[asterisk-users] Distinctive Ring
Good morning, I'm trying to distinctive ring internal/external the channel bank FXS, after some research that has to be checked by dahdi, but I can not use someone could tell me how should I proceed ?? Thanks [ ]'s Willian Castello de Alcantara Ensite Telecom mail: will...@ensite.com.br msn: williancaste...@yahoo.com.br skype: williancastello Fone: (18) 3643-1214 (18) 9143-0691 - (11) 6488-4769 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive Ring on parked call timeout
Hi, Is there a way to have a distinctive ring for the polycom phones when the timeout is reached on a parked call? I have google this questions to no success! Thanks in advance! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
On 19 July 2010 00:35, Anthony Messina amess...@messinet.com wrote: On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Quite a lot seam to follow the Bellcore stand says the rhythmn of the ring tone, but not the tune, so Bellcore-dr2 might be long long short and bellcore-dr3 might be short short. A type or Morse code I guess... But its hard work to notice the difference in a hurry when you need to answer the phone, hence its not normally enough. In an ideal world you should be able to send the ring tone with the call so sending a URL or embedding it in the sip header, but I've not heard any method to do this. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
On Monday, July 19, 2010 01:03:57 am Peter Childs wrote: One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Agreed. I only mentioned what I did since I, along with the OP use Aastra phones. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
On Wed, 14 Jul 2010, bruce bruce wrote: Thanks for the input but that won't be good because people are not going to remember two extensions for one person. People don't have to - that's what computers are for... This wouldn't be hard to do in the dialplan, but it would need some custom dialplan programmng. The sip header should be able to carry alert_info to internal extensions really easily. Anyone else got a thought? I do it in my systems, but have only used Snom and Grandstream phones - you need to know the right SIP header to poke at the phone before you Dial() it. (And you have to set some parameters in the GS phones too) So for a Snom, it's: http://127.0.0.1\;info=alert-external\;x-line-id=0 (the id=0 is the index into the 9 ring tones they have) And in the dialplan: SIPAddHeader(Call-Info: http://127.0.0.1\;info=alert-external\;x-line-id=0) So read the manual for the phone to find out what format it requires the ring-tone in. Grandtreams are: http://127.0.0.1\;info=Ringer However the text 'Ringer' has to be programmed into the 3 ring-tone slots, so I put ring1, ring2 and ring3 into the ones I setup. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go directly to extension for external ringtone to be different. So, I am looking for internal calls ringtones to be different rather than external call ringtones. Anyone has got this working? Thanks, Burce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
At 11:44 AM 7/14/2010, you wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? It's ugly, but you could give the phone two different SIP IDs and give those different ringtones. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
Thanks for the input but that won't be good because people are not going to remember two extensions for one person. The sip header should be able to carry alert_info to internal extensions really easily. Anyone else got a thought? Thanks again, On Wed, Jul 14, 2010 at 5:44 PM, Ira i...@extrasensory.com wrote: At 11:44 AM 7/14/2010, you wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? It's ugly, but you could give the phone two different SIP IDs and give those different ringtones. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring for INTERNAL calls only? How to do it?
At 03:05 PM 7/14/2010, you wrote: Thanks for the input but that won't be good because people are not going to remember two extensions for one person. That's why there's a dialplan. But the piece I'm unsure of is how the second SIP address handles more than one call. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] distinctive ring on sipura
Hi All, I'm trying to move some POTS phones from Zap to sipura. I've searched and read a several articles which suggest that something like this should work: exten = 600,1,Dial(SIP/cordless) exten = 600,n,Hangup() exten = 700,1,Set(ALERT_INFO=Bellcore-r2) exten = 700,n,Dial(SIP/cordless) exten = 700,n,Hangup() Unforutnately though 600 and 700 sound exactly the same -- normal US ring. I found this in the sipura config: Ring1 Cadence: 60(2/4) Ring2 Cadence: 60(.3/.2,1/.2,.3/4) Ring1 Name: Bellcore-r1 Ring2 Name: Bellcore-r2 So I think I'm on the right track, but just passing the variable wrong. Here's the console output during the call: -- Executing [EMAIL PROTECTED]:1] Set(SIP/office-081c5970, ALERT_INFO=Bellcore-r2) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/office-081c5970, SIP/cordless) in new stack -- Called cordless -- SIP/cordless-081cba38 is ringing This is all on 1.4.21.2. When the phone is plugged into the zap card, Zap/1r2 definitely rings differently on the phone. Thanks a bunch in advance for any help. Best regards, David ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distinctive ring on sipura
I am successfully using this in my dialplan for a number of Sipuras (modified to fit your dialplan): exten = 700,1,SIPAddHeader(Alert-Info: info=Bellcore-r2) Not saying there's no other way to get it accomplished, but this is known to work (1.4.21.2). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] distinctive ring
It depends on which type of SIP device you have that determines on how you signal a distinctive ring. You need to change the SIP Header like: exten = s,n,SIPAddHeader(Alert-Info:Bellcore-r8) where the number after the 'r' signifies a different ring tone but some devices uses different names other than Bellcore... If you are on an internal path you would set one ring and if you are on an external path set another. Fidel Garcia wrote: This one! The sound of a phone that signals a call coming from internal/external My phones are SIP, I do not know what ZAP means or what it does. Thanks for your reply! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Tuesday, July 15, 2008 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] distintive ring Am Dienstag, den 15.07.2008, 14:02 -0400 schrieb Fidel Garcia: Need to have a different TONE for any internal call (EXT OR TRANSFER) from an external (outside) call. Any suggestions? Fidel, I do not know what kind of tone you mean: The sound of a phone that signals a call coming from internal/external? The sound in the earpiece after you dialled while you wait for the other end to pick up? In the first case distinctive ring is probably the right term to search for. You will have to decide wether your phones are SIP or ZAP (or both, or different), because methods seem to differ. As a start reading point have a look at http://www.malico.com.tw/voip-info/wiki/view/Asterisk+SIP+channels.html The mailing list archives contain a lot of information *hint* Best regards Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.4.11/1553 - Release Date: 7/15/2008 5:48 AM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring for SIP?
Does anyone have experience with setting distinctive ring in SIP in such a way? I have done this for internal call with ;;;grandstream;;; exten = _12X,1,Set(_ALERT_INFO=http://127.0.0.1\;info=internal) exten = _12X,2,Dial(SIP/${EXTEN},30,tTr) http://www.grandstream.com/asteriskfaqs.html ;;;linksys;;; exten = _1XX,1,SetVar(_ALERT_INFO=Simple-1) exten = _1XX,2,Dial(SIP/${EXTEN},30,Tr) ;;;polycom;;; exten = s,n,Set(_ALERT_INFO=Custom 1) or when it doesnt works best solution is exten = s,n,,SIPAddHeader(Alert-Info: Custom 1) http://threebit.net/mail-archive/asterisk-users/msg02875.html http://www.voip-info.org/wiki/view/Polycom+auto-answer+config LL ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive Ring for SIP?
