Re: [asterisk-users] Is there any way to pass caller id to cell phone?
Hi Ivan,
Check whats CallerID you are getting before initiating dial command.
;Eric on extension 105
exten => 105,1,NoOp( Call ID: ${CALLERID(all)} )
exten => 105,n,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
same => n,VoiceMail(105@default,u)
Also what Caller ID is set on outgoing trunk? Is that enforced in trunk
configuration?
--
regards,
abdul basit
On Thu, 11 Oct 2018 at 22:19, Ivan Demkovitch wrote:
>
> We have following problem. On some of the extentions I call cell phone
> after 10 seconds or so.
> Or, like this one below- we call cell and office phone at the same time
>
> ;Eric on extension 105
> exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)
> same => n,VoiceMail(105@default,u)
>
> Where problem comes in - if person not at the desk - his cell phone shows
> call from OFFICE number and there is no way to tell who is really calling.
>
> We use Callcentric as a trunk if it makes any difference.
>
> I'd like to add info about caller when passing to cell phone if possible.
> Is there any way to do that?
>
> Thank you,
> Ivan
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11! Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> 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 is coming up October 9-11! Signup is available at:
https://www.asterisk.org/community/astricon-user-conference
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] General Kernel practices on CentOS
Olivier If you installed asterisk from source, you need to recompile it after kernel version upgrade. This will compile & install asterisk modules with latest installed kernel sources. -- regards, abdul basit On 19 December 2017 at 08:01, Ron Wheeler wrote: > Linux x.y.com 3.10.0-693.5.2.el7.x86_64 #1 SMP Fri Oct 20 20:32:50 UTC > 2017 x86_64 x86_64 x86_64 GNU/Linux > I try to keep up with the latest versions of everything. > > Ron > > On 15/12/2017 5:59 AM, Olivier wrote: > > Hello Ron, > Which kernel do you run Asterisk/Freepbx with ? > Cheers > > 2017-12-14 16:57 GMT+01:00 Ron Wheeler : > >> CentOS 7 works well with Asterisk. >> Install latest CentOS7 with updates install asterisk >> >> I am running FreePBX on CentOS 7. >> >> Ron >> >> On 14/12/2017 10:38 AM, Olivier wrote: >> >> Hello, >> >> I'm used to install Asterisk on Debian stable platforms. >> >> A customer is asking how I would proceed on a CentOS platform. >> >> After a short research (see [1] as an example), I'm wondering what are >> general kernel practices on CentOS regarding Asterisk and when targeting >> stability: >> >> - Is it recommended to upgrade kernel version(s) (ie moving from linux >> 3.10 to 4.3) just after OS installation ? >> >> Best regards >> >> >> >> >> -- >> Ron Wheeler >> President >> Artifact Software Inc >> email: rwhee...@artifact-software.com >> skype: ronaldmwheeler >> phone: 866-970-2435, ext 102 <%28866%29%20970-2435> >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > Ron Wheeler > President > Artifact Software Inc > email: rwhee...@artifact-software.com > skype: ronaldmwheeler > phone: 866-970-2435, ext 102 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] atcom card: how it is?
