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
>
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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 <rwhee...@artifact-software.com> 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 <rwhee...@artifact-software.com>: > >> 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 <bilmar...@yahoo.com> 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 aster...@voipbusiness.us 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 mi...@enterux.in: Hey Helvio, Would like to check it out as well. Do email me, Mitul On 22-Jun-2015 9:05 AM, Helvio Junior helvio.lis...@gmail.com 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 ing.diasda...@gmail.com 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 jpar...@digium.com 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 mjor...@digium.com wrote: - Original Message - From: Doug Lytle supp...@drdos.info 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 boykin...@gmail.com 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 m...@bellcom.dk 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 m...@bellcom.dk 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 basit.e...@gmail.com 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 vipki...@gmail.com wrote: try using voicemail_odbc On Thu, May 26, 2011 at 2:19 PM, Abdul Basit basit.e...@gmail.comwrote: 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 vipki...@gmail.com wrote: try using voicemail_odbc On Thu, May 26, 2011 at 2:19 PM, Abdul Basit basit.e...@gmail.com 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
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 tareksa...@hotmail.com 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
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 tareksa...@hotmail.comwrote: 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/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] 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] 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 da...@debsinc.com 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] FW: New in asterisk
With best regards Abdul Ahad Anwer Khan, M.Sc(CME, in progress) University of Applied Sciences Offenburg Germany Phone:+497814748226 Mobile:+4917623468462 From: abdulahadan...@hotmail.com To: asterisk-users-boun...@lists.digium.com Subject: New in asterisk Date: Sun, 27 Sep 2009 14:50:59 +0600 Hello All I am a student and doing my thesis which is related to asterisk. I am new in this field and hence facing a little bit problem. I have to work with AMI to do the call generation. I have two sip soft clients '6010' and '6011'. The asterisk I am working with is trixbox 2.6.2.3. To originate the call between the two softphones I have tried to use the following set of commands C:\telnet 192.168.0.72 5038 Asterisk Call Manager/1.1 Action: login Username: manager Secret: password Response: Success Message: Authentication accepted Action: Originate Channle: SIP/6010 Exten: 6011 Priority: 1 Timeout: 6 Context: default Response: Error Message: Premission denied Please let me know the remedy of this problem if it is possible?? or how could I acheive a calling mechanism between two softphones using AMI Waiting for the replies With best regards Abdul Ahad Anwer Khan See all the ways you can stay connected to friends and family _ Windows Live™: Keep your life in sync. Check it out! http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009___ -- 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] Preferred language for Asterisk AGIs development ?
On Tue, May 5, 2009 at 1:34 PM, Kenneth Shaw k...@expitrans.com 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] Preferred language for Asterisk AGIs development ?
On Wed, May 6, 2009 at 1:51 AM, Steve Edwards asterisk@sedwards.comwrote: 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] [asterisk-pakistan] How to connect Asterisk-stat with Asterisk CDRs database
=s3=b=50 Reach customers searching for you. Special K Group on Yahoo! Groupshttp://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
[asterisk-users] Two Leg CDR
Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten = _x.,1,Dial(SIP/[EMAIL PROTECTED]TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming in CDR and some call coming with wrong duration, when we check that wrong duration log in third party carrier CDR page we can see the duration is less than asterisk and it is different in minutes mote than 10 and 20 and some time 1 hr different. Please suggest us if possible to log CDR of ATTIP ? Thank You - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Two Leg CDR
Hi, I tired to use the following configuration but still in cdr table i can see only one record. sip.conf [444] type=friend username=444 secret=444 host=dynamic nat=yes context=vpstoteles disallow=all allow=all Extentions.conf [vpstoteles] exten = _X.,1,Set(CALLERID(accountcode)=212.XXX.XXX.240) exten = _X.,n,Dial(SIP/[EMAIL PROTECTED]) exten = s,2,Hangup In CDR table accountcode for this call is empty and also we can see only one record. Plase advice us how i can get two record in CDR table one for Registrar (444) and second for carrier (212.XXX.XXX.240)? Thank You - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- 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] Two Leg CDR
Hi, I tired to use the following configuration but still in cdr table i can see only one record. sip.conf [444] type=friend username=444 secret=444 host=dynamic nat=yes context=vpstoteles disallow=all allow=all Extentions.conf [vpstoteles] exten = _X.,1,Set(CALLERID(accountcode)=212.XXX.XXX.240) exten = _X.,n,Dial(SIP/[EMAIL PROTECTED]) exten = s,2,Hangup In CDR table accountcode for this call is empty and also we can see only one record. Plase advice us how i can get two record in CDR table one for Registrar (444) and second for carrier (212.XXX.XXX.240)? Thank You - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ -- 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] Toll-Free setup on Asterisk Server
Hi friends, Is their any possibility to setup our own Toll-Free Number in Asterisk using some PCI or FXO Card? I have one number from my local Telecom called 123 and i would like to setup this number in my asterisk if some one called this number from his mobile or land line he should not be charged when the call will come i can route to my SIP or IAX in asterisk internally. In this cause i don't need to search any Toll-Free provider and we can setup our own. Your suggestion and link of information will be high appreciated. Thank You - Never miss a thing. Make Yahoo your homepage.___ -- 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] Backup Route
Good Day All,
Is it possible to put backup route in asterisk dial plan? fro the example if
the first carrier disconnect the call with Congestion or Circuit busy then
asterisk can dial another carrier?
I did the following but it is not working as i need to dial the second one only
on congestions or circuit busy.