Hello, I've done a bit of research, though obviously in the wrong places. :) I'm looking to set up some type of distinctive ring for SIP phones based on the context in which the SIP phone is being called from. If that sounds confusing it isn't. I have two FXOs connected to POTS, so in essence a caller could be dialing us at one of two phone numbers. Each of those FXOs has its own context in extensions.conf. The contexts essentially grab the incoming call and ring a bunch of single line SIP phones. If no one answers then the call is sent to voicemail depending on the number that the caller dialed. The problem is that when the SIP phones ring, there's no way for us humans to tell which number the caller actually dialed. My initial thought was to make some type of distinctive ring to the SIP phones to indicate which number the caller dialed, much in the same way that the telco does it when someone has the distintive ring service. Another idea would be to set the callerid on the SIP phones to indicate which line was dialed. The drawback there is that we lose the incoming callerid from POTS. Maybe set one of line of the callerid to the incoming line and leave the other as the callerid number? Does anyone have experience with setting distinctive ring in SIP in such a way? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring for SIP?
exten = s,1,Wait(1) exten = s,n,Set(Var_Alert=${SIP_HEADER(Alert-Info)}) exten = s,n,GotoIf($[${Var_Alert} = http://127.0.0.1/Bellcore-dr3]?2,1) ;goto ext 2 if distinctive ring exten = s,n,Dial(SIP/1) - Original Message - From: Steve To: asterisk-users@lists.digium.com Subject: [asterisk-users] Distinctive Ring for SIP? Date: Tue, 8 Jul 2008 19:04:31 -0500 Hello, I've done a bit of research, though obviously in the wrong places. :) I'm looking to set up some type of distinctive ring for SIP phones based on the context in which the SIP phone is being called from. If that sounds confusing it isn't. I have two FXOs connected to POTS, so in essence a caller could be dialing us at one of two phone numbers. Each of those FXOs has its own context in extensions.conf. The contexts essentially grab the incoming call and ring a bunch of single line SIP phones. If no one answers then the call is sent to voicemail depending on the number that the caller dialed. The problem is that when the SIP phones ring, there's no way for us humans to tell which number the caller actually dialed. My initial thought was to make some type of distinctive ring to the SIP phones to indicate which number the caller dialed, much in the same way that the telco does it when someone has the distintive ring service. Another idea would be to set the callerid on the SIP phones to indicate which line was dialed. The drawback there is that we lose the incoming callerid from POTS. Maybe set one of line of the callerid to the incoming line and leave the other as the callerid number? Does anyone have experience with setting distinctive ring in SIP in such a way? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring for SIP?
At 05:04 PM 7/8/2008, you wrote: Does anyone have experience with setting distinctive ring in SIP in such a way? On my phones, this changes the ring: exten = s,n,set(_ALERT_INFO=Bellcore-dr1) exten = s,n,set(_ALERT_INFO=Bellcore-dr4) I use dr1-dr5 for the 5 rings I want. I also do this to set the called ID and not loose the actual CID: exten = s,n(prefixCID),Set(CALLERID(Name)=L1_${CALLERID(Name)}) or exten = s,n(prefixCID),Set(CALLERID(Name)=L2_${CALLERID(Name)}) Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive ring detection not detecting ring cadences
On 7/3/07, Exploding Lemur [EMAIL PROTECTED] wrote: I'm using Asterisk 1.4.5 (will try 1.4.6 on Thursday, but I don't see anything in the changelog after the 1.4.5 release dealing with distinctive ring), zaptel 1.4.3, and wanpipe 2.3.4-10 with a Sangoma A200 card. I enabled usedistinctiveringdetection in zapata.conf. However, on the Asterisk console, the output I get is: -- Detected ring pattern: 0,0,0 Okay, I've installed Asterisk 1.4.7.1, and I'm still having the same issue. Any thoughts? (config files unchanged from my previous mailing list posting) Would any other info help troubleshoot this? -Nick K ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive ring detection not detecting ring cadences
I'm using Asterisk 1.4.5 (will try 1.4.6 on Thursday, but I don't see anything in the changelog after the 1.4.5 release dealing with distinctive ring), zaptel 1.4.3, and wanpipe 2.3.4-10 with a Sangoma A200 card. I enabled usedistinctiveringdetection in zapata.conf. However, on the Asterisk console, the output I get is: -- Detected ring pattern: 0,0,0 It would appear that Asterisk is just not getting ring info from the card. It worked a few times, but not very reliably, and now not at all. Any thoughts on what the issue may be? Is there other info I can provide that will help? Config files pasted below: zapata.conf: ;autogenerated by /usr/local/sbin/config-zaptel do not hand edit ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=-7.0 group=1 callgroup=1 pickupgroup=1 immediate=no usedistinctiveringdetection=yes dring1=360,230,193 dring1context=from-intercom dring2=205,93,0 dring2context=from-zaptel dring3=0,0,0 dring3context=from-zaptel ;Sangoma A200 [slot:6 bus:1 span:1] context=from-zaptel group=0 signalling = fxs_ks channel = 1-4 zaptel.conf: # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A200 [slot:6 bus:1 span:1] fxsks=1 fxsks=2 fxsks=3 fxsks=4 wanpipe1.conf: # # WANPIPE1 Configuration File # # # Date: Mon Jul 31 17:10:23 EDT 2006 # # Note: This file was generated automatically # by /usr/local/sbin/setup-sangoma program. # # If you want to edit this file, it is # recommended that you use wancfg program # to do so. # # Sangoma Technologies Inc. # [devices] wanpipe1 = WAN_AFT_ANALOG, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort= PRI AUTO_PCISLOT= NO PCISLOT = 6 PCIBUS = 1 FE_MEDIA= FXO/FXS TDMV_LAW= MULAW TDMV_OPERMODE = FCC MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = NO ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive Ring detection and caller ID
I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called number clears. At the moment, if I try to do this, my IVR autoattendant picks up the ringback, when ideally I want to be able to make every phone in the house ring (distinctively in the case of the Zap devices but I have mastered that). I have read that this has been an issue for users in Argentina, Australia and New Zealand, and have tried patching chan_zap.c and recompiling, and enabling that patch, as people have had success with there. However, it has not made any difference here. It is more important for me to have the caller ID processing at present, but I do need distinctive ring detection as well. I use stable Debian with backports, meaning my asterisk is currently 1.2.10. All my packages are from there, patched as necessary. I hope this can be resolved fairly quickly. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring detection and caller ID
On Tue, Dec 19, 2006 at 11:22:53AM +, Phil Reynolds wrote: I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called number clears. At the moment, if I try to do this, my IVR autoattendant picks up the ringback, when ideally I want to be able to make every phone in the house ring (distinctively in the case of the Zap devices but I have mastered that). I have read that this has been an issue for users in Argentina, Australia and New Zealand, and have tried patching chan_zap.c and recompiling, and enabling that patch, as people have had success with there. However, it has not made any difference here. It is more important for me to have the caller ID processing at present, but I do need distinctive ring detection as well. I use stable Debian with backports, meaning my asterisk is currently 1.2.10. All my packages are from there, patched as necessary. I hope this can be resolved fairly quickly. Don't know about distinctive ring. As for caller ID: Have you set zapata.conf to use v23 signalling for callerid? callerid=asreceived cidsignalling=v23 cidstart=polarity -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distinctive Ring detection and caller ID
