similar to: SMDI support in Asterisk ?

I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------

Hello. I am new to this forum. If this issue has been discussed already, please inform me, but my STFW have indicated that it has not. If SS7 were available on * would it make commercial sense to use it? There are case studies on the web (courtesy of the Death Star :->) for ISPs with 10,000+ subscriber lines showing clear cost savings in transitioning from PRI to SS7 controlled IMT. My

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/a6f17a16/attachment.htm -------------- next part -------------- Hello I found related question on vmail.cgi in the mailing list but that didn't answer my question. I did copy the vmail.cgi to /var/www/cgi-bin/ but still gets the following error message when I access

Hey All, I'm just daydreaming here.. but what's the status of SIP-T in Asterisk? I haven't been able to find a whole lot of info on SIP-T but seems like just an extension of SIP. Right? Now if I had a PSTN Gateway (that is a SS7 gateway) that supported SIP-T, could I signal * with SIP-T from it and have asterisk utilize MGCP to sieze a particular DS0 on a remote DS1? Hmm.. What am

hello, all of users: sorry, resend it again for clarifying the message. I have implemented cha_ss7 in china. initially, the chan_ss7 can not support the call group. i modify the code. now the problem is that, both sides can hear the ring, but i can not hear the voice from each other. i think the ss7 does not send the voice steam to the destination. in chan_ss7, i added:

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

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.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/dd10d5ef/attachment.htm -------------- next part -------------- Hello Today I observed this strange problem, as soon as I called from my SNOM IP phone (910) to CISCO IP Phone 7905G (810), I got following warning messages and call didn't connect. But after couple of minutes this

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/3efdeb9f/attachment.htm -------------- next part -------------- Hello Though my SNOM 200 phone receive a voice mail but it doesn't show MWI on the LCD panel, Instead it keeps displaying "DND SW-REG,Call-log". Though I can access my voicemail using exten

Hi all, Does somebody know what's new with SS7 and * ? I'm very interested. Is it ready ? I'm prepared to pay if necessary. Thanks, Angel.

>Freddi Hansen wrote: >> with boards from Aculab, we are replacing Aculab boards with Digium >> boards BUT we would need more >> Digium boards IF we could use both Digium and Aculab cards in the same >> server. The reason being that >> TE410P doesn't support SS7-ISUP so we continue using only Aculab cards >> in the servers that must support >>

Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. any help appericiated Thanks Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi can someone recommend a book about ISDN and SS7? I've been hacking asterisk for a while, but it'd be good to get some better background within things... thanks roy

I hate to bring this thread back to life, but... > it may be possible to get it supported, do you think the price >point is remotely competitive with Digium hardware? Also as I am not >about to divulge my information to them to look in the downloads >section, what is the licensing of their SDK? What is the licensing of >the driver? > >Steven >On Tue, 2002-11-26 at 14:52,

Hi all, what is the signalling of IAX? Currently I want to connect two switch through IP using asterik signaling, and I want to transfer SS7 over IP (between the 2 asterisk), will IAX can transfer SS7 signalling through IP (like TDMoIP does) If no which solution can I use? see below the architecture switch1-------------Asterisk--------------------------Asterisk --------------------------Switch

Hello: I've been wrestling with an integration issue for which there also seems to be one piece missing. I'm hoping someone on this list can help. Due to a complete lack of cooperation from our current voice provider we are in need of an alternative way to access our Octel 350 voicemail system from our SIP (SER) proxy. I can redirect and relay calls to numerous destinations via

We have recently pulled an ancient Fujitsu-branded Centigram voicemail system out of production use and replaced it with an Asterisk box, which is now serving as our enterprise voicemail system and automated attendant. The Asterisk system is connected to a Fujitsu F9600 PBX and uses the 1.6.1.x SMDI module to communicate between the two systems. Save for one issue, the replacement of the old

Hello! I'm integrating an Asterisk-based voicemail system with an old switch, and I want the call history from SMDI. My understanding is that the terminal number in the SMDI message matches the channel's trunk number. >From within an Asterisk app, how do I get the trunk number? Thanks! Nate