similar to: Transfer and Release of a call out to PSTN

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

I'm making outbound calls by placing call files in the asterisk outgoing directory. At times, the call would be hung by SS7 without even attempting (due to error in the outgoing number). I get the following on console: -- Attempting call on ss7/9297210213 for s at croom:1 (Retry 1) -- Sent IAM CIC=22 ANI=9134904821 DNI=9297210213 RNI= -- SS7 hangup 'SS7/callserver/22'

Hi all, I'm being requested to deploy an?IVR service?using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

hi some time ago, I asked the list of a good book for learning ISDN and SS7. I don't need to know how to write a channel driver or something; I just want to know more about the possibilities and what's really sent back and forth. I was told the book "ISDN and SS7: Architectures for Digital Signaling Networks" by Uyless Black (ISBN 0132591936) was a good choice, but this

Hello all, I have a project which requires me to rout calls from ten blocks of sequential numbers i.e. 02081000100 - 02081000200 (each block - 100 numbers) coming in from a telco gateway via Dahdi-SS7 to 10 specific numbers outside the box through two to three SIP trunks (trunk 2 and 3 will be spare capacity/redundant for trunk 1). CLI is crucial here as I need to forward the CLI of the numbers

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 wrote a proof of concept regarding the cpu capabilities runtime detection and choice of optimized function. I follow design which had been discussed on IRC. Also, i notice a little drawback: we must propagate the arch index through functions which don't have codec state as argument. However, if it's look good, i will continue to implement it. Best regards, -- Aur?lien Zanelli

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

OK here it goes.. Caller ID is two parts or actually three: Part 1 Number only Part 2 Number + Name Part 3 Whole lotta stuff (also known as ADSI) Here is the US, I cannot speak for other countries. When party A places a call to Party B. Party A's Telco picks up the number, either from a table on the switch or passed from the PRI from Party A. Then on the far side (Party B's Telco)

Hi, I am requesting for a E1 connection from my telco. They are asking if I want DSS1 or SS7, and I am stuck here. Could someone tell me the difference between the two? How should I decide which one to use? Thanks in advance for your help. Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Dear All, As per this article http://www.voip-info.org/wiki/view/Asterisk+cmd+Smsasterisk support PSTN to SMS and from SMS support endpoints to PSTN. In my scenario we have SS7 based E'1 on which our SMS provider sending SMS on our DID numbers and my all DID's are registered on OPENSER. What I want to do I want to receive SMS from PSTN on E?1 and forward on Register user?s if Register

Hi Sangoma a104 card have in product specyfication support for Line protocol SS7 , http://www.sangoma.com/products/p_aft-104-specs.htm [..] Line protocols Frame Relay, X.25, HDLC, PPP, SS7, Transparent bit-stream, BSC. [..] Anyone of you guys use line protocol SS7 for E1/T1 termination in asterisk ? As far I know asterisk don't have support for SS7 signaling, but my telco wants to setup

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

Dear R users, When I run the following code, R crashes: require(cclust) x <- matrix(c(0,0,0,1.5,1,-1), ncol=2, byrow=TRUE) cclust(x, centers=x[2:3,], dist="manhattan", method="kmeans") While this works: cclust(x, centers=x[2:3,], dist="euclidean", method="kmeans") I'm posting this here because I am not sure if it is a bug. I've been searching

Looking for some advanced thoughts relative to exten number assignments. We're in the planning stage for rolling out asterisk at multiple small US telco/isp operations. Their typical voip customer has had their pstn line for a looooong time and wants to keep the pstn line and number, but add voip to their existing home/soho arrangement. One approach (from a planning perspective) is to

>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 ppl, Am looking at some PSTN termination providers in US. If this question has been repeated, please point me to the correct link, as I've tried searching the archives but have been unsuccesful so far. I have come across quite a few companies which provide the same, such as : Iconnecthere <http://www.iconnecthere.com> Vonage <http://www.vonage.com> Teliax

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