On Tue, Dec 19, 2006 at 01:45:00PM +0200, Tzafrir Cohen wrote: Don't know about distinctive ring. As for caller ID: Have you set zapata.conf to use v23 signalling for callerid? callerid=asreceived cidsignalling=v23 cidstart=polarity Yes - and it works, but breaks distinctive ring detection, as indicated. -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Distinctive ring
Hi list! I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5 (I know, I should upgrade) and in dial plan I have: exten = _64X,n,Set(_ALERT_INFO=Chirp2) exten = _64X,n,Dial(SIP/${EXTEN},30,wWtT) On Cisco in Settings = Ring type I have Chirp1 and Chirp2. By default phone is ringing sound Chirp1. For internal calls I'm using dial plan I have sent you above. Problem is that Cisco doesn't ring with Chirp2, but with slightly different Chirp1 (instead of ring, pause, ring he sounds ring ring pause). Is there any way that my Cisco 7940, thru dial plan, can ring Chirp2 instead of Chirp1? Thank you for your time! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring on SPA941
exten = 805,1,SET(_ALERT_INFO=Classic-1) exten = 805,2,Dial(SIP/210) Thankx, works a treat -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hadley Rich Sent: Wednesday, 5 April 2006 9:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Distinctive Ring on SPA941 On Wednesday 05 April 2006 11:56, Cory Hawkless wrote: Does anyone know how to set the distinctive ring on the Linksys SPA941? Try; SET(_ALERT_INFO=Classic-1) hads -- bureaucrat, n: A politician who has tenure. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring on SPA941
Does anyone know how to set the distinctive ring on the Linksys SPA941? I want to be able to dial one extension and have the phone ring with a certain tone and then dial another and have the phone ring with a different tone. I have tried the following --- exten = 802,1,SIPAddHeader(call_info=Classic-4) exten = 802,2,Dial(SIP/210) exten = 803,1,SIPAddHeader(call_info:\;ring-tone=Classic-4) exten = 803,2,Dial(SIP/210) exten = 804,1,SIPAddHeader(call_info:\;ring-tone-id=Classic-3) exten = 804,2,Dial(SIP/210) exten = 805,1,SIPAddHeader(call_info:\;ring-tone-id=4) exten = 805,2,Dial(SIP/210) -- I'm out of ideas... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring on SPA941
On Wednesday 05 April 2006 11:56, Cory Hawkless wrote: Does anyone know how to set the distinctive ring on the Linksys SPA941? Try; SET(_ALERT_INFO=Classic-1) hads -- bureaucrat, n: A politician who has tenure. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive ring?
pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Has anyone found a solution to this? I did a similar thing for a client last week. Newer versions of the GXP2000 firmware allow up to 4 accounts, so I've defined separate sip.conf entries for internal vs. external calls. So, for example, a user might be 211 internally, and 211a externally. Set account 2 to register as 211a and modify your queue handling external calls (or however you do it) to include SIP/211a as a member rather than 211. Then use the phone's web config to assign a different ringtone to accounts 1 and 2. It's a bit of a kludge, but it does work, and seems to be reliable. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring detection using SIP - Broadvoice addon line detection
Can * detect distinctive ringing on a SIP line? The reason I ask is I have broadvoice with an add on line. It does not send any type of info that I know of for the two separate lines so I can not determine which number is ringing. Broadvoice can however send distinctive ring tones so if I could intercept that I could tell which line was ringing. Or does anyone have any other ideas to offer? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring detection using SIP - Broadvoice addon line detection
Last time I checked, Broadvoice sent the Alert-Info header in the INVITE message. The main line does not have this header, an add-on line does. On 1/22/06, Robert Mann [EMAIL PROTECTED] wrote: Can * detect distinctive ringing on a SIP line? The reason I ask is I have broadvoice with an add on line. It does not send any type of info that I know of for the two separate lines so I can not determine which number is ringing. Broadvoice can however send distinctive ring tones so if I could intercept that I could tell which line was ringing. Or does anyone have any other ideas to offer? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring?
Has anyone found a solution to this? On Sun, 27 Nov 2005 01:46 am, Kristof Hardy wrote: Kerry Garrison wrote: pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Hi Kerry, I'm also using grandstreams on a few places, have the 'same' issue/question. Afaik it can't be done with the current Grandsteam firmware. (at least, you can't command the phone to use tone X, like you can do with Cisco's) You can use the phone's built-in Distinctive Ring Tone: setting (Advanced settings), but I'm not aware of any 'wildcard' you can fill in there, I only got it to work when filling in an 'exact' number. It could be that the next firmware (should have arrived end of oct) gives us distinctive ring tones and working hint leds.. Let's hope.. If you do find a way to get any working, please report back to the list, meanwhile, i'm eagerly waiting for the firmware :) cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 == ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring and zapata.conf
Thanks. Can anyone explain what the three values for the ring pattern signify? I assume it's a ring cadence pattern (in ms) but shouldn't it be 4 values (ring on, ring off, ring on, ring off) So is Asterisk ignoring the last ring off? And does Asterisk have some tolerance value for the ring timing? ie. 323 ms +/- 10ms or is it exact? e.g. dring1=323,0,0 Ring Patterns (Non-Asterisk) http://resource.intel.com/telecom/support/tnotes/gentnote/dl_soft/tn235.htm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring and zapata.conf
Does anyone have distinctive ring working with Asterisk? Could you share your zapata.conf and relevent extensions.conf? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring and zapata.conf
Robert, This configuration is working fine for me (In ontario with Bell Canada) dring1 is the 2nd ring pattern on our line, it is a double-ring dring3 is the regular ring, which I wanted to ignore but since you cant do that I just send it to a wait loop ZAPATA.CONF [channels] usercallerid=yes signalling=fxs_ks usedistinctiveringdetection=1 faxdetect=both dring1=323,0,0 dring1context=work dring2=90,0,0 dring2context=home dring3=0,0,0 dring3context=home echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 musiconhold=default channel = 1-2 EXTENSIONS.CONF (partial) [home] ;ignore 853-1073 calls unless it is a fax exten = s,1,GotoIf($[${CALLERIDNUM} = 8531073]?5:2) exten = s,2,Wait,30 exten = s,3,system(/var/lib/asterisk/agi-bin/phone_call.sh ${CALLERID}) exten = s,4,goto(s,7) exten = s,5,answer exten = s,6,Goto(fax,1) exten = s,7,Hangup exten = fax,1,Macro(home_faxreceive) [work] ; ; We start with what to do when a call first comes in. ; exten = s,1,Answer ; Answer the line exten = s,2,Wait,2 ; Wait a second, just for fun exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,AGI(openclose.agi) exten = s,6,GotoIf($[${STATUS} = closed]?10:7) exten = s,7,GotoIf($[${STATUS} = holiday]?12:8) exten = s,8,GotoIf($[${STATUS} = afternoon]?14:16) exten = s,9,Goto(s,16) exten = s,10,BackGround(goodevening) exten = s,11,Goto(s,17) exten = s,12,BackGround(holiday) exten = s,13,Goto(s,17) exten = s,14,BackGround(goodafternoon) exten = s,15,Goto(s,17) exten = s,16,BackGround(goodmorning) exten = s,17,BackGround(greeting) exten = s,18,BackGround(instruct) exten = o,1,VoiceMailMain exten = t,1,playback(goodbye) exten = t,2,Hangup exten = i,1,Playback,invalid-exten exten = i,2,Goto,s|17 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert La Ferla Sent: Tuesday, December 20, 2005 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Distinctive Ring and zapata.conf Does anyone have distinctive ring working with Asterisk? Could you share your zapata.conf and relevent extensions.conf? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring and zapata.conf
