similar to: X-IMail-SPAM-Connection DNS Sudo ANI vs True ANI

Hi, I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-) I'm looking for boxes known to implement T.38 properly, and which really work in the real world. Regards, Steve

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Hi there, I'm using ser and asterisk together. Asterisk for voice mail etc and ser for registration of the users usig database. I can restrict forwarding calls from another sip proxy to ser (using proxy_authorize) but how can I restrict access to asterisk ... Now everyone can forward calls to my asterisk and can place pstn calls. Thanks in advance, Pavel -------------- next part

Hello, I would like to use Intel Blade machine for running Asterisk. Is there anyone who already use Intel Blade server for running Asterisk? Can you please explain, how perform Asterisk with Intel Blade machine? I would appreciate for giving me feedback regarding this issue. Regards Nahid -------------- next part -------------- An HTML attachment was scrubbed... URL:

I get seg. fault with my * box. at the crash time i had about 35 Bridged Channel. i have: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with

*spews coffee over keyboard* - FUNNIEST - THREAD - EVER - Also one of the most insightful. Teddy, your gmail invite is on the way.

Hi, all!!! What will Asterisk do in the following case: For example, we have 4 licenses, and have 4 simultaneous calls, using G729. Will asterisk allow incoming calls from peer, that can talk G729 and ulaw, and will it force it somehow to use ulaw in this case? All phones there in LAN behind Asterisk prefer GSM codec, so it does transcoding. So, what I mean is will Asterisk fall back to use

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy

I've successfully got inbound/outbound calling working with our Asterisk using the Asterisk-OH323 channel driver. We are using a parent gatekeeper and the NuFone H323 channel driver would not work with the parent gatekeeper... I'm trying to determine a way to ensure that the line used for outbound calling is always available i.e. like trunking.. >From what I can tell when I place an

Can anyone help me to understand what the significance of this output is? May 17 10:50:23 DEBUG[2030]: Didn't get a frame from channel: SIP/105-1ae4 May 17 10:50:23 DEBUG[2030]: Bridge stops bridging channels SIP/105-1ae4 and SIP/outbound-7dc3 I searched for these phrases but am coming up short on what they really mean. I'm trying to investigate problems we are having with two

Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and

Hi, We've got endpoints and gateways who have T.38 fax support. We now use SER and Asterisk to do our routing and other functionality, but fax doesn't seem to work. Asterisk complains like this: Sep 9 09:25:45 WARNING[467828746]: RTP Read too short Sep 9 09:25:45 WARNING[467828746]: Asked to transmit frame type 256, while native formats is 32 (read/write = 256/256) With lots of the

asterisk-users-bounces@lists.digium.com wrote: > Isn't it possible to use T.38 for interconnecting hardware gates > supporting T.38 with asterisk using SIP REINVITE? > I'm not shure but but think its's might be possible because after > reinvite traffic goes directly from one gate to anotger, not over > Asterisk We've seen a problem here with asterisk. Wehn

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

Hi, Our gateway has asked that we send DTMF as RFC 2833. Although chan_oh323 seems to do this, it doesn't specify the DTMF mode during the H323 setup headers. Is there an easy way around this? Thanks, Andrew

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx