[asterisk-users] Video Softphone
Hi, Any video softphone that will send the video codec in the first INVITE, in eyebem and some other phones like Ekiga first we are getting audio and then there is a button SEND VIDEO, if we click that the re-invite is going with video codec, whereas i need to send the video at first invite itself, is there any softphone for that. I tested with Android phone with a application called IMS droid with Asterisk where in the first INVITE I am able to send the video. Can any one suggest any softphone which can send video codec in first INVITE. 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
[asterisk-users] Video call Setup in Asterisk 1.4.17
Hi, I am planning to setup a video call with Asterisk 1.4.17 and eyebeam softphone, can any one suggest some link or configuration to setup the things. 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
Re: [asterisk-users] mISDN Vs Dahdi
Hi Tamer, Many thanks for your comments, really your comments are useful. And finally I think using dahdi instead of mISDN is better. On Wed, Sep 21, 2011 at 3:10 AM, Tamer Higazi th9...@googlemail.com wrote: Am 20.09.2011 19:47, schrieb Gopal krishnan: What is the difference between using mISDN for BRI and using Dahdi mISDN was at 1st done for ISDN Services and channel driver as I know. It supported like call routing (switch based, not your side on the pbx level). without mISDN? you can use DAHDI without ISDN, for other related telephony interfaces. Like analogue cards. and if there are other telephony hardware interfaces that has nothing common todo you can use it too. and genereal: DAHDI is the native digium hardware telephony interface. Beside mISDN you have the native support for an echo cancellor you could use. for example: oslec or the hpec (high performance echo cancellor) which you can't make use of it natively with mISDN. As long you have no hardware interface boards as PCI modules, and you connect only throug the network to your enddevices, there is no need to startup dahdi at all. end beside: dahdi is an extra service that starts up, mISDN is a channel driver you must activate in the modules.conf. Are all your question answered that far?! Regards Tamer -- _ -- 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] mISDN Vs Dahdi
Ok Thank you Tamer. On Wed, Sep 21, 2011 at 4:32 PM, Tamer Higazi th9...@googlemail.com wrote: Be sure, if you make us of HFC Boards that you have the zapfhfc patches. There is some work for you to accomplish, like patching dahdi to make use with the cheap isdn boards. For office using ISDN Devices it's fairly enough. If you want to make use of a server, I advise you to take the digium or sangoma boards, because of the native support for asterisk. If you don't love patching and searching to get what in the internet, take Gentoo Linux. All included (don't use dahdi 1.5.x, still not working on gentoo). Tamer Am 21.09.2011 12:43, schrieb Gopal krishnan: Hi Tamer, Many thanks for your comments, really your comments are useful. And finally I think using dahdi instead of mISDN is better. On Wed, Sep 21, 2011 at 3:10 AM, Tamer Higazi th9...@googlemail.com mailto:th9...@googlemail.com wrote: Am 20.09.2011 19:47, schrieb Gopal krishnan: What is the difference between using mISDN for BRI and using Dahdi mISDN was at 1st done for ISDN Services and channel driver as I know. It supported like call routing (switch based, not your side on the pbx level). without mISDN? you can use DAHDI without ISDN, for other related telephony interfaces. Like analogue cards. and if there are other telephony hardware interfaces that has nothing common todo you can use it too. and genereal: DAHDI is the native digium hardware telephony interface. Beside mISDN you have the native support for an echo cancellor you could use. for example: oslec or the hpec (high performance echo cancellor) which you can't make use of it natively with mISDN. As long you have no hardware interface boards as PCI modules, and you connect only throug the network to your enddevices, there is no need to startup dahdi at all. end beside: dahdi is an extra service that starts up, mISDN is a channel driver you must activate in the modules.conf. Are all your question answered that far?! Regards Tamer -- _ -- 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 -- _ -- 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] mISDN Vs Dahdi
What is the difference between using mISDN for BRI and using Dahdi without mISDN? 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
[asterisk-users] Asterisk is keep on sending Register request
Hi, *Scenario 1* I am trying to register a VoIP trunk, which is keep on sending the register request and I am not getting any response from the SIP Server, this I am trying from one network. *Scenario 2* From another network if I try the same VoIP trunk, the account got registered. One thing here to notice is the same account has already been worked in *Scenario 1* and now which is not working without any reason. Any comments would be much appreciated. 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] Asterisk Integration with Android device