I am trying to configure zapata.conf to handle distinctive ring. Everytime someone calls my main number, I get a ring pattern of 0,0,0 which works consistently. The problem is that every time someone calls one of the other phone numbers (same number each time), I get a different ring pattern in the console. I also could not find documentation on what the three numbers in the ring pattern mean. My guess is that the first number is time spent ringing (in milliseconds), the second number is the time between rings and the third is how long the second ring rings for. Is this correct? The fact that I'm getting different numbers tells me that perhaps there's a range of values involved in the detection. Someone please elucidate! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [Asterisk-Users] Distinctive Ring Detection not working
It would be good to see the analog cards working 100% here in Aust! later, PaulH Howard Lowndes [EMAIL PROTECTED] wrote: I have a similar problem in Australia and I think it has to do with chan_zap.c Currently Digium are investigating it for me as it is in association with one of their TDM400P cards. Gonzalo Servat wrote: Hi there. I'm having a strange issue with the distinctive ring detection in Asterisk (I have a FXO card). It certainly seems to be enabled as I can see the Asterisk console spitting out the cadences (same cadence every time: 0,0,0) but the problem is that it is not waiting 2 seconds after Starting switch on Zap/1-1 like it used to, long enough to determine the cadences, presumably the reason why it is always 0,0,0 as it hasn't had enough time to detect the ring pattern. My zapata.conf looks like the following: [trunkgroups] [channels] language=es context=incoming-landline signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=8 dring1=334,146,0 dring1context=secondnumber channel = 1 I've looked through some of the chan_zap.c code to try and increase the wait period, but after making a couple of attempts at fixing it decided to leave it alone before I break something :-) Another thing I've noticed is that if I *don't* add a dring pattern for 0,0,0, when a call comes in, it tries to find the dring pattern for 0,0,0, fails to do so, so it tries to go to context ,s,1 (notice the missing context name as the first argument), fails to do so and it supposedly hangs up the chan, then detects the ringing again (it's still the same call, only in its 5th ring by now) and successfully detects a pattern different to 0,0,0. This is the only way to have it somewhat working, although it's pretty unreliable. It's coming up with quite a few different patterns, still, I shouldn't have to do it this way. A lot of people hang up after the 4th or 5th ring. Does anyone have any ideas on this? Any suggestions would be greatly appreciated. Cheers, Gonzalo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring?
Kerry Garrison wrote: pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Hi Kerry, I'm also using grandstreams on a few places, have the 'same' issue/question. Afaik it can't be done with the current Grandsteam firmware. (at least, you can't command the phone to use tone X, like you can do with Cisco's) You can use the phone's built-in Distinctive Ring Tone: setting (Advanced settings), but I'm not aware of any 'wildcard' you can fill in there, I only got it to work when filling in an 'exact' number. It could be that the next firmware (should have arrived end of oct) gives us distinctive ring tones and working hint leds.. Let's hope.. If you do find a way to get any working, please report back to the list, meanwhile, i'm eagerly waiting for the firmware :) cheers! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring?
Don't you love clients that keep asking for features after an install? I have a client that is asking about doing distinctive rings for external vs internal calls. They are using Grandstream GXP-2000 phones which (although a pain to configure) have 4 ring types. I am guessing that I would need to figure out how to tell this particular phone to use a different ring tone unless there is a way to send a stutter type ring to the phones. Any suggestions? -Kerry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Detection not working
Hi there. I'm having a strange issue with the distinctive ring detection in Asterisk (I have a FXO card). It certainly seems to be enabled as I can see the Asterisk console spitting out the cadences (same cadence every time: 0,0,0) but the problem is that it is not waiting 2 seconds after Starting switch on Zap/1-1 like it used to, long enough to determine the cadences, presumably the reason why it is always 0,0,0 as it hasn't had enough time to detect the ring pattern. My zapata.conf looks like the following: [trunkgroups] [channels] language=es context=incoming-landline signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=8 dring1=334,146,0 dring1context=secondnumber channel = 1 I've looked through some of the chan_zap.c code to try and increase the wait period, but after making a couple of attempts at fixing it decided to leave it alone before I break something :-) Another thing I've noticed is that if I *don't* add a dring pattern for 0,0,0, when a call comes in, it tries to find the dring pattern for 0,0,0, fails to do so, so it tries to go to context ,s,1 (notice the missing context name as the first argument), fails to do so and it supposedly hangs up the chan, then detects the ringing again (it's still the same call, only in its 5th ring by now) and successfully detects a pattern different to 0,0,0. This is the only way to have it somewhat working, although it's pretty unreliable. It's coming up with quite a few different patterns, still, I shouldn't have to do it this way. A lot of people hang up after the 4th or 5th ring. Does anyone have any ideas on this? Any suggestions would be greatly appreciated. Cheers, Gonzalo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Detection not working
I have a similar problem in Australia and I think it has to do with chan_zap.c Currently Digium are investigating it for me as it is in association with one of their TDM400P cards. Gonzalo Servat wrote: Hi there. I'm having a strange issue with the distinctive ring detection in Asterisk (I have a FXO card). It certainly seems to be enabled as I can see the Asterisk console spitting out the cadences (same cadence every time: 0,0,0) but the problem is that it is not waiting 2 seconds after Starting switch on Zap/1-1 like it used to, long enough to determine the cadences, presumably the reason why it is always 0,0,0 as it hasn't had enough time to detect the ring pattern. My zapata.conf looks like the following: [trunkgroups] [channels] language=es context=incoming-landline signalling=fxs_ks usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 busydetect=yes busycount=8 dring1=334,146,0 dring1context=secondnumber channel = 1 I've looked through some of the chan_zap.c code to try and increase the wait period, but after making a couple of attempts at fixing it decided to leave it alone before I break something :-) Another thing I've noticed is that if I *don't* add a dring pattern for 0,0,0, when a call comes in, it tries to find the dring pattern for 0,0,0, fails to do so, so it tries to go to context ,s,1 (notice the missing context name as the first argument), fails to do so and it supposedly hangs up the chan, then detects the ringing again (it's still the same call, only in its 5th ring by now) and successfully detects a pattern different to 0,0,0. This is the only way to have it somewhat working, although it's pretty unreliable. It's coming up with quite a few different patterns, still, I shouldn't have to do it this way. A lot of people hang up after the 4th or 5th ring. Does anyone have any ideas on this? Any suggestions would be greatly appreciated. Cheers, Gonzalo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Detection in AU
Has anyone got distinctive ring detection working for PSTN lines in Australia. I am using the latest CVS and have got zapata.conf set up thus: but it appears that the chan_zap modules is not going anywhere near that piece of code and all it returns is the default 0,0,0 [channels] context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes pulsedial = no rxgain = +10% txgain = 0 immediate = no busydetect = yes busycount = 6 progzone = au musiconhold = default usecallerid = yes sendcalleridafter=2 callerid = asreceived usedistinctiveringdetection = yes dring1=0,0,0 dring1context=default dring2=296,235,146 dring2context=default dring3=296,275,266 dring3context=default useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that works, just, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Tones
