similar to: SS7 links

Hi, Does Asterisk support SS7? Google shows an old new post from Feb. 2002 stating that OpenSS7 would help add SS7 support to Asterisk, but presently OpenSS7 seems to be dead and I can't seem to find anything about it at Asterisk or Digium's sites. What happened? -- Regards, Tais M. Hansen ComX

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page

The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:

Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to

Hello.. I have a customer who wants to connect 2 PBX's over IP.. The setup should look like this: [PBX] <-- SS7 --> [Asterisk] <-- IAX --> [Asterisk] <-- SS7 --> [PBX] Since there are no SS7 cards , I was thinking at a way of carrying the E1 data as bulk...Can I do that ? How ? Is possible a scenario like this ? I'm thinking of IAX because I don't

Hi, I've posted this question on OpenSS7 mailing list but I didn't get a clear answer. I have a system running asterisk using Digium T400P card. Can this same card be used with OpenSS7 project to carry SS7 links? Is there any difference between these two cards? -- Matteo D'Amato Network/Telecom Administrator Intelco Communication Inc

Hi, there are still some questions to be answered by OpenSS7.com in order to decide, whether E400P-SS7 is a good choice for the asterisk SS7 support. In the meanwhile I'm also in negotiations with another manufacturer (whose name I currently may not tell due to a NDA) of SS7 hardware, who gets likely persuaded to offer a cheap SS7-PCI-card which would be suitable for asterisk. Asterisk users

I am trying to hook up my Cisco telephones to Asterisk using a Layer 3 switch and am having difficulties it getting it to work. I realize this may not be the proper forum for a discussion on VLAN architecture and configuration so I won't post the question here. I though I had read all the requisite information regarding the configuration for this, but perhaps I am missing something simple.

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, 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.

We have a strange echo problem. Maybe echo isn't the correct term. When we make a call f/ a SIP phone (we have several 7960's, some 3coms, and I've even tried a softphone, all on the same 100BaseTX network) to the pstn, if the person I'm calling has a PRI or channelized T1 f/ Bell, then the sound is perfect, couldn't be better. If I make a call to a person with a plain POTS

Does anyone know of a device that will take an SS7 link and convert it to a PRI? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

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 >

Dear Asterisk Community, I would like to offer my most humble apology for the enormous stream of mail messages currently battering at your mailboxes. I was honestly trying to help a worthy cause. Please blame me and only me. Neither Russell nor Josh are at all to blame for this. Nor, for that matter, is Olle. I am the guilty party. Please direct all anger/outrage/hatred/etc. towards

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

Hello! I wanted to make sure that, in addition to my complaints, I make it very clear: Digium's support is excellent. The jury is still out on the usefulness of the TDM products. However, Digium has worked very hard to make sure that this issue is resolved. I actually got an e-mail from someone at Digium actually asking what they could do to make me happy! She even gave me

You are correct... No glare on a PRI W. Kevin Hunt CCIE #11841 www.huntbrothers.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven Critchfield Sent: Tuesday, May 25, 2004 3:26 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] "Glare" condition - How well does asteriskhandle?

Anyone having problems with LiveVoIP lately? I am seeing failed outgoing calls. Calls that are being routed to wrong numbers. DID's that ring busy. For the pass 2 days I am unable to pass CID. Is anyone else have these problems? Can anyone recommend a Quality VoIP provider?

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

Hello, After reading the lists and taking reccomendations from TC, I have finally given up on the echo can built into asterisk. I am sick of hearing complaints from users, so the money spent on a hardware echo can will be worth its weight in gold. I am curious however, about some setup and component requirements. It seems as if every telecom place I call, either never calls back or doesn't