Thanks for all your comments. Actually I have 3G connection but even then the signal in my mobile automatically changes from 3g to 2G; it is automatically going to Edge signal. Anyways let me try with some other softphone like media5. Regards, Gopal On Thu, Aug 25, 2011 at 9:21 AM, James Perkins ja...@clove.net.au wrote: ** Try media5 fone. I couldn't get 3cx to work on my iphone and tried about 7 different softfones. Media5 is the best by a long shot. Android is still in better and haven't tried it but if its anything like their iphone app it will be worth a look. There is a signup for the better at the website. let us know how you go. James - Original Message - *From:* bakko asannu...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 11:48 PM *Subject:* Re: [asterisk-users] Asterisk Integration with Android device I think don't work with 2G network. Regards - Original Message - *From:* Gopal krishnan gopalakrishnan...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Wednesday, August 24, 2011 4:01 PM *Subject:* [asterisk-users] Asterisk Integration with Android device Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- -- _ -- 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
[asterisk-users] Asterisk Integration with Android device
Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be appreciated. Regards, Gopal -- _ -- 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] Custom Dialplan
Use extensions_custom.conf file to update your custom configurations. On Sun, Aug 7, 2011 at 3:59 AM, Barry L. Kline blkl...@attglobal.netwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 08/05/2011 04:32 AM, Richard Zulu wrote: I would like to import my dialplan into freepbx+asterisk since I am switching to that...how can I create my own custom dialplan in freepbx? I'm not sure why you'd want to... freepbx is anathema to custom dialplans. That said, I believe you end up naming your extensions.conf file to extensions_additional.conf and freepbx will pick it up when it starts. It's been a long, long time since I've dealt with freepbx -- in fact I went the other way: from freepbx+asterisk to pure asterisk. When I was using freepbx that was the solution you seek. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFOPcAxCFu3bIiwtTARAkjKAKCPCgcoaRyPNs7BXhge7xxcy7C2qQCdF6hx 2Bwz/YEUSbKFsfzD9V0xX6Q= =W2Dn -END PGP SIGNATURE- -- _ -- 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] Reconfiguring smoother module in 1.8
Hi all, In Asterisk 1.6 version there was function to re-configure the smoother. This function we were able to call from an external loadable module. How to do the same thing in Asterisk 1.8 to re-configure the smoother and be able to call from an external loadable module. Regards, Gopal -- _ -- 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] Integration of OpenVXI
Hi, Have anybody integrated OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk? Thanks, Gopal -- _ -- 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] RTP Streaming
Hi Users, I would like to know about the RTP audio streaming. I am taking the example as youtube, in youtube if bandwidth is less the application will buffer and will stream the video; likewise how to do with audio buffering and play the file using RTP in asterisk. Any guide of clue will make me understand. Thank you in advance. Thank you, Gopal -- _ -- 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 and Audiocodes PRI card
This card is a standalone SIP media server on a PCI blade. But you can make it work with Asterisk for that you have to tweak Asterisk source and as well as you have to buy API from audiocodes if I am not wrong. Instead of this why can't you use Sangoma or Digium cards? On Wed, Jun 8, 2011 at 11:39 PM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello list, can anyone tell me if this card : http://www.audiocodes.com/product/ipm-260-sip is compatible with Asterisk (DAHDI) for use as PCI PRI card ? Kind regards, Jonas. -- _ -- 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] benefits of asterisk 1.8
1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term support. On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote: can someone explain to me the benefits of upgrading to version 1.8? we are currently running 1.6 I know one benefit of 1.8 is digium supports it also, how stable is version 1.8 compared to 1.6? Thank you for you input. -- _ -- 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] To know if the ISDN PRI E1 is UP?
your PRI is not up. You can see this Status: In Alarm, Down, Active it means you have some error. Some parameter is not correct with the configuration and with your line. If you are using Sangoma card then please check this link http://wiki.sangoma.com/wanpipe-linux-asterisk-debugging On Tue, May 31, 2011 at 6:02 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; This is the output of the pri show status, so I appreciate if to know if that means the E1 is UP? What does it means that the status us (Status: In Alarm, Down, Active)? What in the below result give an indication that it is UP? CC*CLI pri show span 1 Primary D-channel: 16 Status: In Alarm, Down, Active Switchtype: EuroISDN Type: Network Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No 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] AMI extenstion state
Hi,
I would like to get the status of asterisk extension with my php program.
