[asterisk-users] Caller ID Sent in PAI header.
Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -- _ -- 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] Caller ID Sent in PAI header.
Hello, Usually in the P-Asserted you have the network number and in the From the preferred number. In this case the Preferred (from) number is displayed. BR Laurent De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Aziz TestAccount Envoyé : jeudi 28 janvier 2016 15:46 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Caller ID Sent in PAI header. Hi All, When receiving an invite containing two different caller ID, one in FROM header and the other in "P-Asserted Identity" Header, Which one will be used by the callee ? I couldn't find any RFC specifying this detail. Thank you. -- _ -- 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] Caller ID Sent in PAI header.
Hello, Thanks for your reply. Is this mentioned in any RFC ? I checked RFC3325 for PAI and RFC3261, but nothing mentioned there. Best regards On Thu, Jan 28, 2016 at 2:50 PM, Laurent Schweizer < laurent.schwei...@peoplefone.com> wrote: > Hello, > > > > Usually in the P-Asserted you have the network number and in the From the > preferred number. > > > > In this case the Preferred (from) number is displayed. > > > > > > BR > > > > Laurent > > > > *De :* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *De la part de* Aziz TestAccount > *Envoyé :* jeudi 28 janvier 2016 15:46 > *À :* asterisk-users@lists.digium.com > *Objet :* [asterisk-users] Caller ID Sent in PAI header. > > > > Hi All, > > When receiving an invite containing two different caller ID, one in FROM > header and the other in "P-Asserted Identity" Header, Which one will be > used by the callee ? I couldn't find any RFC specifying this detail. > > > Thank you. > > -- > _ > -- 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] caller id spoofing/setting on analog
On Friday 25 Sep 2015, Ryan, Travis wrote: > I've not used analog for quite some time. It seems it's not possible in > asterisk to spoof a phone number/name on an analog call? Probably not if you are using an analogue FXO connection to the exchange; because there is no standardised way of communicating supervisory information over such a link. In any case, changing the caller identity information is a telco-dependent feature. Not all telephone companies support changing it; and even the ones that do, may well restrict you to using only numbers that belong to you. Of course, if you have an analogue telephone plugged into your Asterisk machine with an FXS adaptor, then *you* are the telco -- and can send whatever ident you like to phones thus connected. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk . -- _ -- 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] caller id spoofing/setting on analog
I've not used analog for quite some time. It seems it's not possible in asterisk to spoof a phone number/name on an analog call? -- _ -- 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] Caller ID Names
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: 11 March 2015 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Names Are the phones exposed to the internet (even using NAT)? If so there is a good chance these calls are not coming through your PBX but are coming in direct from someone, usually scammers. Polycom has a config option to disable accepting calls from unknown devices. No idea if Cisco has something similar. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Todd R. Sent: Wednesday, March 11, 2015 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Names To be sure you could setup a soft phone and see if the caller ID name comes in correctly. From a softphone (x-lite) the caller id information comes through as anonymous@anonymous.invalid These are also valid calls - If I disable outbound CLID on my mobile and call in - this happens. However it works fine on calls where I send caller id information. This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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] Caller ID Names
From a softphone (x-lite) the caller id information comes through as anonymous@anonymous.invalid These are also valid calls - If I disable outbound CLID on my mobile and call in - this happens. However it works fine on calls where I send caller id information. Okay, just figured this out - needed to do Set(CALLERID(num-pres)=allowed) Thanks for your help with narrowing this down This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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] Caller ID Names
To be sure you could setup a soft phone and see if the caller ID name comes in correctly. On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.net wrote: Hi, In my dialplan I have the following line. same = n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to ‘Support’ - however, we are getting calls come in as “Anonymous” with the number as something like “unknown@unknown” We’re using Cisco 7945 phones – I possibly wonder if they are displaying this rather than asterisk not changing it? Anyone had similar experiences before? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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-- _ -- 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] Caller ID Names
Are the phones exposed to the internet (even using NAT)? If so there is a good chance these calls are not coming through your PBX but are coming in direct from someone, usually scammers. Polycom has a config option to disable accepting calls from unknown devices. No idea if Cisco has something similar. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd R. Sent: Wednesday, March 11, 2015 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID Names To be sure you could setup a soft phone and see if the caller ID name comes in correctly. On Mar 10, 2015, at 8:41 AM, Jordan Cook - Gyron Networks jordan.c...@gyron.netmailto:jordan.c...@gyron.net wrote: Hi, In my dialplan I have the following line. same = n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to ‘Support’ - however, we are getting calls come in as “Anonymous” with the number as something like “unknown@unknown” We’re using Cisco 7945 phones – I possibly wonder if they are displaying this rather than asterisk not changing it? Anyone had similar experiences before? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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
[asterisk-users] Caller ID Names
Hi, In my dialplan I have the following line. same = n,Set(CALLERID(name)=Support) I am expecting this to always set the caller id name to 'Support' - however, we are getting calls come in as Anonymous with the number as something like unknown@unknown We're using Cisco 7945 phones - I possibly wonder if they are displaying this rather than asterisk not changing it? Anyone had similar experiences before? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may monitor email traffic data and the content of email for the purposes of security and staff training. Gyron Internet Ltd is a limited company registered in England and Wales. Registered number: 4239332. Registered office: 3 Centro, Boundary Way, Hemel Hempstead, HP2 7SU. VAT reg no 804 2532 63. Gyron is a registered trademark. Gyron is a Deloitte Technology Fast 50 ranked company. -- _ -- 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] caller id setting on channel originate
I am trying to make a data channel using ISDN and i need to set the caller id num field. Can any body tell me how i can set the caller id field since i notice in chan_dahdi.conf callerid field doesn't work with channel originate. Thanks, Pawel -- _ -- 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] caller id setting on channel originate
On Fri, May 9, 2014 at 9:52 AM, Pawel Pastuszak pawelpastus...@gmail.comwrote: I am trying to make a data channel using ISDN and i need to set the caller id num field. Can any body tell me how i can set the caller id field since i notice in chan_dahdi.conf callerid field doesn't work with channel originate. Use call files or the AMI Originate action. You can set the caller id using those methods. The CLI channel originate command does not have way to set the caller id. You should not be using CLI commands unless you don't have any other means to do what you want since CLI commands are generally intended for human interaction with Asterisk. Richard -- _ -- 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] Caller ID not real nor showing in call logs.
I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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] Caller ID not real nor showing in call logs.
logs ? full log containing the call? On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote: I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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] Caller ID not real nor showing in call logs.
Does not show up in the cdr log. I am going to enable an asterisk cli dump tonight and try to catch it. I am thinking its a straight sip attack or IP attach on the sip client vs a real call or problem with asterisk. It's also a polycom IP 335 Thanks, --Eric Sent from my phone. On Jan 8, 2014, at 12:02 PM, Tiago Geada tiago.ge...@gmail.com wrote: logs ? full log containing the call? On 8 January 2014 14:56, Eherr email.eherr9...@gmail.com wrote: I have a multi tenant asterisk box where on tenant is receiving calls from the caller ID as1as and they cannot pickup the call. The caller ID also does not show up in the call log. Thoughts? Thanks, --Eric -- _ -- 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 -- _ -- 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] caller id not shown
hello all
i have asterisk 1.8.22 and have problem with caller id. this is my
scenario:
PSTN -- FXO --- FXS --- phone(223)
when i call from a 223 to another phone, every thing is ok and caller id
(223) is shown in called phone. but when i call from another phone to 223,
no caller id is shown and just zero is shown.
if i set callerid=12345 in chan_dahdi.conf file, when another phone call
223, this number (12345) is shown as caller id instead of zero. but i want
to show incoming number as caller id.
this is my chan_dahdi.conf file:
[channels]
;cidsignalling=dtmf
cidstart=polarity;; in gozine takhir dar tamas (aghab boodan yek zang) ra
az beyn mibarad.
callprogress=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
transfer=yes
echocancel=yes
echotraining=yes
callerid=asreceived
group=0
callgroup=1
pickupgroup=1
usecallerid=yes
context=pstn-channels
channel=5-8
group=1
callgroup=1
pickupgroup=1
usecallerid=yes
context=phone-channels
channel=1-4
and this is my extensions.conf file:
[phone-channels]
exten=_.,1,Dial(DAHDI/8/${EXTEN})
[pstn-channels]
exten=_.,1,Dial(DAHDI/2/${EXTEN})
i searched a lot but found nothing useful:( please help me to solve it.
