 at 7:20 AM marek wrote: > hi, > > after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, > its old. customer is very conservative...) > > i have problem with missing 180 Ringing > > flow is easy (PBX -> Asterisk -> SIP SBC) > > Asterisk 11 > PBX - Asterisk > ->

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > BLF is an interaction between the phones and the server. You need to > configure function buttons on the phones to display the presence state of > individual peers that have been set up on the server. > > This command in the asterisk cli will help

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 10:41, Luca Bertoncello wrote: > Ishfaq Malik schrieb: > > > BLF is an interaction between the phones and the server. You need to > > configure function buttons on the phones to display the presence state of > > individual peers

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... Regards Luca Bertoncello (lucab...@lucabert.de) --

Re: [asterisk-users] Signaling ringing on other extension

Hi Look into Busy Lamp Field/Presence Here's a starting point: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Regards Ish On 29 December 2015 at 15:27, Luca Bertoncello wrote: > Hi again! > > With the "call

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 10:03, Luca Bertoncello wrote: > Ishfaq Malik schrieb: > > Hi Ishfaq > > > Look into Busy Lamp Field/Presence > > > > Here's a starting point: > > > > >

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > The hints have to be in the same contexts in extensions.conf as defines in > the sip.conf subscribecontext which can be set per peer. Well, [anika_incoming] will be included in [default], of course... But I tried to define anika_incoming in

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 10:19, Luca Bertoncello wrote: > Ishfaq Malik schrieb: > > > The hints have to be in the same contexts in extensions.conf as defines > in > > the sip.conf subscribecontext which can be set per peer. > > Well, [anika_incoming]

Re: [asterisk-users] Signaling ringing on other extension

On 12/30/15 12:24, Luca Bertoncello wrote: Ishfaq Malik schrieb: Do you have a link to the user guide for your exact phone model? Unfortunately not... I have a Thomson ST2022, but I can just find in Internet manual for the ST2030... The administrator manual can be

Re: [asterisk-users] Signaling ringing on other extension

Patrick Laimbock schrieb: > On 12/30/15 12:24, Luca Bertoncello wrote: > > Ishfaq Malik schrieb: > > > >> Do you have a link to the user guide for your exact phone model? > > > > Unfortunately not... > > I have a Thomson ST2022, but I can just find in

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 15:04, Luca Bertoncello wrote: > Patrick Laimbock schrieb: > > > On 12/30/15 12:24, Luca Bertoncello wrote: > > > Ishfaq Malik schrieb: > > > > > >> Do you have a link to the user guide for your exact phone

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > Looks like your phones do not support it. And it is a very common feature. I think so... Maybe I can write a little program running on my PC to receive a message from Asterisk if someone calls the other phone... I'll think about that... Or maybe is

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 15:09, Luca Bertoncello wrote: > Ishfaq Malik schrieb: > > > Looks like your phones do not support it. And it is a very common > feature. > > I think so... > Maybe I can write a little program running on my PC to receive a

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > Look up fop2 Thank you very much, but I prefer a standalone application, if it's possibile... Any other suggestion? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ --

Re: [asterisk-users] Signaling ringing on other extension

On 30 December 2015 at 15:16, Luca Bertoncello wrote: > Ishfaq Malik schrieb: > > > Look up fop2 > > Thank you very much, but I prefer a standalone application, if it's > possibile... > Any other suggestion? > > Thanks > Luca Bertoncello >

Re: [asterisk-users] Signaling ringing on other extension

Ishfaq Malik schrieb: > If you can programme, create an application that logs into the asterisk box > via the AMI, read the event stream and produce an alert which it sees a > phone ringing. > > https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=4817239 It was my

Re: [asterisk-users] no ringing tone with Dial option r

Hello, >>> I'm not getting any ringing when I use option r with Dial: >>> >>> Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new >>> stack >> >> Warning, options are the 3rd arguments. >> >> You seem to have an extra comma and a non-closed double-quote. >> >>

Re: [asterisk-users] no ringing tone with Dial option r

On 11/04/2015 03:43 AM, Bertrand LUPART - Linkeo.com wrote: Hello, I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Warning, options are the 3rd arguments. You seem to have an extra comma and a non-closed