[wellsip]
exten = _x.,1,AGI(routing.pl)
exten = _x.,2,Set(TIMEOUT(absolute)=${TMO})
exten = _x.,3,Dial(SIP/[EMAIL PROTECTED])
exten = _x.,3,Dial(SIP/[EMAIL PROTECTED])
exten = h,1,DeadAGI(stop.pl)
Regard,
-
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Re: [asterisk-users] Asterisk 1.4.17 crashing more
Where we can see the log when this crashed coming. after that we can investigate for that particularly error. Before 1.4.17 we was using 1.2.X but we faced problem call hanged on console for one day and two day without any media and RTP. Once the call removed it comes in our billing with high duration which damage whole balance of customers for this reason we came to use 1.4.17 and till now we did not get such issue. Thank You Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time asterisk not responding anything for clients for registration either for invitation. Please advice us how we can fix this issue. Upgrade to Asterisk 1.2.X unless you need the features in 1.4. Thanks, Steve Totaro - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- 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] Asterisk 1.4.17 crashing more
Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time asterisk not responding anything for clients for registration either for invitation. Please advice us how we can fix this issue. - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- 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] Perl-AGI process
Hi All,
i have created one prepaid PERL AGI script to integrate asterisk users in our
current Oracle Billing System. I am using $AGI-exec('Dial', $dialstr); in
script after getting the MAX time out for the priticular call.
But when the channels increase on my asterisk more than 50-60 asterisk get
crashed and i am suspecting the cause is of AGI Script. because when i check ps
on server i found lot of process for routing.pl file which is the main to check
balance and max credit time.
[EMAIL PROTECTED] root]# ps -aux|grep asterisk
root 16419 0.0 0.0 5288 1072 pts/1S10:46 0:00 /bin/sh
/usr/sbin/safe_asterisk
root 16421 32.5 0.4 39404 17604 pts/1 S10:46 22:39
/usr/sbin/asterisk -f -vvvg -c
root 24537 0.0 0.2 96392 10472 pts/1 S11:29 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
root 26821 0.0 0.2 96392 10472 pts/1 S11:41 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
root 27177 0.0 0.2 96384 10476 pts/1 S11:43 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
it is more than this just i put only few for example.
I think the routing.pl file continue running till the call hangup.
My question is, Is there anyway to kill the routing.pl once $AGI-exec('Dial',
$dialstr); will run and the call should be continue?
Regard,
-
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Re: [asterisk-users] Perl-AGI process
HI Trevor,
Thank you for your suggestion.
I configured to use what u told as following but not working still.
routes.pl
$dgw = 'SIP/5556'; #A-Z carrier
$opt = 'L(6:1)';
$AGI-set_variable(routecall-destination, $dgw);
$AGI-set_variable(routecall-args, $opt);
Extnenitons.conf
[testwell]
exten = _x.,1,Set(TIMEOUT(absolute)=3660)
exten = _x.,2,AGI(routes.pl)
exten = _x.,3,Dial(${routecall-destination},${routecall-args})
exten = h,1,DeadAGI(stop.pl)
Warnning :
[Jan 12 14:34:22] WARNING[27323]: app_dial.c:863 dial_exec_full: Dial requires
an argument (technology/number)
== Spawn extension (testwell, 9745424620, 9) exited non-zero on
'SIP/8098179675-b726f5e8'
Could you please find out where is the problem?
Abdul [EMAIL PROTECTED] wrote: Hi All,
i have created one prepaid PERL AGI script to integrate asterisk users in our
current Oracle Billing System. I am using $AGI-exec('Dial', $dialstr); in
script after getting the MAX time out for the priticular call.
But when the channels increase on my asterisk more than 50-60 asterisk get
crashed and i am suspecting the cause is of AGI Script. because when i check ps
on server i found lot of process for routing.pl file which is the main to check
balance and max credit time.
[EMAIL PROTECTED] root]# ps -aux|grep asterisk
root 16419 0.0 0.0 5288 1072 pts/1S10:46 0:00 /bin/sh
/usr/sbin/safe_asterisk
root 16421 32.5 0.4 39404 17604 pts/1 S10:46 22:39
/usr/sbin/asterisk -f -vvvg -c
root 24537 0.0 0.2 96392 10472 pts/1S11:29 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
root 26821 0.0 0.2 96392 10472 pts/1 S11:41 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
root 27177 0.0 0.2 96384 10476 pts/1 S11:43 0:00
/usr/local/bin/perl /var/lib/asterisk/agi-bin/routing.pl
it is more than this just i put only few for example.
I think the routing.pl file continue running till the call hangup.
My question is, Is there anyway to kill the routing.pl once $AGI-exec('Dial',
$dialstr); will run and the call should be continue?
Regard,
-
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Regard,
-
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[asterisk-users] Bugs??
Good Day All, I am facing a serious problem since I started to use asterisk. I dont know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI show channels showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from CLI with very high duration which damaged our billing system and customers accounts goes in high negative. First I tried to get call info from asterisk mysql CDR using billsec field but the same result then I create PERL AGI to get duration from ANSWEREDTIME and same result. Please have a look to the following URL which I put the result of show channel channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 20h54m16s which really strange and out of my mind to control such as call. http://www.emafone.net/bugs.html Please advice us if it is Bug and solved in some ver or its need some configuration to avoid this issue. This is in both ver of asterisk 1.2 and 1.4 Regard, - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ --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] Bugs??
We are not using any GSM Gateway for call carriers we have Asterisk TELES(iSWITCH) --- MCI As Teles is world class telecoms product it should not make poor protocol stack. In my AGI script already i am using TIMEOUT(absolute)to limit the call according to registrar balance. I am thinking my be exten = foo,n,Congestion(3) function can solve the issue but how i can call this i should call it after dial or before? is (3) is max Congestion time? Thank You Abdul [EMAIL PROTECTED] wrote: Abdul [EMAIL PROTECTED] wrote: Good Day All, I am facing a serious problem since I started to use asterisk. I dont know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI show channels showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from CLI with very high duration which damaged our billing system and customers accounts goes in high negative. First I tried to get call info from asterisk mysql CDR using billsec field but the same result then I create PERL AGI to get duration from ANSWEREDTIME and same result. Please have a look to the following URL which I put the result of show channel channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time: 20h54m16s which really strange and out of my mind to control such as call. http://www.emafone.net/bugs.html Please advice us if it is Bug and solved in some ver or its need some configuration to avoid this issue. This is in both ver of asterisk 1.2 and 1.4 Regard, - Looking for last minute shopping deals? Find them fast with Yahoo! Search. - Looking for last minute shopping deals? Find them fast with Yahoo! Search. Regard, - Looking for last minute shopping deals? Find them fast with Yahoo! Search.___ -- 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] Bugs??
