similar to: Help Dial extention

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

Does anybody know why am getting a lot of static and sometimes dropped calls from my asterisk server. Vitelity is my number provider if it matters. Thank you Jarga Jallow -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071105/3ea1de51/attachment.htm -------------- next part -------------- A

Under asterisk info: Sip registry 12/12 76.xxx.xxx.xxx D N 5066 UNREACHABLE 11/11 76.xxx.xxx.xxx D N 5064 UNREACHABLE 10/10 76.xxx.xxx.xxx D N 5062 UNREACHABLE All these IP phones are behind NAT. What could be the problem? Thanks in advance. Jarga -------------- next part -------------- An HTML

I need help with my grand stream GXP2000 phones they keep freezing randomly. Any ideas? Jarga -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071101/0f180a43/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 3781

I am getting this error under system info: File Line Command Message common_functions.php 314 file_exists(/proc/scsi/scsi) the file does not exist on your machine Does anybody know how to fix this? Thank you in advance Jarga -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi, I have a trivial peace of dialplan for exten 100. I try to change it to _1XX and the asterisk act according to a different (Default??) dial plan and not the one I want? Is that possible? Where is the other dialplan sits? In my extention.conf I can't see something that look like what asterisk is dialing. How can I trace\debug my dialplan? Thanks, Eyal -------------- next part

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

I have created a rxml template and I am sucessfully generating valid xml. However, the file does not have an extention on it. This is creating a problem for one of the applications running on my computer, when referencing the file off the server it requires the .xml file extention. Despite being vaild xml it needs that extention. I am lost at how to solve this problem. I am still working in a

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040420/952ce2d4/attachment.htm

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Apparently the Scheirer-Ray-Hare extention for the Kruskal-Wallis test is too weak for estimating interaction effects in more than twoway design matrices isn't it? If so, why is it? TIA [[alternative HTML version deleted]]

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give. I use realtime asterisk. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, has someone a clear complete and systematic overview over what libs, compiler versions, etc. are really necessary to enable fax receiving and forwarding via e-mail with asterisk and spandsp? As a "non-kernel-hacker" I'm running a bit confused and find it all a bit unclear and not too systematic?! %) It is remarkable how many and various errors can be produced by different

Hello, trying to implement srtp with already working tls i somehow stuck with srtp. If the extension has successfully registered a call from asterisk to that extension works fine. But the other way round nothing happens. [Aug 13 14:54:16] WARNING[31053]: chan_sip.c:3906 __sip_xmit: sip_xmit of 0x7fc8880467e0 (len 609) to 123.456.789:36785 returned -2: Success [Aug 13 14:54:20] NOTICE[31053]:

At 04:44 PM 6/21/2003 -0400, you wrote: >I think we definetly need to get YP authentication into Icecast2 soon, >but I'd like to propose an extension to the draft now... can you be more specific on what you mean by YP authentication ? >with action=add: > banner=URL > >This allows the Icecast2 server to supply a URL for a graphical banner >to represent the radio stream