Re: [asterisk-users] CURL to post application/json
My mistake, CURLOPT(header) is only to retrieve headers, not to sent. sorry. Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Fri, Oct 5, 2018 at 8:53 AM Nasir Iqbal wrote: > Hi David, > > Have you tried CURLOPT function. > i.e > Set(CURLOPT(header)=Content-Type: application/json) > > Regards > > Nasir Iqbal > > ICTBroadcast - an Auto Dialer software for ITSP > <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> > SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns > http://www.ictbroadcast.com/ > > > On Fri, Oct 5, 2018 at 1:59 AM David P wrote: > >> We tried to use the CURL fn to POST json, but it's sent as form data and >> there seems no support for changing the Content-Type header. We switched to >> invoking curl in the shell. >> >> All the documentation I could find says there is just one parameter for >> the url and an optional second for POST body. Is there an undocumented way >> to set Content-Type? >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Astricon is coming up October 9-11! Signup is available at: >> https://www.asterisk.org/community/astricon-user-conference >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spontaneous reboot due to MySQL lookups ?
Hi Jonas Kellens,
Like everybody else, I also recommend to upgrade your Asterisk version, and
replace mysql driver with odbc. the one big advantage of odbc driver is its
pooling feature, you can configure odbc to create a reusable pool of active
connections, so Asterisk does't needs to reconnect for each query.
For example with following odbc settings you can achieve 500+ concurrent
channels (approx 2500 queries / minute) without any performance issue.
pooling => yes
limit => 16
pre-connect => yes
Regards
Nasir Iqbal
ICTBroadcast - an Auto Dialer software for ITSP
<https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition>
SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
http://www.ictbroadcast.com/
On Fri, Oct 5, 2018 at 4:26 AM John Novack wrote:
> As others have said, clearly it ISN'T "just working" or you would not have
> posted the question
>
> To state again, I am using Version 13, though a few minor revisions
> behind, with MySql, on CentOS 6 and have no rebooting or other MySql
> related issues
>
> Clearly you need to state in more detail what issues remain, once you
> migrate to AT LEAST 13.xx, and state your OS after becoming current with
> Asterisk, MySql and the OS
>
> I use MySql on every incoming call, and also maintain call detail records
> in MySql for every call, and it just simply works, and has for some time.
>
> Although I may be using it quite differently that you, it simply works.
> Is this a newly developing issue, or has it persisted for some time
> What if any changes have been made to the dialplan etc?
>
> Have you considered a strictly hardware issue? Memory? HD? MB??
>
> The crystal ball is very cloudy on this one!
>
> John Novack
>
>
> Jonas Kellens wrote:
>
> Hello
>
> thank you for your answer.
>
> If I read your (and others) reaction correctly I can conclude that this is
> an Asterisk problem and not a problem of MySQL or dialplan logic ?
>
>
> You should know that the MySQL database is heavily questioned :
>
>
> mysql> show status like '%onn%';
> +--++
> | Variable_name| Value |
> +--++
> | Aborted_connects | 469|
> | Connections | 132762 |
> | Max_used_connections | 8 |
> | Ssl_client_connects | 0 |
> | Ssl_connect_renegotiates | 0 |
> | Ssl_finished_connects| 0 |
> | Threads_connected| 3 |
> +--++
> 7 rows in set (0.00 sec)
>
>
>
> I stick to 1.8 because it just works. I had some issues with version 11
> and 13 in the past.
>
>
> Regards
>
> Jonas.
>
> Op 04-10-18 om 17:49 schreef John Novack:
>
> Woefully out of date.
> You really need to put your efforts into at least a modest upgrade
> I use version 13 with MySql queries built into the dialplan on CentOs 6
> and have NO such issues, either performance or any restart or reboot. It
> simply works
>
> I never used either 1.6 or 1.8, going from 1.4 to version 11, which did
> require some syntax changes to the dialplan.
>
> Given that even version 11 is EOL, you really need to put your efforts
> into doing the migration rather than tracking this one down
>
> JMO
>
> John Novack
>
>
>
> Jonas Kellens wrote:
>
> Hello
>
> using Asterisk 1.8.32.
>
> I notice that there is a spontaneous reboot of the Asterisk system from
> time to time.
>
> When I look in the logs (verbose file) I noticed that every time this
> occurs it's at a moment that there is a MySQL action, be it a lookup or an
> insert/update/delete.
>
> I must say I do have some MySQL queries that occur in my dialplan when a
> call comes in, to look up different actions to perform on this call.
>
>
> An idea how to overcome this problem ? Seems a "performance" issue, no ?!
>
> Is it better to have these MySQL queries to be done by an external script
> (like a php script that I call with the System()-command or a
> SHELL()-command) ?
>
>
> Here are some examples from the verbose file.
>
>
>
> [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing
> [s@sub-GetAlertInfo:3] MYSQL("SIP/SipAgenT01-317d", "Connect connid
> localhost myuser mypwd myDB") in new stack
> [Aug 22 15:19:10] VERBOSE[2977] pbx.c: [Aug 22 15:19:10] -- Executing
> [s@sub-GetAlertInfo:5] MYSQL("SIP/SipAgenT01-317d", "Query resultid 1
> SELECT uri, callinfo FROM distringtone WHERE onoff='1'") in new stack
> [Aug 22 15:19:18] VERBOSE[3306] config.c: [Aug 22 15:19:18] == Parsing
> '/
Re: [asterisk-users] CURL to post application/json
Hi David, Have you tried CURLOPT function. i.e Set(CURLOPT(header)=Content-Type: application/json) Regards Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Fri, Oct 5, 2018 at 1:59 AM David P wrote: > We tried to use the CURL fn to POST json, but it's sent as form data and > there seems no support for changing the Content-Type header. We switched to > invoking curl in the shell. > > All the documentation I could find says there is just one parameter for > the url and an optional second for POST body. Is there an undocumented way > to set Content-Type? > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC as Softphone substitute ?
@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is security feature and not a bug. Regards On Tue, Oct 2, 2018, 13:03 Olivier wrote: > @Nasir: > Thanks for replying here. > > Did you met in your deployments, the kind of stability issues Carlos > reported earlier ? > > Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal a > écrit : > >> Hi Olivior, >> >> We have recently worked on a WebRTC based agent panel. As based on my >> experience I think that WebRTC based phones are far better and cheaper then >> those soft / sip phone. the big plus is that they are easy to customize and >> developer can use the power of browser and web to build / offer features >> which are not possible with regular phones. >> >> Regarding your concern about BLF or call history, for me as a being >> developer it is just a matter of customization. >> >> Regards >> >> Nasir Iqbal >> >> ICTBroadcast - an Auto Dialer software for ITSP >> <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> >> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns >> http://www.ictbroadcast.com/ >> >> >> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez >> wrote: >> >>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote: >>> >>> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez >>> wrote: >>> >> On 9/26/2018 4:46 AM, Olivier wrote: >>> >> >>> >>> Hello, >>> >>> >>> >>> This morning, I asked myself if WebRTC could be a viable alternative >>> >>> to softphone deployment. >>> >>> >>> >>> For me, main issue with Softphones is the amount of work needed for >>> >>> installation and configuration. >>> >>> Also, Softphones must be carefully choosen if Deskphone-like quality >>> >>> is expected. >>> >>> >>> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade >>> >>> Softphone features (call history, BLF, ...) for WebRTC deployment >>> >>> simplicity. >>> >>> >>> >>> What do you think of this ? >>> >>> What kind of experience did you met with such WebRTC deployments ? >>> >>> What about classic telephony features (CallTransfer) ? >>> >>> Have you tried Cyber Maga Phone 2K ? >>> >>> >>> >> If you can get it to work WebRTC is a good option. The problem >>> is >>> >> that any changes in your network may disrupt it and even trying to >>> >> replicate your installation is difficult. I have it working fine on >>> my >>> >> website so customers can call us directly from our web page but I >>> never >>> >> could get Cyber Mega Phone 2K to work on the same server. We used >>> JSSIP >>> >> to create the webrtc phone on our website. >>> > We just updated the documentation for how to get CMP2K working on the >>> > wiki [1]. We'd love some feedback if you still have issues getting it >>> > setup so that we can improve the docs. >>> > >>> > [1] >>> https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone >>> > >>> > Best wishes, >>> > Matthew Fredrickson >>> > >>> I followed the procedure indicated in the link but I cannot get >>> remote video. I can only see my own feed. We do have audio for a >>> little while. For some reason the users get disconnected after a few >>> minutes even though you can still see your video feed on screen. This >>> was done with Asterisk 15.6.0 >>> >>> -- >>> Telecomunicaciones Abiertas de México S.A. de C.V. >>> Carlos Chávez >>> +52 (55)8116-9161 >>> >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Astricon is coming up October 9-11! Signup is available at: >>> https:/
Re: [asterisk-users] WebRTC as Softphone substitute ?
Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call history, for me as a being developer it is just a matter of customization. Regards Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez wrote: > On 9/26/18 10:20 AM, Matthew Fredrickson wrote: > > > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez > wrote: > >> On 9/26/2018 4:46 AM, Olivier wrote: > >> > >>> Hello, > >>> > >>> This morning, I asked myself if WebRTC could be a viable alternative > >>> to softphone deployment. > >>> > >>> For me, main issue with Softphones is the amount of work needed for > >>> installation and configuration. > >>> Also, Softphones must be carefully choosen if Deskphone-like quality > >>> is expected. > >>> > >>> Now that WebRTC becomes ubiquitous, it might make sense to trade > >>> Softphone features (call history, BLF, ...) for WebRTC deployment > >>> simplicity. > >>> > >>> What do you think of this ? > >>> What kind of experience did you met with such WebRTC deployments ? > >>> What about classic telephony features (CallTransfer) ? > >>> Have you tried Cyber Maga Phone 2K ? > >>> > >> If you can get it to work WebRTC is a good option. The problem is > >> that any changes in your network may disrupt it and even trying to > >> replicate your installation is difficult. I have it working fine on my > >> website so customers can call us directly from our web page but I never > >> could get Cyber Mega Phone 2K to work on the same server. We used JSSIP > >> to create the webrtc phone on our website. > > We just updated the documentation for how to get CMP2K working on the > > wiki [1]. We'd love some feedback if you still have issues getting it > > setup so that we can improve the docs. > > > > [1] > https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone > > > > Best wishes, > > Matthew Fredrickson > > > I followed the procedure indicated in the link but I cannot get > remote video. I can only see my own feed. We do have audio for a > little while. For some reason the users get disconnected after a few > minutes even though you can still see your video feed on screen. This > was done with Asterisk 15.6.0 > > -- > Telecomunicaciones Abiertas de México S.A. de C.V. > Carlos Chávez > +52 (55)8116-9161 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Astricon is coming up October 9-11! Signup is available at: > https://www.asterisk.org/community/astricon-user-conference > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Outbound on SPA3102 FXO stopped to work. Where to look at ?
Can you confirm if said device is being provisioned correctly by provisioning server. and if there is no Firewall between said device and auto provisioning HTTP server ? And what is PSTN line status ? you can check both from SPA3102 web interface. Regards Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Tue, Dec 20, 2016 at 10:26 PM, Olivier <oza.4...@gmail.com> wrote: > Hello, > > I'm currently facing a quite strange issue. > > On a customer location, an old SPA3102 suddenly stopped to work a couple > of days ago. > More precisely, calls still come in but I can't dial out to PSTN. > This box worked for several years for oubound and inbound calling. > > My setup is: > Asterisk 11 <--SIP--> SPA3102 <---FXS/FXO---> Router with FXS port < @ > ---> PTSN > > It took me days, long time ago, to adapt default SPA3102 config to local > conditions. > The box autoprovision itself from HTTP server. > I didn't change anything lately in its config file. > When I directly plug an analog into my router box, I can dial out. > > > Symptoms are: > I can't hear dialed digits anymore > destination phone doesn't ring, > SPA3102 Info page displays dialed number. > > Either, my provider changed something which bothers my SPA3102 somehow but > not an analog phone, my SPA3102 "is getting old" and needs to be > reconfigured a bit to behave as usual. > > Has anyone met something like this ? > Suggestions ? > > Best regards > > -- > _ > -- 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] Pet project: one step Asterisk compile on Centos 7
Hi All, Thanks to Tzafrir Cohen who described very well that why RPM is better then any other script and how easy to build one. However for those like me still enough lazy to collect dependencies or setup mock builder, I recommend Koji build system for them. Either one can host his/her own Koji server or can freely signup for Community Koji Build system being hosted by CentOs and Fedora After you have access to koji build system it is very simple and quick to make your own custom RPM for any target distribution and architecture of your choice! Regards Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Tue, Jun 14, 2016 at 10:50 PM, Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote: > Off-topic: > > On Tue, Jun 14, 2016 at 11:02:17AM +0200, Lenz Emilitri wrote: > > > Project located at https://github.com/l3nz/CompileAsteriskPBX > > From the build script: > > # build Asterisk > cd $TARGET_DIR/$ASTVERSION > ./configure --libdir=/usr/lib64 > cd $TARGET_DIR/$ASTVERSION/menuselect > make menuselect > cd $TARGET_DIR/$ASTVERSION > make menuselect-tree > > ./menuselect/menuselect \ > > #[snip menuselect parameters] > > # we want mp3's > ./contrib/scripts/get_mp3_source.sh > > make > > (This is not intended to criticise Lenz) > > All of this magic (specifically the bits before the menuselect) is > needed for a proper build? > > * Why is an explicit libdir packameter needed? What break if it is not > passed? > * Why is there a need to build menuselect manually? What about 'make > menuselect'? Do we need a Makefile target to build menuselect but not > run it? > > -- >Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- 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] How to list parameters for applications data
Hi List, According to official documentation Originate application can accept two type of destinations either extension or application, Now if we are using application as destination then in forth parameter we have to provide application data, my question is about this data parameter. *What format should we use if application data contain more then one parameter*, for example if we have to exten => _X.,n,Originate(SIP/,app,ChanSpy,someChannel,oqs) Where "someChannel,ogs" is application data containing two parameters separated by comma, of course above dialplan will generate a syntax error in asterisk, My question is if there anyway to fix this syntax error without compromising extra parameters in application data. Nasir Iqbal ICTBroadcast - an Auto Dialer soft ware for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Abandoned SIP-TCP connection causes Asterisk to crash
Hi All, We are using SIP over TCP transport but often we got an Asterisk crash with following error. [Mar 1 11:23:13] WARNING[1509]: chan_sip.c:3755 __sip_xmit: sip_xmit of 0x7f294000cac0 (len 680) to Soft.Phone.IP.Address:56780 returned -2: Interrupted system call Asterisk uncleanly ending (0). Executing last minute cleanups == Destroying musiconhold processes == Manager unregistered action DBGet == Manager unregistered action DBPut == Manager unregistered action DBDel == Manager unregistered action DBDelTree We have tested this issue with Asterisk 11.20 and it can be reproduced as following 1. From client workstation / labptop register a softphone with Asterisk over TCP 2. Dial into some local asterisk extension. (i.e play voice message in loop ) 3. From client side kill the softphone i.e ( killall zoiper ) 4. Wait some time, Asterisk will crash Further we are unable to get any coredump ! (even running with -g) Any help will be appreciated. Thanks in advance Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play sound file
check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal e...@mcr-m.com wrote: Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- 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] Outbound Dialer, Agent Login and Logout
Check our asterisk based software ICTBroadcast at http://www.ictbroadcast.com, it might fulfil your requirements, It support multiple campaign types including playing voice message to recipients and forwarding call to live agent Regards Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Sat, Dec 31, 2011 at 3:19 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use his help. Regards Bilal -- _ -- 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] Working on web based IVR Designer for asterisk and Freeswitch
We are working to develop a web based IVR Designer that will work with Asterisk as well as Freeswitch using Raphaejs library, Click following link for detail http://sourcecodemania.com/ivr-designer-using-raphaeljs-for-asterisk/ Looking for your valuable suggestions Regards Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues
for which user/number sip reinvite is for? ooh! you are running a direct application without any dialplan or user, may be that is the cause! I think you should first write fax dialplan, reload asterisk and test again with originate but this time with extension not application. Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Oct 8, 2011 at 12:20 AM, James Sharp ja...@fivecats.org wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 -- __**__**_ -- 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-**usershttp://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] Digium FFA + Gafachi T38 outgoing issues
Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Fri, Oct 7, 2011 at 1:37 AM, James Sharp ja...@fivecats.org wrote: Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963@gafachi1a for application SendFAX(/srv/httpd/htdocs/**upload/scantest2.tiff,dz) (Retry 1) == Using UDPTL CoS mark 5 == Using SIP RTP CoS mark 5 Channel SIP/gafachi1a-000a was answered. Launching SendFAX(/srv/httpd/htdocs/**upload/scantest2.tiff,dz) on SIP/gafachi1a-000a -- Channel 'SIP/gafachi1a-000a' sending FAX: --/srv/httpd/htdocs/upload/**scantest2.tiff -- Channel 'SIP/gafachi1a-000a' FAX session '6' started -- FAX handle 0: [ 000.000594 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX -- FAX handle 0: [ 000.001139 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY rt: TRDYNHTY -- FAX handle 0: [ 000.001724 ], P30EVN_SEND_STARTED [Oct 6 04:21:36] ERROR[11616]: res_fax.c:1421 generic_fax_exec: channel 'SIP/gafachi1a-000a' FAX session '6' failure, reason: 'fax session timed-out' (TIMEOUT) [Oct 6 04:21:36] NOTICE[11616]: pbx_spool.c:373 attempt_thread: Call completed to SIP/18884732963@gafachi1a THIS PART HAPPENS ABOUT 15 SECONDS LATER -- FAX handle 0: [ 040.000211 ], STAT_EVT_T1_EXPst: WT_DIS rt: WDISNT1X -- FAX handle 0: [ 042.499953 ], STAT_EVT_HW_CLOSE st: WT_HW_CLS rt: WCLSNCLS -- FAX handle 0: [ 042.500083 ], STAT_SES_COMPLETE -- FAX handle 0: [ 042.500110 ], P30EVN_COMPLETE -- Channel 'SIP/gafachi1a-000a' FAX session '6' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' A tcpdump trace shows the initial invite, ringing, answering, some G711 frames back and forth, the send-T38-invite-after-10-**seconds reinvite (as specified by the Z option), then the far end sends a bunch of T38 traffic until Asterisk times out and drops the call. What also confuses me is this (and this may just be semantics or a true bug): asterisk*CLI fax show stats FAX Statistics: --- Current Sessions : 0 Reserved Sessions: 0 Transmit Attempts: 8 Receive Attempts : 0 Completed FAXes : 7 Failed FAXes : 7 How can I have 8 attempted transmits, 7 completed faxes, and 7 failed faxes? I know 1 transmit didn't go through because I tried to place one call while another was in progess and I only have one licensed channel. Thanks, James Sharp ja...@fivecats.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-**usershttp://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] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]
You can do this by an AMI based transfer (Redirect) to Local channel, and then in local channel's dialplan you need to add your desired custom sip header followed by a dial command. Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Wed, Oct 5, 2011 at 11:36 AM, Olivier oza_4...@yahoo.fr wrote: 2011/10/4 Olivier oza_4...@yahoo.fr Hi, Has anyone heard (or read) about an existing or emerging standard targeting the following feature : 1. a SIP handset receives an incoming call 2. this handset starts ringing 3. then it receives an update asking to autoanswer the ringing call. This feature would help to build software panels complementing or replicating hard phones GUI. (I know you can work around such feature using conference rooms or dealing with hard phones API (really ?) but in order to keep Queue log accurate, this feature would be useful). Cheers Hi, In my quest to allow a software panel to ask an hardphone to answer an incoming call without touching the hardphone itself, I'm wondering if a Reinvite application could exist. I'm thinking about the following use case : Alice is calling Bob Bob's phone starts to ring Bob's GUI app also shows the incoming call asking him if he prefers to reject, transfer or answer the call With the GUI app, Bob replies he whishes to reply Asterisk reinvites Bob's phone with autoanswer option Bob's phone answers Alice phone Thoughts ? -- _ -- 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] Passive wait in dialplan?