Sorry i forget to give my extentions config.
[clientsG]
exten = _x.,1,Set(UserN=${CALLERID(all)})
exten = _x.,2,Set(CalledNum=${EXTEN})
exten = _x.,3,Set(Stime=${DATETIME})
exten = _x.,4,Set(CID=${CALLERID})
exten = _x.,5,Set(HCA=${HANGUPCAUSE})
exten = _x.,6,Set(Cun=${UNIQUEID})
exten = _x.,7,AGI(routing.pl)
exten = h,1,DeadAGI(stop.pl)
exten = h,1,Hangup
routing.pl
$AGI-exec('Set',TIMEOUT(absolute)=$cstatus);
my $dialstr = $gwtype/$gwip/$dialednum;
$AGI-exec('Dial', $dialstr);#//Dial the number
Abdul [EMAIL PROTECTED] wrote: We are not using any GSM Gateway for call
carriers we have
Asterisk TELES(iSWITCH) --- MCI
As Teles is world class telecoms product it should not make poor protocol stack.
In my AGI script already i am using
TIMEOUT(absolute)to limit the call according to registrar balance.
I am thinking my be exten = foo,n,Congestion(3) function can solve
the issue but how i can call this i should call it after dial or before?
is (3) is max Congestion time?
Thank You
Abdul [EMAIL PROTECTED] wrote:
Abdul [EMAIL PROTECTED] wrote: Good Day All,
I am facing a serious problem since I started to use asterisk. I dont know if
it is a bug or some one already solved this.
Currently I am running 120 VoIP SIP channels on my asterisk server but each day
2, 3 calls got hanged in asterisk, and on asterisk CLI show channels
showing us as call UP but in real there is no call.
When asterisk restarted the hanged calls removed from CLI with very high
duration which damaged our billing system and customers accounts goes in high
negative.
First I tried to get call info from asterisk mysql CDR using billsec field but
the same result then I create PERL AGI to get duration from ANSWEREDTIME and
same result.
Please have a look to the following URL which I put the result of show channel
channelname you can see the DIALSTATUS=CONGESTION but Elapsed Time:
20h54m16s which really strange and out of my mind to control such as call.
http://www.emafone.net/bugs.html
Please advice us if it is Bug and solved in some ver or its need some
configuration to avoid this issue.
This is in both ver of asterisk 1.2 and 1.4
Regard,
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[asterisk-users] Active Calls
Hi Friends, Happy New Year I was developing billing system for my end user customers. I need to get Asterisk Active calls in MySQL database with full status of call likem ringing, UP and runtime? i will be thank full for your help and suggestion. Thank You - Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.___ --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] Asterisk CDR Variable
Hi all,
I was coding for Callback application in Perl. I have small question to get the
variable name of duration.
I seen in CDR table of mysql there is two filed one is duration and second is
billsec the billsec value variable is $AGI-get_variable('ANSWEREDTIME') But
could you guys tell me the variable name of duration field.
In this way i want to capture when the first leg was answered the second leg
duration i can get it easy using ANSWEREDTIME.
Thank You
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[asterisk-users] Asterisk Sound File
Hi all, I was playing with asterisk .gsm sound file to work for callback. But the quality is very poor and sound is very low so we cannot clearly hear what is sound played. Is there any option in asterisk to increase the volume of the IVR files or any suggestion to get maximum quality of sound? Thank You Abdul - Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now.___ --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] SIP multi Bindport
Hi, Is it possible to have multi listening bindport in asterisk? Now days mostly ISPs are Blocking the standard 5060 port so we want to keep option if 5060 is blocked we can ask our customers to use another port. Thank You Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] Nokia E65 SIP/2.0 407 Proxy Authentication Required Problem
Hi friends, We have are getting SIP/2.0 407 Proxy Authentication Required on Invite pakcet once Nokia E65 trying to dial number. But it can recive well from other caller. We tried to disable secrete and it worked fine. But we have lot of users and disabling secrete is risky. Interesting thing is Nokia N95, N80 is working well with the secrete the problem is only with Nokia E65. I will be appreciate if some one can help us to solve this issue. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] SIP Peer with Mulit Host
This is little risky, if some one got his account username/pws he will be able to send the traffic allowing only IPs means he need to assign his IP then he can send traffic. Is there no possibilities in asterisk to adding more host? Thank You If you are only going to receive and not send calls to that host then use host=dynamic __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] SIP Peer with Mulit Host
Hi friends, I have one customers as originator but he have more than 3 IPs, I need to assign his all IPs under one account type peer.? i tried the following but it did not work [vps] type=peer context=from-vps host=209.85.24.98, 67.19.85.130, sip.qualityfone.com allow=all Could you please guide me how i can assing these ip address in host field? Thank You __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] Installing Asterisk on to CentOS 4
Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [EMAIL PROTECTED] ~]$ asterisk -vvc -bash: asterisk: command not found [EMAIL PROTECTED] ~]$ I checked modules and other configuration files which are installed correctly. Please help me to locate this problem. Thank You - Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk MESSAGE Method for SMS
