"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas
You need: exten => s/_0.,Answer Where the "." is a wildcard. It would match 001 or 01234567, etc. Regards, Jeff On Mon, Mar 03, 2003 at 03:43:48PM +0100, Tomaz Izanc wrote:> > "In general you can match callerID with the /, but if you don't put > anything after the /, then the rule matches "no caller*ID", and if no > slash is there at all, it matches "any callerid". " > > > > Ok.My question is -> > > how to match callerid from 001... ? > and if don't know how many numbers ? > > exten => s/0_,Answer don't work- > anything else ? > > tnx > Thomas > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
> how to match callerid from 001... ? > and if don't know how many numbers ?You can do it the same way you match extensions: exten => s/_001.,1,blabla exten => s/_00[2-90].,1,bleble regards Martin> > exten => s/0_,Answer don't work- > anything else ? > > tnx > Thomas > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users at lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. Zapata.conf [channels] musiconhold=default callerid=asreceived threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 immediate=no context=default-nga signalling=featd group=2 channel => 5-8 context=default-tne signalling=featd group=1 channel => 1-4 context=default-pb signalling=featd group=3 channel => 9-12 context=default=ctm signalling=featd group=3 channel => 13-14 Any thoughts.... -gcc
Resolved the issue... It turned out to be a problem with the ISP -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of AstGrp Posted At: Monday, April 05, 2004 2:21 PM Posted To: Asterisk User Group Conversation: CallerID Subject: [Asterisk-Users] CallerID I am having an issue with Callerid (INBOUND). I have a system set up with 4 companies sitting behind the system. On all of the companies except of one of them, it displays callerid withh 'asterisk'. The other company displays the callerid of the person calling. Zapata.conf [channels] musiconhold=default callerid=asreceived threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 immediate=no context=default-nga signalling=featd group=2 channel => 5-8 context=default-tne signalling=featd group=1 channel => 1-4 context=default-pb signalling=featd group=3 channel => 9-12 context=default=ctm signalling=featd group=3 channel => 13-14 Any thoughts.... -gcc _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Thank you for the response... After talking with the ISP they did not have CID turned on all of the trunk groups. This has since been resolved. Thanks again, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of C. Maj Posted At: Tuesday, April 06, 2004 1:41 PM Posted To: Asterisk User Group Conversation: [Asterisk-Users] CallerID Subject: Re: [Asterisk-Users] CallerID On Mon, 5 Apr 2004, AstGrp waxed:> I am having an issue with Callerid (INBOUND). I have a system set up > with 4 companies sitting behind the system. On all of the companies > except of one of them, it displays callerid withh 'asterisk'. The > other company displays the callerid of the person calling.> callerid=asreceivedThat's a good line to have.> context=default-nga > signalling=featd > group=2 > channel => 5-8> context=default-tne > signalling=featd > group=1 > channel => 1-4> context=default-pb > signalling=featd > group=3 > channel => 9-12> context=default=ctm > signalling=featd > group=3 > channel => 13-14What context is the company in that gets the cid right ? Maybe you are only receiving the cid on certain channels ? Why do you have 3 groups, but 4 contexts ? Is everything hooked up to a channel bank ? What kind of hardware is installed on the box ? Are you explicitly setting the cid in extensions.conf ? --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
In my zapata.conf, I have callerid="<unknown>" <> so if an incoming call doesn't set or suppresses it's callerid then my phone will show "unknown". I have found that if the callerid on the incoming call is suppressed, then the call goes straight to Voicemail. Has anyone seen this problem? Simon Brown
More on this issue - this is what I see on the console:
-- Called 201
-- Called 202
-- Got SIP response 400 "Bad Request" back from 10.2.2.102
-- Got SIP response 400 "Bad Request" back from 10.2.2.104
== No one is available to answer at this time
I am using Cisco 7940 phones
Simon Brown
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Simon Brown
Sent: Friday, 20 August 2004 13:51
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CallerId
In my zapata.conf, I have
callerid="<unknown>" <>
so if an incoming call doesn't set or suppresses it's callerid then my
phone
will show "unknown". I have found that if the callerid on the
incoming call
is suppressed, then the call goes straight to Voicemail.