What about waiting in queues?
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych panyc...@gmail.com wrote:
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same = n,Set(COMMAND=${CMD_NOOP})
same = n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =
n(COMMAND_SWITCH),GoToIf($[${COMMAND}=${CMD_DOSTUFF1}]?LBL_DO_STUFF1)
same = n,GoToIf($[${COMMAND}=${CMD_DOSTUFF2}]?LBL_DO_STUFF2)
same = n,GoToIf($[${COMMAND}=${CMD_DOSTUFF3}]?LBL_DO_STUFF3)
same = n,Wait(0.2)
same = n,GoTo(COMMAND_SWITCH)
same = n,NoOp(--- NOT REACHED ---)
UserEvent sends blah-event via AMI to high-level UI, user makes
decision and issues some command via Action:SetVar, then dialplan
continues to work.
The problem is, in dialplan there is an active wait loop, i.e. waiting
mechanism which rapidly checks some var(consuming processor resources
and flooding logs). Is it possible to create passive waiting loop
within current abilities of Asterisk 1.8?
regards, Yaroslav
--
_
-- 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] Delay before ringing from PSTN`s call
On some analogs systems caller id is sent after first ring, so removing callerid=asreceived may help Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Tue, Oct 4, 2011 at 4:38 AM, neo haux neo.h...@gmx.com wrote: Hi I am testing a degium TDP400P (2fxo+2fxs) on my asterisk I configured incoming calls from pstn to ring my SIP phone (extension : 100) cat extensions.conf ... [from-pstn] exten = s,1,Dial(SIP/100,10) same = n,VoiceMail(100,u) root@PC-debian:/etc/asterisk# cat dahdi-channels.conf ... ... ... ;;; line=1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default ... ... ... What I don`t understand is why the SIPphone rings after 3 secondes after Astererisk detects the incoming call. Moreover, after hanging off the external caller the SIPphone continue to ring for 3 seconds. I did those modifications in the file /etc/asterisk/chan_dahdi.conf without improuvement ( After restarting Asterisk) [channels] cidstart=ring immediate=yes faxdetect=no usecallerid=no Here is the debug from Asterisk console *CLI -- Starting simple switch on 'DAHDI/1-1' -- Executing [s@from-pstn:1] Dial(DAHDI/1-1, SIP/100,10) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- SIP/100-0001 is ringing == Spawn extension (from-pstn, s, 1) exited non-zero on 'DAHDI/1-1' -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' -- _ -- 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] Database Lookup Advice
have you tried with MYSQL command? http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Tue, Oct 4, 2011 at 11:25 PM, Bryant Zimmerman brya...@zktech.comwrote: Hey all I wanted to get some input on what you all think is the best way to lookup database data from asterisk dial plan. This is a two fold question. 1. I am using fun_odbc to pull settings and values back and it works good but is there a better way. I want to maintain performance and simplicity as much as possible. 2. func_odbc does not appear to allow for reading multiple records in a return set. Is there a way around this or is there a better method. I understand I can always us a mono, pearl, php, lua or some other kind of script to handel some of this but I am looking for better ways to do database access using the built-in dialplan abilities first. I look forward to ideas and input. Thanks Bryant -- _ -- 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] T.38 client for Linux?
Yes, Zoip support T.38 faxing but It is only client application and you need FOIP gateway (asterisk) to transmit a fax to your FXO port Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Thu, Sep 22, 2011 at 3:20 AM, Olivier oza_4...@yahoo.fr wrote: 2011/9/21 Ian Pilcher arequip...@gmail.com I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of softfax that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. This is one of those things that I thought would be relatively straightforward, but a couple of hours of Googling has left my head spinning. I'm posting here in the hope that there is a (fairly) simple way to do this, and someone can point me in the right direction. Thanks! -- Ian Pilcher arequip...@gmail.com If you're going to shift my paradigm ... at least buy me dinner first. -- _ -- 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 Doesn't Zoiper include some T.38 features ? -- _ -- 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] T.38 client for Linux?
You can use ictfax HTTP://www.ictfax.org web interface to send faxes, Ictfax is pure foip software based on t.38 as compared to hylafax No need for iaxmodem and client application On 22-Sep-2011 4:00 AM, Larry Moore lmo...@starwon.com.au wrote: On 22/09/2011 4:12 AM, Ian Pilcher wrote: I am looking for a simple way to send occasional faxes via the FXO port on my SPA3102 -- without having to connect a fax modem to an ATA. In an ideal world, this would be some sort of softfax that runs on my Linux desktop and talks (via Asterisk) to the SPA3102 with T.38. The SPA3102 doesn't use T.38 on the FXO port. You can do it by using HylaFAX with an iaxmodem and a HylaFAX client on your desktop. -- _ -- 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] NC DATA FINDOUT IN AUTO DIALER
Please check offline message Regards Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Tue, Sep 20, 2011 at 2:47 AM, mahesh katta maheshka...@flexydial.comwrote: Thanks for reply, I had check it. in auto dialer whenever dial the number there is no voice to get agent. dialer will dial the number asterisk not getting voice like swo,NA. how we can get the voice in there. Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com On Mon, Sep 19, 2011 at 6:23 PM, Nasir Iqbal na...@ictinnovations.comwrote: Please check our voice sms and fax broadcasting / smart autodialler / smart predictive dialler based on asterisk named ictbroadcast , it provide real time report of busy, answered, congestion , failed, no answer call statistics of running campaign HTTP://www.ictinnovations.com/ictbroadcast Regards On 19-Sep-2011 7:13 PM, mahesh katta maheshka...@flexydial.com wrote: Hi List, I have one query, I am using Go autodial in this using auto dialing. autodial can do only whenever customer pick the call that call will go to agents. but problem autodial dialing the database in that I am not getting NC data means, not reachable,switch off ,outofservice data. how can I get this data. is there any software get the telcovoice and give report ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] NC DATA FINDOUT IN AUTO DIALER
Please check our voice sms and fax broadcasting / smart autodialler / smart predictive dialler based on asterisk named ictbroadcast , it provide real time report of busy, answered, congestion , failed, no answer call statistics of running campaign HTTP://www.ictinnovations.com/ictbroadcast Regards On 19-Sep-2011 7:13 PM, mahesh katta maheshka...@flexydial.com wrote: Hi List, I have one query, I am using Go autodial in this using auto dialing. autodial can do only whenever customer pick the call that call will go to agents. but problem autodial dialing the database in that I am not getting NC data means, not reachable,switch off ,outofservice data. how can I get this data. is there any software get the telcovoice and give report ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZRTP SDK Source
Hi, As zfone download server is offline, is there anyone who can provide me copy / link of libzrtp SDK source? Thanks in Advance Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Processing sip messages
Is there an way (asteisk command / AMI / Agi ) to process incoming SIP messages like ( 100 trying , 183 session progress , 200 Ack) , I am intersted to findout delay between 183 and 200 message Regards Nasir Iqbal -- _ -- 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] Faxes
Try open souce solution ICTFAX for T.38 faxing developed by us available at http://www.sourceforge.net/projects/ictfax Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ On Sat, Sep 4, 2010 at 3:03 AM, Joel Maslak jmas...@antelope.net wrote: g711 across a network without perfect jitter/delay characteristics will not work. You cannot do g711 faxing across the internet - at all. It's not a perfect solution even in an office on a dedicated LAN environment (you'll still get failed faxes). On Fri, Sep 3, 2010 at 12:32 PM, dave george dgeo...@teletoneinc.comwrote: Thanks Kevin, I tried passing it over VOIP using g711U codecs with no success. I will try using the patches that you mentioned and post the results. Thanks, Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, September 03, 2010 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Faxes On 09/03/2010 10:50 AM, dave george wrote: The asterisk box is connected to the PSTN using TE410 cards. Asterisk talk SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the PSTN. The carrier sending the calls wants me to be able to pass faxes to physical fax machines on the PSTN. So far they are failing. We just want ot be able to pass faxes using g711u or t.38 pass through. As I told you on the asterisk-ss7 list, you can't 'pass through' T.38, because the PSTN does not speak T.38. If one side of the call is SIP, and the other side is TDM, then you have only two choices: pass the call through in audio mode (FAX over VoIP), or act as a T.38 gateway (FAX over T.38). At this time, the only option without patching Asterisk is to pass the call through in audio mode, but there are many, many problems with doing FAX over VoIP (Steve Underwood's page on the soft-switch.org site explains them very well). There are patches in the issue tracker at issues.asterisk.org to add T.38 gateway functionality to various releases of Asterisk, and they work well for quite a few people. If you added that, you'd be able to act as a T.38 gateway, which would dramatically increase your chances of success. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Calling Line Identity - any ideas