thanks in advance,
SAM
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[asterisk-users] Caller ID is not persisted when using Channel Redirect
Is there a work around for Caller ID information not being persisted when using the CLI or AMI Channel Redirect. A calls B (caller id is displayed), B transfers call to C (no caller id is displayed on phone c). Jacob Miles Software Engineer jacob.e.mi...@l-3com.com 903.457.4422 image001.png-- _ -- 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] Caller ID to be identical across all lines
I have two SIP lines, 09271 09974 , the 09271 is my publicly list phone line which is also my second pick, when i make an outgoing call , I use 09974, what I want is the caller ID for 09974 to be 09271 to the people I call. The reason is.. when i ring anyone i want them recognise my number (which they will if it is 09271). I have tried to change the caller ID in the trunk routes there was no differnece, what i do at the moment is sent 09974 calls to the IVR of 09271 but it would be nice to have an elegant solution. Is the caller ID generated at the phone exchange , if so, i probably cannot change it. Thanks in advance - Paul E -- _ -- 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] Caller ID to be identical across all lines
On Mon, 18 Feb 2013, Paul Edgar wrote: I have tried to change the caller ID in the trunk routes there was no differnece, what i do at the moment is sent 09974 calls to the IVR of 09271 but it would be nice to have an elegant solution. Is the caller ID generated at the phone exchange , if so, i probably cannot change it. It depends on your PSTN termination company. Some let you set CID to anything, some to any DID you rent from them, some not at all. You should ask them what format they want/allow the CID specified -- how many digits, country code, punctuation, etc. If you don't follow their format, all bets are off. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] Caller ID DTMF is not coming
Alexander Tarasov Sent: Friday, 12 October 2012 12:10 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Caller ID DTMF is not coming Hi. I have a problem with the Caller ID in Ukraine - it is not coming, but I sure that telco is providing it as a DTMF. You'll need to enable DTMF logging in 'logger.conf' dtmf = dtmf Then you can watch /var/log/dtmf or whereever it is on your system. Also since Asterisk 1.8.18.0-rc1 there are some options in dsp.conf that may assist, these are to do with DTMF threshold levels. Have a look at configs/dsp.conf.sample The options are; dtmf_reverse_twist dtmf_normal_twist relax_dtmf_reverse_twist relax_dtmf_normal_twist Initally I'd set all to 100, you may get talkoff when on a call, but atleast you'll know if CID is working. Then set back to the appropraite standards of Ukraine, ETSI ATT etc. Alexander: I checked the wav file, and found that the DTMF rate is 70ms on and 70ms off. So the duration should be OK with versions prior to 1.8.18.0-rc1 What version of asterisk are you using? The reason I ask is that for a few releases, the DTMF acceptance duration had been extended to 4*12.75ms (or 63.5ms including 1 extra block). Again since 1.8.18.0-rc1, the defaults have been set back to Begin = 2*12.75ms and EndDTMF=3*12.75ms. They were Begin = 4 * 12.75ms and End = 4 * 12.75 These can also now (since 1.8.18.0-rc1) be set in dsp.conf as below, the values used below are also the defaults. dtmf_hits_to_begin=2 dtmf_misses_to_end=3 Alec Davis -- _ -- 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] Caller ID DTMF is not coming
Hi. I have a problem with the Caller ID in Ukraine - it is not coming, but I sure that telco is providing it as a DTMF. I have recorded the line using dahdi_monitor: http://dl.dropbox.com/u/2962/dtmf.wav The record contains DTMF codes, first ring and answer of dialplan -- it is just Playtones(1004/1000) . I have set cidsignalling=dtmf and tried all choices for cidstart (dtmf, polarity, ring), but have no luck. If you will open the record in the audio editor, you will see that the first digit (DTMF D) is quite louder than other digits (7495727C). Is this normal? I have adjusted my rxgain parameter on that dahdi channel, so I think it is the telco issue (DTMF sound overload at its output side). I can provide any debug logs, if it they are needed. I sure, that it is possible to set up the CallerID recognition, because I am able to recognise them in Audacity! :-) Thanks! -- Forex Club System Administrator (Branches VoIP) System Support and Services Management skype me: oioki17 -- _ -- 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] Caller ID DTMF is not coming
Alexander Tarasov Sent: Friday, 12 October 2012 12:10 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Caller ID DTMF is not coming Hi. I have a problem with the Caller ID in Ukraine - it is not coming, but I sure that telco is providing it as a DTMF. I have recorded the line using dahdi_monitor: http://dl.dropbox.com/u/2962/dtmf.wav The record contains DTMF codes, first ring and answer of dialplan -- it is just Playtones(1004/1000) . I have set cidsignalling=dtmf and tried all choices for cidstart (dtmf, polarity, ring), but have no luck. If you will open the record in the audio editor, you will see that the first digit (DTMF D) is quite louder than other digits (7495727C). Is this normal? I have adjusted my rxgain parameter on that dahdi channel, so I think it is the telco issue (DTMF sound overload at its output side). You'll need to enable DTMF logging in 'logger.conf' dtmf = dtmf Then you can watch /var/log/dtmf or whereever it is on your system. Also since Asterisk 1.8.18.0-rc1 there are some options in dsp.conf that may assist, these are to do with DTMF threshold levels. Have a look at configs/dsp.conf.sample The options are; dtmf_reverse_twist dtmf_normal_twist relax_dtmf_reverse_twist relax_dtmf_normal_twist Initally I'd set all to 100, you may get talkoff when on a call, but atleast you'll know if CID is working. Then set back to the appropraite standards of Ukraine, ETSI ATT etc. Alec Davis -- _ -- 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] Caller ID : FSK ETSI or FSK US
Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after buying this converter hardware and upgrade to dahdi 2.6.1 still not solve my caller id problem, I really dont know what to do and feel hopeless :( Thanks, Anam. -- _ -- 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] Caller ID : FSK ETSI or FSK US
Welcome to da Matrix :) Look at this issue : https://issues.asterisk.org/view.php?id=6683 And try different combinations suggested over there, you might get lucky :) Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta anam.satri...@gmail.comwrote: Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after buying this converter hardware and upgrade to dahdi 2.6.1 still not solve my caller id problem, I really dont know what to do and feel hopeless :( Thanks, Anam. -- _ -- 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] Caller ID : FSK ETSI or FSK US
Thanks Mitul :) The patch on the link is so old (2006-2007) so I think it's already implemented in the newest version. Honestly to say, I already try any combitions but still the caller id doesn't work :( cidsignalling=bell,dtmf,v23 cidstart=ring,polarity,dtmf with some parameter if we set it to dtmf Hopeless :(( On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani mi...@enterux.in wrote: Welcome to da Matrix :) Look at this issue : https://issues.asterisk.org/view.php?id=6683 And try different combinations suggested over there, you might get lucky :) Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta anam.satri...@gmail.comwrote: Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after buying this converter hardware and upgrade to dahdi 2.6.1 still not solve my caller id problem, I really dont know what to do and feel hopeless :( Thanks, Anam. -- _ -- 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 -- _ -- 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] Caller ID : FSK ETSI or FSK US
Hopeless: Do you know that your provider is delivering CLID? In the US, conventional providers charge extra for CLID and even more for CLID with name Have you ever determined WHICH delivery system is used in your as yet undefined country? Most systems are coded into Asterisk, but require Asterisk to be told which one to use Some locations may not be covered There should be no need for external hardware. Quit thrashing about and resolve this issue in a methodical manner Peg Leg O'Brien Satria Anamarta wrote: Thanks Mitul :) The patch on the link is so old (2006-2007) so I think it's already implemented in the newest version. Honestly to say, I already try any combitions but still the caller id doesn't work :( cidsignalling=bell,dtmf,v23 cidstart=ring,polarity,dtmf with some parameter if we set it to dtmf Hopeless :(( On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani mi...@enterux.in mailto:mi...@enterux.in wrote: Welcome to da Matrix :) Look at this issue : https://issues.asterisk.org/view.php?id=6683 And try different combinations suggested over there, you might get lucky :) Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in mailto:mi...@enterux.in DID: +91-22-61447605 tel:%2B91-22-61447605 Cell: +91-9820332422 tel:%2B91-9820332422 On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta anam.satri...@gmail.com mailto:anam.satri...@gmail.com wrote: Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after buying this converter hardware and upgrade to dahdi 2.6.1 still not solve my caller id problem, I really dont know what to do and feel hopeless :( Thanks, Anam. -- _ -- 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 -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] Caller ID : FSK ETSI or FSK US
Thanks John :) Yes, the telco company does deliver the CLID because I pay for this service every month and when I test it using analog phone, the caller ID displayed on the phone perfectly. 10 out of 10 try the caller id displayed on the analog phone. I cant found a official information from the telco company but based on paper released by a student I found on the net (he did a experiment on caller id system in our country), it's FSK. I understand that this feature is not auto detected and need to be told on the conf file. I did post the conf file on this list few weeks ago but still no solution so I try something else (buy a caller id converter hardware, upgrade the dahdi to 2.6.1). I'm from indonesia. Is there is somebody here ever working with this issue in my country, please let me know :) And I need to know: is it possible that echo canceller (epsecially OSLEC) can mess the caller id detection? I'm asking this because somebody is posting this in a forum: I just solved the problem by re-install dahdi with custom configs, it seems like the problem is the echo canceller. The default echo canceller oslec seems to cancel my caller id and therefore, no caller id was received. So I change echo_can oslec to echo_can mg2 in the etc/dadhi/genconf_parameters, restart elastix, re-detect hardware and restart again, everything works! If anyone from Taiwan also has the same problem, you can refer to here Thanks :) Best regards, Anam. On Sun, Jun 3, 2012 at 6:59 PM, John Novack jnov...@stromberg-carlson.org wrote: Hopeless: Do you know that your provider is delivering CLID? In the US, conventional providers charge extra for CLID and even more for CLID with name Have you ever determined WHICH delivery system is used in your as yet undefined country? Most systems are coded into Asterisk, but require Asterisk to be told which one to use Some locations may not be covered There should be no need for external hardware. Quit thrashing about and resolve this issue in a methodical manner Peg Leg O'Brien Satria Anamarta wrote: Thanks Mitul :) The patch on the link is so old (2006-2007) so I think it's already implemented in the newest version. Honestly to say, I already try any combitions but still the caller id doesn't work :( cidsignalling=bell,dtmf,v23 cidstart=ring,polarity,dtmf with some parameter if we set it to dtmf Hopeless :(( On Sun, Jun 3, 2012 at 4:51 PM, Mitul Limbani mi...@enterux.in wrote: Welcome to da Matrix :) Look at this issue : https://issues.asterisk.org/view.php?id=6683 And try different combinations suggested over there, you might get lucky :) Regards, Mitul Limbani, Chief Architech Founder, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-61447605 Cell: +91-9820332422 On Sun, Jun 3, 2012 at 3:08 PM, Satria Anamarta anam.satri...@gmail.com wrote: Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after buying this converter hardware and upgrade to dahdi 2.6.1 still not solve my caller id problem, I really dont know what to do and feel hopeless :( Thanks, Anam. -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by
[asterisk-users] Caller ID : FSK ETSI or FSK US
