similar to: callerid name inboune from PRI

XO CAN supply callerid NAME on a NI2 PRI connection. We have three of them and they work great. Its takes a little doing to get to someone at XO that knows what they are doing but XO does have some VERY good tech support people that know how to get things done. It just takes a bit of work to find them. Outgoing CNAM is a different beast however. They can't take it via IE. You need to get

XO fixed my caller ID name. I am using FreePBX and I can include a "wait" to my custom extensions. Is there a way to add a wait to the whole PRI? I assume that if I set immediate to yes, I can then have a "s" extension do the wait, but how would it get from the "s" to the DID extension? (also, I would rather not answer every call) Is there a "magic" spot

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

so here it is, the problem that's been nagging me for the past 2 days: connected a box to my telco's NTBA <-> zap/asterisk. which works: box:/etc/asterisk# cat /proc/zaptel/1 Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In

Hi I have recently upgraded my Asterisk system (from 1.2 to 1.4) and I have started to notice the following messages when I recieve a call on my Zap channel :- [Mar 11 09:20:17] WARNING[4294]: chan_zap.c:8851 zt_pri_error: 1 copying 5 bytes LLC I have a single PRI ISDN 30 link to a Siemens Realitis DX. Here is my zapata.conf :- [channels] echocancel=no echocancelwhenbridged=no rxgain=-5.0

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

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",

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

Hello, On a Asterisk 1.6.1 powered system, I've just discovered that using Busy() application in dialplan was no enough to send a Busy signal on incoming Dahdi channel. On this specific install, adding an Answer()) and a Playtone() statement in dialplan triggered sending of busy tone but I'm still surprised by my findings. Should I expect public switch to send a Busy tone to caller

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how

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, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is