In simple words , Paddy should go with my trick, that is what i got from this reply Regards On Sat, Aug 21, 2010 at 5:14 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Nasir Iqbal na...@ictinnovations.com wrote: With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ? also it would be quiet helpful for us if you understand well the requirement of post *snip* Nasir, You don't need my permission to post on a public forum...However, neither do I, and I took issue with what you said, and found that your comment about those who are dealing with high load traffic offensive, since it made the assumption that I was just some new guy who deals with hobby/small Asterisk systems and doesn't know what he's talking aboutTherefore, I made it abundantly clear that I wasn't, and that I definitely took issue with that statement. However, I will say that yes, I did mis-take something the OP said... Paddy: Now, here's idea I came up with (haven't tested yet, too busy writing a system for an international interpretation company's telecom needs) First of all, you should have a separate context for outbound calls made by internal extensions... so, in THAT context have code to set the CID to what you wish (you can do logic control and if you're feeling spiffy you can even lookup what CLID to use based on the extension making the call). Second, calls that are being passed from the outside world onto should pass through a different context, performing pretty much the same function... Third, both of THOSE contexts should then pass to a third context that performs the dialout using the multiple targets... Let me know if that works...I know I can make this do what you want, but I'm not trying to do all the work, just point you in a direction, since I get paid to actually do the work ;-) Cheers all, and remember, some of us have been doing this a while, and get grumpy... ;-) there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? 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 First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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
Re: [asterisk-users] Calling Line Identity - any ideas
With all honor and respect you deserve, Do I need your permission to express my point of view on community forum ? also it would be quiet helpful for us if you understand well the requirement of post Regards On Fri, Aug 20, 2010 at 1:34 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: Paddy, I believe I have a solution, let me sober a bit ;) and rum it through (typo not intended but funny) my test server to doublecheck Sent from my iPhone On Aug 20, 2010, at 12:20 AM, Paddy Grice pa...@wizaner.com wrote: Hi Sherwood I actually do want dynamic CLID as I tried to make clearer I don't know if this makes it any clearer - An internal call from Ext123 should send 123 as the CLID to SIP/ Ext400 but should send 442071110123 to SIP/TheWorld but an external call from 44123455667788 should send the received CLID 44123455667788 to both. So over the provider connection the CLID will be different for different calls. Setting the main office number in sip.conf is fine as a default but as the code/dialplan needs to set cli for some calls I actually set CLID for all calls. This setting and onward transmission by provider works fine. So what I am trying to do is call 2 different sip endpoints AT THE SAME TIME presenting different AND VARIABLE CLIs. If Nasir's trick is not recommended what is the best way to achieve this. As a newbie to Asterisk advise and best practice gained from user experience is always welcome. Paddy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sherwood McGowan Sent: 20 August 2010 04:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling Line Identity - any ideas On Thu, Aug 19, 2010 at 7:46 PM, Nasir Iqbal na...@ictinnovations.com wrote: Hi, there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? Regards -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users First, the reason is, why use a BAD IDEA when there's perfectly good solutions in front of the user There was no mention on this ONE call going outbound over the trunk needing a different CID...the request was as follows: Client needs to call an INTERNAL extension, where the INTERNAL CallerID will be used, and at the SAME TIME, a call to an EXTERNAL number (which would necessitate USING THEIR PROVIDER TRUNK), using the EXTERNAL CallerID Now, p-lease tell me how just configuring the damned trunk's outbound CID is NOT more sensible, efficient, and just friggin' COMMON SENSE TO START WITH...over using a Local channel call, which would require slightly more typing, and using something that I've almost NEVER found a good reason to use, and if you'd care to search the damn archives, you'll see that I was pushing upwards of 5k CONCURRENT CALLS back in 2005, WITH 1.4 Trunk and the RealTime addiiton (which was experimental)... For the love of whatever you find holy and good and true...don't come at me like that...I'm really not in the mood anymore...I put 3-4 solid years of helpjng newbies figure out why shit didn't work, reporting REAL bugs and issues to thew developers and even assisting with some of the fixesI feel entitled (yes, I know that's an asshole thing to say) to a little common respect Now...anyone for a pint? I'm off to vent some frustration with people who jump on the WRONG bandwagon and try to take over Sherwood Mother-F'in' McGowanb... Telecommunications and Tattooing You konw anyone else who combines those two professions? I'd like to buy that guy a drink! -- _ -- 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
Re: [asterisk-users] Calling Line Identity - any ideas
Hi,
here is a trick for you!
exten = s,1,Dial(SIP/Ext400Local/${EXTEN}/home-context)
[home-context]
exten = s,1,Set(CALLERID(num)=44112233445566)
exten = s,1,Dial(SIP/TheWorld/441234567890)
Regards
On Thu, Aug 19, 2010 at 12:21 PM, Paddy Grice pa...@wizaner.com wrote:
Hi list
I have a requirement that I just don't know how to address - I don't think
its strange but can't find any pointers anywhere.
I have a user that wishes to have a multi phone divert. By that I mean
calls made to his extension say Ext200 can be redirected to a different
extension say Ext400 and also to his home landline.
Doing the dial is fine using Dial(SIP/Ext400SIP/TheWorld/441234567890)
The problem is CLID -
At the moment internal calls (Ext to Ext) show a CLID EXTxxx and External
Calls show the received CLID.
When the phone is redirected to both Internal and external numbers he wants
the correct CLI displayed on both phones.
So with the redirect operational
1) a call from the outside world to his DID number will show the received
CLI(ANI) on both devices - this works
BUT
2) a call from an office extension needs to show EXTxxx on the extension
(Ext400) but show the office telephone number on the landline
so in pseudo code I want to do something like
Dial ( SIP/Ext400 using CLID EXT123 SIP/TheWorld/441234567890 using
CLID 44112233445566 )
Any ideas ?
Paddy
--
_
-- 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
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling Line Identity - any ideas
little syntax mistake, try this
exten = s,1,Dial(SIP/Ext400Local/${ext...@home-context)
[home-context]
exten = s,1,Set(CALLERID(num)=44112233445566)
exten = s,n,Dial(SIP/TheWorld/441234567890)
--
_
-- 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] Calling Line Identity - any ideas
Hi, there's still no conceivable reason What can be? except performance! (as asterisk has to create one additional leg and bridge it) Which is very conceivable to those who are dealing with high load traffic. And what will be the option, if other outgoing call requires different custom CLI while using the same trunk? Regards -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 debug of registration - Only getting RX and there is no TX response from Asterisk - is that normal?
Hi, Use requirecalltoken=no in your peer configuration Regards On Wed, Aug 11, 2010 at 4:28 AM, bruce bruce bruceb...@gmail.com wrote: Hello Everyone, I am trying to diagnose issue with my IAX2 extension not working. When I have iax2 set debug on all I see is this: *Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ * * Timestamp: 3ms SCall: 00130 DCall: 0 [64.229.229.111:64823]* * USERNAME: 100* * REFRESH : 60* * * *Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK* * Timestamp: 3ms SCall: 00130 DCall: 1 [64.229.229.111:64823]* So, all the packets are coming in, but there is no Tx response. Is that normal and is that how IAX2 works according to RFC to not respond back? I have checked my firewall and all is set fine. I have any WAN address to come in through port 4569 to map to the server and it worked last week but now it doesn't. Any suggestions? Thanks -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MySQL Connect problem...
Avoid to use MySQL dialplan application, instead write an AGI script for
this purpose
On Tue, Aug 17, 2010 at 4:59 PM, Geraint Lee gera...@gmail.com wrote:
Right, I'm baffled.
I have:
exten = s,1,MYSQL(Connect DB1 127.0.0.1 geraint xxx amis2)
exten = s,n,MYSQL(Query NORESULT ${DB1} INSERT\ INTO\ recordings\
(caller_number\,called_number\,date_created\,date_started\,in_use\,server_id)\
VALUES\ (\'${CALLERID(number)}\'\,\'${ARG1}\'\,NOW()\,NOW()\,\'Yes\'\,12))
exten = s,n,MYSQL(Query RESULT1 ${DB1} SELECT\ LAST_INSERT_ID())
exten = s,n,MYSQL(Fetch FOUND1 ${RESULT1} VALUE1)
exten = s,n,MYSQL(Clear ${RESULT1})
exten = s,n,MYSQL(Disconnect ${DB1})
exten = s,n,MixMonitor(${VALUE1}.wav)
exten = s,n,Set(CALLERID(all)=xxx)
exten = s,n,Dial(SIP/prov1/${ARG1})
in a macro to dial numbers...
Every few hours or so every call hangs on the s,1 MYSQL(Connect) and won't
work until i restart asterisk.
The mysql server has a maximum connections of 2048 (of which around 90 are
in use) so it's not a mysql connection limit problem from what i can tell
since while asterisk is stuck i can still log in to mysql just fine, as
can the web server.
Does anyone have any suggestions what could be causing asterisk to get
stuck here? i don't see anything in cli and core show channels just shows
everyone stuck in state ring on the connect string with no errors.
Cheers
Geraint
--
_
-- 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
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime sip peers : musiconhold class
Hi to convert wav file use following sox 'orgFile' -w -r 8000 -c 1 -s 'fixedFile' while replace orgFile and fixedFile with actual filenames If still now luck try with mp3 Regards -- _ -- 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] Prepay Limited Calls.