Dear In essence Caller ID ETSI and FSK US (Bellcore) is based on the same pattern as; _ ___ _ _ |First Ring burst |_500ms_|Channel seizure 300 bits|__|Mark Signal|__| Caller ID Message|_200 ms_|Second ring burst | So basically any kind of device should be work without any problem, unfortunately during these process if some noises (as miss ground connection or others) happens during the process can make failed to process caller-id information, by the modem. Mc GRATH Ricardo E-Mail mcgra...@mail2web.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID problem
Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Caller ID problem
Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Caller ID problem
Hi Arthur, read the article and understand,thanks :) Btw, is there any patch for this problem without need to upgrade to version 10.x ? On 4/16/12, Arthur Stanfield a...@dmcip.com wrote: Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] Caller ID problem
No - if someone figures out a way, let me know since my receptionist doesn't like blind transfers. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Monday, April 16, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Caller ID problem Hi Arthur, read the article and understand,thanks :) Btw, is there any patch for this problem without need to upgrade to version 10.x ? On 4/16/12, Arthur Stanfield a...@dmcip.com wrote: Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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
[asterisk-users] Caller ID problem
Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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] Caller ID problem
Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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
[asterisk-users] Caller id issues
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[asterisk-users] Caller id issues
First of all, I want apologize for the first two blank emails that I sent out by mistake. I have Xorcom USB fxo channel bank, asterisk 1.6, freepbx 2.8. Up to now, the lines connected from Telekom did not have caller id feature enabled, now that we enabled we cannot see incoming caller id shown. However it shows up if I connect a normal analogue phone with LCD screen to show caller id feature. So my question is - is there any specific settings I need to do for it to show caller id? Thanks! Sent from my iPhone -- _ -- 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] Caller ID not working in DAHDI 2.6.0
I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2 without any problems. A couple of weeks ago I upgraded to 2.6.0 and found that caller ID was no long working for me. All calls came in with a blank caller id. I reverted back to 2.5.0.2 and everything was happy again. I tried upgrading to 2.6.0 again this morning and got the same results. I'm compiling from source on an Ubuntu 10.04.4 box. I was very careful when I merged my settings from my old 2.5.0.2 setup with the new 2.6.0 configuration files. I've searched the bugs and read through the Changes file but didn't see anything obvious. Should I file a bug? -- Chris -- _ -- 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] Caller ID not working in DAHDI 2.6.0
On Thu, Mar 15, 2012 at 10:04:56AM -0500, Chris Gentle wrote: I have a TDM410 with one FXO and one FXS. I've been running dahdi 2.5.0.2 without any problems. A couple of weeks ago I upgraded to 2.6.0 and found that caller ID was no long working for me. All calls came in with a blank caller id. I reverted back to 2.5.0.2 and everything was happy again. I tried upgrading to 2.6.0 again this morning and got the same results. I'm compiling from source on an Ubuntu 10.04.4 box. I was very careful when I merged my settings from my old 2.5.0.2 setup with the new 2.6.0 configuration files. I've searched the bugs and read through the Changes file but didn't see anything obvious. Should I file a bug? Hi Chris, I believe this is fixed in the head of the 2.6 branch. We're prepping a 2.6.0.1 release now... [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10481 If you could try out the branch and let me know if it *doesn't* work for you, I would be appreciative. Thanks, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Caller ID not working in DAHDI 2.6.0
On Thu, Mar 15, 2012 at 10:08 AM, Shaun Ruffell sruff...@digium.com wrote: Hi Chris, I believe this is fixed in the head of the 2.6 branch. We're prepping a 2.6.0.1 release now... Hey Shaun. Thanks for the quick reply. I applied the patch for the bug to my 2.6.0 and it works fine. I've made five test calls and the caller ID came through fine. It wasn't coming through at all before, not even intermittently. Thanks for the help! I'll be watching for the 2.6.0.1 release. -- Chris -- _ -- 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] Asterisk-users caller ID
Hello,
I have a server that connects to my Voice Server provider so far is working
great! I have a second server that I want to set caller id to a different
number second server I'm going to call it server B. And server B will go
through server A which is connected to my Voice Server Provider. Thus far
I'm unsussessful! Can some one help?
A$ ee extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=8006332211)
exten = _91NXXNXX,2,Dial(SIP/VSP/${EXTEN:1},80)
exten = _9NXX,1,Set(CALLERID(num)=8006332211)
exten = _9NXX,2,Dial(SIP/VSP/${EXTEN:1},80)
B$ ee extensions.conf
[outbound]
exten = _91NXXNXX,1,Set(CALLERID(num)=8007342323)
exten = _91NXXNXX,2,Dial(SIP/ServerA/${EXTEN}@serverBout)
exten = _9NXX,1,Set(CALLERID(num)=8007342323)
exten = _9NXX,2,Dial(SIP/ServerA/${EXTEN:1}@serverBout)
Every time I call my cell phone from server B I get the caller id from
server A, please help!
Thanks,
Motty
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[asterisk-users] Caller ID
We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets that ring at odd hours with Asterisk before the phone is picked up, and Out of Area after it is picked up. I have read that Asterisk is what is reported by Asterisk for 0 length caller ID number. But since we don't subscribe to Caller ID Name, I am wondering where the Out of Area is coming from? Could these be hacking attempts via IP? Perhaps they are doing the caller ID name? It only happens to a few extensions as far a we know. TIA for any input or knowledge. -- _ -- 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] Caller ID
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 01, 2011 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Caller ID We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets that ring at odd hours with Asterisk before the phone is picked up, and Out of Area after it is picked up. I have read that Asterisk is what is reported by Asterisk for 0 length caller ID number. But since we don't subscribe to Caller ID Name, I am wondering where the Out of Area is coming from? Could these be hacking attempts via IP? Perhaps they are doing the caller ID name? It only happens to a few extensions as far a we know. TIA for any input or knowledge. IP Hacking should not apply on your Telco lines. I'd start with your CDR file (/var/log/asterisk/cdr-csv/Master.csv) and go from there. -- _ -- 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] Caller ID
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, March 01, 2011 11:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 01, 2011 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Caller ID We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets that ring at odd hours with Asterisk before the phone is picked up, and Out of Area after it is picked up. I have read that Asterisk is what is reported by Asterisk for 0 length caller ID number. But since we don't subscribe to Caller ID Name, I am wondering where the Out of Area is coming from? Could these be hacking attempts via IP? Perhaps they are doing the caller ID name? It only happens to a few extensions as far a we know. TIA for any input or knowledge. IP Hacking should not apply on your Telco lines. I'd start with your CDR file (/var/log/asterisk/cdr-csv/Master.csv) and go from there. I had in mind a hacking attempt IP calling extension lines that exist, but thanks, I will look at the cdrs. -- _ -- 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] Caller id is not proper when I do call forward
Hi Caller id is not show showing proper when I do call forward from asterisk,bellow is the example. 1001 called 1002 and 1002 forwarded call to 1003 then callerid in 1003 phone is showing 1002,this is wrong it shound be 1001(he is actual caller). If u do blind transfer instead of call forward it will show properly.Please correct me I am wrong Thanks NIkhil -- _ -- 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] Caller ID issue
Sorry for the delay - I lost this message in the middle of a digest. I tried Answer(2000) and was getting an annoying warning: [Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to DAHDI/1-1 So I changed it back to Wait(2). I'll try shorter wait intervals and see what happens. Cassius Subject: Re: [asterisk-users] Caller ID issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlkti=s6fboeqysvpvw25tmevkdpnjbvjmsvniwu...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 In most cases wait(.5) will do. I would not recommend using answer(2000) as that answers the channel, which means you start getting billed. On 8/2/10, Peder pe...@networkoblivion.com wrote: I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Some PRI take that long too because the telco sends the name in a followup message, not in the initial call setup. -- -- _ -- 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] Caller ID issue
In most cases wait(.5) will do. I would not recommend using answer(2000) as that answers the channel, which means you start getting billed. On 8/2/10, Peder pe...@networkoblivion.com wrote: I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Some PRI take that long too because the telco sends the name in a followup message, not in the initial call setup. -- _ -- 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
[asterisk-users] Caller ID issue
Hi list,
I'm having a problem with CallerID names not showing up when calls come
in. I have dialplan code to store the callerid(name) away and it is
blank (null). However, the voicemail variable ${VM_CALLERID} has the
name field populated. For example, here is some of the dialplan code:
2. Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
3. Set(CALLER_ID_INFO_NAME=${CALLERID(name)})
4. Set(CALLER_ID_INFO_NUM=${CALLERID(num)})
5. Set(CALLER_ID_INFO_ANI=${CALLERID(ANI)})
6. Set(CALLER_ID_INFO_DNID=${CALLERID(DNID)})
Which yields this at the CLI:
-- Executing [3...@from_outside:2] Set(DAHDI/1-1,
CALLER_ID_INFO_ALL= 2565551212) in new stack
-- Executing [3...@from_outside:3] Set(DAHDI/1-1,
CALLER_ID_INFO_NAME=) in new stack
-- Executing [3...@from_outside:4] Set(DAHDI/1-1,
CALLER_ID_INFO_NUM=2565551212) in new stack
-- Executing [3...@from_outside:5] Set(DAHDI/1-1,
CALLER_ID_INFO_ANI=2565551212) in new stack
Note the first line should have the name field with the number, but does
not.
HOWEVER the voicemail notification contains:
Just wanted to let you know you were just left a 0:04 long message
(number 1) in mailbox 3703 from SMITH CASSIUS 2565551212
So - I know the NAME field is getting into the system, but it's not
showing up on the phones (and with telemarketers, that annoys my
users).
I'm using Asterisk 1.6.2.9, DAHDI 2.3.0
I have added callerid=asreceived to chan_dahdi.conf for my inbound
trunks, and shrinkcallerid=no to my sip.conf. (without effect)
Any ideas?
THANKS
Cassius
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Re: [asterisk-users] Caller ID issue
On Mon, Aug 2, 2010 at 2:56 PM, Cassius Smith cass...@cassius.org wrote: Any ideas? THANKS Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. -- Thanks, --Warren Selby http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID issue
Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. Ciao, Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. -- _ -- 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] Caller ID issue
Un-top-posting... On Mon, 2 Aug 2010, Cassius Smith wrote: I'm having a problem with CallerID names not showing up when calls come in. On Mon, 2 Aug 2010, Warren Selby wrote: Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. On Mon, 2 Aug 2010, Cassius Smith wrote: Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Using answer(2000) should also work. Can you try it and reply with your results? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Caller ID issue
I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Some PRI take that long too because the telco sends the name in a followup message, not in the initial call setup. -- _ -- 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] Caller ID on analog line
Hello,
I'm new to asterisk and trying to set up a PBX that's connected to ISDN
on the telecom operator side (Swisscom, Switzerland) and has analog line
on the local side. I use Digium B410PF and AEX2460EF cards.
Globally, everything is working well except that I can't get the CID
sent to the analog phone.