*My program as follows,*
html
!--meta http-equiv=refresh content=1 /--
?php
$fp = fsockopen(xxx.xxx.xxx.xxx, 5038, $errno, $errstr, 30);
if (!$fp)
{
echo $errstr ($errno)br /\n;
}
else
{
$out = Action: Login\r\n;
$out .= UserName: admin\r\n;
$out .= Secret: amp111\r\n\r\n;
fwrite($fp, $out);
$in = Action: ExtensionState\r\n;
$in .= Exten: 777\r\n\r\n;
$in .= Context: ext-did-custom\r\n\r\n;
$in .= ActionID: 1\r\n;
//$in.= Status: State\r\n;
$in .= Action: Logoff\r\n\r\n\r\n;
fwrite($fp,$in);
while (!feof($fp))
{
echo fgets($fp, 256);
//echo $fp;
}
echo \r\n\r\n\r\n\r\n;
fclose($fp);
}
?
/html
*When i try to run this program i am getting the following output*
Asterisk Call Manager/1.0 Response: Success Message: Authentication accepted
Response: Success Message: Extension Status Exten: 777 Context: default
Hint: Status: -1 Response: Error Message: Missing action in request
Response: Goodbye ActionID: 1 Message: Thanks for all the fish.
*My dialplan as follows*
[outbound-allroutes-custom]
exten = 5101,1,Dial(SIP/5101)
exten = 5101,2,Hangup
exten = 5102,1,Dial(SIP/5102)
exten = 5102,2,Hangup
exten = 777,1,Wait(2)
exten = 777,hint,SIP/5101SIP/5102
*Scenario*
The exten that I have used in the program is 777 that is my hint
extension. Instead of the 777 if i try 5101 also I am getting the same -1
status.
Any help would be appreciated.
--
Thank you with regards,
Gopal,
PeopleTech Systems Private Limited
www.peopletech.co.in
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Re: [asterisk-users] Clustering Meetme over multiple boxes?
Hi Matt, As you said, is this will work like this? 1. Student A will login in a conference room no 7789 2. Student B will login in a conference room no 7789 3. Student C will login in a conference room no 7789 4. Instructor for student A,B and C will login in a conference room no. 6689 5. When the instructor click a button the 7789 conference and 6689 conference will be merged in a listen mode Am I correct? If I am wrong please correct me. Thanks On Tue, Mar 4, 2008 at 11:22 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, I have actually done this both ways, with many small conferences and few large conferences. The best example of both is the voice_lab feature that is included with VICIDIAL(although not very well documented). What this feature does is it has students log into individual meetme rooms and then have an instructor dial into their own meetme room. When the students are all logged in and the instructor is ready the instructor clicks a button to initiate calls from all of the student meetme rooms to the instructor meetme room where they are in listen-only mode and the instructor speaks english phrases which the students then repeat in their own conference. The reason this is set up this way is to allow for supervisor monitoring of individual students as well as recording of each student individually as they hear and repeat the phrases. This application is in use in telemarketing schools in the Philippines to help students learn to better speak American English. I have tested this to 120 channels going into the instructor meetme room across 6 servers. Hope that helps, MATT--- On 3/4/08, Tony Mountifield [EMAIL PROTECTED] wrote: Hi Matt, thanks for your reply. In article [EMAIL PROTECTED] , Matt Florell [EMAIL PROTECTED] wrote: Hello, We have done this using IAX trunks between Asterisk servers to connect a PRI line on server A with a meetme room on server B. We have had hundreds of participants in meetme rooms across a dozen Asterisk servers using this method. Not knowing your setup I'm not sure if this would work easily for you, but this is a somewhat-easy, scalable method for expanding meetme capacity. Is it correct to understand that in your setup, a given conference only ever exists on a single server (presumably the one used by the first caller), and that calls arriving on a different server are proxied individually to whichever server is hosting the requested conference? I can see that this would be quite easy, and would work well for lots of smallish conferences, but might be a bit heavy for a system running a small number of huge conferences. In the latter case, I want to look at having a local conference on each box and bridging the conferences together. This is the bit that gets complicated for making conf-wide decisions :-) Cheers Tony MATT--- On 3/4/08, Tony Mountifield [EMAIL PROTECTED] wrote: Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should all be conferenced together, by somehow linking matching Meetme conferences on both/all boxes. Particular complications I can envisage are the handling of marked users (A, w and x options), call recording (r option), and MeetmeAdmin operations such as mute all and unmute all. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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 ___ -- 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 -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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 ___ -- Bandwidth and Colocation
Re: [asterisk-users] Help: dtmf mode
