similar to: Busy out lines...

Hi all Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? -- Cheers Wayne

Hi folks, this is specifically directed to Australia Asterisk users...... We are having a roblem with x100p 's when dialing STD. Upon receipt of the approximately the 5th (out of the ten) PIP's asterisk will hang up.... Now I am wondering if others are suffering the same problem ?? Any ideas ?? (it might exist on other cards, but so far I have only noticed the problem on x100p's).

Hello all, This afternoon I had a BRI line installed by Telstra (our telco in Australia). I'm using an Eicon Diva 2.02 PCI ISDN card with the isdn4linux driver. Incoming and outgoing calls with Asterisk work fine (and with no echo - my main reason for getting ISDN). However, I can't seem to get outgoing DTMF working (incoming works fine). I made a call from my desk phone (Cisco 7940G)

Hello Group, Current Setup: - Eicon Quad BRI ISDN Card. - 4 ISDN BRI (Telco: Telstra) Onramp2 services. - Mode: Point2Point. - 100 Indial Number ranges. Full National Number (9 digit format): BAAAAAAXX where: B (Area code): 2/3/7/8 A (Normal Numbers) X (99 Indial extensions) eg: BAAAAAA00 BAAAAAA20 etc Requirement: - To be able send Indial numbers as Caller ID when dialing out. Configration:

Hello, We have 4 'Onramp-2' Telstra ISDN BRI services operating on Asterisk Server with Eicon 4BRI card. For most part the service is okay. However, we are are having problems with passing callerID to internal extensions. This is the set of command executed. exten => <pattern>,1,Answer ; Answer the line exten => <pattern>,2,NoOp(${DNIS}) ; debug statements exten =>

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

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

I've seen this question asked before but haven't seen a definative answer. Does the e100p work in australia? Did any one who was asking the question before bite the bullet and get one? I can get a te410 if i really have to but would prefer to stay with the cheaper option. Any comments appreciated, Thanks ------------------------------------------------- This mail sent through IMP:

Hi does anyone have a definitive list of countries other than those listed on the wiki which are supporting app_SMS on landlines using ETSI ES 201 912 ?? Thanks Tim Robinson Basingstoke, UK

I may have found a source of an A-Ticked HFC 4BRI PCI adapter in Australia, and will be testing one next week if all goes well. I don't want to post the details of the reseller online unless invited to do so, so if nobody replies and says they are interested then I won't :) I'll follow up once I've tested it. Let me know if you want the details. James

Hi, I have a strange problem with our Asterisk installation. Outgoing calls are handled by the following lines: exten => _0[2-9]X.,1,Set(CALLERID(num)=09999403${CALLERID(num)}) exten => _0[2-9]X.,2,SET(CALLERID(num)=${IF($[ ${CALLERID(num)} = 0999940321]?099994030:${CALLERID(num)})}) exten => _0[2-9]X.,3,DIAL(CAPI/g1/${CALLERID(num)}:${EXTEN},180,tr) exten =>

Hello everyone: using chan_capi 0.7 and asterisk 1.4 Quick question: How can I detect the number of "voice" channels (B channels) in use at a given time. I'd like to call "Busy" if two B channels are used on my BRI to give the calling customer a Busy signal. Long question: On my single-line BRI (two channels) I'd like to give the 3rd simultaneous incoming call an

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.

Hi all, I'm new to asterisk and not too knowledgeable on ISDN, so please be gentle :) I have a dual-channel Eicon Diehl Diva card in a Debian Woody box with kernel 2.4.27, connecting to a Telstra (Australia) Onramp Home Highway ISDN line. I'm pretty certain the card and line both work since they've been used in this machine for PPP before this (but with an older kernel with DoV

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

Telstra (Australian Telco) has recently introduced a feature to allow the sending of SMS direct to fixed analogue lines, with an appropriate handset. As best as I can figure out, this appears to use CID type signalling, or at least on a line that otherwise has no CID on it, CID is sent, but with a standard modem I can only receive the date, time, and phone number (eg normal CID info). After that

Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] --> channel:0 mode:TE cause:0

<div>Is there any chance to create feature request for that useful functionality?</div><div>š</div><div>02.07.2015, 14:03, "Rusty Newton" <rnewton@digium.com>:</div><blockquote type="cite"><div><div><div>On Wed, Jul 1, 2015 at 4:46 AM, <span><<a href="mailto:royj@yandex.ru"

On Wed, Jul 1, 2015 at 4:46 AM, <royj at yandex.ru> wrote: > Hi, all > > Is there someway ability to insert custom Header to "SIP 486" message, > when HANGUP application is invoked? > > Our use case is to set that Header, when call-limit is reached, to analyze > elsewhere, but we do not want to set some custom causecode in HANGUP > application because this

> Is there any chance to create feature request for that useful functionality? > 02.07.2015, 14:03, "Rusty Newton" <rnewton at digium.com>: >> On Wed, Jul 1, 2015 at 4:46 AM, <royj at yandex.ru <mailto:royj at yandex.ru>> wrote: >> >> Hi, all >> >> Is there someway ability to insert custom Header to "SIP 486"