I tried with all combinations of options I could think of and several
asterisk version (the SVN-trunk-r27331, 1.6.2.6 and 1.6.2.8). I called
Swissccom (in case they knew something about that) but they couldn't
help me. I also did my homework, but didn't find any solution on
Asterisk wiki, mailing lists archives and Google.
Here is the actual working configuration (stripped down at the minimum
required) of chan_dahdi.conf (commented is the options I've played
with):
[channels]
tonezone=30
progzone=30
internationalprefix=00
nationalprefix=0
dialplan=unknown
pridialplan=unknown
prilocaldialplan=unknown
;cidstart=polarity
;cidsignalling=v23
;sendcalleridafter=0
usecallerid=yes
hidecallerid=no
;mwimonitor=fsk
; group 1 is incoming swisscom isdn line
signalling =bri_cpe
group=1
context=incoming
channel = 1-2
channel = 4-5
;Analog channel
signalling=fxo_ks
group=3
context=from-inside
channel=13
Here in the dialplan (down to the minimum) :
[incoming]
exten = 21,1,Verbose(${CALLERID(num)})
exten = 21,n,Dial(Dahdi/g3/13)
I tried to set manually the cid, but it didn't work. The Verbose
display the caller id correctly but it doesn't go any further.
So I must have missed something, but I don't know what and I don't know
where to look. If someone can help me ...
Thanks,
Etienne.
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[asterisk-users] Caller id, sip header from problem
Hello all, My pbx server is connected to a sip gateway, when I call an originate command from the asterisk console, to establish a sip connection, the gateway doesn't accept URL with white spaces, for example: * Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport * * From: PBX SERVER sip:PBX ser...@10.10.1.10;tag=as2512881b * * To: sip:927817...@10.10.1.250:5060;tag=2615730116 * * Contact: sip:PBX ser...@10.10.1.10 * * Call-ID: 454df9c904486e7647231af102a05...@10.10.1.10 * * CSeq: 102 ACK* * Max-Forwards: 70* The sip gateway will respond with the following message: *SIP/2.0 400 Bad Request * * Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport * * From: PBX SERVER sip:PBX ser...@10.10.1.10;tag=as2512881b * * To: sip:927817...@10.10.1.250:5060;tag=2615730116 * * Call-ID: 454df9c904486e7647231af102a05...@10.10.1.10 * * CSeq: 102 INVITE * * Content-Type: text/plain * * Content-Length: 23 * The PBX SERVER name is set in the sip.conf in the callerid parameter. Question: Is it possible, without trimming the callerid parameter, to set some type of variable in asterisk to trim automatically. Thanks in advance, Alex -- _ -- 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] Caller ID questions
Hello Folks; I have a dilemma: I have a client with Asterisk 1.4x and he needs to have a record of all incoming calls - caller ID and date/time is sufficient. Since I am not an Asterisk wizard, I am doing it this way. I set a cron job to tailf the last 10 lines of the Master.csv file and package those nicely in an email. However, I can see some inefficiencies in this. Main one is what if there are more than 10 incoming calls between cron runs? So, questions: 1. has anyone done this? 2. is there a better way? 3. if so, can you 'skool' me ? Thanks B -- _ -- 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] Caller ID questions
On Sat, 22 May 2010, GlenM wrote: Hello Folks; I have a dilemma: I have a client with Asterisk 1.4x and he needs to have a record of all incoming calls - caller ID and date/time is sufficient. Since I am not an Asterisk wizard, I am doing it this way. I set a cron job to tailf the last 10 lines of the Master.csv file and package those nicely in an email. However, I can see some inefficiencies in this. Main one is what if there are more than 10 incoming calls between cron runs? So, questions: 1. has anyone done this? 2. is there a better way? 3. if so, can you 'skool' me ? AIUI, Asterisk opens for append the Master.csv file, (fopen (... a)) which creates the file if it doesn't existis.. writes a line to it then closes it for each CDR recorded, so ... You can rename the Master.csv file then email the file then delete it... Pseudocode: Once every 10 miuntes from cron: if Master.csv does not exist, then exit // No calls rename Master.csv work.csv sleep 1 process and email work.csv to whoever delete work.csv exit The sleep may not be needed, but it won't do any harm in the event that you rename the file after asterisk opens it but before it writes the line into and closed it. And instead of deleting the work.csv you could append it to some other file for a permanent log... 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
Re: [asterisk-users] Caller ID questions
On Sat, May 22, 2010 at 11:28 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 22 May 2010, GlenM wrote: Hello Folks; I have a dilemma: I have a client with Asterisk 1.4x and he needs to have a record of all incoming calls - caller ID and date/time is sufficient. Since I am not an Asterisk wizard, I am doing it this way. I set a cron job to tailf the last 10 lines of the Master.csv file and package those nicely in an email. However, I can see some inefficiencies in this. Main one is what if there are more than 10 incoming calls between cron runs? So, questions: 1. has anyone done this? 2. is there a better way? 3. if so, can you 'skool' me ? AIUI, Asterisk opens for append the Master.csv file, (fopen (... a)) which creates the file if it doesn't existis.. writes a line to it then closes it for each CDR recorded, so ... You can rename the Master.csv file then email the file then delete it... Pseudocode: Once every 10 miuntes from cron: if Master.csv does not exist, then exit // No calls rename Master.csv work.csv sleep 1 process and email work.csv to whoever delete work.csv exit The sleep may not be needed, but it won't do any harm in the event that you rename the file after asterisk opens it but before it writes the line into and closed it. And instead of deleting the work.csv you could append it to some other file for a permanent log... Gordon -- I use asterisk-addons with mysql to store cdr data. I process this data and insert it into the companies call database link to users, you could just email it. I basically added a column to mysql and mark each row as processed. Ryan -- _ -- 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] Caller ID questions
Gentlemen! Telemarketers!! - RELEASE THE PHONE SPIDERS!!! (Dr Weird - of AquaTeen Hungerforce ) Seriously, thank you all for the excellent suggestions. I will try them both and present them to the client. Nice to have some helpful folks! Glen Ryan Wagoner said the following on 5/22/2010 11:39 AM: On Sat, May 22, 2010 at 11:28 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 22 May 2010, GlenM wrote: Hello Folks; I have a dilemma: I have a client with Asterisk 1.4x and he needs to have a record of all incoming calls - caller ID and date/time is sufficient. Since I am not an Asterisk wizard, I am doing it this way. I set a cron job to tailf the last 10 lines of the Master.csv file and package those nicely in an email. However, I can see some inefficiencies in this. Main one is what if there are more than 10 incoming calls between cron runs? So, questions: 1. has anyone done this? 2. is there a better way? 3. if so, can you 'skool' me ? AIUI, Asterisk opens for append the Master.csv file, (fopen (... a)) which creates the file if it doesn't existis.. writes a line to it then closes it for each CDR recorded, so ... You can rename the Master.csv file then email the file then delete it... Pseudocode: Once every 10 miuntes from cron: if Master.csv does not exist, then exit // No calls rename Master.csv work.csv sleep 1 process and email work.csv to whoever delete work.csv exit The sleep may not be needed, but it won't do any harm in the event that you rename the file after asterisk opens it but before it writes the line into and closed it. And instead of deleting the work.csv you could append it to some other file for a permanent log... Gordon -- I use asterisk-addons with mysql to store cdr data. I process this data and insert it into the companies call database link to users, you could just email it. I basically added a column to mysql and mark each row as processed. Ryan -- _ -- 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] Caller ID on Asterisk and Astribank
Hi all...
I have a problem with caller id on my asterisk server.
here is my configuration :
centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2
ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting)
everything fine until I try to feed my app with caller id.
My extensions.conf :
[incoming1]
exten = s,1,AGI(/var/apps/core/runagi,incoming,${CALLERID(num)})
exten = s,n,QUEUE(${que},trkd)
exten = h,1,Hangup()
here is the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:24:40] ERROR[30895]: callerid.c:562 callerid_feed: No start bit
found in fsk data.
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8712 ss_thread: CallerID feed
failed: Success
[Apr 30 11:24:40] WARNING[30895]: chan_dahdi.c:8816 ss_thread: CallerID
returned with error on channel 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-- Playing 'en/0006' (escape_digits=) (sample_offset 0)
I read instructions from a few forums
then I made a change on 'chan_dahdi.conf' like :
-
1:
cidsignalling=v23, cidstart=ring, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18
(Ring Begin)...
[Apr 30 11:42:03] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 2
(Ring/Answered)...
[Apr 30 11:42:05] NOTICE[31296]: chan_dahdi.c:8672 ss_thread: Got event 18
(Ring Begin)...
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:42:06] WARNING[31296]: chan_dahdi.c:6174 dahdi_handle_event:
Ring/Off-hook in strange state 6 on channel 15
== Spawn extension (default, s, 2) exited non-zero on 'DAHDI/15-1'
-
2:
cidsignalling=dtmf', cidstart=ring, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:28] WARNING[31491]: chan_dahdi.c:8610 ss_thread: DTMFCID timed
out waiting for ring. Exiting simple switch
-- Hungup 'DAHDI/15-1'
-- Starting simple switch on 'DAHDI/15-1'
[Apr 30 11:49:34] DEBUG[31492]: chan_dahdi.c:8630 ss_thread: CID is '', flags 8
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
[Apr 30 11:49:35] WARNING[31492]: chan_dahdi.c:6174 dahdi_handle_event:
Ring/Off-hook in strange state 6 on channel 15
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-
3:
cidsignalling=dtmf', cidstart=polarity, hidecallerid=no, callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1] Answer(DAHDI/15-1, ) in new stack
-- Executing [...@default:2] Wait(DAHDI/15-1, 6) in new stack
-- Registered IAX2 '9009' (AUTHENTICATED) at 127.0.0.1:48961
-- Executing [...@default:3] Goto(DAHDI/15-1, im-incoming,s,1) in new
stack
-- Goto (im-incoming,s,1)
-- Executing [...@im-incoming:1] AGI(DAHDI/15-1,
/var/apps/core/runagi,incoming,) in new stack
-- Launched AGI Script /var/apps/core/runagi
-
4:
cidsignalling=v23', cidstart=polarity, hidecallerid=no,
callerid=asreceived
Here's the log :
-- Starting simple switch on 'DAHDI/15-1'
== Starting DAHDI/15-1 at from-pstn,s,1 failed so falling back to exten 's'
== Starting DAHDI/15-1 at from-pstn,s,1 still failed so falling back to
context 'default'
-- Executing [...@default:1]
Re: [asterisk-users] Caller ID in Asterisk
As far as I know, you should set up the callerid in the chan_dahdi.conf
with the usecallerid=yes and the callerid=8001234001 options where you
are setting the each channels.