Has anyone seen this problem?
Simon Brown
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That did it - thanks. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Darryl Ross Sent: Friday, 20 August 2004 14:05 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] CallerId Simon Brown wrote:> In my zapata.conf, I have > callerid="<unknown>" <>That doesn't look right to me. Try: callerid="Unknown" <> Cheers Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: darryl@oeg.com.au Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole "There is more than one way to do it" slogan, you have to give someone a swiss army chainsaw ... _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all, I was wondering how the easiest way to restrict the users ability to set caller ID would be ? I have some users that uses IAX to connect with me. multiple numers via iax. on outgoing calls I would like the user to only be able to set his "range" of numbers on the outgoing calls. Is there an easy way to do this ? /Mike
Use a separate context for the outbound calls for that customer, check the caller ID in the dialplan before completing an outbound call using a PATTERN MATCH, and IF the pattern does not match the pattern of the customers numbers GOTO a step that sets the caller ID to the customers main phone number, then resume (GOTO) where you left off in the dialplan. Advise your customer that the caller ID they transmit must match known numbers or it will be changed by * before the call is completed. Make sure your terms of service agreement explains this carefully because it is not typical, with a lot of commercial PRIs you can set your caller ID to anything you wish.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > micke@party.pp.se > Sent: Thursday, December 30, 2004 8:11 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] callerid > > > Hi all, > > I was wondering how the easiest way to restrict the users > ability to set caller ID would be ? > > I have some users that uses IAX to connect with me. multiple > numers via iax. > > on outgoing calls I would like the user to only be able to > set his "range" of numbers on the outgoing calls. > > Is there an easy way to do this ? > > /Mike > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Damon Estep wrote:> Use a separate context for the outbound calls for that customer, > check the caller ID in the dialplan before completing an outbound > call using a PATTERN MATCH, and IF the pattern does not match the > pattern of the customers numbers GOTO a step that sets the caller ID > to the customers main phone number, then resume (GOTO) where you left > off in the dialplan. > > Advise your customer that the caller ID they transmit must match > known numbers or it will be changed by * before the call is > completed. Make sure your terms of service agreement explains this > carefully because it is not typical, with a lot of commercial PRIs > you can set your caller ID to anything you wish. >Ok I see. IS there an example to look at somewhere ? /Mike
On Thu, 30 Dec 2004 micke@party.pp.se wrote:> I was wondering how the easiest way to restrict the users ability to set > caller ID would be ? > > I have some users that uses IAX to connect with me. multiple numers via > iax. > > on outgoing calls I would like the user to only be able to set his > "range" of numbers on the outgoing calls. > > Is there an easy way to do this ?Either use different contexts with the tests in the dialplan like another poster suggested or do a database lookup and check if the number is valid. Peter
Guys... I see there is a callerid parameter on zapata.conf... what does that cid modify? the callerid people see when you call them using any PSTN line? Is there a way to send the SIP phone the incoming callerid frpm PSTN lines asrecevied and append some string depending on the line it is coming from? __________________________________________________________________ Anton Krall
What re you guys doing for windows callerid from Asterisk besides using yac? Any other working software? I have the tapi driver installed but all software I have tried doesn't seem to work or doesn't support the asterisk tapi driver. Any suggestions?