Re: [asterisk-users] no ringing tone with Dial option r

On 11/03/2015 01:11 PM, John Kiniston wrote: Have you checked your indications.conf? I've seen a missing or misconfiguration in the zone definition cause this. On Tue, Nov 3, 2015 at 11:07 AM, sean darcy > wrote: On 11/01/2015 12:38 PM,

Re: [asterisk-users] no ringing tone with Dial option r

On 11/04/2015 03:40 AM, A J Stiles wrote: On Tuesday 03 Nov 2015, sean darcy wrote: On 11/01/2015 12:38 PM, sean darcy wrote: I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Otherwise all works. The call goes

Re: [asterisk-users] no ringing tone with Dial option r

On Tuesday 03 Nov 2015, sean darcy wrote: > On 11/01/2015 12:38 PM, sean darcy wrote: > > I'm not getting any ringing when I use option r with Dial: > > > > Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in > > new stack > > > > Otherwise all works. The call goes through, good audio. >

Re: [asterisk-users] no ringing tone with Dial option r

Hello, > I'm not getting any ringing when I use option r with Dial: > > Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new > stack Warning, options are the 3rd arguments. You seem to have an extra comma and a non-closed double-quote.

Re: [asterisk-users] no ringing tone with Dial option r

On 11/01/2015 12:38 PM, sean darcy wrote: I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean FWIW, 11.18.0 on Fedora 22. sean --

Re: [asterisk-users] no ringing tone with Dial option r

Have you checked your indications.conf? I've seen a missing or misconfiguration in the zone definition cause this. On Tue, Nov 3, 2015 at 11:07 AM, sean darcy wrote: > On 11/01/2015 12:38 PM, sean darcy wrote: > >> I'm not getting any ringing when I use option r with Dial:

Re: [asterisk-users] Disabling Ringing/Alerting

Can anyone please guide us if there's any way of disabling alerting/ringing in asterisk when a call is placed to any subscriber. What we want is the channel establishment as it happens during a call progress but the subscriber should not ring. Is this possible in asterisk? I think this is

Re: [asterisk-users] Normal ringing tone for the caller, while call waiting.

On Wednesday 14 Mar 2012, NaJIm wrote: When I make a call to an extension, which is on another call, the called party (who is on call waiting) will get a BEEP sound. But the caller gets the normal ringing tone. Is there any way to have a different dialer tone for the Caller, which lets him

Re: [asterisk-users] Ghost ringing

Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, Is there any chance that you still have this firmware around or can get it? It's unavailable through the Polycom site and through their support. Thanks much in advance, Joe It`s possible the firmware

Re: [asterisk-users] Ghost ringing

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of

Re: [asterisk-users] Ghost ringing

Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We are having the strangest issue

Re: [asterisk-users] Ghost ringing

. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 12:58 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ghost ringing We

Re: [asterisk-users] Ghost ringing

...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jfratant...@iswan.net Sent: Friday, January 14, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, I originally thought the same thing; however, I

Re: [asterisk-users] Ghost ringing

: Friday, January 14, 2011 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ghost ringing Hey Mike, I originally thought the same thing; however, I have swapped their phone with another one here in the office. The one on-site is still

Re: [asterisk-users] Relay ringing sip message 180

On 6 Apr 2009, at 15:40, Khaled W. Chehab wrote: Dears Asterisk is a median server between the caller and the terminations gateway The caller send the call to asterisk à asterisk will play music on hold untill the termination gateway send 200 OK and the RTP establish My problem

Re: [asterisk-users] distictive Ringing in SIP

I'm using exten = s,42,SIPAddHeader(Alert-Info: info=Bellcore-r2) and it works just fine. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Different ringing tones ...

