similar to: RE: Aculab SS7/ISUP

>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 >>

Hello Trevor, I wonder how I can find out for sure what is the H/W version for a PROSODY ACULAB SS7 Card? I dought that I have a ver 1.1 which have may issues with recently made computers. I have a case opened for my problem with aculab but sysdiag shows that I have ver 1.1 and aculab says I have ver1.5! can sysdiag be inaccurate? I tried config summary and I had the same result! wish you

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, Does anybody have experience working with Aculab groomer II, to convert between ISDN E1 and non-ISDN T1, or anything similar. I am looking for sample config files. We have asterisk as ISDN E1, but for testing we set it up as regular T1 if we get sample config files. Zeeshan A Zakaria -- www.ilovetovoip.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the

Hello, Is SS7 and PRI in any way compatible in that if the interface is configured one it will work for the other (granted, it will not have any of the ISUP, etc. parameters available if the line is PRI) or are they two distince protocols that have incompatible signalling? Thanks, Elliot

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

Hi, one month ago, I announced, that I will look at the openss7 project in order to use it together with asterisk. It took a while for me to check the capabilities of and around the project. Since the openss7 project consists of only one person, and when there was just silence for a long time in the project's mailing list, I doubted, whether it was a good idea to consider using openss7. In

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, 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 --------------

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

On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.

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

All, I am trying to find a SIP ITSP that honors dial around compensation. We are adding a Flex ANI code to our outgoing SIP invites by appending an isup-oli tag to our From: address, like this: INVITE sip:18889996563@carriers.icall.net SIP/2.0 Via: SIP/2.0/UDP xxx.y.34.201:5060;branch=z9hG4bK7f314484;rport From: "Dougs Payphone"

Greetings to one and all on this fine list; We have the current system: Meridian 1 Option 11 +-------------------+ | | ISDN/30 (DASS/2) ===> |NTAK79BB (2MB Pri) | | |<-->4x16 port Digital / 1x16 port Analogue ISDN/30 (EUROIDSN) ===> |NTBK50AA (2MB Pri)

hi below is sysvinit script for handling tcng-script .... please excuse my bad bash skils, correct me where i''m doing stupid things :") what is left... many things, if i have time i will implement them too : - start [devices] - and then correct handling of service lock files i.e. per device lock file - stats blah ... - all ideas are welcome !! - test or some such, i mean the

Hello Everyone, We are currently having talks with various service providers, and trying to determine what the best way is to interconnect in order to have access to the PSTN network. As you know there are two ways of doing this: Traditional PRI: Have trunks grouped into a transport layer such as OC3/12. With DIDs attached to the group. As you many know, this approach would also require a POP

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

I checked https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information But I cannot find a way to insert CALLERID(ani2), which I can read, but when I try to set it for a new call, I get a runtime error. This information, known as isup-oli comes embedded in the From header,like this <sip:9087311878 at 64.45.157.104:5060;isup-oli=62>;tag=sansay1724414rdb124 and it can be read

Hello list, I'd like to know if there is a function that tells in which package is a given function. Something like : which.package("lda") [1] "MASS" Thank you. Romain -- ~~~~~~~~ ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ~~~~~~~~ ~~~~~~ Romain FRANCOIS - http://addictedtor.free.fr ~~~~~~ ~~~~ Etudiant ISUP - CS3 - Industrie et