similar to: te410p and Telstra Onramp 10

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hello, We have an Asterisk 1.0.9 machine with a Sangoma A101 card fitted, it is connected to a Telstra OnRamp E1 in Melbourne, Australia. The problem we are experiencing is extreme echo and clicking noises. These are only audible to the calling party, e.g. the person calling in from the PSTN to Asterisk via the E1, the person on the SIP handset cannot hear any echo or clicking, only the

hi all, i've now installed a TE410P (Quad T1/E1 Primary Card - Digium) and connected it to a primary line. My telco (eTel) only told me that they are using hdb3 and crc4. So i still don't know which coding i have to use (cas or ccs) - and what timing options i have to use. Have someone already get a card like this up in Austria with a line from eTel or from the telekom Austria ? Another

Hi, I am currently trying to find a replacement for a dinosaur PBX and want to replace it with a VoIP solution. We have just moved our lines over to an Optus Multiline from a Telstra ISDN Onramp 30 service with 100 lines. My question for you good people is what sort of hardware do I need to interface Asterix into the Optus Multiline? The Optus service is terminated in my office to a SHDSL

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

Hi all, Does somebody know how I have to setup my TE405P ? Is it correct my configuration below ? Otherwise, can somebody help me ? Thanks, Angel. zaptel.conf span=1,1,0,ccs,hdb3 span=2,0,1,ccs,hdb3 span=3,0,1,ccs,hdb3 span=4,0,1,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31 bchan=1-15 dchan=16 bchan=17-31

Hello all. I have an NEC PBX connected via a TE210p E1 line to an asterisk 1.6 box. NEC -> E1 -> TE210P:1 -> * -> TE210P:2 -> E1 -> Telco Incomming calls from the telco to the asterisk box to the NEC work fine with indials and everything. Works sweet. Outbound from the NEC to the Asterisk box fail. Giving an long dial tone that then times out. Ie, pick up NEC handset, dial

We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forwarded to a SIP

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

We have a machine with a TE410P in it acting as a client to route calls via iax2 to our central server, caller --> ( zap -> iax ) ---> ( iax -> whatever ) --> called client server often the called can't hear the caller (both machines on public ip) 'iax2 show netstats" on client machine shows more and more dropped packets on the

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

Dear All, we have installed a TE410P card on a Dell Poweredge 1750 running a slackware 10 with 2.4.26 Kernel. Then we have made two loops on the card and we have configurated all the 120 channels. Our goals was to perform some stess tests even if in this scenario we used the same box as generator and target. The stress test comprised to generate up to 60 calls at the same time by placing a file

hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'.

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

Hello I've got small PBX (30 simultaneous connections) based on asterisk (1.6.2.6), which uses Stargate 2N ISDN to GSM gate. It runs ok for day or two, but then I get: dahdi: HDLC Receiver overrun on channel TE4/0/1/16 (master=TE4/0/1/16) in my kernel logs, in asterisk i get: pri show spans PRI span 1/0: Provisioned, Down, Active PRI span 3/0: Provisioned, In Alarm, Down, Active (span 3

Hello Asterisk users. We have a quadBRI card installed and have the following problem. When starting Asterisk, the card is up and works perfect. But if no one uses it for 2-3 hours, the card seems to change status. If I try making a call from my sip phone to an extarnal telephone then asterisk registers that I'm trying to call, i.e. it's not a bridged but a show channels gives this

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

We've changed E1 providers and I'm trying to reconfigure an E400P to make it work with the new lines. They're supposedly "standard" EuroISDN lines (in the UK). I'm initially just trying to get a single line up. I have the following in /etc/zaptel.conf: span=1,0,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 loadzone=uk defaultzone=uk The LED on the back

Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to "nirwana" also no indication is hearable, calling a second internal subcribes bridges them to the