similar to: Re: Asterisk-Users digest, Vol 1 #1835 - 12 msgs

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

Message: 5 To: asterisk-users@lists.digium.com From: Doug Meredith <doug.meredith@skyridge.com> Date: Wed, 21 Jan 2004 20:05:19 -0400 Organization: Skyridge Systems Inc. Subject: [Asterisk-Users] Re: What technology could my phone company be using? Reply-To: asterisk-users@lists.digium.com >>Mark Hazlewood <lists@idontknow.com> wrote: >>Sounds like Centrex services, we

We currently use Asterisk to provide a SIP-to-PSTN service. The actual conversion takes place somewhere in a softswitch owned by our SIP-to-PSTN provider, where we have an SS7 link. We would like to do that conversion ourselves. Is it possible to replace a softswitch with a Cisco AS5xx0 only (ie. AS5300, 5350, 5400), or is a *real* softswitch (ie. Cisco PGW2200) needed? Does anyone have any

>I spoke with someone today who is interested in an IP Centrex solution that >starts with about 3500 extensions in a multi-tenant application. And >growing from there. > >I'm wondering about scalability of Asterisk. I'm trying to put my head >around how to put the whole thing together, if it can be put together. > >The nice thing about it is that if I can show

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

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?

I'm presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Is anyone out there using Lucent brand equipment to handle an incomming DS3, converting all 672 calls to SIP (as G729) and sending those to Asterisk/SER over ethernet? If you are and are willing to speak to my boss about your experiences (over the phone) with it, please contact me off list. We have a possible contract with a local CLEC to handle their long distance, and they want to send to

Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural "monopoly". From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them,

Does anybody know, if asterisk support the rfc 3323 "A Privacy Mechanism for the Session Initiation Protocol (SIP)"?? I'm working with a Softswitch wich works with this rfc, and I don't know jet how to dissable this functionality. This is a problem becouse the SS do not pass the ANI in the interworking SIP-SS7 (only in this direcction). Regards Miguel

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

OK, that "VOIP dialtone?" thread was getting really out of hand, so I'll condense my answers into one big ugly message: 1) 911 service. Yes, that is one of three reasons to keep your PSTN line. The other two reasons are: Inbound calls from local callers still should work on a POTS line, for now. You can't find VOIP providers in most area codes, so you'll most

> > >> I'm giving a talk at SCALE 2009 (Southern CAlifornia Linux Expo) on > >> Sunday in Los Angeles, and the topic of my talk is "Open Source in an > >> Economic Downturn". I've got lots of talking points for this talk, > >> but it would be interesting to hear some short anecdotes about how you > >> in the Asterisk community are

Has anyone setup SIP services with ss7 and lis trunks? If so .. what was used hardware and software.. we're trying to do a SIP -> pstn setup and not having much luck as QWEST keeps pushing dates off (aka trying to screw us over) for our pri lines due to the recent court and fcc activity in regards to unbundled switching and I'm looking for solutions/ideas involving SS7..

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

we're evaluating the use of a Lucent APX8100 E3/SS7 to SIP gateway for use in conjunction with asterisk, serving something like 4000+ lines. does anyone have experience with the APX8100 and it's integration with SIP on asterisk ? does the APX8100 handle SS7<->SIP signalling well enough to be used ? any anecdotes would be well appreciated. -- Regards,

The national guard and/or army routinely implements VoIP over wireless in situations where comm is lost, I did see an news release that the Guard started this project in the south the day after the disaster hit. The key is not the VoIP infrastructure, that is the easy part (one ss7 Sonus softswitch and a DS3!), the key is distributing IP over a wide area, which is best done on the quick with WiFI

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

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.

We have been pretty much able to solve our echo problems, except for the primary mode in which we desire to operate our system. See system diagram at bottom. Prior to making adjustments to cancel echos (all echocancel=no): Call Type Result (Before) --------- --------------- CP <- LEC PRI * TDMoE * FXO -> AP