similar to: CAS signalling and FAX solution

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going

How do I set configure my voicemail notification so that when I'm left a voicemail message it: 1) sends an e-mail to my inbox with the voicemail message attached 2) sends a message to my cellphone without the message attached I get notifications when I've got attachments turned off, but my cell doesn't like attachments in the messages and doesn't send them. An even better

Can someone point me in the right direction to find documentation on best practices when setting up a new Asterisk server? I'm using RHES4 and Dell 1750 with TE412P. My current problems are frequent crashes and choppy audio so I think I can easily tweak these out of the picture. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

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

Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under

I still can't get this thing to work. Outgoing calls are fine, incoming just hangup straight away. I get a lot of & characters in my logs, I'm wondering whether they're caused by some sort of bad data ? Mar 3 16:03:01 VERBOSE[10437]:

Hi, now that these cards have approval in Australia, has anyone had any luck in connecting them to a Telstra Onramp 10 service? Are these configured as a PRI, bchan=1-10, dchan=16 in zaptel.conf and switchtype=euroisdn in zapata.conf? Better yet anyone have any of the above .conf files they could share? Is an onramp 10 what is referred to as a 'channel bank'? Craig

Greetings; We are trying to get Asterisk up and happy at our site-we tried VOIP using Sphere about a year ago, spent a *boodle* on expensive hardware and services from a local "expert", but it never was happy. I'm brand-spanking new at VOIP, and I've learned a *ton* getting Asterisk breathing in the last couple of days. I have three Polycom Soundpoint IP 500 SIP phones, which

Hi all, I want to link three incoming Bell Canada centrex pstn lines (which currently go to an old norstar pbx) into asterisk. Can anyone suggest the most "painless" (i.e., "just works") way to do this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter, or the twice-as-costly Audiocode MP104-FXO-C3S? I know the Digium TDM400P or Sangoma A200 are options

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +612421100000 to something like 024221100000 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XXXXXXXXXX but that still seems wrong to me. it could match other numbers.

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

Hi, Using MeetMee(1234,M) on asterisk 1.6.2.9 (debian/testing) we have terrible sound: the MOH is unrecognizable and speakers can't be understood; it sounds "ghostly". However the prompts ("your are the only one in this conference, etc.") sound fine. Our server has a Digium T410P card with two E1 lines going in and the wct4xxp dahdi module. Any idea?

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

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 =>

who can tell me how to do TDM over enthernet ? pc a connect pc b only use TDM card? thank you John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?23? 11:47 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5, Issue 336 Send Asterisk-Users mailing list

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, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to