Hi used atcom phones and cards few years ago. I found them good. But i didn't use their GSM cards. I believe these will be good as well. -- regards, abdul basit On 27 October 2017 at 22:29, bilal ghayyad wrote: > Hello; > > I am thinking to use atcom card which can be shown in this link: > AXE2G4AN - GSM card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products > Manufacturer <http://www.atcom.cn/gsm90.html> > > AXE2G4AN - GSM card - Atcom_Ip phone,IP PBX,Asterisk Cards,Voip Products > Ma... > ATCOM is the leading VoIP hardware manufacturer in global market. We have > been keeping innovating with customer’... > <http://www.atcom.cn/gsm90.html> > > > But I am afraid, because I used to use digium and I am afraid of the > quality. > Maybe someone will ask me why not to use digium? The answer: because I > need the card to has one GSM sim port and 1 FXO port and did not find this > with digium or sangoma. But I am afraid from ATCOM that it might be low > quality. > Also, I need to know how the quality will be in case there is GSM and FXO > at the same card, will there be a noise or distortion? > > Appreciate the kindly help and advise. > Regards > Bilal > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Product CDR/Queue/Meetme
Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support wrote: > Please keep the “me to” emails off the list. > > Regards; > > JV > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Magno Guimarães > *Sent:* Monday, June 22, 2015 3:55 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Product CDR/Queue/Meetme > > > > Hello, > > > > > > I am interested, too. > > > > > > Att, > > Welinghton > > > > > Citando Mitul Limbani : > > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > > On 22-Jun-2015 9:05 AM, "Helvio Junior" wrote: > > Gentleman, > > Moderators, i don't know if this topic if OFF-Topic, if yes, please tell > me. > > I had some difficult looking for a Asterisk software that provide me some > functions (For exemple: CDR, Queue control, MeetMe Control) all-in-one. So > i decided to develop than. > > In a few weeks i'll deploy a Beta version of this software and i'd like to > know if is somebody available to try this beta and free version? > > If you don't want to try this version but would like to see/suggest any > feature in this software, let me know. > > Forecast functions to Beta Version: > >- Realtime view for: > > >- Queues; > - Peers (Similar as BLF); > - Trunk calls/utilization; > > >- MeetMe > > >- Create, modify, delete and schedule; > - Real time view of members; > - Delete members; > - Mute/Unmute; > - Send Invite by e-mail (with .VCS file) > > >- Dialer > > >- Create dialer (by campaign with contacts) > - Monitoring of campaig, calls, and status; > - Time control to retry failed call > - Control of day time to call (commercial time, full time, etc...) > > >- Charts and reports: > > >- Trunk utilization; > - CDR; > - Queues (Most common reports and charts, distributions, times, > etc...) > - Export to Excel Spreadsheet and PDF File > - Report Scheduler > - Much more... > > >- REST API for 100% of functionalities; >- Admin and User Console 100% Web HTML5; >- Developed in Windows with C#; >- Integrate with Asterisk using AMI only; >- Allow manage many Asterisk that you want using same instance of this >software (One software and one installation); > > > Obs.: I'll provide a Full License for everybody that help me trying the > Beta version. > > > > -- > > > > Att, > > Hélvio Junior > > SafeId - Gestão de identidades e Acessos > > +55 41 | 9893-2694, single-sign-on.com.br > > helvio.jun...@safetrend.com.br > > > > -- > > > > Att, > > Hélvio Junior > > SafeId - Gestão de identidades e Acessos > > +55 41 | 9893-2694, single-sign-on.com.br > > helvio.jun...@safetrend.com.br > > > -- > _ > -- 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] Deleting OLD Voicemails
You can delete old voicemails. Why not your install webvmail? This is web based GUI for voicemails. You can select and delete from font end without breaking anything. http://www.voip-info.org/wiki/view/Asterisk+gui+vmail.cgi -- regards, abdul basit On Wed, May 23, 2012 at 3:03 AM, Danny Dias wrote: > Thanks Jason, > > But how to delete them? there are a lot of old voicemails, but i don't > want to break the app_voicemail. > > > > 2012/5/22 Jason Parker > >> On 05/22/2012 04:54 PM, Danny Dias wrote: >> > There are 4 files for each voicemail: >> > >> > msg.gsm >> > msg.txt >> > msg.wav >> > msg.WAV >> > >> >> That is perfectly normal. The .txt file is metadata that contains things >> like >> caller ID and duration. Asterisk will also save voicemails into every >> format >> you have specified in voicemail.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 >> > > > > -- > www.danntel.net > *sip:danny4...@thesipschool.com* > sip:dann...@opensips.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 -- _ -- 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] Asterisk 1.8.x app_meetme.so