Having just gotten into this today, here's what I got for Grandstream, Linksys Polycom phones to all work: exten = _7XX,1,SIPAddHeader(Call-Info: sip:\;answer-after=0) ;this works for Linksys Grandstream exten = _7XX,n,SIPAddHeader(Alert-Info: Ring Answer) ;this works for

Re: [asterisk-users] 180 Ringing with SDP

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, June 18, 2007 8:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 180 Ringing with SDP On Mon, 18 Jun 2007, Jared Smith wrote

Re: [asterisk-users] 180 Ringing with SDP

-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Monday, June 18, 2007 5:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 180 Ringing with SDP On Mon, 18 Jun 2007

Re: [asterisk-users] 180 Ringing with SDP

On Tue, 19 Jun 2007, Douglas Garstang wrote: Tell that to level 3. :) Is Level3, or, more precisely, the implementors of the SIP stacks for the vendors of Level3's equipment, of a different persuasion? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel:

Re: [asterisk-users] 180 Ringing with SDP

A cursory interpretation of the RFC suggests that 180 Ringing is a message designed solely to convey ringback, and that it is the payload of the 183 response that may be used to convey additional details about the nature of the call's progress. Therefore, a 180 would be an

Re: [asterisk-users] 180 Ringing with SDP

On 6/18/07, Douglas Garstang [EMAIL PROTECTED] wrote: We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just

Re: [asterisk-users] 180 Ringing with SDP

On Mon, 18 Jun 2007, Jared Smith wrote: I could be totally off base here, but it's my understanding that a 180 is telling Asterisk to generate ringing on it's side, and that a 183 (with SDP) would tell Asterisk that the call is progressing and that it should play the early media specified

Re: [asterisk-users] Pickup() ringing extension and call waiting

On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote: Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup()

Re: [asterisk-users] Pickup() ringing extension and call waiting

On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote: Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup()

Re: [asterisk-users] Pickup() ringing extension and call waiting

On Monday 29 January 2007 03:20:16 pm Steve Davies wrote: On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote: Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to

Re: [asterisk-users] Dual Ringing Tones

Hi guys, Leo Ann Boon wrote: I have a couple of interconnected asterisk boxes connected to several providers. With one provider in particular (ATP in Australia) there are two ringing tones heard on outbound calls. It is not the end of the earth - I am not reselling our services yet - but it

Re: [asterisk-users] Dual Ringing Tones

Troy - Purple Oranges wrote: Hi all and Happy New Year. I have a couple of interconnected asterisk boxes connected to several providers. With one provider in particular (ATP in Australia) there are two ringing tones heard on outbound calls. It is not the end of the earth - I am not reselling

Re: [asterisk-users] No ringing on outgoing SIP calls.

Perhaps you can check on the dtmf code. On 7/14/06, Robert La Ferla [EMAIL PROTECTED] wrote: When I dial out, I can't hear any ringing.I am using the latest SVNcode (SVN-branch-1.2-r37458M ).Is this a problem with Asterisk? Orwith my VOIP

Re: [Asterisk-Users] No ringing being played to remote caller?

If there is an Answer(), Playback(), Background(), or anything else that answers the line before Dial(), you must have set up /etc/asterisk/indications.conf or Asterisk will not know how to do inband ringback. Derek wrote: Hi all, I've got a fairly simple setup, a 4 port zaptel T1 card with

RE: [Asterisk-Users] Re: Ringing indication not working as expected

Thanks for the advice. indications.conf is now existent and Asterisk is reloaded but the problem still persists. Reloaded? Peraphs restarted is better... DV ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

Re: [Asterisk-Users] Re: Ringing indication not working as expected

On 05/18/06 18:45 Sebastian Kayser said the following: So although the Zap interface is used for both types of external calls (snom - POST, snom - PSTN) the ringing indication to my snoms fails for calls to the PSTN. we've got the following: E1 PRI --- Asterisk ---+--- FXS Gateway --- Analog

Re: [Asterisk-Users] NO ringing tone while dialing

Hi! I have tried when the option 'm' , but I don?t want the default music on hold to be listened neither. I want nothing (silence) to be heard instead of ring, ring. Any idea how to do this??? The answer is in your question: Create a MOH music file with... silence in it. Then make a new MOH

Re: [Asterisk-Users] double ringing tone on asterisk 1.2 (workaround)

You must change in the indication.conf the country [general] country=it ; default location Simone Cittadini wrote: After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem changing

Re: [Asterisk-Users] double ringing tone on asterisk 1.2 ((better) workaround)

Matteo Piazza ha scritto: You must change in the indication.conf the country [general] country=it ; default location After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

On Sat, 14 Jan 2006, Brian Capouch wrote: Rich Adamson wrote: Since there does not seem to be anyone else complaining about the same problem, there must be something in your config that is causing it. Without specific copy/paste samples of what you've configured, no one is going