Hi Jarga, What type of connection you are using is it VoIP or ISDN PRI, if it is VoIP check your dtmfmode in sip.conf if it is PRI check zapata.conf On Jan 25, 2008 12:13 AM, Jarga Jallow [EMAIL PROTECTED] wrote: Hi, I am having trouble making a selection when I call a number and need to make a selection to go to an extension with my polycom phones 301. Anybody have an idea how to fix this problem? Thanks in advance. Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.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 -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in image003.gifimage002.gifimage001.jpg___ -- 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] Finding difficulty in installing Asterisk
Hi Pandey, What type of OS you are using, is it redhat or fedora. and install with latest version. On Jan 25, 2008 9:55 AM, Lyle Giese [EMAIL PROTECTED] wrote: You need to do a 'make' before the 'make install'. Lyle [EMAIL PROTECTED] wrote: Hi all, Please help me in installing Asterisk. I am getting the following error when trying to install Libpri [EMAIL PROTECTED] Asterisk]$ cd libpri-1.4.2 [EMAIL PROTECTED] libpri-1.4.2]$ make clean rm -f *.o *.so *.lo *.so.1 *.so.1.0 rm -f testprilib libpri.a libpri.so.1.0 rm -f pritest pridump rm -f .depend [EMAIL PROTECTED] libpri-1.4.2]$ make install gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC -c -o copy_string.o copy_string.c ___ -- 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 -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- 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] Need sample configuration files for sipura/linksys ata
Hi, Try this http://www.kcip.com/support/pap2uk.html On Jan 25, 2008 4:18 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, i need sample xml configuration files for linksys pap2, linksys pap-2t, sipura 2100, sipura 2102, 1001, 3000 and 3102. All of these are linksys/sipura products. So if anyone has these sample files then plz share. -- Best Regards Rizwan Hisham Software Engineer Axvoice Inc. www.axvoice.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 -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] two zaptel card
Hi , No need of any special requierments. On Jan 24, 2008 12:52 PM, Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all I want to use two zaptel card(TE210p) in pc for asterisk. Is there any special requirement for this configuratin. any suggestion. thanks , Bhrugu Mehta ___ -- 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 -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- 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 optimalization
Hi, Dell is not a recomeded server for linux. Its only compatible with windows. On Jan 24, 2008 12:02 PM, Goke Aruna [EMAIL PROTECTED] wrote: ram wrote: http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm check this link may help you ram On Jan 23, 2008 10:23 PM, marek cervenka [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: hi, i'm testing asterisk 1.4/1.2 in the following scenario centos5/cpu quad xeon E5335 2.0Ghz - test clients behind nat - 1500+ testing instances - reregister option from 1min to 1hour - qualify set to 5000 top shows over 100% cpu. cpu cores sometimes go to 95% with htop i see ~16threads but only one child have ~95% cpu (how i can get info about that thread? what he is doing?) what is major bottleneck? qualify imho not. i'm tried set qualify=no, does not help SIP REGISTER packets? this problem persist if no calls are active after restart cpu usage slowly increase. after a ~hour is about 100% which optimalizations do you recommend for ~1500 peers scenario? (behind nat, reregistrations) --- Marek Cervenka === ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 http://www.transnexus.com/White%20Papers/asterisk_V1-4-11_performance.htm That result is suprising! but i have DELL 2950 with 2 X 3.0GHz CPU on 6GB ram, equiped with 8e1 link (2 sangoma A104D) running FC5. I installed chan_ss7-1.0 with asterisk-1.2.25 doing transcoding, and each time calls get to 120+ the cpu is fully utilized. the calls come from sip to the ss7 link. can someone advice me on what I can do to improve the performance. goksie NB. I felt we re talking on the same topic thats why i added my own experience. ___ -- 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 -- Thank you with regards, Gopal, PeopleTech Systems Private Limited www.peopletech.co.in ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id issue for INDIA
Hi, For the caller id there is a patch available for digium cards. you can patch that file. I am not aware about those files. so please refer some googleing. On Jan 18, 2008 2:57 PM, sandeep [EMAIL PROTECTED] wrote: hi all, how to set the caller id facility for the TDM400p card in INDIA. thanks sandeep.s ___ -- 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 -- Thank you with regards, Gopal, ___ -- 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] extension.conf with mysql
Hi,
I am trying to connect the outbound dialing with mysql with the following
code,
exten = 88,1,MYSQL(Connect connid hostname username password dbname)
exten = 88,2,GotoIf($[${connid} = ]?error,1)
exten = 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\
tablename\ WHERE\ phone =${a})
exten = 88,4,MYSQL(Fetch fetchid ${resultid} ph\ sa)
...