This is an Australian situation. I have a PSTN connection that has CLID presentation enabled and has two numbers assigned to it, the primary number with the standard ring cadence: 400,200,400,2000 and the secondary number with the alternative cadence: 200,400,200,400,200,1600 CLID presentation is working fine and in zapata.conf I have: usecallerid = yes usedistinctiveringdetection = yes I am trying to set up the dring and dringcontext variables in zapata.conf and am trying to identify the returned codes for the two ring tones. Unfortunately, what gets returned in the asterisk console (verbose) is: -- Starting simple switch on 'Zap/4-1' -- Detected ring pattern: 0,0,0 for both ring cadences. I have looked at the code, chan_zap.c, and can see where this gets zeroed out, but not being a C programmer I am at a loss to identify what is not happening to get the correct numbers for the two situations. All cluesticks welcomed. -- Howard. LANNet Computing Associates - Your Linux people http://lannet.com.au -- When you just want a system that works, you choose Linux; When you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Problems
Title: Message Is anyone aware problems with it not recognising ring patterns when UK CID is enabled. If I enable distinctive with without UK CID, I see a ring pattern of "246,97,0". However as soon as I enable my UK CID with the following settings, the ring pattern fails as I see "0,0,0" My UK CID settings: callerid=asreceivedusecallerid=yescidsignalling=v23cidstart=polarityukcallerid=yes Cheers Graham ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring for Agents (Was: Re: Asterisk 1.0.8)
Russell Bryant wrote: Greetings! Version 1.0.8 has been released of Asterisk, Asterisk-addons, Zaptel, and Libpri. This release contains a significant amount of bug fixes (possibly the most of the 1.0.X releases). Tarballs are available on the asterisk web site as well as the asterisk ftp server. Thanks! I appreciate the effort you put into these releases. I've upgraded from 1.0.7 to 1.0.8 and have had no problems so far. I was happy to see that the distinctive rings for queues bug was patched in this version. However, I have a related question about distinctive rings for agents, and I'm not sure whether it's a bug or working as intended. Currently, if I setvar(ALERT_INFO...) for the purposes of setting up a SIP distinctive ring, and then dial an agent extension, the ALERT_INFO variable does not make it to the SIP channel which the agent is logged in on. When I added debugging to the extension chan_agent is dialing, the ALERT_INFO didn't even make it that far. Is this the way it's supposed to work? Is there any known way around this problem? My goal is to have an agent who is a member of two queues, hear a different ring depending on what queue the call is coming from. (Yes, I'm aware of the queue announcement, and I intend to use it. But this is another requested feature which I'd like to implement if it's feasible.) Thanks, Alan Ferrency pair Networks, Inc. [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring
On Sun, May 08, 2005 at 12:15:09AM -0500, Anton Krall said: How do you configure asterisk to recognize distingtive ringing using x100p cards? Can this be done and how? Check the example in the zapata.conf file, and the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20zapata.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring
Guys. How do you configure asterisk to recognize distingtive ringing using x100p cards? Can this be done and how? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring on BT100
Hello, Is it possible to make BT100 phones ring in different ways based on where the call is coming from? The general idea is that I need the BT100 ring in 2 different ways depending on whether the call come from Zap1 or Zap2. It's because this system is for a receptionist answering two different phone lines for two separate companies and she needs to know how to greet the person on the other side ... one way that could be useful for her to recognize which line is ringing is by having a different ring tone for each. If BT100 cannot do it .. which phone can? Or is there some alternative way of helping the receptionist in this situation distinguish between the two lines? (Flash Operator Panel would not work well since she would not have it on all the time) Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring on BT100
You can only set 1 distinctive ring if by caller id. There is a tool on the website to record custom ring tone. - Original Message - From: Tomas Florian [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, April 26, 2005 1:39 AM Subject: [Asterisk-Users] Distinctive ring on BT100 Hello, Is it possible to make BT100 phones ring in different ways based on where the call is coming from? The general idea is that I need the BT100 ring in 2 different ways depending on whether the call come from Zap1 or Zap2. It's because this system is for a receptionist answering two different phone lines for two separate companies and she needs to know how to greet the person on the other side ... one way that could be useful for her to recognize which line is ringing is by having a different ring tone for each. If BT100 cannot do it .. which phone can? Or is there some alternative way of helping the receptionist in this situation distinguish between the two lines? (Flash Operator Panel would not work well since she would not have it on all the time) Thanks, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring
If I have a Wildcard X100P and Asterisk, it is possible to make it answer only the distinctive ring call of two short rings and ignore the regular incoming ring? Bill Lohr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring detection problem
I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default channel = 1 here is the debug output: Aug 18 10:36:20 NOTICE[1112767280]: chan_zap.c:5053 ss_thread: Got event 2 (Ring/Answered)... -- Detected ring pattern: 137,0,0 -- Distinctive Ring matched context internal2 -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack -- Executing VoiceMailMain("Zap/1-1", "") in new stack -- Playing 'vm-login' (language 'en') this should have passed on to the default context... 134,0,0 also hits the internal2 context. any ideas? Paul Budden [EMAIL PROTECTED] EarthLink Revolves Around You. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
just store the cids of your high paying accs and give them vip treatment or a different did to call in =) On Thu, 26 Aug 2004 12:39:49 +1200, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On 25 Aug 2004 at 21:34, Nicolas Gudino wrote: On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will always hit 1 g That's why you kinda need to make it an after call thing. LOL you could even use it in a queue... I.E. caller id starts with rating of 50 (max 100, min 0) After call press 1 for annoying, 2 for useful Then every time you press 1 their rating goes down...which could cause the queue priority to be higher...so if someone calls in with a rating of 25 and someone else with 75 you answer the 75 first! :-) Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Cadences
Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-December/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Cadences
This is something you can do when you use Sipuras (chose one of eight cadences) for your FXS ports. Once you search the list, you will find that it also works on Cisco IP Phones. In order to change the cadence based on the user/caller (or caller-id, or...) you will need some clever AGI script and a optionally database. -Original Message- From: Mike Meyer [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 4:42 PM To: Asterisk Users Group Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-Decembe r/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