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

the problem appears no matter where I terminate the call (IAX or Zap), and I don't have that problem on a 1.0.7 connected to the same PRI lines and IAX servers , what I have to check ? looked in confif files but appears to be the same (indications, modules loaded, iax, zapata, the

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

Rich Adamson ha scritto: the problem appears no matter where I terminate the call (IAX or Zap), and I don't have that problem on a 1.0.7 connected to the same PRI lines and IAX servers , what I have to check ? looked in confif files but appears to be the same (indications, modules loaded,

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 14, 2006 6:32 AM Subject: Re: [Asterisk-Users] double ringing tone on asterisk 1.2 Rich Adamson ha scritto: the problem appears no matter where I terminate the call (IAX or Zap), and I

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

Rich Adamson wrote: Since there does not seem to be anyone else complaining about the same problem, there must be something in your config that is causing it. Without specific copy/paste samples of what you've configured, no one is going to be able to guess at what you are doing. Given the

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

Simone Cittadini wrote: While I wait for the call to be answered I hear a double ringing tone, like : expected tone : tuuu tuuu tuuu tuuu what I hear : tuuu tuuu tuuu tuuu tuuu tuuu tuuu tuuu the second tuuu I think is generated somewhere and not true, since it

Re: [Asterisk-Users] double ringing tone on asterisk 1.2

Rich Adamson ha scritto: Simone Cittadini wrote: the problem appears no matter where I terminate the call (IAX or Zap), and I don't have that problem on a 1.0.7 connected to the same PRI lines and IAX servers , what I have to check ? looked in confif files but appears to be the same

Re: [Asterisk-Users] Problem ringing multiple extensions when one is forwarded

I'm not so sure that the entire group is diverted. Lets see first: 1. How are you calling these phones? 2. Are you using Zap? 3. If the forward is to a local extensions, does the same thing happen? Also please post your CLI output. For some reason I think you are using Zap channels, and the Cisco

Re: [Asterisk-Users] Problem ringing multiple extensions when one is forwarded

Here is a bit more information. First, to clarify, you are correct the entire group is not diverted, however, since the forward is going to a direct to voicemail extension it answers immediately and that stops the group from ringing. What we need is for asterisk to completely ignore the forward

Re: [Asterisk-Users] Delayed ringing on some SIP phones

Hi :) Chris Bagnall wrote: Hello all, What I'd like to do is implement a delayed ringing strategy - i.e. if the call comes in for Company 1, only their SIP phones will ring for the first 15 seconds, then if there's not been an answer, company 2's SIP phones will also start ringing. Is there

Re: [Asterisk-Users] Delayed ringing on some SIP phones

One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI

Re: [Asterisk-Users] Delayed ringing on some SIP phones

On 10/17/2005, Chris Bagnall [EMAIL PROTECTED] wrote: Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is

Re: [Asterisk-Users] Delayed ringing on some SIP phones

On Mon, 2005-10-17 at 02:50 -0600, Rich Adamson wrote: Is there any way to do this without stopping Company 1's phones ringing (i.e. timing out the dial statement after 15 seconds)? Either this is a very simple question or I'm missing something... Wouldn't something like this work for

Re: [Asterisk-Users] Delayed ringing on some SIP phones

On Mon, 2005-10-17 at 03:01 -0500, [EMAIL PROTECTED] wrote: Bingo! You got it! Timeout the dial after X seconds - and then do a Dial to both companies for another another X seconds. Remember - busy does a jump to n+101 (some one is there...) and unavailable just goes to the next step.

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Wouldn't something like this work for you? [incoming-bri-one] exten = s,1,SetCIDName(Company 1) exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 1's 2's phones exten = s,4,Voicemail(su200) [incoming-bri-two]

Re: [Asterisk-Users] Delayed ringing on some SIP phones

Why dont you make a special extension where you could provide the delay and the numbers you want to dial? for example exten = _900X,1,Wait(${EXTEN:4:2}) exten = _900X,2,Dial(SIP/${EXTEN:5}) then in the incoming context you could dial exten =