Let mek now what you need Florian and Ill send it offlist. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Florian Overkamp |Sent: Domingo, 22 de Mayo de 2005 03:56 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] CallerID | |Hi, | |Citeren Anton Krall <akrall-lists@intruder.com.mx>: | |> Seems to me Im been displayed both... How can I control it? | |No way to know that without more in-depth knowledge about your |configuration (i.e.dialplan, what channel have you configured |in asttapi etc.) | |Florian |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
I am having a problem communicating with my asterisk box behind a Cisco router. I am running NAT on the inside and wanted to port forward to the asterisk IP but it is not working. I must be missing something.. This is the NAT statement I am using: ip nat inside source static udp 10.2.1.50 4569 interface Serial0 4569 ip nat inside source static udp 10.2.1.50 22 interface Serial0 7022 I realize that this may not be a totally asterisk question, so if anyone can offer a suggestion, a reply off-list is appreciated. Kevin
Hi, Citeren Anton Krall <akrall-lists@intruder.com.mx>:> Let mek now what you need Florian and Ill send it offlist.> |> Seems to me Im been displayed both... How can I control it? > | > |No way to know that without more in-depth knowledge about your > |configuration (i.e.dialplan, what channel have you configured > |in asttapi etc.)Show me extensions.conf (and possibly files included there) Also please tell me the following - What channel is making the call - What context is that channel making the call in - What channel is receiving the call - What channel do you have configured in ASTTAPI -- Met vriendelijke groet, Florian Overkamp ObSimRef BV
Anyway that works would be nice :) What I would like to do Is get the callerid and then pass that parameter in order to open a http url containing the callerid to pass or open a CRM webpage like SugarCRM. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Peter Bowyer |Sent: Martes, 24 de Mayo de 2005 10:50 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] CallerID | |On 24/05/05, Anton Krall <akrall-lists@intruder.com.mx> wrote: |> I would like yac to have the hability of running any |external command |> line for example, opening a http:// url or running any command line. | |OK - presumably with parameter substitution from the incoming |callerid and/or extension - or would you want to send the |entire command line / URL from the dialplan? | |Peter | | |-- |Peter Bowyer |Email: peter@bowyer.org |Tel: +44 1296 768003 |VoIP: sip:peter@bowyer.org |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
Matt Riddell wrote:> Do you want me to write you something?Heh, looks like you have a way of interesting developers! -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
We are mostly trying to figure out ways to enable web apps to display callerid in realtime and also run crm apps passing parameters, etc. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Matt Riddell |Sent: Mi?rcoles, 25 de Mayo de 2005 11:02 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] CallerID | |Anton Krall wrote: |> Do you know any apps that can receive informatio non tcpip ports and |> display it and maybe run an external app upon receiving |something as an event? | |:) | |Did you have something in mind? | |Do you want me to write you something? | |-- |Cheers, | |Matt Riddell |_______________________________________________ | |http://www.sineapps.com/news.php (Daily Asterisk News - html) |http://www.sineapps.com/rssfeed.php (Daily Asterisk News - |rss) _______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
:) |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Matt Riddell |Sent: Mi?rcoles, 25 de Mayo de 2005 11:10 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] CallerID | |Matt Riddell wrote: |> Do you want me to write you something? | |Heh, looks like you have a way of interesting developers! | |-- |Cheers, | |Matt Riddell |_______________________________________________ | |http://www.sineapps.com/news.php (Daily Asterisk News - html) |http://www.sineapps.com/rssfeed.php (Daily Asterisk News - |rss) _______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |
Has anybody been having problems with caller id with the current CVS? I jumped from Asterisk CVS-v1-0-02/21/05-19:04:26 to CVS HEAD as of 07/27/2005 7:30EST and all caller id info going from our Definity G3 to the IP phones or out our PRI longer shows information. Incoming from the IP phones to the Definity is fine. From the Asterisk console, I see caller id info going both ways. I moved back to the Asterisk CVS-v1-0-02/21/05-19:04:26 version and the caller id info returned. On this machine I'm running a TE110P under Linux 2.6.8.1-12 Thoughts? Doug
Hello All, My Asterisk is configured like a Soft Switch, I have many incoming and outgoing traffic to different providers. How can I forward CallerID to my providers. My providers are complaining big time as the CallerID is set to Zero... Please help!!! Thanks, Neal