- Original Message - From: Mike Meyer [EMAIL PROTECTED] To: Asterisk Users Group [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:41 PM Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-December/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Mike, I am in no way a SIP expert so please take my comments with a grain of salt, but this would require something attached to the INVITE message sent to the phone. I know that * (with a little patching) can transmit intercom=yes to a SNOM phone indicating that the incoming session wants to use the intercom function... If phones have something similar to this like ringcadence=3 then it could be passed on... I would imagine that the value you pass to the phone is specific to the make and model of phone however... Not all phones would use ringcadence= and not all models would support '3' for example... There has been some discussion of this in the bug lists and here on list, but I'm not sure as to the status of this yet in *... I know for the Cisco phones, there is a variable called ALERT_INFO you can set to change the ring cadence but that's the only phone I know of so far... It would be s cool if Digium made their own brand of phones, or at least commissioned someone like Sayson to do it for them... then we could get a completely *-compatible solution with all of the bells and whistles we want... If we want to go with our own phones, then we would have to sacrifice the fancy features but at least it would still work! Maybe they already have something like this in mind? I would be totally behind it! Especially if it spoke IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Cadences
This is what you're looking for (Sipura SPA-2000): exten = 201,1,SetVar(ALERT_INFO=bellcore-r1) exten = 201,2,Dial(SIP/201,40) exten = 301,1,SetVar(ALERT_INFO=bellcore-r4) exten = 301,2,Dial(SIP/201,40) This dials SIP-ext 201 with a different ring-cadence when you dial 301. Sipura Supports bellcore-r1..r8, but even those names (and the cadences) are configurable in config. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Distinctive Ring Cadences - Original Message - From: Mike Meyer [EMAIL PROTECTED] To: Asterisk Users Group [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:41 PM Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-Decembe r/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Mike, I am in no way a SIP expert so please take my comments with a grain of salt, but this would require something attached to the INVITE message sent to the phone. I know that * (with a little patching) can transmit intercom=yes to a SNOM phone indicating that the incoming session wants to use the intercom function... If phones have something similar to this like ringcadence=3 then it could be passed on... I would imagine that the value you pass to the phone is specific to the make and model of phone however... Not all phones would use ringcadence= and not all models would support '3' for example... There has been some discussion of this in the bug lists and here on list, but I'm not sure as to the status of this yet in *... I know for the Cisco phones, there is a variable called ALERT_INFO you can set to change the ring cadence but that's the only phone I know of so far... It would be s cool if Digium made their own brand of phones, or at least commissioned someone like Sayson to do it for them... then we could get a completely *-compatible solution with all of the bells and whistles we want... If we want to go with our own phones, then we would have to sacrifice the fancy features but at least it would still work! Maybe they already have something like this in mind? I would be totally behind it! Especially if it spoke IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Cadences
Again, sorry for the top-posting, just following the thread. Re: you can do cadences with FXS of Sipuras - this can also be done with the FXS modules for TDM400P cards. I.E. in my house there is an extension for me, 1 for my work, 1 for my wife and 1 for our tenant. Then in extensions.conf I have: [extensions] ;== MATT == exten = 1,1,macro(stdexten|1|${MATT}${STUDIO}r2${PETE}r2${ROOM}r2ZAP/5r2) exten = 4,1,macro(stdexten|4|${MATT}${STUDIO}r2ZAP/3r2${ROOM}r2ZAP/5r2) ;== PETE == exten = 2,1,macro(stdexten|2|ZAP/3r4ZAP/5r4) ;== CYNTHIA == exten = 3,1,macro(stdexten|3|${STUDIO}r3ZAP/3r3${ROOM}r3ZAP/5r3) (Sorry for the wrapping, I've seperated the lines so there is a gap between them. Oh and BTW ${MATT} is a sip connection to my computer - if I have to call somewhere I know I'll be on hold, I use my counterstrike headset and x-lite so that I don't get a sore neck from tilting my head to hold the phone for hours on end) You will note that there is an r2,r3 or r4 after the FXS lines which means that even though one of the phones is in the lounge, we all know who it's for because of the ringing. Also, you will note that Pete's (our boarder) extension does not ring in the room or studio (which has helped no end when he has received calls at 5am!). I personally would like to see more cadence types (as far as I'm aware you can change the cadences of the 4 ring patterns, but can not add more than 4 - correct me if I'm wrong). Cheers, Matt Riddell P.S. The above could easily be controlled by callerid (although it depends on how many callerid's you want to check for). If you want to check for a lot of numbers, you're probably best using a database. You could maybe use the asterisk db. I'm using a plugin in winamp that lets you press + or - to rate up or down the visual you are viewing. It would be nice if you could do something similar with asterisk. I.E. at some time (maybe after the call) you can press a number between 1 and 4 and then the number will be added to the database with that code. When a call comes in, check in db, if there call using the correct cadence. End result?: Call comes in, annoying person on line, press 4 after, next time you know it's an annoying person. Your new girlfriend rings...press 1...next time she calls allow the phones to ring for 60 seconds instead of the usual 5 :-) On 25 Aug 2004 at 17:50, Jay Milk wrote: This is something you can do when you use Sipuras (chose one of eight cadences) for your FXS ports. Once you search the list, you will find that it also works on Cisco IP Phones. In order to change the cadence based on the user/caller (or caller-id, or...) you will need some clever AGI script and a optionally database. -Original Message- From: Mike Meyer [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 4:42 PM To: Asterisk Users Group Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-Decembe r/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! -Chris - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Chris Shaw' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 4:42 PM Subject: RE: [Asterisk-Users] Distinctive Ring Cadences This is what you're looking for (Sipura SPA-2000): exten = 201,1,SetVar(ALERT_INFO=bellcore-r1) exten = 201,2,Dial(SIP/201,40) exten = 301,1,SetVar(ALERT_INFO=bellcore-r4) exten = 301,2,Dial(SIP/201,40) This dials SIP-ext 201 with a different ring-cadence when you dial 301. Sipura Supports bellcore-r1..r8, but even those names (and the cadences) are configurable in config. -Original Message- From: Chris Shaw [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Distinctive Ring Cadences - Original Message - From: Mike Meyer [EMAIL PROTECTED] To: Asterisk Users Group [EMAIL PROTECTED] Sent: Wednesday, August 25, 2004 2:41 PM Subject: [Asterisk-Users] Distinctive Ring Cadences Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-Decembe r/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Mike, I am in no way a SIP expert so please take my comments with a grain of salt, but this would require something attached to the INVITE message sent to the phone. I know that * (with a little patching) can transmit intercom=yes to a SNOM phone indicating that the incoming session wants to use the intercom function... If phones have something similar to this like ringcadence=3 then it could be passed on... I would imagine that the value you pass to the phone is specific to the make and model of phone however... Not all phones would use ringcadence= and not all models would support '3' for example... There has been some discussion of this in the bug lists and here on list, but I'm not sure as to the status of this yet in *... I know for the Cisco phones, there is a variable called ALERT_INFO you can set to change the ring cadence but that's the only phone I know of so far... It would be s cool if Digium made their own brand of phones, or at least commissioned someone like Sayson to do it for them... then we could get a completely *-compatible solution with all of the bells and whistles we want... If we want to go with our own phones, then we would have to sacrifice the fancy features but at least it would still work! Maybe they already have something like this in mind? I would be totally behind it! Especially if it spoke IAX! :) -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will allways hit 1 g -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Cadences