Regards,
Peter Gelencser
2010.03.05. 7:54 keltezéssel, Gopalakrishnaiyer Venugopal-Q16770 írta:
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten =
_988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn’t help.
I have 8 lines configured as below and need the callerID of the
individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
*From:* Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thursday, March 04, 2010 6:36 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
*Subject:* RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private
Call and Out of Area in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)
exten = 8001234004,n,Set(CALLERID(name)=Line 4)
exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was
not going through
Venugopal
*From:* asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
*Sent:* Thu 3/4/2010 5:53 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I
guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)
exten = 8001234003,n,Set(CALLERID(name)=Line 5)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not
put them in ${}.
Jimmy
-Original Message-
*From:* venui...@motorola.com
*Sent:* Thu, 4 Mar 2010 19:50:03 +0800
*To:* asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com, asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
*From:* asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
*Sent:* Thu 3/4/2010 3:59 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP
server.When i call from the Asterisk machine to the SIP server i am
not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf
file is attached.
Cheers
venu
Re: [asterisk-users] Caller ID in Asterisk
Hi,
Well, if you replicate the line that set the callerid for every extension than you can set each one manually.
Jimmy
-Original Message-From: venui...@motorola.comSent: Fri, 5 Mar 2010 14:54:56 +0800To: venui...@motorola.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten = _988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn’t help.
I have 8 lines configured as below and need the callerID of the individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 *
From: Gopalakrishnaiyer Venugopal-Q16770 Sent: Thursday, March 04, 2010 6:36 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting "Private Call" and "Out of Area" in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]static=yeswriteprotect=noautofallthrough=noextenpatternmatchnew=noclearglobalvars=nopriorityjumping=yesuserscontext=default
[globals]CONSOLE=Console/dsp ; Console interface for demo;CONSOLE=DAHDI/1;CONSOLE=Phone/phone0IAXINFO=guest ; IAXtel username/password;IAXINFO=myuser:mypassTRUNK=DAHDI/G1TRUNKMSD=1
[Internal]include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)exten = 8001234004,n,Set(CALLERID(name)="Line 4")exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]exten = s,1,Answerexten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten = _988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was not going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy GodboutSent: Thu 3/4/2010 5:53 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)exten = 8001234003,n,Set(CALLERID(name)="Line 5")
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them in ${}.
Jimmy
-Original Message-From: venui...@motorola.comSent: Thu, 4 Mar 2010 19:50:03 +0800To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer Venugopal-Q16770Sent: Thu 3/4/2010 3:59 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.comSubject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is attached.
Cheers
venu
Receive Notifications of Incoming MessagesEasily monitor multiple email accounts access them with a click. Visit www.inbox.com/notifier and check it out!
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http://www.asteris
[asterisk-users] Caller ID in Asterisk
Hi All, I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side. Please help me to identify how this can be set.The extensions.conf file is attached. Cheers venu extensions.conf Description: extensions.conf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When
i call from the Asterisk machine to the SIP server i am not getting the caller
id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is
attached.
Cheers
venu
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)exten = 8001234003,n,Set(CALLERID(name)="Line 5")
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them in ${}.
Jimmy
-Original Message-From: venui...@motorola.comSent: Thu, 4 Mar 2010 19:50:03 +0800To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com, asterisk-users@lists.digium.comSubject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer Venugopal-Q16770Sent: Thu 3/4/2010 3:59 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.comSubject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When i call from the Asterisk machine to the SIP server i am not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is attached.
Cheers
venu
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Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Please note that this is a lab setup and we are not connected to any external
telcos
Rgds
Venu
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 5:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion; asterisk-us...@lists.digium.comhi
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnaiyer
Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP server.When
i call from the Asterisk machine to the SIP server i am not getting the caller
id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file is
attached.
Cheers
venu
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Re: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private Call
and Out of Area in the sip phone display when i call from asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)
exten = 8001234004,n,Set(CALLERID(name)=Line 4)
exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was not
going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
Sent: Thu 3/4/2010 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I guess
it's a typo that the priority in this dialplan is all 1; it should be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)
exten = 8001234003,n,Set(CALLERID(name)=Line 5)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not put them
in ${}.
Jimmy
-Original Message-
From: venui...@motorola.com
Sent: Thu, 4 Mar 2010 19:50:03 +0800
To: asterisk-users@lists.digium.com, asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the SIP
server.When i call from the Asterisk machine to the SIP server i am not getting
the caller id of the lines at the sip side.
Please help me to identify how this can be set.The extensions.conf file
is attached.
Cheers
venu
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Re: [asterisk-users] Caller ID in Asterisk
Hi All,
Finally I am able to get the number displayed at the SIP side using
exten = _988.,1,Set(CALLERID(num)=8001234000)
exten = _988.,n,Dial(DAHDI/g1/${EXTEN},20)
However this number is fixed and I want to display the number of the
individual lines whoever is calling. I tried with
exten = _988.,1,Set(CALLERID(num)=${exten}) and exten =
_988.,1,Set(CALLERID(num)=${EXTEN})
Both the above lines didn't help.
I have 8 lines configured as below and need the callerID of the
individual lines to be displayed at the SIP side
exten = 8001234001,n,Dial(DAHDI/32,,rt)
exten = 8001234002,n,Dial(DAHDI/33,,rt)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
exten = 8001234004,n,Dial(DAHDI/35,,rt)
exten = 8001234005,n,Dial(DAHDI/36,,rt)
exten = 8001234006,n,Dial(DAHDI/37,,rt)
exten = 8001234007,n,Dial(DAHDI/38,,rt)
exten = 8001234008,n,Dial(DAHDI/39,,rt)
Warm Regards
Warm Regards
Venugopal G
HNM-SO WiMAX CPE VoIP IOT Team
Cell : +91-99723-99437
*
From: Gopalakrishnaiyer Venugopal-Q16770
Sent: Thursday, March 04, 2010 6:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Caller ID in Asterisk
Hi Jimmy,
Appreciate your help.
I tried the one below and cudnt get the caller ID.I am getting Private
Call and Out of Area in the sip phone display when i call from
asterisk.
My current extensions.conf looks like below
[general]
static=yes
writeprotect=no
autofallthrough=no
extenpatternmatchnew=no
clearglobalvars=no
priorityjumping=yes
userscontext=default
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=DAHDI/G1
TRUNKMSD=1
[Internal]
include = Incoming
exten = 8001234001,1,Dial(DAHDI/32,,rt)
exten = 8001234002,1,Dial(DAHDI/33,,rt)
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234004,1,Set(CALLERID(num)=8001234004)
exten = 8001234004,n,Set(CALLERID(name)=Line 4)
exten = 8001234004,3,Dial(DAHDI/35,,rt)
exten = 8001234005,1,Dial(DAHDI/36,,rt)
[Incoming]
exten = s,1,Answer
exten = s,2,Dial(DAHDI/g1,20,rt)
exten = _988.,1,Dial(DAHDI/g1/${EXTEN},20)
I also tried changing the dial plan to exten =
_988.,3,Dial(DAHDI/g1/${EXTEN},20) and in that case the call itself was
not going through
Venugopal
From: asterisk-users-boun...@lists.digium.com on behalf of Jimmy Godbout
Sent: Thu 3/4/2010 5:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID in Asterisk
Hi,
You need to set the callerid before making the call, not after. Also, I
guess it's a typo that the priority in this dialplan is all 1; it should
be
exten = 8001234003,1,Set(CALLERID(num)=8001234003)
exten = 8001234003,n,Set(CALLERID(name)=Line 5)
exten = 8001234003,n,Dial(DAHDI/34,,rt)
Unless your using variable for the name and the number, you should not
put them in ${}.
Jimmy
-Original Message-
From: venui...@motorola.com
Sent: Thu, 4 Mar 2010 19:50:03 +0800
To: asterisk-users@lists.digium.com,
asterisk-users@lists.digium.com, asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Caller ID in Asterisk
HI All,
Below is the ones i tried
exten = 8001234003,1,Dial(DAHDI/34,,rt)
exten = 8001234003,1,Set(CALLERID(num)=${8001234003})
exten = 8001234003,1,Set(CALLERID(name)=${Line 5})
However i got an error message sayinfg Function CallerID not
registered.
Kindly help me...
From: asterisk-users-boun...@lists.digium.com on behalf of
Gopalakrishnaiyer Venugopal-Q16770
Sent: Thu 3/4/2010 3:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
asterisk-users@lists.digium.com
Subject: [asterisk-users] Caller ID in Asterisk
Hi All,
I have an asterik machine which is connected via a PRI to the
SIP server.When i call from the Asterisk machine to the SIP server i am
not getting the caller id of the lines at the sip side.
Please help me to identify how this can be set.The
extensions.conf file is attached.
Cheers
venu
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[asterisk-users] Caller ID question
Hiya - quick question.. When an external call is answered by an extension and the person answering the call wants to forward it to a different extension, is there any way to change the caller ID when the call is transferred? If someone is transferring a call to me, I see the caller ID of the other person in the office. When the call is transferred, could the caller ID be set back to the caller ID of the original incoming call? Staff members here often want to see the number of the last person they spoke to but when they check the call history on the (snom) phone, all they can see is the extension of the person that forwarded the call to them.. I doubt it's possible but thought I'd check Thanks, Will -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID question
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Monday, February 22, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Caller ID question
Hiya - quick question..
When an external call is answered by an extension and the person answering
the call wants to forward it to a different extension, is there any way to
change the caller ID when the call is transferred?
If someone is transferring a call to me, I see the caller ID of the other
person in the office. When the call is transferred, could the caller ID be
set back to the caller ID of the original incoming call? Staff members here
often want to see the number of the last person they spoke to but when they
check the call history on the (snom) phone, all they can see is the
extension of the person that forwarded the call to them..