Hi Use Set(TIMEOUT(absolute)=XYZ) in your dialplan or timeout parameter in Dial and Originate commands. Get maximum available seconds from your db for calling peer and use it as timeout. But after every call you have to deduct used time from you db for calling peer. Regards On Mon, Aug 9, 2010 at 2:01 PM, Catalin S. jonsonpla...@gmail.com wrote: Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really want to install another software to make this or modify all my settup. I'm wonder if someone is using something simple to limmit calls. Anyway if someone is using some other programs/software/scripts and another settup/method please let me know how is yours. I want to check few methods to realize that limmit. Thank you for help guys. -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to track a call result originated from originate AMI command
Hi, Confusing! you are not alone here. Actually there is no unified development approach exist in Asterisk, every module, application introduce a new way to handle same things!! And the monitoring is most difficult part! you have to write different parsing algos to get each bit of information, and unfortunately you have to rewrite most of your code for every new release! And regarding your question, I recommend you to use AGI for monitoring here is some tips for you - in originate command use extension as destination. - create failed extension in same context. - you can include some variables in originate command which can be used later in dialplan. - use AGI scripts in destination and failed extensions to get and save call status in database. Regards On Sun, Aug 8, 2010 at 6:10 PM, thiyagu venkatesan thiyagu.v...@gmail.comwrote: Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were very confusing and I m getting various events with different uniqueid value. For a single call I m getting events with four or five uniqueid. I also filtered using specific channel but also I m getting events with different uniqueid. How can I find the below status for the call generated using originate command through AMI events, 1. Answer 2. No Answer 3. Busy Can any one help me for this. Thanks, Thiyagu VOIP -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
I agree with you and suggest you to use CLI command via AMI, for example Command core show channels I prefer CLI commands when they are available, as they return an aggregate response as compared to AMI you do not need to filter, identity, and group multiple responses / events to get result of a single command! Regards On Sun, Aug 8, 2010 at 11:28 PM, Richard Zulu richard.z...@time.co.ugwrote: Thanks Nasri, I don't want to only be able to use the CLI because I need the Helpdesk and application support Unit to be able to monitor, and they are not all the techy with CLI and stuff.. On Sun, Aug 8, 2010 at 5:00 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi following asterisk cli commands can help show channels, show uptime and show sysinfo here is an example asterisk -x core show sysinfo On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote: Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMD setup in Astersik
Hi Tino,
I think you can do it by using dummy queue number. for example create 500
queue in freepbx. and replace your goto command in ext-queues-custom with
exten = 5000,n,Goto(ext-queues,500,1)
Regards
On Sat, Aug 7, 2010 at 7:06 PM, Tino t...@sparksupport.com wrote:
In my Asterisk server following things have been done to detect answering
machines before the answered call connects to the agents in queue.
In extension_additional.conf
==
[ext-queues]
include = ext-queues-custom
exten = 5000,20,Macro(user-callerid,); changed the priority to 20
...
==
In extension_custom.conf added following amd dialplan
===
[ext-queues-custom]
exten = 5000,1,Answer()
exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384)
exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human)
exten = 5000,n(machine),Verbose(3, We found an answring machine)
exten = 5000,n,Set(AMP=${CALLERID(num)})
exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = 5000,n,System(not showing the actual command)
exten = 5000,n,Goto(ext-queues,5000,20)
exten = 5000,n(human),Verbose(3, We've got a human on the line!)
exten = 5000,n,Goto(ext-queues,5000,20)
===
This setup is working fine but the problem is that when i reload freepbx,
extension_additional.conf will go to its original form
and the changes made will be lost. Is there any way to make the changes in
extension_additional.conf conf permanent . Or is there any alternative
method for this ?
--
_
-- 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
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
Hi following asterisk cli commands can help show channels, show uptime and show sysinfo here is an example asterisk -x core show sysinfo On Sun, Aug 8, 2010 at 12:25 AM, Richard Zulu richard.z...@time.co.ugwrote: Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to reuse mysql connection between AGI's
Hi Faheem, You need to build some daemonized application, here FastAGI will help you Regards On Fri, Aug 6, 2010 at 10:54 AM, Faheem faheem_...@yahoo.com wrote: Hey, Is there any way to share MySQL connection between different agi's. Actually when call comes to asterisk box it executes various agi scripts sequentially. Each script checks various values by making a new MySQL connection and then execute query and then disconnects. So, Ideally there should be one connection, and it should be reused between each agi and when a call is over it should be disconnected. Is there any mechanism to reuse single MySQL connection between agi scripts? The agi scripts are written in Perl Thanks, Faheem, M. -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] How do I install speex for asterisk?
Hi, May you also need to install *speex-tools* . if problem retain then let us know about your Linux distribution and Asterisk version. Regards On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] How do I install speex for
Hi, Currently CentOS yum repository does not provide speex-tools so you have to install it manaully. follow the steps given below. 1. first remove existing speex pacages yum remove speex* 2. run following to install required rpms rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-1.2-0.10.rc1.i386.rpm rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-devel-1.2-0.10.rc1.i386.rpm rpm -ivh http://www.lfarkas.org/linux/packages/centos/5/i386/gstreamer/speex-tools-1.2-0.10.rc1.i386.rpm 3. reconfigure asterisk 4. verify speex codec via make menuselect under Codec Translators. if speex is enabled and selected continue with make Enjoy! On Fri, Aug 6, 2010 at 7:13 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi Chandrakant I have checked and it shows func_speex module is enabled. Where can I install speex-tools from ? Asterisk version 1.6.2.10 and Centos 5.5 are installed. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use for Invoicing?
Hi Bruce, We have build an Invoicing module (ICTInovice) for Elastix. It is Free, Open Source, Generate PDF Invoices, and can mail invoices to clients! You can download it from http://sourceforge.net/projects/ictinvoice/ http://sourceforge.net/projects/ictinvoice/Note: Currently ICTInvoice only work with Elastix 1.6 Regards On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What do you use for Invoicing?
On Tue, Aug 3, 2010 at 6:24 PM, bruce bruce bruceb...@gmail.com wrote: Oh, you seem to be right on. It's actually an install of Elastix. I will be testing this for sure. Hope it doesn't do any damages though. I guess the installation material is inside the tar ball? It is very easy to install. just upload it into Elastix using module installation interface!!. for further information you can check user manual. No. you have to download it separately from sourceforge. Thanks On Tue, Aug 3, 2010 at 2:01 AM, Nasir Iqbal na...@ictinnovations.comwrote: Hi Bruce, We have build an Invoicing module (ICTInovice) for Elastix. It is Free, Open Source, Generate PDF Invoices, and can mail invoices to clients! You can download it from http://sourceforge.net/projects/ictinvoice/ http://sourceforge.net/projects/ictinvoice/Note: Currently ICTInvoice only work with Elastix 1.6 Regards On Mon, Aug 2, 2010 at 11:26 PM, bruce bruce bruceb...@gmail.com wrote: Hi Everyone, Sorry, if it's not directly related to Asterisk. Some of people on this list might have PBX deployed for their clients. What software do you use to invoice them so the invoice looks like a proper telecom invoice maybe? Prefer: -opensource with Windows binary available. -able to create .pdf invoices rather than printable ones. Thanks -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX softphone
I use http://www.voixphone.com/ On Mon, Aug 2, 2010 at 9:41 PM, Alan Lord (News) alansli...@gmail.comwrote: On 02/08/10 17:35, Ronaldo Zacarias Afonso wrote: Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. This one is great on Ubuntu/Linux. http://www.sflphone.org/ Unfortunately I know not about Windows though, I never use it. Cheers Al -- The Open Learning Centre http://www.theopenlearningcentre.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR and custom name fields
MSSQL Updateable Views can help you On Thu, Jul 29, 2010 at 7:15 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I have Asterisk 1.4, trought ODBC I'm savind CDR to MSSQL. How can I change the fields name of database? You will want the adaptive CDR backport to Asterisk 1.4: https://issues.asterisk.org/view.php?id=1 http://svncommunity.digium.com/view/tilghman/branches/1.4/ Philipp -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] urgent:how to transfer a call using asterisk FAGI
Use dial application along with (agi command) exec for more see http://www.voip-info.org/wiki/view/exec On Tue, Jul 27, 2010 at 10:38 AM, Janu Mukherjee janu.mu...@gmail.comwrote: Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==1500,1,AGI(localhost//hello.agi So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent. Thanks Regards, Jahnavi. -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preserving CDR(accountcode) in Local channels
while setting up accountcode value try two underscores just before variable name like '__accountcode' for more google for Inheritance of Channel Variables On Tue, Jul 20, 2010 at 5:05 PM, Chris Bagnall aster...@lists.minotaur.ccwrote: Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten = 123,1,Set(CDR(accountcode)=foo) exten = 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member = Local/4...@outbound member = Local/5...@outbound member = Local/5...@outbound The 'accountcode' field is empty in the CDRs, even if I add /n to the queue members. So, a couple of quick questions if I may: 1) Is this behaviour expected? 2) How would one go about the above scenario whilst preserving the 'accountcode' field? Thanks in advance! Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Different source IP address for each peer
As per my knowledge Asterisk uses system network settings to pick appropriate source address so you can not do it within Asterisk until you configure Linux firewall / routes accordingly. Please correct me if I am wrong! On Tue, Jul 20, 2010 at 5:19 PM, AC a57m...@gmail.com wrote: Hi, I can configure multiple source IP addresses on a Ethernet interface. Is it possible to configure asterisk to bind to a different source IP address for each peer? Thank you, AC -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Logging registration/unregistration of peers/extensions in database
I am not aware with any logging option, but If you want to monitor registration status. Asterisk Realitme can help you. For example if you are using Realtime SIP configuration then you can find registration info at regserver and regseconds fields On Sun, Jul 18, 2010 at 7:28 PM, Bram Bosboom b.bosb...@prompt.nl wrote: Can asterisk log the registration date/time in a database? Is there a standard option to do this? I know it being logged in the asterisks 'full' (debug) log and we are probably able to script something with the API interface but there might be somewhat easier if there is a option to make asterisk log this information directly into a database. Thanks in advance, Bram -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T.30 fax receiving problem with app_fax
Try 3 second wait between Answer and ReceiveFAX
On Mon, Jul 19, 2010 at 9:27 AM, Stefan Schmidt s...@sil.at wrote:
Alexander Aksarin schrieb:
Hello, All. I have a problem with receiving fax through T.30. I'm
calling 543 number from fax machine, then start sending fax and fax
machine send document without problem. But asterisk don't receive fax.