Hi Experts, I was trying to send SMS using Perl AGI with MESSAGE Method. I can see Asterisk is performing well, But i have a small questions How i can capture the sent message in AGI Perl? For the example sender send Hello World now i need to put this Hello World in Perl Variable using AGI so from there i can send SMS to the regular mobile phones using SMS Provider using their provided Perl API. I will be appreciate for your kind of help. Thank You Abdul Lateef Regard, Abdul Lateef Computer Programmer Mob: +974 - 5405022 MSN: [EMAIL PROTECTED] YM!: abdul_zu - Pinpoint customers who are looking for what you sell. ___ --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] Linksys (PAP2) Registration problem
Hi all, I have very small issue for PAP2 registration issue. I hope some one already faced this problem and solved. I have more than 40 PAP2, first time it registered well and making call but after some times like 5-10 mintes its not able to register on Asterisk till customer manually restart PAP2. I have already changed registrationtimeout = 60 in sip.conf but it dones not give us any benifit. Could you please guys help to solve this issue? Regards - Food fight? Enjoy some healthy debate in the Yahoo! Answers Food Drink QA.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OH323 Fake ring
Hi All,I need your urgent help. i installed OH323 channel and it is working well with samll problem fake ring. I have my VoIP provider MCI and ATT when i am routing the call via OH323 from the SIP ATA like Linksys i am getting too much fake ring even some time real RBT is there and also i can hear fake ring.Could u please guide from which configuration i can disable the fake ring. one thing more funny when i am dialing using SIP channel to these provider i am not getting any fake ring, just we are getting real RBT.this is my dial option:AGI Script Executing Application: (Dial) Options: (OH323/MCI/phone-number|350|S(max talk time))Regards, Abdul Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-addons-1.2.4 Installation Problem
Hi all,I was trying to install asterisk-addons-1.2.4 on Redhat EP, where MySQL is already installed and running for my Billing System.But i am little confiuse why i am not able to install MySQL Real-Time. here is the Error when i am trying to "make all" for asterisk-addons-1.2.4.[EMAIL PROTECTED] asterisk-addons-1.2.4]# make all./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directorycdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directoryres_config_mysql.c:55:20: errmsg.h: No such file or directoryPlease give me some idea how i can install it.Regards Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime; User not registering {Urgent}
Hi all,I have EMG 202 dailup VoIP Gateway. Which is not registering to our Asterisk server and generating following debug.Unable to find key '113685' in family 'SIP/Registry'I stored 113685 user and password in MySQL, but first time when i connect to the dialup its registered and once i made first call after the call its unregister from asterisk with the above error.But the same EMG has Broadband inteface once i use this the problem is never happened and always it registered to asterisk. So the problem is only for the dialup interface.We order 1000 EMG but not a single working. really we are so upset please try to help us to solve this issue.Regards,Abdul Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Silence Call {very very urgent plz}
Hi all,I was running my asterisk from one year. today i upgrade with 1.2.12.1 once the caller is dialing the destination number caller can hear well real RBT from telecom and once called party pickup the phone the call became silence no voice both side.Please try to help me ASAP 150 calls waiting couse of this issue.Regards, Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk mysql cdr
Hi all,I am using MySQL for mysql_cdr. I have very strange issue, while destination is ringing and caller disconnect the phone without any conversation, i can see in cdr of mysql the duration is starting and for this customer are charged without any calls.Any can suggest me how i can stop this issue i checked with my sipprovider (MCI) and they monitor the call but the told our GW is not sending call connect message while remote is ringing so it means there is some mis configuration my Asterisk.I will appriciate your kinds of help and idea.Regards,Khan Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Max Time
Hello,Could you tell how i can use it in PERL AGI script?currently i am using in my AGI with this format, but some time call is not disconnecting customers talking without money.$dialstr = "SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial', $dialstr);regards,
Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business.
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RE: [asterisk-users] Call Max Time
Hi,i am using the same calculating ((BALANCE / RATE) / 1000) method to return tTL.and i am sure my GAI is working well. but could u tell me how i can set Verbose() sepecial for my dialstring?Regards, Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Max Time
Hi All,Could anyone give me idea, How i can set Call Max Time, so in pariticular time the call should disconnect automatically.I will be appriciate for your helps.Abdul Get your email and more, right on the new Yahoo.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
RE: [asterisk-users] quintum Calling Card
Hello Jonathan,I tried in quintum to route my server with any dialed number. but i am not agble to get in quintum FXO line configuration, so i can route the call to my asterisk.do u have any about quintum how i can route calls to server once FXO line will be called?Abdul Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quintum Calling Card
Hi all,Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk.Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card number.i have add [EMAIL PROTECTED] server, so if some one trying to call FXO line of quintum then quintum should dial automatically this URI and rest my AGI will do. even i don't wnat to use quintum IVR.I will be appriciate for your helps.Regards Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] monmp3thread: Request to schedule in the past?!?!