Hi all, How i can change the CallerId format in plan id? for the example i can see the value of CALLERID variable like "lateef" <110001> I want to let asterisk do in plain id like lateef any idea? -- Thank You, Code Lover
Assuming its a SIP based device [110001] user=something allow=whatever callerid= lateef Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Code Lover wrote:> Hi all, > > How i can change the CallerId format in plan id? > for the example i can see the value of CALLERID variable like > > "lateef" <110001> > > I want to let asterisk do in plain id like > > lateef > > > any idea? > > -- > Thank You, > Code Lover > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hi, I am using SIPS softphoe. and i tested with another SIP Gatekeeper and i can see callerid in plain format. But when i am trying using Asterisk it is apearing "callerid", <username>. So i don't think this is from client side or softphone. -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________ Yahoo! DSL ? Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com
how do you set two types of caller id one for internal calling and one for external? Basically everyone calling out from asterisk from one context I want to assign a single callerid. On all other contexts I want to assign a caller ID specific to each line for all calls going out to asterisk. Finally for all calls that remain behind the asterisk box (ext to ext) the Caller ID is set to the specific extension of the caller. Thanks Miles
In article <4434BCFD.1010205@garnetweb.com>, asterisk@garnetweb.com says...> how do you set two types of caller id one for internal calling and one > for external? Basically everyone calling out from asterisk from one > context I want to assign a single callerid. On all other contexts I > want to assign a caller ID specific to each line for all calls going out > to asterisk. > > Finally for all calls that remain behind the asterisk box (ext to ext) > the Caller ID is set to the specific extension of the caller.That's easy. In sip.conf define caller id for every telephone that you wont them to have in internal calls. In every context put something like this. exten => _0.,1,Set(CALLERID(name)=Lama.hr) exten => _0.,n,Set(CALLERID(number)=00038521495148) exten => _0.,n,Dial(OOH323/${EXTEN:1}@xxx.xxx.xxx.xxx,60,TW) exten => _0.,n,Hangup Hope it helps. -- Tomislav Parcina tparcina#lama.hr
I do the same as well. From my SQL server, I have my Asterisk box query my customer and B2B contacts using ODBCSock and compose them as as CSV on the Asterisk box; a script then parses the CSV and DBPut's them into Asterisk itself. The nice thing about it is you can modify the CallerID with rich data, for example, when a customer calls, I prepend the customer ID number for our CRM into the CallerID so the staff member can type in the customer ID in to the CRM before they pick up. I have an .awk that parses the CSV and DBPut's it into Asterisk, if you are interested email me offlist. -----Original Message----- From: Alejandro Vargas [mailto:alejandro.anv@gmail.com] Sent: Friday, April 07, 2006 2:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID 2006/4/7, Miles Scruggs <asterisk@garnetweb.com>:> Could you give me an example code of how this would work, and how to > setup the database, I'm pretty new and while what you have written makes > sense, and sounds like a good plan I'm not sure I can implement it.I'm using my own agi-bin for "patching" callerid and adding the name if the number is found in a table (a csv that is mantained with a spreadsheet), it adds the name taken from this table. Then you can see the name in the display of the phones. -- Alejandro Vargas _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Michelle, you sent a single message containing suggestions to me on 11/02/2005. Your claim to have contacted me "many times" is clearly false. Due to demands outside the asterisk world, I have not been monitoring the list, but I doubt that should have been necessary, considering that contact information and even a mailing list are available for cid-rewrite. Nobody at all contacted me about reverse lookup not working, and since the script has published was "in production" here for over a month as well as on many other servers, I have to question the validity of that claim. My comments about spelling and commercial use are very productive. Much unlike you seem to, I take pride in the work I do, and being associated with something so poorly written as your changes to the readme is an embarrassment to both of parties. Additionally, programming is a very exact process, and the quality of your documentation betrays your ability. I do maintain that you are in fact misleading potential downloaders on the origin of the script. You have removed contact information and effectively taken credit for this work. You furthermore are offering paid support, which qualifies as "commercial use" and you have neither asked for nor been granted permission for commercial use of my intellectual property. Expect no help or cooperation from me in integrating your changes -- your changes are hacks at best, and a far cry from the properly architected changes