ConfBridge is not much flexible as MeetMe. On Wed, Feb 22, 2012 at 7:19 PM, Matthew Jordan wrote: > > - Original Message - > > From: "Doug Lytle" > > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > > Sent: Wednesday, February 22, 2012 7:22:20 AM > > Subject: Re: [asterisk-users] Asterisk 1.8.x app_meetme.so > > > > You mentioned that the meetme source was there, I was guessing that > > the > > option to compile wasn't checked so the binary wasn't available. > > > > I just ran into this myself yesterday when converting a 1.4x box > > (Still > > in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme > > was available. > > > > Doug > > > > Just a few points of clarification: > 1. MeetMe is still the preferred conferencing application in Asterisk 1.8. > In Asterisk 10, the preferred conferencing application is ConfBridge. > Even still, in Asterisk, 10, you can compile and install MeetMe using > menuselect. > 2. In the screenshot you attached, you cannot choose to compile MeetMe > as one of its dependencies is not available, in this case, DAHDI. > > Note that DAHDI being a dependency for MeetMe was one of the reasons > Asterisk 10 moved to using ConfBridge as the default conferencing > application. > > Matthew Jordan > Digium, Inc. | Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://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 > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] Skype For Asterisk (SFA)
Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] Asterisk scaling
On Fri, Aug 19, 2011 at 6:39 AM, Jim Boykin wrote: > convert mp3 to sln, this itself will give you quiet a big capacity boost. > How does sln boost capacity? > > On Wed, Aug 17, 2011 at 12:21 PM, Morten M. Hansen wrote: > > On 2011-08-16 21:14, Warren Selby wrote: > >> Is it going to be just one mp3 stream that is shared across all users > (I.e everyone hears the same thing at the same time), or is it 1000 separate > mp3 streams (everyone always starts at the beginning of whatever they are > hearing). > > > > It's a shared stream. When testing now, new listeners doesn't spawn new > > mpg123 processes. > > > >> Are you going to have reliable timing generation on an EC2 instance, > since IAX streams and music on hold playback will sound bad if the timing > isn't good. > > > > We are using the zaptel and ztdummy kernel module, and we haven't > > noticed any problems with the audio quality yet. Should we be worried > > about this when the load gets higher? > > > >> Will you have sufficient bandwidth allocated to you for that many > simultaneous calls? > > > > Good point. We will have to do some calculation and research on what EC2 > > offers here. > > > >> Is there going to be any codec transcoding going on? Can you generate > your streams in the preferred codec, instead of mp3? > > > > The source is an icecast server streaming mp3. I haven't figured out a > > way to get around that. But from what I understand its just one > > reencoding for all the listeners. > > > >> I think if you're just using one stream spread across all the callers, > you'll have much better performance from the system as a whole. You may want > to look at the quality differences between a SIP trunk and an IAX trunk as > well. > > > > I had a talk with our IAX2 trunk provider and they told me that we could > > expect better performance from a SIP trunk. They also had a limit on > > 2000 channels, so we may have to look for another trunk. > > > > Are there any tools or services to simulate a lot of IAX2 or SIP users > > that you can recommend? How do you test how many users an asterisk > > system can handle? > > > > Thank you for taking the time to reply. > > Morten > > > >> Thanks, > >> --Warren Selby, dCAP > >> > >> On Aug 16, 2011, at 10:16 AM, "Morten M. Hansen" > wrote: > >> > >>> Hi > >>> > >>> I'm hoping someone could comment on how our setup will perform under > >>> larger loads. > >>> Its a quite simple setup, with Asterisk 1.6.2 on Debian 6 on an EC2 > large > >>> instance (7GB RAM, 2 virtual cores with EC2 compute units). > >>> Using an IAX2 trunk we offer normal phones to dial in and listen to a > mp3 > >>> stream using music on hold. > >>> > >>> If we wanted to let 1000 users listen to the stream at the same time, > >>> would that be possible? What limits will we hit? How about 1 users? > >>> > >>> Regards > >>> Morten > >>> > >>> > >>> -- > >>> _ > >>> -- 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 > >>> > > > > -- > > _ > > -- 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 > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] DB driven voicemail