.
...
after this I am getting confused. My moto is to display the number in the
database and need to check with my outgoing number.
how to display fetched number from the database.
--
Thank you with regards,
Gopal
http://www.peopletech.co.in
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Re: [asterisk-users] detect asterisk pbx via sip
hi, What is the useragent that you have specified in the sip.conf? if you specified useragent=asterisk change that that to something like useragent=eyebeam or leave it empty. On Nov 11, 2007 4:08 PM, Giedrius Augys [EMAIL PROTECTED] wrote: Hello, My situation is that , I can't make calls with asterisk, but with x-lite works fine. Asterisk shows , that successfully registers with another SIP server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408 Request timeout. And as I said , with x-lite no problems. I heard that for comercial purposes, this SIP server detects asterisk , and ignores him. Or maybe it check is it server or device? Maybe somebody can give me some advices... Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you, with regards, Gopal, www.peopletech.co.in ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk direct dialing
Hi,
I am using Asterisk 1.2.24, I have written my dialplan to land
with an IVR with the same time if the customer knows the parties
extensions they can dial directly, but what happens is sometimes its
working and sometime its not working.
My extensions.conf as follows,
[incoming]
exten = 052477302,1,Wait(2)
exten = 052477302,2,NoOp(${CALLERIDNUM})
exten = 052477302,3,Goto(from-internal,s,1)
[from-internal]
exten = s,1,Answer()
exten = s,2,Background(welcome_pride5)
exten = 501,1,Dial(Zap/1)
exten = 502,1,Dial(Zap/2)
exten = 503,1,Dial(Zap/3)
exten = 504,1,Dial(Zap/4)
exten = 505,1,Dial(Zap/5)
I dont know what could be the reason. Is there any other way that i can use.
--
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Gopal,
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Re: [asterisk-users] sidetone
hi, try this with zapata.conf echocancel=yes On Nov 10, 2007 6:34 PM, Todd [EMAIL PROTECTED] wrote: Hi - I've got a new install with a Sangoma A200 and a few GXP2000's. When users are talking over the Sangoma, they get a lot of sidetone (local echo). Internal calls are fine. Where do I adjust that? I assume its in zapata.conf somewhere? thanks Todd ___ --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 -- Thank you, with regards, Gopal, www.peopletech.co.in ___ --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] 'Traditional' Faxing
Hi, For that you need to purchase sangoma FXS card and connect your fax machine in the fxs card. so that the fax machine will seems as an extension for asterisk. whatever fax comes in it we can write a rule where a fax nmber will reach the extension connected fax machine. On Nov 10, 2007 6:04 PM, Greg Cockburn [EMAIL PROTECTED] wrote: Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there maybe someway to bridge an Analogue FXS port to a channel on the E1? Basically I want to mimic what we have now. 1. Any person can send a fax using the fax machine, and the PBX picks the next free channel on the E1. 2. A fax call can come over any channel on the E1, and the dialed number is matched and sent to the analogue FXS port of the PBX to be received by the fax machine. Is there anyway I can do this in Asterisk that will work seamlessly? I have not yet purchased any hardware, so recommendations would be greatly appreciated. (I believe some of the problem exists due to timing, does any hardware; E1 card / Analogue card; support linking a timing signal together?) Sangoma, Digium, Pika? Thanks all for any help on this one. Greg. ___ --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 -- Thank you, with regards, Gopal, www.peopletech.co.in ___ --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] agi
hi asterisk' ians How to write agi scripts, how to see the output.. thanks in advance... Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soft phone code
hi,Is it possible to write a program for softphone... pls any idea about that Blab-away for as little as 1ยข/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice.___ --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