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Wouldn't something like this work for you? [incoming-bri-one] exten = s,1,SetCIDName(Company 1) exten = s,2,Dial(SIP/200SIP/201etc.,15) ; comapny 1's phones exten = s,3,Dial(SIP/200SIP/201SIP/300SIP/301etc.,15) ; company 1's 2's phones exten = s,4,Voicemail(su200)

Re: [Asterisk-Users] Double Ringing for PRI Calls

Yes, Go into sip.conf and add this line: progressinband=no On 10/17/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, We've recently upgraded to 1.2 and for outgoing PRI calls we are now getting a SIP 180 Ringing message generated by asterisk along with the RTP audio stream with the PRI

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Why dont you make a special extension where you could provide the delay and the numbers you want to dial? exten = _900X,1,Wait(${EXTEN:4:2}) exten = _900X,2,Dial(SIP/${EXTEN:5}) then in the incoming context you could dial exten = s,1,Dial(SIP/200SIP/201LOCAL/90015300LOCAL/90015301)

Re: [Asterisk-Users] Double Ringing for PRI Calls

Matt wrote: Yes, Go into sip.conf and add this line: progressinband=no Thank you!!! My Cisco 7960's started acting weird with SIP version 7.5, so I kept them at 7.4 for this reason. Works great now! Mark ___ --Bandwidth and Colocation

RE: [Asterisk-Users] Delayed ringing on some SIP phones

Discussion' Subject: RE: [Asterisk-Users] Delayed ringing on some SIP phones Why dont you make a special extension where you could provide the delay and the numbers you want to dial? exten = _900X,1,Wait(${EXTEN:4:2}) exten = _900X,2,Dial(SIP/${EXTEN:5}) then in the incoming context you

Re: [Asterisk-Users] Distinctive ringing on Cisco 79xx

Youneed to set_ALERT_INFO. I'm using these values for the Cisco 79xx: Bellcore-BusyVerifyBellcore-StutterBellcore-MsgWaitingBellcore-dr1Bellcore-dr2Bellcore-dr3Bellcore-dr4 Bellcore-dr5 On 9/8/05, Bruce Komito [EMAIL PROTECTED] wrote: Greetings, I am trying to implement distinctive ringing on a

Re: [Asterisk-Users] Distinctive ringing on Cisco 79xx

exten = s,1,SetVar(ALERT_INFO=Bellcore-dr4) Bruce Komito wrote: Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing

Re: [Asterisk-Users] Distinctive ringing on Cisco 79xx

James, isn't there a _ before alert_info? I thought there was... On 9/8/05, James Sizemore [EMAIL PROTECTED] wrote: exten = s,1,SetVar(ALERT_INFO=Bellcore-dr4)Bruce Komito wrote: Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or

RE: [Asterisk-Users] queue - ringing members in order

Hi Sr. rrmemory i same like roundrobin, but this policy store which is the next when a call get into your system. For example with next queue: SIP/1 SIP/2 SIP/3 and roundrobin, all calls stars with SIP/1 and with rrmemory first call starts with SIP/1, second call with SIP/2 and so on.

Re: [Asterisk-Users] queue - ringing members in order

thanks for reply but i tried it out: roundrobin - is doing like this first Call rings sip/1 then sip/2 then sip/3 second Call rings sip/2 then sip/3 then sip/1 third Call rings sip/3 then sip/1 then sip/2 i want that: (this should be roundrobin) first Call rings sip/1 then sip/2 then sip/3

RE: [Asterisk-Users] queue - ringing members in order

 : Re: [Asterisk-Users] queue - ringing members in order thanks for reply but i tried it out: roundrobin - is doing like this first Call rings sip/1 then sip/2 then sip/3 second Call rings sip/2 then sip/3 then sip/1 third Call rings sip/3 then sip/1 then sip/2 i want that: (this should

Re: [Asterisk-Users] Choppy Ringing

Todd Weiser wrote: We recently changed our asterisk system to begin using G.729a as the primary codec. We have a Cisco 1700-series router which connects to the PSTN via FXO ports, along with Cisco 7940 SIP phones. Everything is working great, except... When an inbound caller calls into our

Re: [Asterisk-Users] 180 Ringing? (BUG?)