I have planned and partially integrated in my production script. In the meantime, either remove the download of this bastardized script from your site, or add full contact information back into the readme file and offer FREE support for it. Please comply within 72 hours of receipt of this message. Regards, -- Jay Milk Technical Support wrote:> Jay, > > I contacted you many times regarding the script, whether you planned to > update it, suggestions for features, etc. You did not respond to any of my > later emails. Similarly, there was discussion between list members > regarding whether this script was orphaned after changes to 411.com made the > reverse lookup non-functional - for a long time. I assumed responsibility > for updating the script as a courtesy to Asterisk users. > > Your comments about spelling, resale, etc. are abrasive, unproductive, and > misleading. Not only is the script available without charge on the web > site, credit to you remains with the script - in fact even the download link > of the web site gives you credit! And of course, why would I update the > script and then encourage users to download an older version from another > site? > > If you have time to dedicate to the cid_rewrite project terrific - I would > rather see one stream benefit all users. Let's work to integrate changes > going forward. If you would prefer not to, I would be pleased to rename the > script so that there is no confusion. > > Regards, > Michelle > > -----Original Message----- > From: Jay Milk [mailto:ast-users@skimmilk.net] > Sent: Saturday, April 08, 2006 1:05 AM > To: Technical Support; Asterisk Users Mailing List - Non-Commercial > Discussion; Michael Stahl > Subject: Re: [Asterisk-Users] CallerID > > Michelle, > > 1. Courtesy would suggest that you would have contacted the author of the > script (me) to ask permission to modify this and host it elsewhere. > 2. What possessed you to remove ALL credits and original download location > from the readme file? Are you trying to pawn other people's work off as > yours? > 3. It's not exactly smart to continue someone else's versioning scheme if > you're intending to make a "fork". > 4. Your spelling is atrocious. > 5. The script is not orphaned, even though you seem to imply this in the > readme file. > > Since you are selling support for this script, that qualifies as commercial > use and is expressly prohibited by the micro-license included in the > original script. Please remove it from your download page until you have > made arrangements for further distribution with me. I'm utterly amazed at > the bad form I see here. > > Downloads of the original script are available here: > http://www.muware.com/asterisk/ > > The script is alive and working well, and I've made various enhancements to > user-requests in the recent past. > > -- JM > > > Technical Support wrote: > >> Miles, >> >> You can also download cid_rewrite from www.generationd.com This PHP >> script looks up the phone numbers in a local MySQL table, and/or uses >> reverse 411 on the web to lookup the name, and/or more options. >> >> Michelle >> >> -----Original Message----- >> From: asterisk-users-bounces@lists.digium.com >> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >> Alejandro Vargas >> Sent: Friday, April 07, 2006 4:07 AM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] CallerID >> >> 2006/4/7, Miles Scruggs <asterisk@garnetweb.com>: >> >> >>> Could you give me an example code of how this would work, and how to >>> setup the database, I'm pretty new and while what you have written >>> makes sense, and sounds like a good plan I'm not sure I can implement it. >>> >>> >> I'm using my own agi-bin for "patching" callerid and adding the name >> if the number is found in a table (a csv that is mantained with a >> spreadsheet), it adds the name taken from this table. Then you can see >> the name in the display of the phones. >> >> -- >> Alejandro Vargas >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> >> > > > > > > >
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by another PBX seems to fill in something when asterisk does not.. If I set a number either in the sip channel for the phone, or from extensions.con, it is realized.. If I try to leave them blank, or even Not Defined, the main number of the pri gets sent out.. I am trying to debug a glitvh in or software and I need to be able to make a test call with unknown (blank callerid).. I can successfully set it to private, but that is not the same.. Any ideas? TIA -Greg
Here in the UK on pri, setting the callerid to 0, withholds it.> I am trying to set CIDNum to nothing, but my outgoing PRI controlled by > another PBX seems to fill in something when asterisk does not.. If I > set a number either in the sip channel for the phone, or from > extensions.con, it is realized.. If I try to leave them blank, or even > Not Defined, the main number of the pri gets sent out.. > > I am trying to debug a glitvh in or software and I need to be able to > make a test call with unknown (blank callerid).. I can successfully set > it to private, but that is not the same..