I doubt it's possible but thought I'd check
Thanks,
Will
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Re: [asterisk-users] Caller ID question
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor staff :)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally don't
blind transfer).
I need to change the CID on an already-established SIP channel and have no idea
if it's doable..
W
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Re: [asterisk-users] Caller ID question
The ID at dial/transfer time is what you are stuck with.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Will Payne
Sent: Monday, February 22, 2010 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID question
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor
staff :)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally
don't blind transfer).
I need to change the CID on an already-established SIP channel and have no
idea if it's doable..
W
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Re: [asterisk-users] Caller ID question
On 22 February 2010 15:59, Will Payne w...@teambadger.co.uk wrote:
On 22 Feb 2010, at 15:38, Danny Nicholas wrote:
What you need to do is set a channel variable with callerid(num) from the
external number, then reset callerid(num) whenever you do an internal dial
to transfer - something like this
[from-pstn]
Exten = s,1,answer
Exten = s,n,Set(passcallID=callerid(num))
[transfer]
Exten = s,1,set(callerid(num)=${passcallID})
Exten = s,n,dial(SIP/123)
I thought about doing something like that but it would confuse the poor staff
:)
They'd have a call from what appeared to be an external number but it would
turn out to be an internal extension that was calling them (we generally
don't blind transfer).
I need to change the CID on an already-established SIP channel and have no
idea if it's doable..
W
--
I believe what you want is called COLP Connected Line Presentation.
I was also if the opinion that it had been merged into all of the
newer versions of the Asterisk code.
If you are using Asterisk 1.4, you may find a usable patch here:
https://issues.asterisk.org/view.php?id=8824
Regards,
Steve
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[asterisk-users] Caller ID not working properly on some phones...
I have a strange problem with CallerID that only affects some phones. The problem is that whenever I receive a call the Callerid Name is correct but the Callerid number is always my own extension. It does not matter if the call is internal or external. So far only Aastra phones and Linksys PAP2T adapters seem to have this problem. Other phones like Snom and Cisco SPA525 display the correct number. I am using Asterisk 1.6.2.1 on two different servers that have the same problem. I guess there is a setting on Asterisk that the phones do not like. One of the servers was upgraded from 1.4.28 last week and we never had that problem. If I do a NoOP on the Dialplan I can see that the correct CallerID info is set but the phone will always say the number is my own extension no matter what. This is a problem because I cannot call back from the call history on the phone. CDR is correct. Any ideas what may be happening? Why would this only affect some phones and not others? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
hi all, i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE GUYS UR HELP N SUPPORT WILL BE HIGHLY APPRECIATED. THX___ -- 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
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
- Original Message - From: aster...@opensourcesolution.in To: asterisk-users@lists.digium.com Sent: Friday, November 13, 2009 9:47 AM Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE hi all, i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE guys ur help n support will be highly appreciated. There are many fine explanations on the net. Read and try, if you then have problems with the details, come back. Or you can pay a consultant to do your work Leif ___ -- 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
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
On 13 Nov 2009, at 08:47, aster...@opensourcesolution.in aster...@opensourcesolution.in wrote: I had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE guys ur help n support will be highly appreciated. thx What is it with you! ARGH. Steve ___ -- 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
Re: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE
On Fri, 13 Nov 2009, aster...@opensourcesolution.in wrote: i had installed and configured asterisk on centos 5.3, i had made a minimum dial plan in which i had made two extentions. i am easily able to make call from one extention to other extention. i know its just a basic thing which i had done n i had done from this place only. now i want to features of dial plan.i want to implement these features in my dial plan. HOLD MUSIC ON HOLD CALLER-ID QUEUE GUYS UR HELP N SUPPORT WILL BE HIGHLY APPRECIATED. THX You are amazing. You could spend your time the reading many references you have been supplied yet you prefer to beg for others to do your job for you. Please go away. UR ABSENCE WILL BE HIGHLY APPRECIATED. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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
Re: [asterisk-users] Caller ID from POTS lines
and how are those POTS lines connected to Asterisk?
In any event doing something like:
Set(CALLERID(num)=${CALLERID(num):0:10}) should do the trick.
On Tue, Sep 8, 2009 at 12:27 PM, Jeremy Taylor jer...@getwiredright.com wrote:
Hi,
I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When
calls come in on our POTS lines, the caller id shows up like
555-555-1...@192.168.1.10 where 555-555-1234 is the correct phone
number and 192.168.1.10 is my pbx server IP. This format does not work
for redialing on outbound calls.
While there may be an outbound dialing change that could be made, it
seems like the correct solution would be to change the format of the
caller id string sent to the phones. I verified from the snom sip
trace that the caller id is always sent with @192.168.1.10 on it.
What configuration change can be made in asterisk to correct this and
only send the phone number as the caller id to the VOIP phone?
Thanks, Jeremy
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[asterisk-users] Caller ID from POTS lines
Hi, I'm using asterisk 1.4.22-4 in Trixbox with snom 360 phones. When calls come in on our POTS lines, the caller id shows up like 555-555-1...@192.168.1.10 where 555-555-1234 is the correct phone number and 192.168.1.10 is my pbx server IP. This format does not work for redialing on outbound calls. While there may be an outbound dialing change that could be made, it seems like the correct solution would be to change the format of the caller id string sent to the phones. I verified from the snom sip trace that the caller id is always sent with @192.168.1.10 on it. What configuration change can be made in asterisk to correct this and only send the phone number as the caller id to the VOIP phone? Thanks, Jeremy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] caller id problem
I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] caller id problem
Yes, the issue(s) is/are: 1. The VOIP provider may be masking the callerID for their own cost allocation reasons. That is some of the issue. 2. Your Asterisk box may forward some of the regular phone line calls with their caller ID. 3. Somehow, the number you want to use may leak through sometimes. :-) What you need to do is put in a simple, absolute CallerID(num) = 3216540987 type of statement before sending the call out. Make it apply to every call no matter what. That isn't the syntax but you get the idea. Of course you won't have true caller ID then, but do you want cheap or real? Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan Sent: Friday, August 07, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] caller id problem I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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] caller id problem
Hi Cary, Thanks for the quick reply :D I get what you're saying. I have a suspicion that it is the telco's fault since every other number that receives a call from my Asterisk box displays the correct number. I'll give setting the caller id another go and play with that. I guess what I am looking for is a) confirmation that this problem has happened to other people and b) a suggestion of how to point the tech support in the right direction so they can fix this problem for me, or how I can just override this problem myself. Thanks again for your help and quick reply. Cary Fitch wrote: Yes, the issue(s) is/are: 1. The VOIP provider may be masking the callerID for their own cost allocation reasons. That is some of the issue. 2. Your Asterisk box may forward some of the regular phone line calls with their caller ID. 3. Somehow, the number you want to use may leak through sometimes. :-) What you need to do is put in a simple, absolute CallerID(num) = 3216540987 type of statement before sending the call out. Make it apply to every call no matter what. That isn't the syntax but you get the idea. Of course you won't have true caller ID then, but do you want cheap or real? Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan Sent: Friday, August 07, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] caller id problem I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] caller id problem
On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca wrote: I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. Since this is already a little off-topic, care to share which provider you are using? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] caller id problem
David Backeberg wrote: On Fri, Aug 7, 2009 at 1:48 PM, Terry Nathantnat...@aiinc.ca wrote: I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. Since this is already a little off-topic, care to share which provider you are using? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 Yeah, no problem. The telco is Telus in British Columbia, Canada and Digital Voice in Vancouver is my VOIP provider. I'd rather not have to switch telcos as there is always some nice fees and charges when you sign up. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Caller ID (name) - where does it come from?
Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote: Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards Whether true or not, I was told that nearly 80% of people in the US have caller ID. I would say that number is much higher for business, especially on PRI circuits. I think the two big motivators there were packaging of services, for X amount extra, you get caller ID, call waiting, voicemail on at the telco, etc The other factor was the proliferation of telemarketing. Before the DNC, a white pages listed home phone could ring a dozen times a day by people selling stuff. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
On 7/8/09, Steve Totaro stot...@first-notification.com wrote: On Wed, Jul 8, 2009 at 2:14 AM, Olivieroza-4...@myamail.com wrote: Hi, Reading this thread, is this correct to say CallerName is widely used in the US ? Here in France, this service is optional but I don't think many companies are subscribing to it and I'm not aware of any non-Telco CNAM providers. I would curious to know how the situation is elsewhere. Regards Whether true or not, I was told that nearly 80% of people in the US have caller ID. I would say that number is much higher for business, especially on PRI circuits. I think the two big motivators there were packaging of services, for X amount extra, you get caller ID, call waiting, voicemail on at the telco, etc The other factor was the proliferation of telemarketing. Before the DNC, a white pages listed home phone could ring a dozen times a day by people selling stuff. -- Thanks, Steve Totaro In Canada, their telephone network is set up to allow for dynamic CallerIDname on PRIs just like how CallerIDnumber works here in the USA. We didn't believe it at first until we tried it, but they seem to be the only country we've worked in, out of a few dozen countries, that allows dynamic CallerIDname defined on a per-call basis. MATT--- ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
CALLERID(name) is a TELCO specific field. In the long run, you will be best served using your own lookup of a database using CALLERID(num), since CID(name) is unreliable and in some cases costly. IMO, you would be well served with an app (AGI?) that recorded valid names into the database and let you insert the names where they aren't. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry D. Hassler Sent: Tuesday, July 07, 2009 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID (name) - where does it come from? Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
Well, Teliax says they have no access to the PSTN's database, but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). On Wed, Jul 8, 2009 at 8:51 AM, Danny Nicholas da...@debsinc.com wrote: CALLERID(name) is a TELCO specific field. In the long run, you will be best served using your own lookup of a database using CALLERID(num), since CID(name) is unreliable and in some cases costly. IMO, you would be well served with an app (AGI?) that recorded valid names into the database and let you insert the names where they aren’t. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Barry D. Hassler *Sent:* Tuesday, July 07, 2009 12:41 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Caller ID (name) - where does it come from? Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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 -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
Barry D. Hassler wrote: Well, Teliax says they have no access to the PSTN's database, but I'm suggesting they check out TargusInfo as mentioned above. One of their suggestions, is to contact the local ILEC to get the number published in their white pages. Will that accomplish the same thing (I doubt it). As I understand it, if they got a document signed by their origination provider granting them authorization to do CNAM hosting on their own numbers, they could then hire someone such as Verisign to host their CNAM records in the so-called PSTN database. They'd even profit from this assuming they have enough subscribers. There are probably several reasons for why they don't do this, possibly starting with administrative overhead and/or their provider is not willing to relinquish control of the records. If someone has experience with this, feel free to correct me. However, this is my understanding from my previous experience with looking up Caller Name information via CNAM/LIDB/SS7. Regards, -- Trevor Peirce Digital Conceptions Canada http://www.digitalcon.ca 1-888-606-3030 / 250 483-0386 ___ -- 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] Caller ID (name) - where does it come from?
Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
On Tue, Jul 7, 2009 at 1:40 PM, Barry D. Hasslerbarry.hass...@gmail.com wrote: Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 It is in a database or CNAM dip. You just need to contact your provider and tell them to have it changed. What you send is moot on the PSTN. Also call 911 and make sure they have the correct address and information on file. I do it all the time for liability reasons, just make sure you tell them right off that bat that there is no emergency and you want to verify what they have in their database is correct. I have never had a problem doing this and try to do it in front of the big boss to show them it is correct and that I am thorough and looking out for them. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
The Caller ID name, CNAM is a separate database owned and maintained
cooperatively by the bell operating companies.
Your ITSP is not doing these CNAM lookups for you because they would have to
pay the BOC's for the 'dips' into the CNAM database. CNAM is a little cash
cow that the BOC's are quick to protect. As such CNAM dips may not be
cached or re-sold as a term service that you must agree to with your CNAM
provider.
As far as solving your CNAM problem, you would need to either choose an ITSP
that will provide you with CNAM data on a per-call basis, OR you need to do
CNAM dips yourself as I (and many others) do. Beware that some ITSP's
provide best-effort name data culled from various sources. It's not
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
As a call comes in to your dial plan you can populate the CALLERID(name)
channel variable using the CURL function in your dialplan as so:
exten =
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=Cf=Sdn=${CALLERID(num)})})
AND let's not forget the completely separate issue with getting your
ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
so people see Karl Fife rather than the city, state or worse, some
string of arcane LATA information. There's a solution to this problem too
but I digress...
I've posted my personal notes below from about 18 months ago when I was
searchign for CNAM providers:
-Karl
CNAM PROVIDRES:
Metrostat.com
about 1.5¢ per dip,
$30 minimum deposit, refundable
CNAM service not well documented on web site
A registerd CLEC
Got Name - Out of business?
1.5¢ per dip. no minimums, no setup
ClearReach Networks
.67¢ per dip $200 monthly minimum, resell ok, significant setup fees
411xml.com
more expensive than ClearReach.
- Original Message -
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 07, 2009 12:40 PM
Subject: [asterisk-users] Caller ID (name) - where does it come from?
Hi Folks, having an issue with outbound calls through a VOIP provider. Calls
get sent out with the CallerID(number), but where does callerID(name) come
from? Apparently not from provider, as we are seeing different (sometime
missing) names on inbound calls, different than what we have configured.
Apparently this comes from some telco database somewhere? Numbers were
ported from a wired-telco.
--
Barry D. Hassler
President, HCST
http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Caller ID (name) - where does it come from?
p.s. Once you've got a reliable CNAM source, you can save a few bucks per month on all of your POTS lines PRI spans by opting out of the carrier-provided CNAM. IIRC, We save something like $40 per month per span on our PRI's $3 per month per line by opting out of CNAM. When a call comes in we populate it ourselves using a quick HTTP GET. -Karl - Original Message - From: Barry D. Hassler To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, July 07, 2009 12:40 PM Subject: [asterisk-users] Caller ID (name) - where does it come from? Hi Folks, having an issue with outbound calls through a VOIP provider. Calls get sent out with the CallerID(number), but where does callerID(name) come from? Apparently not from provider, as we are seeing different (sometime missing) names on inbound calls, different than what we have configured. Apparently this comes from some telco database somewhere? Numbers were ported from a wired-telco. -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -- ___ -- 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___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
There's a bit of oversimplification going on here -- it's not a ...
database. Different CNAM providers have different databases which are
populated from many sources. Most of the data probably matches, but not all
of it.
If the Calling Name is incorrect, the person who received the call will have
to check with their telephony provider (or, if they do their own CNAM
lookups, with their CNAM provider) to get the name for the calling party
fixed up (this presumes that the calling party has already verified with
their own telephony provider that their name is correctly listed). But
that's not all of it, either, because the next time the CNAM provider
refreshes their records, the local fix could be overridden (I'm not sure if
any CNAM providers have the capability to ignore old/bad data for a record,
but perhaps so). Ideally the CNAM provider shares with the calling party
which database the CNAM provider is using for the calling party, so that the
calling party can try to get it fixed directly with the database provider
(if that's even possible).
In short, it's a mess.
But because accuracy rates are one of the elements that CNAM providers
compete on, these usually do get cleaned up.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
The Caller ID name, CNAM is a separate database owned and maintained
cooperatively by the bell operating companies.
Your ITSP is not doing these CNAM lookups for you because they would have to
pay the BOC's for the 'dips' into the CNAM database. CNAM is a little cash
cow that the BOC's are quick to protect. As such CNAM dips may not be
cached or re-sold as a term service that you must agree to with your CNAM
provider.
As far as solving your CNAM problem, you would need to either choose an ITSP
that will provide you with CNAM data on a per-call basis, OR you need to do
CNAM dips yourself as I (and many others) do. Beware that some ITSP's
provide best-effort name data culled from various sources. It's not
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
As a call comes in to your dial plan you can populate the CALLERID(name)
channel variable using the CURL function in your dialplan as so:
exten =
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=Cf=Sdn
=${CALLERID(num)})})
AND let's not forget the completely separate issue with getting your
ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
so people see Karl Fife rather than the city, state or worse, some
string of arcane LATA information. There's a solution to this problem too
but I digress...
I've posted my personal notes below from about 18 months ago when I was
searchign for CNAM providers:
-Karl
CNAM PROVIDRES:
Metrostat.com
about 1.5¢ per dip,
$30 minimum deposit, refundable
CNAM service not well documented on web site
A registerd CLEC
Got Name - Out of business?
1.5¢ per dip. no minimums, no setup
ClearReach Networks
.67¢ per dip $200 monthly minimum, resell ok, significant setup fees
411xml.com
more expensive than ClearReach.
- Original Message -
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 07, 2009 12:40 PM
Subject: [asterisk-users] Caller ID (name) - where does it come from?
Hi Folks, having an issue with outbound calls through a VOIP provider. Calls
get sent out with the CallerID(number), but where does callerID(name) come
from? Apparently not from provider, as we are seeing different (sometime
missing) names on inbound calls, different than what we have configured.
Apparently this comes from some telco database somewhere? Numbers were
ported from a wired-telco.
--
Barry D. Hassler
President, HCST
http://www.hcst.net/
937-427-9000
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Re: [asterisk-users] Caller ID (name) - where does it come from?
This is all excellent information. My primary issue is for calls that are
placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of
those calls are the ones that are not getting the proper CNAM information as
the call comes in.
We just recently ported the client's POTS lines to VOIP, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.
On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk frnk...@iname.com wrote:
There's a bit of oversimplification going on here -- it's not a ...
database. Different CNAM providers have different databases which are
populated from many sources. Most of the data probably matches, but not
all
of it.
If the Calling Name is incorrect, the person who received the call will
have
to check with their telephony provider (or, if they do their own CNAM
lookups, with their CNAM provider) to get the name for the calling party
fixed up (this presumes that the calling party has already verified with
their own telephony provider that their name is correctly listed). But
that's not all of it, either, because the next time the CNAM provider
refreshes their records, the local fix could be overridden (I'm not sure if
any CNAM providers have the capability to ignore old/bad data for a record,
but perhaps so). Ideally the CNAM provider shares with the calling party
which database the CNAM provider is using for the calling party, so that
the
calling party can try to get it fixed directly with the database provider
(if that's even possible).
In short, it's a mess.
But because accuracy rates are one of the elements that CNAM providers
compete on, these usually do get cleaned up.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
The Caller ID name, CNAM is a separate database owned and maintained
cooperatively by the bell operating companies.
Your ITSP is not doing these CNAM lookups for you because they would have
to
pay the BOC's for the 'dips' into the CNAM database. CNAM is a little cash
cow that the BOC's are quick to protect. As such CNAM dips may not be
cached or re-sold as a term service that you must agree to with your CNAM
provider.
As far as solving your CNAM problem, you would need to either choose an
ITSP
that will provide you with CNAM data on a per-call basis, OR you need to do
CNAM dips yourself as I (and many others) do. Beware that some ITSP's
provide best-effort name data culled from various sources. It's not
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
As a call comes in to your dial plan you can populate the CALLERID(name)
channel variable using the CURL function in your dialplan as so:
exten =
s,n,Set(CALLERID(name)=${CURL(
http://cnam1.edicentral.net/getcnam?q=Cf=Sdn
=${CALLERID(num)})})
AND let's not forget the completely separate issue with getting your
ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
so people see Karl Fife rather than the city, state or worse, some
string of arcane LATA information. There's a solution to this problem too
but I digress...
I've posted my personal notes below from about 18 months ago when I was
searchign for CNAM providers:
-Karl
CNAM PROVIDRES:
Metrostat.com
about 1.5¢ per dip,
$30 minimum deposit, refundable
CNAM service not well documented on web site
A registerd CLEC
Got Name - Out of business?
1.5¢ per dip. no minimums, no setup
ClearReach Networks
.67¢ per dip $200 monthly minimum, resell ok, significant setup fees
411xml.com
more expensive than ClearReach.
- Original Message -
From: Barry D. Hassler
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, July 07, 2009 12:40 PM
Subject: [asterisk-users] Caller ID (name) - where does it come from?
Hi Folks, having an issue with outbound calls through a VOIP provider.