Hi all,I am getting lot of NOTICE about this message. always my screen is full from this notice.Could anyone tell me how i can disable this message or is it some dangrus problem.Aug 22 17:21:19 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:19 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:21 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:22 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:22 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:22 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:22 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?!Aug 22 17:21:23 NOTICE[23412]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2¢/min or less.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Auto fallthrough
Hi all,Could anyone help me, why my calls of some clients disconnecting with the following error message: i have more than 500 IAX users but it is happening with very few customers.I will be appricate for u kind of help.Error::Auto fallthrough, channel 'IAX2/2001@2001/1' status is 'UNKNOWN' Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR Variable
Hi all,Could any one tell me how i can change CDR variable value from extentions.conf file.for the example i would like to change the src field value different that caller phone on the first attempt of call?Regards, Abdul How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRTP enabling
Hi everyone, I was trying to support SRTP in asterisk for our Linksys IP Phones to prevent of ISP blocking issue. I compiled successfully SRTP from http://srtp.sourceforge.net/srtp.html But i don't know from where i should start to configure in Asterisk. Could someone please give me the example sip.conf for the way how i can support? You replies will be high appriciated. Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [asterisk-users] SRTP enabling
Hello,In some countries i found that they are blocking SIP port 5060so instead of this i change to another port 1221, and its workwell. But in one country the are not blocking SIP but they areplaying with RTP packets, if they filtered it is VoIP RTP theyare doing something called party cannot hear or some time callercannot hear but called party can hear well.So i cosider to use SRTP to make encryption. and i am usingmy asterisk in VPS so i have full control to manage the server.If you guys have better Idea to prevent such kind of issue, itwill be good for us.AbdulMost of the blocking in other countries, was not for RTP traffic, but for signaling traffic (SIP usually, Mexico x Vonage comes to mind). You are sure they are blocking RTP traffic ? And, from what I understand, in some places the gov. forced the ISPs to remove the blocking (at least, I heard of one such a case in Brazil, a DSL provider started to block SIP, and Anatel, Brazil gov. entity that regulate telephony and others, asked them to remove the blocking, others with more knowledge of the case may be able to add their remarks) Blocking SIP if you control the server is somewhat easy to prevent (if is a plain dumb UDP port 5060 filtering), just have your server listen in another UDP port... See the all-new, redesigned Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and VAD
Hi all, does Asterisk 1.2.7.1 supporting VAD? because i am running my asterisk on VPS and i want to save badwidth. Khan, __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[asterisk-users] Suggesstion Required
Hi all, I want to setup asterisk box to do the following jobs. 1- 100 cuncurent calls 2- 1000 User Registration 3- MySQL Realtim 4- PerlAGI Here is my question could u please reply it: 1- No RTP only singnaling, Is it possible? Ans: 2- How much RAM? Ans: 3- How much bandhwidth per month with G729 Ans: 4- Proccessor? Ans: I will be appriciate for your kind of replies. Abdul, __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] SIP Multi Call Generation
Hi all, Is there any such as tools for multi call generation to test, how much call can be done via Asterisk? _ Best Regards, --- Abdul Lateef Nepal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Call Not Disconnecting
Hi all,
We are running more than 40 active calls on our
Asterisk Box. But some time we are facing problem,
call is not disconnecting for a long time more than 2
and 2 hrs. in this cuase our customers charged for 1,2
hrs. even they made very small calls.
i have already set rtptimeout = 60, but not
disconnecting
Here is my extentions.
[main-ext]
exten = _x.,1,AGI(main-ext.pl)
exten =
h,1,DeadAGI(/var/lib/asterisk/agi-bin/main-stop.pl)
AGI Script:
my $dialstr = $gwtype/$gwip/ . $dialednum .
|350|tTL( . ($credit_time*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
Could please advice me how i can prevent such kind of
issue?
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[Asterisk-Users] Asterisk MySQL
Hi all,
I am using MySQL query inside my extentions.conf. i
have more than 200 agents using the same extentions
and i can see in each request asterisk try to connect
mysql.
My question is, Is there any way to make only one
connection for all users who is using the same
extentions.
Here is my example working extentions:
[mysqlt]
exten = _X.,1,MYSQL(Connect connid 192.168.1.65
username password database)
exten = _X.,2,MYSQL(Query r ${connid} INSERT\ INTO\
Userstabl\ set\ user=921)
exten = s,n,MYSQL(Disconnect ${connid})
Please advice me how i can make one connection for all
users?
Thank You
Abdul
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[Asterisk-Users] DID Provider via Asterisk
Hi all, I have my asterisk server in USA. and i want to be a DID provider, not the reseller from any other provider. i need to connect my server via T1/E1 line, after that i can sell the DID to my customers, and they can route the DID where they want. I do not have much information about DID, so i am not sure T1/E1 connection can help us to be DID provider. Please give me some information and some USA telecom web site, who can provide us these connection? Thank You Abdul Lateef __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Quintum ASM400 FXO configuration
Hi All, This is my first day i brought ASM400 for Calling Card porpuse, I created AGI script for calling crad, so if some one is dialing 12345 our Calling Card AGI script will start to asking PIN,Phone number etc The Script is working well with SIPURA 3000. But i wanted to configure in quintum because this model is already having 4FXO line. So if any once can give me some usefull link or the idea for FXO configuration i will be appricate. I am looking the following diagram: PSTN FXO Line (Quintum) FXO Line [EMAIL PROTECTED] Thats all. Please help me for this issue. Thank very much in advance. Thank You Abdul __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Accept Unregistered GK Calls
Hi everyone, Could any tell me How i can accept unregistered Gatekeepers calls to my Asterisk Box? My customer is using another Gatekeeper and he want to use my Asterisk as a gateway for him to terminate the call using SIP protocol. and his Gatekeeper is not supported as end point to register my Asterisk Box. Here is waht i did the configuration but getting error: Error : SIP/2.0 404 Not Found sif.conf [from-SIPGK] type=friend host=cutomer_SIP_GK_IP_Address port=5060 nat=yes qualify=yes context=ivr-bal disallow=all allow=g729 extentions.con [ivr-bal] ;exten = _x.,1,Answer exten = _x.,2,AGI(ivr-bal.pl) Where ivr-bal.pl file is having very semple gsm file to play some voice. I will be appricate for your replies/ Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [Asterisk-Users] OH323 Peer
Hi, i treid this OH323/ipgateway:port and working well for me. But i need to add some more featurres, like some of my H323 GW supporting only G.7231 codec and some one G.729 and others feature like rtptimeout etc So if i am direct dialing without these feautres, the GW are not able to handel my calls. Any more suggestion..? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] OH323 Peer
Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry with all feature. Please let me know how i can add H.323 GW type peer? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] OOH323 Configuration
Hi all, I am using OOH323 channel to dial our H.323 carriers. I downloaed it from the latest svn. this my extentions.conf how i am dialing to h.323 destination. exten = _x.,1,SetCallerID(700700) exten = _x.,2,Dial(OOH323/[EMAIL PROTECTED]) This is the error what i am getting in h323_log 05:32:08:878 Trying to connect to remote endpoint(:0) to setup H2250 channel (outgoing, ooh323c_o_1) 05:32:08:878 ERROR:Failed to connect to remote destination for transmit H2250 channel(outgoing, ooh323c_o_1) 05:32:08:878 ERROR:Failed to create H225 connection to :0 05:32:08:974 Cleaning Call (outgoing, ooh323c_o_1)- reason:OO_REASON_NOUSER I will be very thankfull if anyone give usefull hint. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
RE : RE : [Asterisk-Users] Codec Selection
What will be the g729 and g723 codec capacity from Intel IPP liberary without License? Because still i am developing all billing and other application for asterisk so first i want to use these codecs for test once all our system become stable i will buy the license. S0 please let me know how many cuncurent calls can be handel using Intel IPP? Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Codec Selection
Hi All, I have one Carrier which is supporting only G.723.1, how i can put in my extentions.conf to send calls to this GW using G.723.1, because for Clients i can specify the codec from sip.conf but i am little confiuse how i can give specific codec for carriers. your ideas will be appriciated. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum .
|30|tTL( . ($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
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RE : [Asterisk-Users] Codec Selection
Hi,
Is there any special configuration for transcoding on
asterisk? Or Asterisk will do it automatically?
---
Olivier Taylor
Sun, 05 Feb 2006 11:51:51 -0800
Hi,
Just forget to choose the Codec on asterisk :(
Only solution is :
Disallow=all
Allow=YourCodec
If client doesn't have that codec you will need to
transcode on asterisk.
If client has that codec,asterisk will do pass-thru
and it will work.
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Abdul Lateef
Envoyé : dimanche 5 février 2006 20:00
À : asterisk-users@lists.digium.com
Objet : Re: [Asterisk-Users] Codec Selection
Hi,
I am using Perl AGI to dial the carrier (Gateway), i
am little experiencing how to do TRUN in Perl AGI.
this is my script how i am dialing the number to
Gateways, So before dialing the number i want to
select the codecs according to our Gateway.
my $discr = $AGI-get_variable(DIALSTATUS);
if ($discr == CONGESTION || $discr == NOANSWER ||
$discr == CHANUNAVAIL)
{
my $dialstr = $gwtype/$gwip/ . $dialednum .
|30|tTL( . ($crdeit*1000) .:7000:5000);
$AGI-exec('Dial', $dialstr);
$discr = ;
}
Any idea?
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
ICQ: 276994704
MSN: [EMAIL PROTECTED]
GoogleTalk: [EMAIL PROTECTED]
YM!: abdul_zu
Doha Qatar
http://www.hatif.com
__
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http://mail.yahoo.com
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[Asterisk-Users] How to Unregister?
Hi all, When i am using database show command, i can see more than 100 users are registered but actually they are not 100 some IP Phones are continue registered even i closed and switch off the IP Phone. Actually i am doing Windows based GUI, so i want to display all real registered users. I am using mySQL relatime for authuntication. I will be appriciate if any one can tell me how i can unregister so i will make some code to do unregisteration which ip phones are not registered. I will be appriciate for your replys. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Gateway TIMEOUT
HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
RE: [Asterisk-Users] Gateway TIMEOUT
Hello All, Is there any idea please? HI All, I have three a-to-z gateway from different terminators, I want to add in extensions some timeout condition. for the example my timeout=2 seconds if first gateway will not response in 2 second automatically it should dial using second gateway, respectively#133; I will be appreciate if any can provide me the configuration how I should add it. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] SIP IP Phone is not registering [urgent]
Hi guys, I have one serius problem, some time our customers IP Phones are not able to register, when i start to geting the following logs. WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! I am usuing realtime for sip registration the ttl of phone is 10 or 20. Please advise me to solve this issue, i will be appricate for your replies. Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Server Specification
Hi All, I was making plan to set an VoIP Gateway in India. And found some copanies who offered me to host my Asterisk server. I will be appriciated if anyone can suggest me how much simultaneous calls can be handeled with the following server specification? CPU : Dual Intel® Xeon® Processor at 2.8GHz Memory : 512 MB Hard Drive : 2 x 40GB 7.2K RPM Serial ATA Hard Drive Bandwidth : 100GB/MONTH HD Configuration : 2 Hard drives, Motherboard SATA RAID1 : Yes Port : 10/100MBPS SWITCHED VLAN Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Some WARNINGS
Hi all, I am getting some warnnings in Asterisk's logs. I am not familiar with this error, could anyone please tell me what is this error, is it danger..? Jan 4 17:58:35 WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:40 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:41 WARNING[30665] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:49 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 17:58:57 WARNING[5478] channel.c: Avoided initial deadlock for '0x9106ef8', 10 retries! Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '30 sip:[EMAIL PROTECTED]:1220' Jan 4 12:27:46 NOTICE[5482] chan_sip.c: stale nonce received from '30 sip:[EMAIL PROTECTED]:1220' Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.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
[Asterisk-Users] Cisco PGW-2200 OR Asterisk
Hi all, I need your golden openion about to set an VoIP softswitch. We decided to set Asterisk either Cisco PGW-2200 SS7/C7 PTSN SoftSwitch. Till now i am not fimiliar with cisco but Asterisk i did well configuration. My question is: Which will reliable to handel more than 600 cuncurent call with all kinds feature like CallBack,Calling Card,SS7 etc... I don't mean about the cost because Asterisk is open source and cisco is commercial, just i need to know which one will be better and why? 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! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid
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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Voip provider
hello, You can check this compnay. http://www.hatif.com 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! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk cdr mysql
Hi all, Did anyone installed asterisk-addons successfull? Becuase i am getting some error in installation. Error: cdr_addon_mysql.c: In function `my_load_module': cdr_addon_mysql.c:292: warning: assignment makes pointer from integer without a cast cc -shared -Xlinker -x -o cdr_addon_mysql.so cdr_addon_mysql.o -lmysqlclient -lz -L/usr/lib/mysql cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro AST_LIST_REMOVE requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: `AST_LIST_REMOVE' undeclared (first use in this function) app_addon_sql_mysql.c:164: (Each undeclared identifier is reported only once app_addon_sql_mysql.c:164: for each function it appears in.) make: *** [app_addon_sql_mysql.o] Error 1 rm app_saycountpl.o Please help me how i can load this mysql cdr module? -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGIphp Installation
Hi friends,
I was trying to execute ring.php using AGIphp but i am not able to
ring another extention i am getting this error:
- Executing AGI(SIP/123456-6e57, ring.php) in new stack
Failed to execute '/var/lib/asterisk/agi-bin/ring.php': Exec format error
-- Launched AGI Script /var/lib/asterisk/agi-bin/ring.php
-- AGI Script ring.php completed, returning 0
Here is my Configuration
[sip] ; i want to ring this ext.