I can't find good documentation for app_fax and I'am googled this
errors.
Please help me.
software: asterisk-1.6.2
OS: ALT Linux 5.0.1 Ark Server
hardware: Digium Wildcard TE110P T1/E1
fax - avaya T1 asterisk
Part with fax from extensions.conf:
exten = fax,1,Goto(543,1)
exten = 543,1,Answer()
exten = 543,n,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num)}.tif)
exten =
543,n,Set(FAXFILENOEXT=/var/spool/asterisk/fax/${CALLERID(num)})
exten = 543,n,ReceiveFAX(${FAXFILE})
debug log: http://pastebin.ca/1903349
Hello,
plz have a look at issue 17641
(https://issues.asterisk.org/view.php?id=17641) and try the patch from
there, or atleast dont use answer before ReceiveFax cause its buggy
without this patch.
what i also see in your log is that dahdi doesnt recognize the CED Tone
from your fax so the echo canceler isnt turned off, which could also be
a problem. (Row 21 to 26 in your log)
best regards
steve
--
_
-- 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
--
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI execution after Dial
Try with something like action.setChannel(SIP/99051000xxx...@yourtrunkname); On Sat, Jul 17, 2010 at 10:19 PM, Felipe Kurkowski felipekurkow...@gmail.com wrote: Hello, I'm currently developing a simple asterisk application using SFS (Skype For SIP) which tries to call to an outbound number, play a message and read DMTF digits. My first approach used the Manager to originate calls and then called an agi script to deal with the rest. Anyway, this ended up being not so clear because the call did not start on the Originate extension that it was supposed to. Instead it would go to the Skype ID number extension. For example, if I originate a call with the code below, it will go first to the 9051000XX extension and then to 1. Is it possible to use the CONSOLE (somehow like console dial number) channel to originate calls? This might be a solution. action.setChannel(SIP/99051000XX); action.setCallerId(99051000XX); action.setContext(autodialer); action.setExten(1); action.setPriority(new Integer(1)); action.setVariable(numero, 555); Then, I figured I could place the calls from within an AGI script. Obviously, I got stuck again. Now, when I execute the application Dial, the script pauses until the called party hangs up. This behavior is expected but I'd like to know if there's any way to continue the execution of the script so I can play the message and read the digits. I tried to create multiple threads to see if I could continue with the script even after the dial, but it would not run the second thread until the call ended. Any help on this subject is welcome. Kindly, Felipe KUrkowski -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan question when using a round-robin
as a quick option you can use
Dial(IAX2/server1/${EXTEN}IAX2/server2/${EXTEN}IAX2/server3/${EXTEN})
call will connect to whichever is available answer, others simply ignored!
On Sun, Jul 11, 2010 at 8:09 PM, unsero...@aol.com wrote:
But how can i determine on which physical server user B is registered?
Or is there an other, better way to achieve this? Maybe in replicating the
registrations between all 3 servers?
DUNDi is an options, same with DNS SRV records.
--
Could you please give me some more info?
Or is there a tutorial available somewhere?
--
_
-- 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] UCS-2 Problem
Hi List, Recently I tried sending sms using app_sms (hardware TDM400P) in Singapore with land line telco provider singtel it worked fine and can send sms in Latin characters 7-bits/8-bits but I am unable to send Unicode (UCS-2 or 16-bits) sms in Arabic or Chinese. the problem is that my mobile show the message with invalid character I have managed to capture the outgoing sms data as following hex values 91 31 01 00 0C 91 29 33 63 81 06 69 00 08 24 D9 85 D8 B4 D8 B1 D9 81 20 D8 AC D8 A7 D8 A6 DB 92 D8 8C 20 D8 AC D8 B3 D9 B9 D8 B3 20 D8 A2 D8 A6 DB 92 0A and for your quick reference I have segmented the sms data as following ?91 1001 0001 = ? len 31 = 48 mix 01 0001 = SMS-SUBMIT mr 00 = 0 da len 0C = 12 da type 91 1001 0001 = International number + ISDN/telephone numbering plan da 293363810669 = +92333618 pid 00 = 0 dcs 08 1000 = UCS-2 ud len 24 = 36 ud D985D8B4D8B1D98120D8ACD8A7D8A6DB92D88C20D8ACD8B3D9B9D8B320D8A2D8A6DB920A as you can see that app_sms sending this message with dsc set to UCS-2 ud HEX string is also in correct format (I have tested it with third party web2sms service). but it can not be shown on my mobile corectly. so I am unable to determince why I am getting invlide charecters instead of a chines Message. Can you please help me Thanks ___ -- 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] A Leg Control on Asterisk Callback
Hi, On Mon, 2007-10-29 at 10:29 -0700, Douglas Garstang wrote: I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 6 Exten: callback Channel: SIP/[EMAIL PROTECTED] Variable: destination=SIP/[EMAIL PROTECTED] Callerid: 5551212 Context: default ActionID: 849120 Priority: 1 Asterisk first goes and dials the Channel parameter, SIP/[EMAIL PROTECTED] This is where it gets confusing. You have no control over what happens here. The actions don't even appear on the Asterisk console debug. It isn't until this party has picked up, and control jumps to the 'callback' extension, that Asterisk shows you what it is doing. So, I went and changed the Channel parmeter to Channel: Local/[EMAIL PROTECTED], and made a LegA context: [LegA] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,n,Playback(tt-monkeys) I wanted to have control over the call both before and after it is placed. I wanted to be able to play a prompt to the caller before the call is placed to the destination number. However, since we've dialled the A party already, we have no control over the dial plan anymore after they have answered, and I can't play prompts. What can I do here? You can use dial macro here like exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],M(a_leg)) and [macro_a_leg] exten = s,1,Playback(tt-monkeys) you can run most of asterisk dialplan commands in macro. as soon as your macro finished your call will be connected to Leg B you can read more at http://www.voip-info.org/wiki-Asterisk+cmd+Dial#Dialmacros and http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Macro Nasir Iqbal ICT Innovations http://www.ictinnovations.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] T.38 Faxing and Asterisk
Hi, Have you tried Callweaver http://www.callweaver.org ___ --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 realtime error
Hi Renzzo, Which linux distribution you are using? maybe problem is due to mysql.sock try with direct tcp connection. 1. remove dbsock = /var/lib/mysql/mysql.sock line 2. change mysql user host permission from localhost % 3. dump mysql permissions or restart mysql; 4. reload / restart asterisk. Regards Nasir Iqbal http://www.ictinnovations.com On Wed, 2007-09-26 at 23:25 -0500, RENZZO SOTOMAYOR wrote: Peder, I have all the permissions in mysql user. I can query my database from the local box. Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/ Asterisk and Mysql are in the same PC I still have the same error and don't know what to do. help plz! thanks in advance, Renzzo Mik Cheez wrote: Is your mysql.sock actually in /var/lib/mysql/ ? Peder wrote: Could be a mysql permission issue. Try this from the local box: mysql -u root -p enter asterisk as the password use asterisk; select * from sip_buddies; select * from iax_buddies; If you get that far and can see the entries in iax_buddies and sip_buddies, you know it isn't a permissions issue. If you can't, then you know where to look. RENZZO SOTOMAYOR wrote: Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of how to of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host 127.0.0.1 http://127.0.0.1/ failed MD5 authentication for '101' (9a43a82001dfa49d84e8facb765f7 d e2 != 31610d29241e861816b83998501ee223) I configure extconfig.conf as: [settings] iaxusers = mysql,asterisk,iax_buddies iaxpeers = mysql,asterisk,iax_buddies sipusers = mysql,asterisk,sip_buddies sippeers = mysql,asterisk,sip_buddies res_mysql.conf as: [general] dbhost = localhost dbname = asterisk dbuser = root dbpass = asterisk dbport = 3306 dbsock = /var/lib/mysql/mysql.sock My table as: CREATE TABLE iax_buddies ( name varchar(30) primary key NOT NULL, username varchar(30), type varchar(6) NOT NULL, secret varchar(50), callerid varchar(100), context varchar(100), host varchar(31) NOT NULL default 'dynamic', disallow varchar(100), allow varchar(100) ); I'm running asterisk on Fedora 6. Plz help thanks in advance Renzzo ___ 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 ___ 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 ___ 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
Re: [asterisk-users] AGI/PHP: missing arguments
Hi Michael,
On Sun, 2007-09-16 at 15:10 +0200, Michael Kamleitner wrote:
thx very much Nasir Philipp, I'm gonna try this tomorrow when I'm
back at the server...
however, I wonder if this behavior has changed recently, as I swear
[ ;) ] that this script has been working before...
yes you are right.
but this is only true if your are using FastAGI. its not available in
regular AGI
please visit http://www.voip-info.org/wiki-Asterisk+FastAGI for more
info.