Greg Oliver wrote:> I am trying to set CIDNum to nothing, but my outgoing PRI controlled by > another PBX seems to fill in something when asterisk does not.. If I > set a number either in the sip channel for the phone, or from > extensions.con, it is realized.. If I try to leave them blank, or even > Not Defined, the main number of the pri gets sent out.. > > I am trying to debug a glitvh in or software and I need to be able to > make a test call with unknown (blank callerid).. I can successfully set > it to private, but that is not the same.. > > Any ideas? > > TIA > > -Greg > >On one of my T1 circuits, ten digits always appear on the other side. I can set all ten digits to zero. If I set less than ten digits then the last digits of the default ten digit string (which is our billing phone number) are overwritten with what is set. On our T3 (different provider), we can set any length of digits but I have never tried to send blank or null values. Does your other PBX send blank callerID? Is the PRI from the same provider? When you have CIDNum= do you see errors in the log that the value must not be null? Thanks, Steve
On Tue, 2006-05-23 at 09:32 +0100, Mark Ackroyd wrote:> Here in the UK on pri, setting the callerid to 0, withholds it. > > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by > > another PBX seems to fill in something when asterisk does not.. If I > > set a number either in the sip channel for the phone, or from > > extensions.con, it is realized.. If I try to leave them blank, or even > > Not Defined, the main number of the pri gets sent out.. > > > > I am trying to debug a glitvh in or software and I need to be able to > > make a test call with unknown (blank callerid).. I can successfully set > > it to private, but that is not the same..Tried that already - the PBX the PRI is connected to fills it in when it is invalid.. -Greg
You should set the presentation flags to private. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres On 5/23/06, Greg Oliver <goliver@cistera.com> wrote:> I am trying to set CIDNum to nothing, but my outgoing PRI controlled by > another PBX seems to fill in something when asterisk does not.. If I > set a number either in the sip channel for the phone, or from > extensions.con, it is realized.. If I try to leave them blank, or even > Not Defined, the main number of the pri gets sent out.. > > I am trying to debug a glitvh in or software and I need to be able to > make a test call with unknown (blank callerid).. I can successfully set > it to private, but that is not the same.. > > Any ideas? > > TIA > > -Greg > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I have a problem with BT in the UK. Using setcallerpres I can change the number shown on the recipents phones to Private or unknown but no matter what I set my asterisk cid and callerpres to it still displays the base number of my PRI ddi range. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: 23 May 2006 15:05 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID You should set the presentation flags to private. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres On 5/23/06, Greg Oliver <goliver@cistera.com> wrote:> I am trying to set CIDNum to nothing, but my outgoing PRI controlled > by another PBX seems to fill in something when asterisk does not.. If> I set a number either in the sip channel for the phone, or from > extensions.con, it is realized.. If I try to leave them blank, or > even Not Defined, the main number of the pri gets sent out.. > > I am trying to debug a glitvh in or software and I need to be able to > make a test call with unknown (blank callerid).. I can successfully > set it to private, but that is not the same.. > > Any ideas? > > TIA > > -Greg > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/
It appears that the PBX sitting between Asterisk and your provider is not passing on the calling pres flags. On 5/23/06, Lee Archer <lee.archer@pentagon-systems.com> wrote:> I have a problem with BT in the UK. Using setcallerpres I can change > the number shown on the recipents phones to Private or unknown but no > matter what I set my asterisk cid and callerpres to it still displays > the base number of my PRI ddi range. > > Regards > > Lee > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > Sent: 23 May 2006 15:05 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] CallerID > > You should set the presentation flags to private. > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres > > On 5/23/06, Greg Oliver <goliver@cistera.com> wrote: > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled > > by another PBX seems to fill in something when asterisk does not.. If > > > I set a number either in the sip channel for the phone, or from > > extensions.con, it is realized.. If I try to leave them blank, or > > even Not Defined, the main number of the pri gets sent out.. > > > > I am trying to debug a glitvh in or software and I need to be able to > > make a test call with unknown (blank callerid).. I can successfully > > set it to private, but that is not the same.. > > > > Any ideas? > > > > TIA > > > > -Greg > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ########################################### > > This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. > For more information, connect to http://www.f-secure.com/ > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I don't have a PBX sitting between Asterisk and the telco. Asterisk is the PBX. I'm using a TE110P card. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F Sent: 23 May 2006 17:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID It appears that the PBX sitting between Asterisk and your provider is not passing on the calling pres flags. On 5/23/06, Lee Archer <lee.archer@pentagon-systems.com> wrote:> I have a problem with BT in the UK. Using setcallerpres I can change > the number shown on the recipents phones to Private or unknown but no > matter what I set my asterisk cid and callerpres to it still displays > the base number of my PRI ddi range. > > Regards > > Lee > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > Sent: 23 May 2006 15:05 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] CallerID > > You should set the presentation flags to private. > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres > > On 5/23/06, Greg Oliver <goliver@cistera.com> wrote: > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled> > by another PBX seems to fill in something when asterisk does not.. > > If > > > I set a number either in the sip channel for the phone, or from > > extensions.con, it is realized.. If I try to leave them blank, or > > even Not Defined, the main number of the pri gets sent out.. > > > > I am trying to debug a glitvh in or software and I need to be able > > to make a test call with unknown (blank callerid).. I can > > successfully set it to private, but that is not the same.. > > > > Any ideas? > > > > TIA > > > > -Greg > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ########################################### > > This message has been scanned by F-Secure Anti-Virus for MicrosoftExchange.> For more information, connect to http://www.f-secure.com/ > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/
Yeah - I have tried everything - even turning it off on the other PBX for the entire system - then XO kindly just put in the last 4 and passes it on - which would normally be OK, but the other PBX I am calling accepts that as valid and therefore I still get data.. I am going to have to get XO to turn it off momentarily or ask a bill collector to call the number for me :) Thanks, Greg On Tue, 2006-05-23 at 12:55 -0400, C F wrote:> It appears that the PBX sitting between Asterisk and your provider is > not passing on the calling pres flags. > > On 5/23/06, Lee Archer <lee.archer@pentagon-systems.com> wrote: > > I have a problem with BT in the UK. Using setcallerpres I can change > > the number shown on the recipents phones to Private or unknown but no > > matter what I set my asterisk cid and callerpres to it still displays > > the base number of my PRI ddi range. > > > > Regards > > > > Lee > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of C F > > Sent: 23 May 2006 15:05 > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: Re: [Asterisk-Users] CallerID > > > > You should set the presentation flags to private. > > http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+CallingPres > > > > On 5/23/06, Greg Oliver <goliver@cistera.com> wrote: > > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled > > > by another PBX seems to fill in something when asterisk does not.. If > > > > > I set a number either in the sip channel for the phone, or from > > > extensions.con, it is realized.. If I try to leave them blank, or > > > even Not Defined, the main number of the pri gets sent out.. > > > > > > I am trying to debug a glitvh in or software and I need to be able to > > > make a test call with unknown (blank callerid).. I can successfully > > > set it to private, but that is not the same.. > > > > > > Any ideas? > > > > > > TIA > > > > > > -Greg > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ########################################### > > > > This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. > > For more information, connect to http://www.f-secure.com/ > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hope someone call help me . I have 2 POTs line coming into Asterisk. We have callerid feature from Verizon on one of the lines. I am not able to track any CallerID coming in, in the log. I am pretty green with asterisk, and it's not clear if I have to activate for CallerID in the dialplan. The voicemail keeps saying " call from an unknown caller " etc. Eventually, i would like to pass on the callerID and name to a pager, if the call is not picked up, at the extension, after hanging up the call. Thanks, braman Ryder Brook Pediatrics P.O.Box 608 Morrisville, VT 05661 --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2?/min or less. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060708/c40932c0/attachment.htm