i implemented voicemail with ODBC. I can now setup mailboxes in voicemail table and can save recordings in voiemailmessages table in blog field. Still have issues with join in voicemails and cdrs. How can we identify the voicemail with corresponding cdr? Has any one tested? On Fri, May 27, 2011 at 4:32 PM, Abdul Basit wrote: > OK. Im trying to setup voicemail on ODBC. > > My objective is to create some relation in voicemail_data and cdr table > based on uniqueid. > > -- > regards, > > Abdul Basit > > > On Fri, May 27, 2011 at 12:12 AM, vip killa wrote: > >> try using voicemail_odbc >> >> On Thu, May 26, 2011 at 2:19 PM, Abdul Basit wrote: >> >>> Have anyone setup voicemail using DB? >>> I am facing problems with asterisk realtime voicemail setup. >>> >>> Asterisk authenticate and saves new voicemail records in mysql with voice >>> file path. >>> >>> /var/spool/asterisk/voicemail/default/337/INBOX/msg0001 >>> >>> When we listen voiemails, app_voicemail deletes old record from >>> voicemail_data and inserts a new one with new file name in Old folder. >>> >>> /var/spool/asterisk/voicemail/default/337/Old/msg >>> >>> Please note that message name also changed. >>> >>> How to handle voicemail in asterisk realtime? >>> >>> -- >>> Regards, >>> >>> Abdul Basit >>> >>> -- >>> _ >>> -- 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 >> > > > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] DB driven voicemail
OK. Im trying to setup voicemail on ODBC. My objective is to create some relation in voicemail_data and cdr table based on uniqueid. -- regards, Abdul Basit On Fri, May 27, 2011 at 12:12 AM, vip killa wrote: > try using voicemail_odbc > > On Thu, May 26, 2011 at 2:19 PM, Abdul Basit wrote: > >> Have anyone setup voicemail using DB? >> I am facing problems with asterisk realtime voicemail setup. >> >> Asterisk authenticate and saves new voicemail records in mysql with voice >> file path. >> >> /var/spool/asterisk/voicemail/default/337/INBOX/msg0001 >> >> When we listen voiemails, app_voicemail deletes old record from >> voicemail_data and inserts a new one with new file name in Old folder. >> >> /var/spool/asterisk/voicemail/default/337/Old/msg >> >> Please note that message name also changed. >> >> How to handle voicemail in asterisk realtime? >> >> -- >> Regards, >> >> Abdul Basit >> >> -- >> _ >> -- 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 > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- 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] DB driven voicemail
Have anyone setup voicemail using DB? I am facing problems with asterisk realtime voicemail setup. Asterisk authenticate and saves new voicemail records in mysql with voice file path. /var/spool/asterisk/voicemail/default/337/INBOX/msg0001 When we listen voiemails, app_voicemail deletes old record from voicemail_data and inserts a new one with new file name in Old folder. /var/spool/asterisk/voicemail/default/337/Old/msg Please note that message name also changed. How to handle voicemail in asterisk realtime? -- Regards, Abdul Basit -- _ -- 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] differential billing
Yes. you are right. I was thinking to avoid reinventing the wheel. Will write AGIs. Trick is how to charge at 3min 59 sec or 4 min 01 sec during live call. We can monitor channel variables over AMI. But this will be a CPU overhead (say for 100 or 200 calls) if we monitor channel variables on every second. I want some thing to push channel details on each transition (or events like IVR level changed, call duration updated to next minute) rather than i request on AMI. Don't know if this logic is workable. Just want a right direction. -- Regards, Abdul Basit | +92 32 1416 4196 On Sat, Sep 25, 2010 at 11:37 PM, Tarek Sawah wrote: > > if you are deploying your own system.. then you can build a small > application (AGI) that would do the math for you .. will devide the call > duration into the stages you want .. and does the calculation.. i think > MYSQL already can do that.. but a PHP script will do it faster and easier.. > or like our billing system.. C# application interacting with Asterisk doing > all the math. after all it's all SQL and Asterisk working. you can do that > with a dial plan i believe.. so why not build an AGI to do it for you? > > > > -- Tarek Sawah > > Integrated Digital Systems > > CCNA, MCSE, RHCE, VoIP USA: +13864929993 > > > > > > > > > > > From: basit.e...@gmail.com > > Date: Sat, 25 Sep 2010 23:27:56 +0500 > > To: asterisk-users@lists.digium.com > > Subject: Re: [asterisk-users] differential billing > > > > Tarek, > > > > I already tested this feature with a2billing. > > > > This is difficult to extract the working code from a2billing. > > Also we are developing billing system so this is not a good idea > > to deploy another billing system in parallel. > > > > Any idea or link might help full. > > > > > > > > > > On Fri, Sep 24, 2010 at 9:30 PM, Tarek Sawah > > > wrote: > > > > A quick answer? A2billing. > > > > It has what you call it differential billing.. but they call it > > progressive billing.. 