Calls
get sent out with the CallerID(number), but where does callerID(name) come
from? Apparently not from provider, as we are seeing different (sometime
missing) names on inbound calls, different than what we have configured.
Apparently this comes from some telco database somewhere? Numbers were
ported from a wired-telco.
--
Barry D. Hassler
President, HCST
http://www.hcst.net/
937-427-9000 http://www.hcst.net/%0A937-427-9000
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Re: [asterisk-users] Caller ID (name) - where does it come from?
On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote: This is all excellent information. My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. We just recently ported the client's POTS lines to VOIP, and with the exception of this issue, all is working well. But, my client is really unhappy that their callerID NAME isn't showing up. snip I was very curious about this myself. We successfully set the CallerID number by creating different contexts for our various offices and using a Set(CALLERID(num)=x) call. But we could not set the name so I asked our new carrier (Vitelity - with whom we have been quite pleased thus far). This is their response to us: We can have the name set for this number, however there is a one time passthrough charge of $xx per number for the update. Outbound caller ID is updated into a national database called LIDB (line information database), it is the final terminating provider that is responsible for querying this database and delivering it to their customers. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
If the calling number shows up correctly for the called party (an obvious
first step), the called party will need to get in contact with their
telephony provider/CNAM vendor to get the calling name fixed. Its possible
that because your client (the calling party) ported their number, the called
partys CNAM source reflects no information because the line was
disconnected. The called partys CNAM source is obviously not getting
directory listing directly or indirectly from Teliax.
It may be helpful to speak to Teliax and find out where they sell/provide
their directory listings. Somehow the called partys CNAM source needs to
get that information from Teliax, either directly, or more likely, via one
or more intermediate parties that aggregates the data. TARGUSinfo
(http://targusinfo.com/solutions/identification/caller_name/default.aspx),
for example, collects from over 90 sources
(http://targusinfo.com/solutions/identification/caller_name/faq/).
Ive heard that Vonage Canada does not sell/provide their directory
listings, so youll never obtain a name-like calling number unless your CNAM
provider collects the data from other sources, and they do (e.g. department
store credit cards applications).
Frank
From: Barry D. Hassler [mailto:barry.hass...@gmail.com]
Sent: Tuesday, July 07, 2009 3:54 PM
To: frnk...@iname.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
This is all excellent information. My primary issue is for calls that are
placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of
those calls are the ones that are not getting the proper CNAM information as
the call comes in.
We just recently ported the client's POTS lines to VOIP, and with the
exception of this issue, all is working well. But, my client is really
unhappy that their callerID NAME isn't showing up.
On Tue, Jul 7, 2009 at 3:42 PM, Frank Bulk frnk...@iname.com wrote:
There's a bit of oversimplification going on here -- it's not a ...
database. Different CNAM providers have different databases which are
populated from many sources. Most of the data probably matches, but not all
of it.
If the Calling Name is incorrect, the person who received the call will have
to check with their telephony provider (or, if they do their own CNAM
lookups, with their CNAM provider) to get the name for the calling party
fixed up (this presumes that the calling party has already verified with
their own telephony provider that their name is correctly listed). But
that's not all of it, either, because the next time the CNAM provider
refreshes their records, the local fix could be overridden (I'm not sure if
any CNAM providers have the capability to ignore old/bad data for a record,
but perhaps so). Ideally the CNAM provider shares with the calling party
which database the CNAM provider is using for the calling party, so that the
calling party can try to get it fixed directly with the database provider
(if that's even possible).
In short, it's a mess.
But because accuracy rates are one of the elements that CNAM providers
compete on, these usually do get cleaned up.
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife
Sent: Tuesday, July 07, 2009 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID (name) - where does it come from?
The Caller ID name, CNAM is a separate database owned and maintained
cooperatively by the bell operating companies.
Your ITSP is not doing these CNAM lookups for you because they would have to
pay the BOC's for the 'dips' into the CNAM database. CNAM is a little cash
cow that the BOC's are quick to protect. As such CNAM dips may not be
cached or re-sold as a term service that you must agree to with your CNAM
provider.
As far as solving your CNAM problem, you would need to either choose an ITSP
that will provide you with CNAM data on a per-call basis, OR you need to do
CNAM dips yourself as I (and many others) do. Beware that some ITSP's
provide best-effort name data culled from various sources. It's not
always terrible but it's not 'coke' it's more like 'dollar store' cola. :-)
As a call comes in to your dial plan you can populate the CALLERID(name)
channel variable using the CURL function in your dialplan as so:
exten =
s,n,Set(CALLERID(name)=${CURL(http://cnam1.edicentral.net/getcnam?q=C
http://cnam1.edicentral.net/getcnam?q=Cf=Sdn f=Sdn
=${CALLERID(num)})})
AND let's not forget the completely separate issue with getting your
ITSP-provisioned number ENTERED INTO the CNAM database in the first place,
so people see Karl Fife rather than the city, state or worse, some
string of arcane LATA information. There's a solution to this problem too
but I digress...
I've posted my personal notes below from about 18 months ago when I
Re: [asterisk-users] Caller ID (name) - where does it come from?
Intersting. Vitelity is charging for something that they might already be getting paid for. Of course, updating a name for a number takes time, and so that's probably why they can justify charging the customer something. Most times when you sign up you specify how you want the directory listing to look, and that's what is sold/delivered to CNAM vendors and aggregators. I'm not sure what Vitelity means by a national database and this database. As I discussed before, a telephony provider can choose pretty well any CNAM vendor they want. Beyond the ones that were mentioned by someone else in an e-mail, there's also VeriSign, Neustar, and Syniverse. It's an oversimplification to tell the customer that it's *a* database -- Vitelity may sell their data to just one CNAM vendor/aggregator, but that doesn't mean every CNAM vendor's database has now been updated. Frank -Original Message- From: John A. Sullivan III [mailto:jsulli...@opensourcedevel.com] Sent: Tuesday, July 07, 2009 7:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: frnk...@iname.com Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote: This is all excellent information. My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. We just recently ported the client's POTS lines to VOIP, and with the exception of this issue, all is working well. But, my client is really unhappy that their callerID NAME isn't showing up. snip I was very curious about this myself. We successfully set the CallerID number by creating different contexts for our various offices and using a Set(CALLERID(num)=x) call. But we could not set the name so I asked our new carrier (Vitelity - with whom we have been quite pleased thus far). This is their response to us: We can have the name set for this number, however there is a one time passthrough charge of $xx per number for the update. Outbound caller ID is updated into a national database called LIDB (line information database), it is the final terminating provider that is responsible for querying this database and delivering it to their customers. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. I neglected to go into detail on this point at the end of my last post because I thought it was out of scope. But now that you ask... While I am not an expert in the specific architecture of CNAM database, I do know that (to Frank's point) it is not at all a database in the 'MySQL' or 'Oracle' sense of the word. It's a database more analagous to the DNS where data can be located in many places (cached) but there is a single source considered authoritative that ultimately propogates out to cache. This authoritative source is the Telco that provides your DID number--after all, they the only ones with a billing relationship to validate the name information. So historically, *normally* your Telco is the authoritative source of the CNAM data that populates the 'screens' of the people you call, and *normally* the Telco of the calling party is ultimately compensated by the Telco of the called party for providing the CNAM data, but this model has broken down in the world if IP telephohy. Your ITSP (Teliax) is one of them-thar new-fangled ITSPs and the big boys have exactly ZERO interest in compensating them for CNAM dips. Meanwhile they are excluded from the holy brotherhood of 'real' CNAM. This is why your name is not populated in the CNAM database. Teliax is not one of the CNAM insiders who exchange name data and compensate each other for said data. That's also why it would never make sense to ask your CNAM lookup serive provider to make corrections to errant CNAM data. It just doesn't work that way. It used to be that you could work around this problem by using LNP to port your number temporarily to an ILEC . Your TN would get a CNAM record which would persist as an orphan for years. Recently this has changed, and NOW when you port your TN away from the losing LEC, they purge your CNAM record. :-( Recently there are some good solutions to this problem. One is to ask your ITSP if they can put your number in the LIDB for a fee or alternatively you can just buy a white pages entry (also from your ITSP) which accomplishes the same thing. I've seen this for $5 per month, and the BONUS you get a white pages entry (which you may or may not want). I hope this helps. -Karl http://www.hcst.net/ 937-427-9000 ___ -- 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 ___ -- 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
Re: [asterisk-users] Caller ID (name) - where does it come from?
On Tue, Jul 7, 2009 at 9:06 PM, Frank Bulkfrnk...@iname.com wrote: Intersting. Vitelity is charging for something that they might already be getting paid for. Of course, updating a name for a number takes time, and so that's probably why they can justify charging the customer something. Most times when you sign up you specify how you want the directory listing to look, and that's what is sold/delivered to CNAM vendors and aggregators. I'm not sure what Vitelity means by a national database and this database. As I discussed before, a telephony provider can choose pretty well any CNAM vendor they want. Beyond the ones that were mentioned by someone else in an e-mail, there's also VeriSign, Neustar, and Syniverse. It's an oversimplification to tell the customer that it's *a* database -- Vitelity may sell their data to just one CNAM vendor/aggregator, but that doesn't mean every CNAM vendor's database has now been updated. Frank -Original Message- From: John A. Sullivan III [mailto:jsulli...@opensourcedevel.com] Sent: Tuesday, July 07, 2009 7:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: frnk...@iname.com Subject: Re: [asterisk-users] Caller ID (name) - where does it come from? On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote: This is all excellent information. My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. We just recently ported the client's POTS lines to VOIP, and with the exception of this issue, all is working well. But, my client is really unhappy that their callerID NAME isn't showing up. snip I was very curious about this myself. We successfully set the CallerID number by creating different contexts for our various offices and using a Set(CALLERID(num)=x) call. But we could not set the name so I asked our new carrier (Vitelity - with whom we have been quite pleased thus far). This is their response to us: We can have the name set for this number, however there is a one time passthrough charge of $xx per number for the update. Outbound caller ID is updated into a national database called LIDB (line information database), it is the final terminating provider that is responsible for querying this database and delivering it to their customers. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com I get paid every time I call someone that subscribes to caller ID. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
