similar to: Asterisk not relaying back the SIP response messages

> -----Original Message----- > From: Christopher [mailto:chris.robinson@voipsupply.com] > I see that there is a lot of discussion on the web about a > common error > after installing and modprobe'ing the zaptel driver. However I don't > see any resolution, anyone found a solution? > > Here's the output I get after modprobe: > > Uhhuh. NMI received.

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to

Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Pardon my ignorance, but I was hoping someone could clear up something for me. - For a few POTS lines, digium has a single port card for that, or a T1 card to a channel bank. - For 10 or more lines, digium has a T1 or E1 card for that too based on PRI channels - For 100's to 1000's of lines, I suspect a soft-switch is in order??? A traditional phone company will sell: - POTS lines for

Hello averybody, I'm looking the softswitch in digium website, anyone test the softswitch? Best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071202/2440f782/attachment.htm

Hi List; If I need to do a trunk between Asterisk and another SIP softswitch (so Asterisk will send a SIP calls to that softswitch), then I have to configure this on the sip.conf file or where exactly? And is it the same when I configure iax trunk? Should I determine the context in this case for this SIP trunk? Regards Bilal

Hi guys. I'm working on a VOIP service provider. We have two customers running asterisk. Customer A and B. When A call to B everything is ok. When B call to A the call ring but sip messages didn't arrive on asterisk A. In my softswitch i see the invite sip message sended to A. When every other numbers(TDM and SIP) call do A everything is ok. Have any issue in asterisk that can resolve

I've been working in the VoIP industry for just a bit over a year now... Mostly taking care of the underlying systems. I've now reached the point where I'm being drawn more and more into the call processing side of things. My background is in computer and "classic" telephony systems (DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor modules and

Hello, ? I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hello friends: I am facing cutoffs randomly when negotiating calls. The PBX dials the destination, the provider (softswitch) receives the request *[1]* and sudenly the PBX hangs up the call* [2]* while the provider is still dialing it, as a consequence the remote peer receives a ghost call. Along the atempt I could see six times a messages regarding NAT isuues *[3]* I hope anyone can give me an

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

Hi folks! I have a problem with our Sip provider that have a Softswitch Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working with G711 with them. They start sending calls to our pbx, some time after they start to receive 408 messages from asterisk and some time after this they start to complete calls normally, I don?t know what can be wrong. Someone has configured asterisk to

When we make calls out of asterisk to the PSTN via a SIP termination service provider the called party gets a slight echo of their voice. Here is the setup; analog phone <> Linksys ata <> asterisk <> sip provider sonus GSX 9000 <> PSTN <> called party. The caller on the analog phone connected to the ATA hears no echo at all. The called party has a slight