3 steps .. for 3 different rates .. > > > > Go for it.. easy to setup and quick to learn and use. > > > > Regards > > > > > > > > From: > > asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] > > On Behalf Of Danny Nicholas > > Sent: Friday, September 24, 2010 4:19 PM > > > > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > > Subject: Re: [asterisk-users] differential billing > > > > > > > > > > > > From: > > asterisk-users-boun...@lists.digium.com > > [mailto:asterisk-users-boun...@lists.digium.com] > > On Behalf Of Abdul Basit > > > > Sent: Friday, September 24, 2010 8:13 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [asterisk-users] differential billing > > > > > > > > Hi All, > > > > > > > > How can we develop a differential charging setup using asterisk like > > for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next > > 30 sec charge @15cent, etc? > > > > > > > > Any idea, suggestion. > > > > -- > > Regards, > > > > Abdul Basit | +92 32 1416 4196 > > > > > > > > Since the CDR records the call duration in seconds, this should be a > > relative “no-brainer”, assuming you are billing post-call. If you are > > wanting to generate the charges during the live calls, AMI would be > > your best option for getting a running duration of the connection. > > > > -- > > _ > > -- 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 > > > > > > > > -- > > Regards, > > > > Abdul Basit | +92 32 1416 4196 > > > > -- > > _ -- > > 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/ast
Re: [asterisk-users] differential billing
Tarek, I already tested this feature with a2billing. This is difficult to extract the working code from a2billing. Also we are developing billing system so this is not a good idea to deploy another billing system in parallel. Any idea or link might help full. On Fri, Sep 24, 2010 at 9:30 PM, Tarek Sawah wrote: > A quick answer? A2billing. > > It has what you call it differential billing.. but they call it progressive > billing.. 3 steps .. for 3 different rates .. > > Go for it.. easy to setup and quick to learn and use. > > Regards > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas > *Sent:* Friday, September 24, 2010 4:19 PM > > *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > *Subject:* Re: [asterisk-users] differential billing > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Abdul Basit > > *Sent:* Friday, September 24, 2010 8:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] differential billing > > > > Hi All, > > > > How can we develop a differential charging setup using asterisk like for > 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec > charge @15cent, etc? > > > > Any idea, suggestion. > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 > > > > Since the CDR records the call duration in seconds, this should be a > relative “no-brainer”, assuming you are billing post-call. If you are > wanting to generate the charges during the live calls, AMI would be your > best option for getting a running duration of the connection. > > -- > _ > -- 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 > -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- 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] differential billing
Thank you Danny. I am thinking for AMI events. Do we need some code level change? As i want asterisk to push events to some listener rather than i ask via AMI. For hight call volume read from AMI may be an over head on asterisk, i think. On Fri, Sep 24, 2010 at 6:19 PM, Danny Nicholas wrote: > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *Abdul Basit > *Sent:* Friday, September 24, 2010 8:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] differential billing > > > > Hi All, > > > > How can we develop a differential charging setup using asterisk like for > 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec > charge @15cent, etc? > > > > Any idea, suggestion. > > -- > Regards, > > Abdul Basit | +92 32 1416 4196 > > > > Since the CDR records the call duration in seconds, this should be a > relative “no-brainer”, assuming you are billing post-call. If you are > wanting to generate the charges during the live calls, AMI would be your > best option for getting a running duration of the connection. > > -- > _ > -- 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 > -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- 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] differential billing
Hi All, How can we develop a differential charging setup using asterisk like for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next 30 sec charge @15cent, etc? Any idea, suggestion. -- Regards, Abdul Basit | +92 32 1416 4196 -- _ -- 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] Preferred language for Asterisk AGIs development ?
On Wed, May 6, 2009 at 1:51 AM, Steve Edwards wrote: > On Tue, 5 May 2009, Abdul Basit wrote: > > > I wrote the php code for asterisk that was two page long and wasim baig > sb > > wrote the same stuff in 1/2 page line of code using python with > > implementation of python libraries. > > > > yeee! > > If I wrote it in a single very long line of C would you be even happier? > > I guess it depends on the metrics you use to judge a solution :) yap! > > 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 > -- Regards, Abdul Basit | Manager Support | +92-321-416-4196 | +92-42-588-7833 | www.convergence.pk ___ -- 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] Preferred language for Asterisk AGIs development ?
On Tue, May 5, 2009 at 1:34 PM, Kenneth Shaw wrote: > Drop Asterisk, move to Freeswitch. Much easier to interact with external > code bases, and it has more than one language interpreter built in > (javascript, lua, etc.). agreed but FS is newer and under test environment. > If you're intent on staying on Asterisk, I would suggest skipping AGI, > and write a client that monitors the state of asterisk via the manager > interface. AGIs are messy in my opinion. There isn't really any "best" > language for AGIs, as AGIs just communicate with Asterisk via a pipe. So > really, the best language for AGIs are the language you like the best > and/or best fits the application domain/requirements for your project. AMI is good way for developing interactive applications. I wrote the php code for asterisk that was two page long and wasim baig sb wrote the same stuff in 1/2 page line of code using python with implementation of python libraries. yeee! > > On Tue, 2009-05-05 at 11:52 +0500, Kashif Naeem wrote: > > Hello, > > > > We are going to start development for a product based over Asterisk. > > According to you, which is the preferred language for AGIs / IVRs > > development in Asterisk. I got opinions that Perl is going to > > be replaced by PHP for all future developments. > > > > > > > > -- > > Kashif Naeem > > Business Development Manager > > Hadi Telecom > > www.haditelecom.com > > > > Cell: +92 (0)345 4226006 > > Office: +92 (0)42 5692766 > > > > Email: kas...@haditelecom.com > > MSN: kashif__na...@hotmail.com > > Gmail: meet.kas...@gmail.com > > Skype: kashif.naeem > > > > 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. > > ___ > > -- 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 > -- > Kenneth Shaw > ExpiTrans, Inc. > 129 W. Wilson St., Suite 204 > Costa Mesa, CA 92627 > tel: 949.650.4600 > fax: 949.642.6044 > k...@expitrans.com > > > _______ > -- 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 > -- Regards, Abdul Basit | Manager Support | +92-321-416-4196 | +92-42-588-7833 | www.convergence.pk ___ -- 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] [asterisk-pakistan] How to connect Asterisk-stat with Asterisk CDRs database
228478942411287/A=4507179/R=0/SIG=12de4rskk/*http://us.rd.yahoo.com/evt=50284/*http://finance.yahoo.com/personal-finance> > > Guides, news, > > advice & more. > Ads on Yahoo! > > Learn more > now.<http://us.ard.yahoo.com/SIG=13o4lhoja/M=493064.12016308.12445700.8674578/D=groups/S=1705004726:NC/Y=YAHOO/EXP=1228486142/L=/B=0.ILJkLaX9M-/J=1228478942411287/A=3848643/R=0/SIG=131q47hek/*http://searchmarketing.yahoo.com/arp/srchv2.php?o=US2005&cmp=Yahoo&ctv=Groups4&s=Y&s2=&s3=&b=50> > > Reach customers > > searching for you. > Special K Group > > on Yahoo! > Groups<http://us.ard.yahoo.com/SIG=13pjs93tn/M=493064.12016300.12445692.11323196/D=groups/S=1705004726:NC/Y=YAHOO/EXP=1228486142/L=/B=1OILJkLaX9M-/J=1228478942411287/A=5170420/R=0/SIG=11b5gu1oe/*http://new.groups.yahoo.com/specialKgroup> > > Join the challenge > > and lose weight. > . > > __,_._,___ > -- Regards, Abdul Basit | Manager Support | +92-321-416-4196 | +92-42-588-7833 | www.convergence.pk ___ -- 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