On 25 Aug 2004 at 21:34, Nicolas Gudino wrote: On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will always hit 1 g That's why you kinda need to make it an after call thing. LOL you could even use it in a queue... I.E. caller id starts with rating of 50 (max 100, min 0) After call press 1 for annoying, 2 for useful Then every time you press 1 their rating goes down...which could cause the queue priority to be higher...so if someone calls in with a rating of 25 and someone else with 75 you answer the 75 first! :-) Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] distinctive ring on SNOM 200
Hi, I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. I've defined two extensions for it in Asterisk and in the SNOM 200 configuration. In the SNOM homesettingsSIPLines config page, I have set the ringer for the first extension to ringer1, and ringer6 for the second. In homesettingsmiscellaneous, I've set Ringer selection to destination and default ringer to ringer6. Despite this, the SNOM always uses ringer6, regardless of which extension is dialed. Ringer1 is never used. What am I missing? The SNOM firmware is snom200-SIP 2.04g. Thanks for any help! - Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distinctive ring on SNOM 200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 01 August 2004 09:32 am, Dr. Michael J. Chudobiak wrote: Hi, I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. I've defined two extensions for it in Asterisk and in the SNOM 200 configuration. In the SNOM homesettingsSIPLines config page, I have set the ringer for the first extension to ringer1, and ringer6 for the second. In homesettingsmiscellaneous, I've set Ringer selection to destination and default ringer to ringer6. Despite this, the SNOM always uses ringer6, regardless of which extension is dialed. Ringer1 is never used. What am I missing? The SNOM firmware is snom200-SIP 2.04g. Thanks for any help! You may want to try the newest version 3.35. http://www.snom.com/download/share/snom200-3.35-SIP.bin. Release notes at http://www.snom.com/snom200_release_notes_en.php. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBDQ3uljK16xgETzkRAi28AJ4g11sZk0XlaYOGchhvzO41UERA7ACg3Ezg 6t2EYeaWz18efUmfhdoWlL4= =jKAq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distinctive ring on SNOM 200
The distinctive rings still fail to work after upgrading to 3.35. (However, the message-waiting indicator is much more reliable now!) - Mike I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. You may want to try the newest version 3.35. http://www.snom.com/download/share/snom200-3.35-SIP.bin. Release notes at http://www.snom.com/snom200_release_notes_en.php. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distinctive ring on SNOM 200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 01 August 2004 03:26 pm, you wrote: The distinctive rings still fail to work after upgrading to 3.35. (However, the message-waiting indicator is much more reliable now!) - Mike I'm trying to set up my SNOM 200 with extensions, with different ringtones - but it doesn't seem to work. You may want to try the newest version 3.35. http://www.snom.com/download/share/snom200-3.35-SIP.bin. Release notes at http://www.snom.com/snom200_release_notes_en.php. Under Preference (web interface) you set the rings. Note that these people need to be in the phone book as one of those types. Use Address Book to add them. I use a cvs file. Though you still have to specify Type manually through the web interface. Make one entry and save it to ge the record layout. I've reported the shortcoming to Snom. You may also notice that it now says VMail instead of WMI. : ) - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBDVtBljK16xgETzkRAotoAKDA26GL532Z6WaqOS/1Vx4H120qjQCeL/do E119bZn4xZalgP5W60XblB8= =fhD2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distinctive ring on SNOM 200
Steve, Yes, I can set distinctive-ringing-by-contact (SetupPreferences) just fine, but I would prefer to use distinctive-ringing-by-line (SetupLine2Ringtone, for instance). I far as I can tell, the per-contact ringing works and the per-line ringtone settings don't actually do anything. I guess I'll email SNOM support... - Mike Under Preference (web interface) you set the rings. Note that these people need to be in the phone book as one of those types. Use Address Book to add them. I use a cvs file. Though you still have to specify Type manually through the web interface. Make one entry and save it to ge the record layout. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] distinctive ring on SNOM 200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Sunday 01 August 2004 05:33 pm, Dr. Michael J. Chudobiak wrote: Steve, Yes, I can set distinctive-ringing-by-contact (SetupPreferences) just fine, but I would prefer to use distinctive-ringing-by-line (SetupLine2Ringtone, for instance). I far as I can tell, the per-contact ringing works and the per-line ringtone settings don't actually do anything. It's only setup to work with their own Address Book. It's not a bug, it's a missing feature. Hehe. I have contacted them about it. I guess I'll email SNOM support... - Mike Under Preference (web interface) you set the rings. Note that these people need to be in the phone book as one of those types. Use Address Book to add them. I use a cvs file. Though you still have to specify Type manually through the web interface. Make one entry and save it to ge the record layout. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBDWsMljK16xgETzkRAuF1AKDLLNOdol7U+xgFRSHLVthQP5CbTQCdH0xH 66xhQIIyuaw7IpY1mP1uNP8= =8zeQ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring availability on an IOS sip gateway
Does anyone know if a 2600 series router supports distinctive ring on an FXS as well as the alert message to send? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Detection On incoming calls
I've got a single inbound analogue line setup with 2 phone numbers and distinctive ring and I'm trying to setup distinctive ring detection to separate calls and put a distinctive ring to the extensions based on what number was called... Problem is it seems most countries send a distinctive ring then the caller ID, however here it appears a short ~50ms ring is sent, followed by a pause with caller ID *then* the proper ring/distinctive ring is sent, is there any simple way to get asterisk to ignore trying to match a distinctive ring with the first 50ms segment, and do it on the 2nd segment instead? Needless to say it was showing up as 0,0,0 ever time no matter which phone number was called... Both myself and a friend have tried coding in methods to shorten rings and flags to try and make asterisk try to set a context for the call on the first ring but this doesn't seem to work and it still gets passed off... Any help would be greatly appreciated... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive Ring Detection On incoming calls
Duane, On Thu, 25 Mar 2004, Duane wrote: Problem is it seems most countries send a distinctive ring then the caller ID, however here it appears a short ~50ms ring is sent, followed by a pause with caller ID *then* the proper ring/distinctive ring is sent, is there any simple way to get asterisk to ignore trying to match a distinctive ring with the first 50ms segment, and do it on the 2nd segment instead? http://bugs.digium.com/bug_view_page.php?bug_id=0001007 I lodged this patch some time ago, but no action -- I think it's brutally ugly, but it worked for me. By all means, try it out (or improve on it). It would be much better to change the way the code works by controlling where the CID and/or distinctive ring detection is done via configuration, rather than just repeating a block of code like I did. I never got around to making a version two (yet). There may also be a conflict with the #define DEFAULT_CIDRINGS 2 that we require here to generate proper CID data for analogue handsets attached to zaptel FXS channels. Cheers, Vic Cross ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring on 2 channels?