Voilà. Now i know where is the problem. I use 2 ISDN channels with a with a fritz! card and the junghanns capi drivers. The problem appears with SIP to ISDN calls. The SIP 180 ringing message doesn't appear because the ISDN PBX sends the ALERT message in-band (channel B), and not in the D

Re: [Asterisk-Users] 180 Ringing? (BUG?)

I guess that's Early Media Connect, i.e., if the phone supports that (not all do), the channels get bridged just after dial completed, (SIP 183), and what you hear is the remote ring tones (from your telco), not locally generated (as if it received SIP 180 Ringing). What IP phones are you using?

Re: [Asterisk-Users] 180 Ringing? (BUG?)

Mirko Marghitola wrote: Asterisk don't send the 180 Ringing SIP message to the calling phone when the called party is ringing. How can I force asterisk to send the ringing messages? The option 'r' in the Dial() command or the Ringing() command didn't solve the problem. Mirko Did the sip

RE: [Asterisk-Users] 180 Ringing? (BUG?)

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 180 Ringing? (BUG?) Mirko Marghitola wrote: Asterisk don't send the 180 Ringing SIP message to the calling phone when the called party is ringing. How can I force asterisk to send the ringing messages

Re: [Asterisk-Users] 180 Ringing? (BUG?)

) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Manousos Sent: Tuesday, June 07, 2005 8:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 180 Ringing? (BUG?) Mirko Marghitola wrote: Asterisk don't

Re: [Asterisk-Users] 180 Ringing?

Mirko Marghitola wrote: Asterisk don't send the 180 Ringing SIP message to the calling phone when the called party is ringing. How can I force asterisk to send the ringing messages? The option 'r' in the Dial() command or the Ringing() command didn't solve the problem. I think you have to

RE: [Asterisk-Users] Supply ringing noise to IAX callers

Whooppss after research for several hours before posting, another asterisk user passed on the answer to me. Add ,r to the Dial string over the E1 to hear the ringing on the line. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Bean Sent:

Re: [Asterisk-Users] constant ringing on Zap channels

On March 29, 2005 08:40 am, Richard Reina wrote: This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? -A. ___

Re: [Asterisk-Users] constant ringing on Zap channels

Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? I have no idea if it's configured correctly. We just kind of hooked it up when the install was done a couple months ago. -A. ___ Asterisk-Users mailing

Re: [Asterisk-Users] constant ringing on Zap channels

Since signalling info is carried in the A/B bits which is how * talks to the Adit regarding the state of each channel, any framing misconfig or timing misconfig will cause this. Perform a print config on the adit and closely compare with zapata.conf and zaptel.conf On Mar 29, 2005, at 8:02

Re: [Asterisk-Users] Extension ringing but no ringing sound asterisk

Paul Dracevich wrote: When I call from extension A on Box and to Extension A on Box B I get no ringing sound. That is under your control. The Dial command has options and one of the options (r) is to add ringing sound. ___ Asterisk-Users mailing

Re: [Asterisk-Users] Phone ringing and not going to voicemail?

On Thu, 2005-03-17 at 14:16 -0500, Matt wrote: Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable'

Re: [Asterisk-Users] Phone ringing and not going to voicemail?

Matt wrote: Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and

Re: [Asterisk-Users] Phone ringing and not going to voicemail?

Not really anything of use.. but here they are: voicemail.conf: 201 = 5929,Matt Hoppes,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes 202 = 202202,Michael Eck,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes 203 = 203203,Matthew Kiessling,[EMAIL PROTECTED],[EMAIL PROTECTED],attach=yes sip.conf:

Re: [Asterisk-Users] Phone ringing and not going to voicemail?

extensions.conf: ; Asterisk Management Portal (AMP) ; Copyright (C) 2004 Coalescent Systems Inc ; dialparties.agi (http://www.sprackett.com/asterisk/) ; Asterisk::AGI (http://asterisk.gnuinter.net/) ; gsm (http://www.ibiblio.org/pub/Linux/utils/compress/!INDEX.short.html) ; loligo sounds

Re: [Asterisk-Users] Not ringing phone that are in use

Which one did you try? the first one or the second? what was the CLI output? On Tue, 15 Mar 2005 18:40:29 -0600, Derrick Shoemake [EMAIL PROTECTED] wrote: We have a small number of phones, when a call comes in we want all the phones that aren't in use to ring. Is there a simple way to test

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