similar to: No call progress from * to E1

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

I've been beating my head a bit against the 1.0.6 Debian builds of Asterisk, using an E100P (E1, single span) board. In machines I've built in the past (back in 1.0.0 days), config I'm using and that card and 1.0.0 driver combo worked fine. ztcfg reports no problems: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. And zttool sees the card, and

Hi all. The SPA941 and friends have pretty multicoloured LEDs, but there doesn't appear to be any support for SUBSCRIBE/NOTIFY as * as implemented for extension hinting. Has anyone managed to get the phone to support this? Thanks! -- David Zanetti <david.zanetti@catalyst.net.nz> Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 -------------- next

Hi, I've looked around the archives and have spent a while on voip-info.org but not found an answer so forgive me if this is in a FAQ somewhere. We've got several Asterisk servers with E1 cards (some Digium, some Sangoma). We provide non geographic numbers for customers and route calls to their existing phone numbers. Calls come in over the PSTN and into Asterisk. This works perfectly

Hi.... Apologies for the length of this, but any help would be greatly appreciated ;-)... I have installed and configured a Digium E100P on my Asterisk PBX and I have connected this to an 8 port E1 VoIP gateway on a Cisco 6509. There is also an external E1 link from our Telco plugged into one of the other ports on this gateway and this gateway is in turn registered with a Cisco CallManager

Hello, I have several customers describing something like annoying "silence suppression". So I did some tests and I can confirm. After disabling echo cancellation in zapata.conf the "silence suppression" effect is gone, but there is a little echo of course. I do not have this problem on other boxes where I am using oslec as an echo canceller. All my calls are SIP to ZAP (E1).

Hi there! I am having some difficult in receiving calls from my E1 link using mfcr2. I can make calls normally , but when I receive an incoming calls, the phone ring I answer it ,so, I listen busy tone and then the phone ring again and again. look the log: -- Executing [4801 at from-pstn-TE1:1] NoOp("DAHDI/2-1", ""1233220567" <1233220567>") in new stack

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

Hi all. I have and E1 channel bank from Loop Telecom. there's a little issue with it, I cannot ring the phones on fxs interface, but can connect without issue them. What happens: I dial the phone on port 1, asterisk says "Zap/1 is ringing", but the phone on the analog port doesn't ring. but if I take off hook the ringed phone, asterisk detects the answer at they're bridged

Hi all, My scenario is this one: LandLine------------------E1---------------|-------------| |-------------------| |OLDPBX|-------E1-----------|Asterisk1.2.5|-----VoIPusers GSMGateway---------Analogue------ |-------------| |-------------------| What is happening: 1- SipUserAgent "A" Dials a call to a Local Extension "B" in the OldPbx 2- "B" , the called party

Hi everybody. I'm about to ask a newbie question, be warned! I have a NEC 2000 IPS PBX connected to a E1. Now I want to set up an asterisk, with some digium card connected to that E1. (suggestions about the card? I'll have maybe another E1 more). The "newbie question" is.. How can I connect the asterisk PC with the PBX? I don't know so much about E1, so I don't know

Hi, Change my Dynastar E1 gateway to Cisco with E1 module, but can't make easiest dialplan. All my routing i made on asterisk, so i need that cisco all calls from E1 send via sip to Asterisk and all calls came from Asterisk by sip send to E1. From E1 to Asterisk already work, but can't understand how send all from Asterisk SIP to E1 ? Can you help ? Thank you

THe wiki doesn't seem to have any user reports. If your using them, how are the working, better, worse about the same. Also what hardware seems to be stable with them installed. Alan

Hello Every one! I am little bit new to asterisk. I am doing research on different telecom options as well. I have question for you professionals In order to get E1 line working with Asterisk. What E1 line parameters need to be specified in Asterisk(configuration files). They vary from country to country. What is difference in one countries E1 and others E1. the most important If we want to

hi list, how can i organize several pcs installed with asterisk and e1 cards to be seen from an asterisk server as one? so if there is a voip call that needs to be forwarded towards the pstn the asterisk server should find a pc that has free channel on it's e1 cards that is connected to the pstn side. /-- Pc1 - E1-\ VOIP -> Asterisk--- Pc2 - E1--PBX->PSTN

Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk

As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too).

After days of searching i've finally figured out that E1 lines in india use multiple types of signalling from EuroISDN to R2. Digium E1 cards dont work for R2 type signalling. Can anyone suggest me asterisk compatible E1 cards which would work on R2. Also if anyone on this list is from INDIA and uses asterisk and E1 lines please let me know what type of cards (which vendor) do you use and what

If I have a euroisdn pri E1 line from my telco (BT), can I use a cisco 3620 router (with 1 Channelized E1/PRI port) to connect to the E1 line, and connect * to the same lan as the router ? I suppose what I'm really asking is "what is a 1 Channelized E1/PRI port" and can it connect to a E1 line. I have gathered from the web/wiki that the cisco works with *. Julian

Hello list, (Resending this email due to a typo in previous copy) I need to do E1 to T1 conversion for a project, and was wondering if there exists a card with both E1 and T1 on it. Or is it possible to use two separate cards in an asterisk box, one for E1 and one for T1? (Please don't mention aculab or adtran, dealt with them in the past, won't deal again.) I talked to Digium and the