Here's what I'm trying to do. I have 2 zap channels (x100p). One is the house line, and the other is the business line. I have call forwarding on busy setup on BOTH lines, to call a distinctive ring number on each other line. This way, no matter which line is busy, calls roll over to the other. Basicaly a poor man's DID. dring1=0,0,0 dring1context=bell2 dring2=335,0,0 dring2context=bell context=bell2 signalling=fxs_ks callerid=asreceived channel = 1 dring1=0,0,0 dring1context=bell dring2=335,0,0 dring2context=bell2 context=bell signalling=fxs_ks callerid=asreceived channel = 2 The problem seems to be that only the first dring1 context is set, so the second channel does not see it's own dring settings... When the dring1 pattern is seen, it always goes to the bell2 context even on channel 2. Is there a better way of doing what I am trying to do? or is this a limitation of the distinctive ring code? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive ring Issues
Steven. I played a bit with the distinct ring function and noticed that * doesn't detetect disctinct ring on the very first ring. Check your log and you will see that the distinct ring output is 0,0,0 After the 2nd or so ring the actual distinct ring pattern shows up. So what happens is that on the first ring * signals 0,0,0 and you are connected to your default ringcontext. Disable the default (no pattern) setting, and make your Standard ring one of the dring patterns. Hope this helps. Alfred. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Ringwald Sent: Tuesday, January 27, 2004 8:49 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Distinctive ring Issues Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0 dring1context=dist_ring1 dring2=95,325,95 dring2context=dist_ring2 dring3=325,0 dring3context=dist_ring3 ; If no pattern is matched here is where we go. context=dist_ring0 channel = 1 I am assuming that 95 ms is a short ring and 325 ms is a long ring. In my extensions.conf file, I have the following contexts defined: [dist_ring1] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7002,SIP/ringwald) [dist_ring2] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7003,SIP/ringwald) [dist_ring3] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7005,SIP/ringwald) [default] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7001,SIP/ringwald) No matter which number I dial, I always get the [default] context on answer. Can anyone shed any light on what I am doing wrong? The PSTN line is through Qwest Business, and uses US format distinctive ring tones. Show version in the asterisk console returns: Asterisk CVS-01/27/04-19:07:39 Thank you in advance for any help! Steve Ringwald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring Issues
Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0 dring1context=dist_ring1 dring2=95,325,95 dring2context=dist_ring2 dring3=325,0 dring3context=dist_ring3 ; If no pattern is matched here is where we go. context=dist_ring0 channel = 1 I am assuming that 95 ms is a short ring and 325 ms is a long ring. In my extensions.conf file, I have the following contexts defined: [dist_ring1] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7002,SIP/ringwald) [dist_ring2] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7003,SIP/ringwald) [dist_ring3] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7005,SIP/ringwald) [default] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,5,Macro(exten-vm,7001,SIP/ringwald) No matter which number I dial, I always get the [default] context on answer. Can anyone shed any light on what I am doing wrong? The PSTN line is through Qwest Business, and uses US format distinctive ring tones. Show version in the asterisk console returns: Asterisk CVS-01/27/04-19:07:39 Thank you in advance for any help! Steve Ringwald ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring confusion
On Thursday 27 November 2003 02:14, Olle E. Johansson wrote: Richard Scobie wrote: Thanks for all the help and I found the different cadences in chan_zap.c. And for non-source code readers I believe the codes are: Cadences to choose from: 1: Quick chirp followed by normal ring 2: British style ring 3: Three short bursts 4: Long ring Right? http://www.voip-info.org/tiki-index.php?page=Asterisk+ZAP+channels Also, there's a patch in the bugtracker to allow you to define your own ring cadences in zapata.conf: http://bugs.digium.com/bug_view_page.php?bug_id=372 -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring confusion
Thanks for all the help and I found the different cadences in chan_zap.c. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] distinctive ring doesn't work
In my extensions.conf file I'm attempting to distinctively ring one of my zap channels with a different ring depending upon whether the call is received from the DID or inside extension. The DID extension looks like so: exten = 5551236543,1,Dial,Zap/28r1|20 However, when I dial in on the DID the phone is not ringing at all. The CLI shows that Zap/28 is ringing and you can hear it ringing on the outside line you call from but you can't hear it on the actual phone that is being dialed. It then goes to voicemail. Can anyone tell me what I am doing wrong here? Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive ring confusion
I am somewhat unsure as to the definition of Distinctive Ring. What I am trying to achieve is to have Zap connected phones (TDM400P) ring with different cadences depending on whether the call is incoming on the PSTN context or an IAX2 context. Googling, I find this from Mark: I've added distinctive ring support to Asterisk now (also I've added answer confirmation which is an essoterric feature that nobody but oliver will likely ever use). Now you can do the following: exten = 1,1,Dial,Zap/28 ; Ring Zap/28 normally exten = 2,1,Dial,Zap/28r1 ; Ring Zap/28 with ring #1 exten = 3,1,Dial,Zap/28r2 ; Ring Zap/28 with ring #2 and when I do a show application Dial, I see : 'r' -- indicate ringing to the calling party, pass no audio until answered Which doesn't seem to match up with what I have in mind. Assuming the former usage, and that it does what I am trying to do, do I have to define ring #1 and ring #2 somewhere, or are they hardcoded in? Thanks, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring confusion
On Wed, Nov 26, 2003 at 10:23:33AM +1300, Richard Scobie wrote: exten = 1,1,Dial,Zap/28 ; Ring Zap/28 normally exten = 2,1,Dial,Zap/28r1; Ring Zap/28 with ring #1 exten = 3,1,Dial,Zap/28r2; Ring Zap/28 with ring #2 and when I do a show application Dial, I see : 'r' -- indicate ringing to the calling party, pass no audio until answered Which doesn't seem to match up with what I have in mind. You are confusing the other parameters to Dial with modifiers that chan_zap knows about. The Dial documentation refers to something like this: exten = 1,1,Dial,Zap/1|60|r (where 60 is the timeout) The Zap/foo just gets passed to chan_zap, Dial doesn't know anything about distinctive rings. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring
Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to, say, 2 SIP clients and incoming calls with ring #2 to 1 SIP client, but somehow label incoming calls so the SIP client knows whether the call was for ring #1 or ring #2. Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring
I do not know the answer for #1, but for #2, I highly doubt it. What you could do is add something to the callerid to distinguish the calls. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gonzalo Servat Sent: Sunday, November 16, 2003 11:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Distinctive Ring Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to, say, 2 SIP clients and incoming calls with ring #2 to 1 SIP client, but somehow label incoming calls so the SIP client knows whether the call was for ring #1 or ring #2. Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring Macro Example
Cool trick! You could simplify this: [macro-std-exten] ; Caller*ID is 4 digits (internal call) exten = s/_,1,Dial(${ARG1}r2,${ARG2}) ; Caller*ID is not 4 digits (external call) exten = s,1,Dial(${ARG1},${ARG2}) ; Both of the above lines go here next exten = s,2,Voicemail(u${MACRO_EXTEN}) exten = s,3,Hangup exten = s,102,Voicemail(b${MACRO_EXTEN}) exten = s,103,Hangup For those of you running Cisco 7960's and using the ALERT_INFO stuff, You can use this version of the same thing. (I am now using this in my config thanks to Eric's example): [macro-std-exten] ; Caller*ID is 4 digits (internal call) exten = s/_,1,SetVar(ALERT_INFO=1) ; Caller*ID is not 4 digits (external call) exten = s,1,NoOp ; Both of the above lines go here next exten = s,2,Dial(${ARG1},${ARG2}) exten = s,3,Voicemail(u${MACRO_EXTEN}) exten = s,4,Hangup exten = s,103,Voicemail(b${MACRO_EXTEN}) exten = s,104,Hangup John -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, June 24, 2003 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Distinctive Ring Macro Example I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten = 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten = s/_,1,Dial(${ARG1}r2,${ARG2}) exten = s/_,2,Voicemail(u${MACRO_EXTEN}) exten = s/_,3,Hangup exten = s/_,102,Voicemail(b${MACRO_EXTEN}) exten = s/_,103,Hangup ; ; Caller*ID is not 4 digits (external call) ; exten = s,1,Dial(${ARG1},${ARG2}) exten = s,2,Voicemail(u${MACRO_EXTEN}) exten = s,3,Hangup exten = s,102,Voicemail(b${MACRO_EXTEN}) exten = s,103,Hangup --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users