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
[ppp]
exten = 111,1,agi(ring.php)
this my ring.php code.
?php
require_once('phpagi-asmanager.php');
$number = '9745405022';
$asm = new AGI_AsteriskManager();
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=SIP/$number,
'Context'='sip',
'Priority'=1,
'Callerid'=$number));
$asm-disconnect();
}
?
Please anyone can explain me why i am getting this error?
--
Best Regards,
Abdul Lateef Khan
Computer Programmer
Mobile No. : +974 - 5405022
ICQ : 276-994-704
YM! : [EMAIL PROTECTED]
MSN : [EMAIL PROTECTED]
Google Talk : [EMAIL PROTECTED]
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Re: [Asterisk-Users] RE: return Credit Time
Hi, I already install the agiphp from the following steps, i want to be sure, is my agiphp installation is correct or not. i copied all following files into /var/lib/asterisk/agi-bin folder phpagi.php phpagi-asmanager.php phpagi-fastagi.php dtmf.php ;For test i crated one extention [ppp] exten = 111,1,agi(dtmf.php) These are all modification which i did for phpagi, Is another configurations need to be done to work properly? When i am dialing this 111 extentions i am getting the error: Nov 21 06:44:30 WARNING[8266]: Timeout, but no rule 't' in context 'ppp' i will be very thank full if anyone can help me. -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return Credit Time
How i can return maximum credit time to terminate the call under his credit. In CISCO NAS i found h323-credit-time which is returning maximum credit time for calls when the call reached to this time, it will disconnect automatically. I did a lot of google but i am not able to find the commond which can return max calling credit. i will be really apriciate if any one can tell me this commond. -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Abdul Lateef Khan wants to talk to you using Google Talk
I've been using Google Talk and thought you might like to try it out. We can use it to call each other for free over the internet. Here's an invitation to download Google Talk. Give it a try! --- Abdul Lateef Khan wants to talk to you for free using Google Talk. If you already have Gmail or Google Talk, visit: http://mail.google.com/mail/b-96175a09a9-18176bcf70-0af60adddc366227 You'll need to click this link in order to add Abdul Lateef Khan to your Friends list and talk with each other for free. To try Google Talk (and get Gmail, a free Google email account with over 2,500 megabytes of storage) visit: http://mail.google.com/mail/a-96175a09a9-18176bcf70-dcb87e27db Google Talk is a downloadable Windows* application that lets you send instant messages to your friends and make free phone calls over an internet connection. Google Talk offers excellent voice quality and works with any computer speaker and microphone. Gmail is Google's free email service, offering lots of free storage, powerful spam protection, built-in search for finding your messages, and a helpful way of organizing email into conversations. And there are no pop-up ads or untargeted banners -- just text ads and related information that are relevant to the content of your messages. Once you sign up, we'll notify Abdul Lateef Khan of your new Gmail address and add you to each others' Friends lists so you can start talking right away. Gmail and Google Talk are still in beta. We're working hard to add new features and make improvements, so we might also ask for your comments and suggestions periodically. We appreciate your help in making our products even better! Thanks, The Gmail and Google Talk Teams To learn more about Gmail and Google Talk, visit: http://mail.google.com/mail/help/benefits.html http://www.google.com/talk/about.html (If clicking the URLs in this message does not work, copy and paste them into the address bar of your browser). * Not a Windows user? No problem. You can also connect to the Google Talk service from any platform using third-party clients (http://www.google.com/talk/otherclients.html). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: return Credit Time
Hi Are, Thank you for your reply, Actually i have my own billing system with freeradius which is running for our customers. and i wanted to integrate Callback system with our Billing System. So if i am going to use AstBill or any others billing system i cannot make connection to my real billing system. For this i start to work with Asterisk and PHP to work with my old database. First as i am begner in Asterisk i wanted to ask how i can include PHP file and retrive the value from PHP variable into sip.conf or extentions.conf. for the example as you give me the example to send max calling time, if i want to take this time value from php variable how i can define into Dial format, is this configuration will work? #include myphp.php Dial(SIP/70103-dc7a, SIP/70108|30|tTL($phpvar:$phpvar1:$phpvar2)|20) -- Best Regards, Abdul Lateef Khan Computer Programmer Mobile No. : +974 - 5405022 ICQ : 276-994-704 YM! : [EMAIL PROTECTED] MSN : [EMAIL PROTECTED] Google Talk : [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Forwarding
Hi all, I have one external VoIP terminator, I need to forward all calls to that terminator i did some configuration in sip.conf but i am confiused what will be the configuration in extentions.conf to forward all calls to that terminator. sip.conf [general] register = 450102:201079:[EMAIL PROTECTED]:5060/450102 i found that 450102 user successfully registered on terminator. Now i want to register Grandstreem using 450102 user on Asterisk Server and using this want to forward call using the same username to the terminator. [user] type=friend username=450102 secret=201079 fromuser=450102 authuser=450102 context=allcall allow=g729 extentions.conf [allcall] exten = Please advice me how i can run this configuration. 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! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A-Z carrier Registration
Hi all, I have 1 a-z carrier i want to forward all calls to that carrier, can any one hint me where i should add this carrier information? I will be appricate if any one give me direction way? 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! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP = H.323 Terminator
Hi all, I have H.323 Terminator and i want to terminate our all SIP clients to this terminator, Is it possible to add H.323 Terminator in Asterisk? Please give me a little hint os i can start to configure. 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! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP = H.323 Terminator