Regards
Nasir Iqbal
http://www.ictinnovations.com
regards,
michael
On 9/15/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Michael Kamleitner wrote:
I've built a simple PHP-script utilizing the AGI-interface.
in
extensions.conf I trigger the script and pass a single value
as first
argument:
exten = h,1,DeadAGI(process.php|${Enter})
On the Asterisk-console, I can actually see that the script
is called
correctly (something like DeadAGI(process.php|1234)).
However, when I read
stdin in the PHP script, I receive all AGI-environment
variables
(agi_request, agi_callerid etc.) correctly, but I'm missing
the actual
passed value (which should be in agi_arg_1 etc.). the last
thing I get from
stdin is the environment-variable agi_accountcode, after
this it seems to
stop.
You don't append the argument to STDIN (which is fine).
In the PHP script check the $argv array. The first argument
(after the name of the script itself) should be in $argv[1] .
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied -
http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
My pick of the month: rfc 2822 3.6.5
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
___
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
--
Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
https://www.xing.com/profile/Michael_Kamleitner
-
+43 699 116 07 923
-
http://www.kamleitner.com
___
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
___
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
Re: [asterisk-users] AGI/PHP: missing arguments
Hi Michael,
Actually parameter passed to AGI script are not Channel Variables and
they passed to PHP/AGI directly so you cannot access them using STDIN.
to access passed parameters simply use global variable argv like.
global $argv;
//Getting input data (Parameter Passed to Script)
$callerID = $argv[1];
Regards
Nasir Iqbal
ICT Innovations
http://ictinnovations.com
On Sat, 2007-09-15 at 18:21 +0200, Michael Kamleitner wrote:
hi folks,
I've built a simple PHP-script utilizing the AGI-interface. in
extensions.conf I trigger the script and pass a single value as first
argument:
exten = h,1,DeadAGI(process.php|${Enter})
On the Asterisk-console, I can actually see that the script is called
correctly (something like DeadAGI(process.php|1234)). However, when
I read stdin in the PHP script, I receive all AGI-environment
variables (agi_request, agi_callerid etc.) correctly, but I'm missing
the actual passed value (which should be in agi_arg_1 etc.). the last
thing I get from stdin is the environment-variable agi_accountcode,
after this it seems to stop.
what's really strange is, that the exact same script has been working
correctly on a different machine...
any suggestions highly appreciated, thx!
regards,
michael
___
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
___
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
Re: [asterisk-users] Astribank and caller ID from PSTN
Hi, uncomment immediate=no Regards Nasir Iqbal ICT Innovations http://ictinnovations.com On Sat, 2007-09-15 at 13:18 -0500, Guillermo Salas M. wrote: Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell ;cidstart=ring hidecallerid=no callwaiting=yes ;usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes ;callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes echotraining=800 relaxdtmf=yes rxgain=3.0 txgain=3.0 callgroup=1 pickupgroup=1 ;immediate=no callerid=asreceived ;amaflags=default busydetect=yes busycount=8 ;busypattern=500,500 answeronpolarityswitch=no hanguponpolarityswitch=no faxdetect=both ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: FXO ;;; line=1 XPP_FXO/0/0/0 FXSKS signalling=fxs_ks callerid=asreceived group=1 context=from-zaptel channel = 1 When replacing callerid=phone-number I get on my ipphone phone-number as callerid: ; Span 1: XBUS-00/XPD-00 Xorcom XPD #0/0: FXO ;;; line=1 XPP_FXO/0/0/0 FXSKS signalling=fxs_ks callerid=2627839 group=1 context=from-zaptel channel = 1 Regards, ___ 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] SpanDSP/TxFAX FAX Status
Hi List, I wonder that how I can check that FAX is delivered successfully or not, in my dialplan while using TxFAX. Which variable contain fax status? as PHASEESTATUS and PHASEESTRING in Callweaver. Regards Nasir Iqbal ___ --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] FW: Can you reload only one conf file?
Hi Mike, Consider ARA www.voip-info.org/wiki/index.php?page=Asterisk+RealTime www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions or you can use dialplan add extension cli command from Asterisk Manager Interface. see http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action +Command Regards Nasir Iqbal ICT Innovations On Fri, 2007-08-10 at 12:19 -0400, Mike wrote: Well, if you really must know (this is OT for everybody else I guess) I have a custom Web GUI used for my customers, and when some settings are modified, a conf file is created. This conf file must be reloaded at this point, therefore I call the reload command externally. Why do I do this? Because the %*$%/$ hint fonctionnality can't accommodate variables fetched from a DB like the rest of my dialplan. Mike ___ --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] Prblem with Page Hight While Faxing over uLaw
Hi List, Me setup for faxing is Asterisk (TxFAX) = ATA = FAX Machine And SIP setting is Codec uLaw dtmfmode inband but I am facing a problem when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX Machine Print two pages (Enlarging the page) but shows it received one page. Please help me Thanks ___ --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] Prblem with Page Hight While Faxing over uLaw
Hi, can anybody help me?. Hi List, Thanks for all replies. IAX Modem + HylaFAX T.38 Modem + HylaFAX T.38 (using Callweaver) all is ok but please help me that what is wrong with my setting? I think there is speed difference between Asterisk and FAX Machine due to improper Negotiation (Hand Shaking). But how I can solve it? I am currently using Asterisk 1.2 Thanks On Tue, 2007-08-07 at 18:24 +0500, Nasir Iqbal wrote: Hi List, Me setup for faxing is Asterisk (TxFAX) = ATA = FAX Machine And SIP setting is Codec uLaw dtmfmode inband but I am facing a problem when I send a FAX of one page from Asterisk to ATA+FAX Machine the FAX Machine Print two pages (Enlarging the page) but shows it received one page. Please help me Thanks ___ --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] PhpAgi call generation
Hi Nitesh,
you are missing Extension
try with
$call = $asm-send_request('Originate',
array('Channel'=SIP/xo-out/$supervisor_num,
'Context'='default',
'Exten'= your_extensions_here,
'Priority'=1,
'Callerid'=$cid));
or you must put an s extensions in your desired context in this case
it is default.
Regards
Nasir Iqbal
On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote:
Hello All,
Can anyone help me with this... This is what my program does: -
1) At certain time the system generates a .call and make a call to User A.
2) When User A picks up the phone call, system will play a menu select
option.
a) Press 1 to call your supervisor.
b) Press 2 to call your manager.
c) Press 3 to leave a voice message.
3) When the User A press 1 to call his supervisor... The system has to
put the User A on hold and place a call to the supervisor.
4) Once the supervisor picks up the call, User A has to be in session
with his supervisor.
Now I have already got part 1 and 2 done... but I am stuck with part 3
and 4.
This is how I generate my call to the supervisor: -
===
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=SIP/xo-out/$supervisor_num,
'Context'='default',
'Priority'=1,
'Callerid'=$cid));
$asm-disconnect();
}
One the *CLI I do see the call, but its failing: -
AGI Rx STREAM FILE
/var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 0
AGI Tx 200 result=0 endpos=26224
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
Channel SIP/xo-out-08f8ae10 was answered.
== Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back
to exten 's'
== Manager 'phpagi' logged off from 127.0.0.1
AGI Rx STREAM FILE goodbye 0
Can anyone put some light what I am missing here... Why the call is
dropped on both end...?
Cheers,
Nitesh
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PhpAgi call generation
Oh,
you need Dial application instead of origination.
so no need to AGI Script simply add
the dialplan called for .call should look like this
exten = yourexten,1,BackGround(your_menu_ivr)
exten = yourexten,n,WaitExten()
exten = 1,1,Dial(SIP/xo-out/$supervisor_num) ;for Supervisor
exten = 2,1,Dial(SIP/xo-out/$manager_num) ;for Manager
exten = 3,1,Voicemail(your_voice_mail_box)
Regards
Nasir Iqbal
On Tue, 2007-07-31 at 12:21 -0400, Nitesh Divecha wrote:
Thanks Nasir,
By putting 'Exten'= your_extensions_here it will create a new channel
to that extension, correct?
What I want to do is to join two channels... Join the User A channel
which is active with supervisor.
Cheers,
Nitesh
Nasir Iqbal wrote:
Hi Nitesh,
you are missing Extension
try with
$call = $asm-send_request('Originate',
array('Channel'=SIP/xo-out/$supervisor_num,
'Context'='default',
'Exten'= your_extensions_here,
'Priority'=1,
'Callerid'=$cid));
or you must put an s extensions in your desired context in this case
it is default.
Regards
Nasir Iqbal
On Tue, 2007-07-31 at 10:08 -0400, Nitesh Divecha wrote:
Hello All,
Can anyone help me with this... This is what my program does: -
1) At certain time the system generates a .call and make a call to User
A.
2) When User A picks up the phone call, system will play a menu select
option.
a) Press 1 to call your supervisor.
b) Press 2 to call your manager.
c) Press 3 to leave a voice message.
3) When the User A press 1 to call his supervisor... The system has to
put the User A on hold and place a call to the supervisor.
4) Once the supervisor picks up the call, User A has to be in session
with his supervisor.
Now I have already got part 1 and 2 done... but I am stuck with part 3
and 4.
