similar to: Top Posting

Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm

Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/

Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.

I'm filtering that is being deliverd to postfix mail server with amavisd-new . I want spam with spam f level 1 - 8 to ad a tag any everything above to be delete is this posebol? If yes how? Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:

Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm

Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost

Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

After opening a few windows, some start to open completely black. So I have a frame, and a black rectangle. If I resize the window some display ok, some don't. if I maximize them again, they go black What is causing this? Does anyone know how to fix this? I am running compiz + Ubuntu 7.10 gutsy Xeon 2.4ghz + nvidia quadro nvs 280 Thanks IMPORTANT NOTICE: This message may

Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

Dears Can anyone guide me ?? I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards _____ ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1"><font face="Bitstream Vera Sans"><br>     Hello,<br> <br>     I am trying to have dns <b>automatically</b> configured through

Hi all Sorry if this is a dumb question: does the input_frame passed to speex_encode_int *have* to be frame_size samples long? e.g., If I only have 100 samples left to encode (which is less than the frame_size of 160 samples), can I just use an array that contains 100 samples, or do I need to create an array containing the 100 "real" samples plus 60 null samples at the end?

Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of

I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither

Hi, I tried to install R-2.4.1 on cygwin system. "./configure" succeeded, but make failed. Below, I provide the output from the process: error message, and info from configure output, in that order. I appreciate that someone can guide me (technically in-sophisticated) through this process. Again, thanks for your help. Michael Niu (1). Output from make make[3]:

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14