Hello Reli, If i am going to install chan_h323 with different port instead of 1719 and 1720, is it will work? Becuase already i have MVTS (Mera Softswitch) which is running on 1719 and 1720 port on the same server. 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! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Changing 5060 port
Hi friends, I want to change the standard 5060 sip port to our any defined port. i made some change in sip.conf but it is not working, I have 2 softphone which are able to register with 81 port but the any kind of hardphone is not able to register using 81 port. here is my sip.conf configuration [general] port=5060 [123456] type=friend username=123456 host=dynamic port=81 ;the hardphone should be register with 81 port context=voip allow=g729 allow=alaw allow=g723.1 Please help me how i can register with 81 port? 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! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Configuration
Hi friends, I am new in asterisk, i installed the Asterisk on my Redhat EP. But i am not able to register any SIP softphone. i am getting Unathurize message when in SIP debug. Here is my sip.conf configuration [general] context=default realm=asterisk port=5060 bindaddr=0.0.0.0 srvlookup=yes [123] type=friend username=123 secret=123 nat=yes host=dynamic ;port=81 reinvite=no canreinvite=no qualify=1000 dtmfmode=rfc2833 disallow=all allow=all context=inbound-from-local Please help me to find the problem. 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! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP/H.323 suggestion
HI all, Is Asterisk able to work as SIP and H.323 Gatekeeper same time? If it has the capability to work which i should open? Yours suggestion will be high appriciated. 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! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallBack Suggestion
Hi friends, I am new in asterisk, i came for CallBack purpose, i read from Voip-info.org aboue callback with asterisk and i am near to collect all information about to start developing callback system. Just i have a samall question, Is Callback needs some special hardware? i have my PSTN phone number i want to call this number after two ring the call will be disconnect and the Callback will start to call back to the caller ID and it should prompt to enter pin id which will authunticate via freeradius.if the authuntication is valid it will give some beep for dialing the international number. Any kind of suggestion will be hearty appriciated. 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! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic callback feature *66
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+reverse+hold ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording channels
Try using filename:wav instead of filename:WAV ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto CallBack on busy
___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto CallBack on busy
Auto Callback on Busy
Register on Busy
I have implemented it as 1- I store Caller and Called party
numbers in database when Called part is busy 2- I retrieve it from
database and Caller is called by called party when Called party hangs up
It is working fine with all kind of SIP phones I have with me
basic configuration for extensions.conf is given and can be accommodated
according to requirements default ;Store Called Number
in DB if he is Busy exten = _.,1,Dial(SIP/${EXTEN},20,Ttr)
exten = _.,2,Congestion() exten =
_.,102,DBPut(CallBack/${EXTEN}=${CALLERIDNUM}) exten =
_.,103,Busy() ;Auto CallBack Caller on hang up of dialed
party exten = H,1,Goto(h,1) exten =
h,1,DBget(temp=CallBack/${CALLERIDNUM}) exten =
h,2,DBdel(CallBack/${CALLERIDNUM}) exten =
h,3,Dial(SIP/${temp},20,tr) exten = h,4,Congestion() exten
= h,102,NoOp(Not Registered for CallBack) exten = h,104,Busy()
Enjoy Abdul Ghafoor [EMAIL PROTECTED]
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[Asterisk-Users] Auto CallBack on busy
Auto Callback on Busy
Register on Busy
I have implemented it as 1- I store Caller and Called party
numbers in database when Called part is busy 2- I retrieve it from
database and Caller is called by called party when Called party hangs up
It is working fine with all kind of SIP phones I have with me
basic configuration for extensions.conf is given and can be accommodated
according to requirements
[default];Store Called Number in DB if he is Busy
exten = _.,1,Dial(SIP/${EXTEN},20,Ttr) exten =
_.,2,Congestion() exten =
_.,102,DBPut(CallBack/${EXTEN}=${CALLERIDNUM}) exten =
_.,103,Busy() ;Auto CallBack Caller on hang up of dialed
party exten = H,1,Goto(h,1) exten =
h,1,DBget(temp=CallBack/${CALLERIDNUM}) exten =
h,2,DBdel(CallBack/${CALLERIDNUM}) exten =
h,3,Dial(SIP/${temp},20,tr) exten = h,4,Congestion() exten
= h,102,NoOp(Not Registered for CallBack) exten = h,104,Busy()
Enjoy Abdul Ghafoor [EMAIL PROTECTED]
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RE: [Asterisk-Users] From 0 to PBX in 2 hours
Hello, Does anyone know if a GUI for Asterisk exists ? Regards, Abdul Hakeem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: 10 March 2004 01:43 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] From 0 to PBX in 2 hours what is your easy start up guide? - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 09, 2004 12:16 PM Subject: [Asterisk-Users] From 0 to PBX in 2 hours Hi everyone, As an newbie in * and knowing how hard Linux can be for starters (more if you come from the world of MSwindows.), I triyed a way to teach my co-workers how to deploy fast and in less than two hours a small but perfect working PBX. I used the following equipment: 1 Clarent CPG 101 (A.K.A. D-link 104s) 4 FXS ports 1 PC Celeron 500Mhz 128MB Ram The following software: 1 Mepis Installation CD, and a cable Internet link. Booted up the computer with the Mepis (Debian based) CD, installed Linux in the Hard disk, 15 minutes or less. Updated the firmware and the boot prom to versions 3.0B35-C and 3.0B14-C in the Clarents respectively, runned apt-get install asterisk, and then begun to figure out how to configure the mgcp gateways. So far whe have deployed in two days 5 locations and starting to connect Barquisimeto, Venezuela, with Buenos Aires, Argentina. Hope that helps someone and encourage others to start in this great world of telecomunications AMG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