This is how I generate my call to the supervisor: -
===
if($asm-connect())
{
$call = $asm-send_request('Originate',
array('Channel'=SIP/xo-out/$supervisor_num,
'Context'='default',
'Priority'=1,
'Callerid'=$cid));
$asm-disconnect();
}
One the *CLI I do see the call, but its failing: -
AGI Rx STREAM FILE
/var/spool/asterisk//tmp//text2wav_e08db16aede0af38ebb90a1c69ee19e3 0
AGI Tx 200 result=0 endpos=26224
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
Channel SIP/xo-out-08f8ae10 was answered.
== Starting SIP/xo-out-08f8ae10 at default,,1 failed so falling back
to exten 's'
== Manager 'phpagi' logged off from 127.0.0.1
AGI Rx STREAM FILE goodbye 0
Can anyone put some light what I am missing here... Why the call is
dropped on both end...?
Cheers,
Nitesh
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--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 Supported Harware Architecture
Hi Saqib, Architecture is depend on what service you want to deliver. Voip is more cheaper then pstn for interoffice connectivity. But consider regulatory issue before using it. visit http://www.voip-info.org/wiki-Asterisk for complete detail. Regards Nasir iqbal On Wed, 2007-07-25 at 22:48 +0500, saqib butt wrote: HI Kindly can anyone plz tell me what will be the broadband architecture for Asterisk, e.g; for a multinational company having offices in different far location. What will the best solution or architecture to setup to go over external PSTN lines accross many locations. Is ISDN is ok or it may need DSL brodband service. kindly guide me about it as i dont know much about establishing asterisk harware/network infrastructure, can u plz forward me to any website for this. THANX ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Default Asterisk Numbers
also have a look on http://www.voip-info.org/wiki/view/Asterisk+standard+extensions On Thu, 2007-07-26 at 20:57 -0600, Al lists wrote: features.conf On 7/26/07, GNUbie [EMAIL PROTECTED] wrote: Hello all, Where can I find the complete list of default Asterisk (telephone) numbers and maybe the other special numbers that are need to be preserve and not use for setting up own dial plan? Thank you. GNUbie ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] MySQL components in asterisk-addons not being built
Hi, please see your ./configure output especially few last lines. and note missing thins. Regards Nasir Iqbal On Tue, 2007-07-24 at 10:45 -0400, hugolivude wrote: Thanks Tahir. I already got the asterisk-addons though - that's what I'm having trouble with! BTW - asterisk-addons also provides a menuselect now. The problem is that the MySQL components all show up XXX even though I have MySQL installed. Hugh On 7/24/07, Tahir Almas [EMAIL PROTECTED] wrote: Hi Hugh, MySQL CDR is not included in default asterisk distribution. so there is no entry for MySQL CDR in make menuselect. you must install additional Addons after asterisk installation. you can download from http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.4.2.tar.gz 1. Extract them 2. Make 3. Make Install 4. cdr_addon_mysql.so will be installed including all other modules. Regards Nasir Iqbal On Tue, 2007-07-24 at 08:17 -0400, hugolivude wrote: I'm trying to add MySQL CDR recording in Asterisk 1.4.6. I'm following the instructions posted here: http://www.voip-info.org/wiki-Asterisk+cdr+mysql I have MySQL installed and it works fine - starts on stratup, I can create DBs, tables and so on and I can connect through php. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 However I still get XXX for all of the MySQL add ons when I do: make menuselect Any pointers for me on how to troubleshoot and fix this problem? Thanks, Hugh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM04B FIOS No Hangups Often
Hi Mike, I think that callprogress=yes is right. but Noah Miller also right. so the solution will be. callprogress=yes busydetect=yes busycount=4 ; suitable values is above then 4 choose minimum ; as More value more time to wait before hangup. rxgain=7; you can adjust this value by error trial ; method check with different values. and ; choose what works best for you. ; note: if you face invalid hangup try to reduce rxgain ; or If you face no-hangup problem try to increase rxgain I hope it will work for you. Regards Nasir Iqbal On Tue, 2007-07-24 at 16:03 -0500, Eric ManxPower Wieling wrote: Noah Miller wrote: 2) Set callprogress=yes in zapata.conf (if you haven't already done that). If you set callprogress=yes you will have the opposite problem -- active calls will be randomly disconnected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answering Machine Beep Detection for *
Hi dave, you can use AMD application for more info please visit www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD Regards Nasir Iqbal On Tue, 2007-07-24 at 17:22 -0400, dave cantera wrote: hi, can anyone point me to answering machine beep detection methods or writeups for *? thanks, daveC ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pass Dialed number to a script
Hi,
add new line in
[serviceinterruption]
exten = s,1,Set(TIMEOUT(response)=10)
exten = s,2,Set(dialedno=${EXTEN}) //Add This Line
and change
[callback]
exten = s,1,Playback(outboundmsgs/customerrepwillcall)
exten = s,n,System(${SCRIPTS_DIR}/rep_callback.sh ${dialedno}) //This Line
Changed
I think that you know how to get arg from shell script.
cheers
Nasir Iqbal
ICT Innovations
On Thu, 2007-07-19 at 08:43 -0400, [EMAIL PROTECTED] wrote:
I'm in the process of writing a simple autodialer to dial a list of numbers
and play a message. One of the options I want to give them is a way to
dial X to have a customer service representative call you
Looking for a simple way to pass the number that I dialed to a script in
extensions.conf... something like this:
[serviceinterruption]
exten = s,1,Set(TIMEOUT(response)=10)
exten = s,2,Answer
exten = s,3,Playback(outboundmsgs/serviceinterrupt)
exten = s,4,Playback(outboundmsgs/choice)
exten = s,5,wait(3)
exten = 1,1,Goto(s,3) ; replay message
exten = 2,1,Goto(msgack,s,1); acknowledge message
exten = 3,1,Goto(callback,s,1)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup
[callback]
exten = s,1,Playback(outboundmsgs/customerrepwillcall)
-- exten = s,n,system(${SCRIPTS_DIR}/rep_callback.sh ${} )
exten = s,n,Playback(vm-goodbye)
exten = s,n,Hangup
how do I pass the number that was dialed (from the creation of a .call file)
to the rep_callback.sh script ?
thanks
Shawn
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] different codec for different extensions
Hi Mojo, I dont have control our calling party. and also called extension is only configured in extensions.conf not sip.conf etc. So I must select the codec within my dialplan (extensions.com) I found one solution by using SIP_CODEC variable like [fax] exten = 605,1,ringing() exten = 605,n,set(SIP_CODEC=ulaw) exten = 605,n,RxFAX(/tmp/nasir.tiff|ecm) exten = 605,n,hangup() but Thanks for your answer Thanks Nasir Iqbal [userX] ... context=internal disallow=all allow=gsm allow=ulaw ... [fax] ... disallow=all allow=ulaw ... Then any IVRs that userX accesses should be in gsm because it's the preferred codec? Assuming that the gsm sound files ARE installed? You might experiment with this. But when userX is bridged to the fax channel, ulaw is the only one the fax channel allows, so it's chosen on both ends. Shouldn't this work? Mojo Nasir Iqbal wrote: Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] different codec for different extensions
Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal ___ --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] Ex-Girlfriend Logic in 1.4.4
Hi,
exten = 5000/19256002182,1,Answer
exten = 5000/19256002182,n,Wait(1)
exten = 5000/19256002182,n,NoOp(${CALLERID(num)})
exten = 5000/19256002182,n,Playback(tt-monkeys)
nothing appears on the console and I get no match. You can see the ca
Try with underscore before extension like.
exten = _5000/19256002182,1,Answer
Nasir Iqbal
ICT Innovations
___
--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] Dynamically adding Context in dialplan?
Hi, How about using asterisk real time ? http://www.voip-info.org/wiki-Asterisk+RealTime We can write a switch command for existing context but I want to new context dynamically. From asterisk CLI we can add extensions in dial-plan dynamically using dialplan add extension command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal ___ --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] background dialing
Hi, Is it possible to dial in background 2 or more different numbers while the same uninterrupted soundfile is playing? Try to use asterisk queues. with queue you can play music on hold etc/ IVR to caller while trying to connect it to the available agent. you can use your target number(s with ring group) as agent. Regards Nasir Iqbal ICT Innovations ___ --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] Dynamically adding Context in dialplan?
Hi everybody, From asterisk CLI we can add extensions in dial-plan dynamically using dialplan add extension command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal ___ --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] Auto Dial Problem
Hi,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
No problem with Auto Call
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try callprogress with yes in zapata.conf
It may cause another problem, after remote party has picked up the
call and asterisk still does not know it. and in ringing status.
if your dial plan work fine now, then no need to change rxgain.
otherwise.
Just Increase your rxgain value. try with different values and choose
best one.
if rxgain greater then desired value ?? you my receive invalid report
that remote party has picked up.
if rxgain less then desired value ?? you my receive invalid ringing
report after call is answered.
so adjust it according your requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dial Problem
Hi,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
No problem with Auto Call
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try callprogress with yes in zapata.conf
It may cause another problem, after remote party has picked up the
call and asterisk still does not know it. and in ringing status.
if your dial plan work fine now, then no need to change rxgain.
otherwise.
Just Increase your rxgain value. try with different values and choose
best one.
if rxgain greater then desired value ?? you my receive invalid report
that remote party has picked up.
if rxgain less then desired value ?? you my receive invalid ringing
report after call is answered.
so adjust it according your requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
___
--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] Dynamically adding Context in dialplan?
Hi everybody, From asterisk CLI we can add extensions in dial-plan dynamically using dialplan add extension command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal ___ --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] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --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] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --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] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --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] how to use sable (festival) markup with asterisk
Hi, I want to use festival with asterisk to play a text with sable tags. have some body any idea about it Nasir Iqbal ___ --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
