similar to: dialing trunk-to-trunk

Hello All,I'm very new in asterisk.Please help - how I can write conf files (or some example) for to delete one ext. and to add another, it means for example: I need to call from one asterisk to another by trunk to trunk and my dialing (for ex.) 100#100 at 1.2.1.2 when the the trunk of first asterisk is 100 at 1.2.1.2 and trunk of second 100 at 1.3.1.3, symbol '#' in this case

Hi, Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of Asterisk B (registered in Asterisk A as extension) to incoming call across another trunk of Asterisk B to extension of Asterisk C What the dial plan should be? Thanks -- We never did too much talking anyway So don't think twice, it's all right -------------- next part -------------- An HTML attachment was

<DIV><FONT face=Arial size=2>hello All,</FONT></DIV> <DIV><FONT face=Arial size=2></FONT>&nbsp;</DIV> <DIV><FONT face=Arial size=2>our Asterisk pbx is sitting behind a normal analog hardware pbx, we have to dial 9</FONT></DIV> <DIV><FONT face=Arial size=2>to take an outside call through the hardware pbx, our

Hi! I have a runing asterisk box and i want to dial to a analog. pbx using a 4FXS Welltech. Let's say that my pbx have no. 700. If i want to dial to a person in that direction i have to dial pbx prefix (ex. 700), wait for pbx to ansear with hello message and after that to dial internal number(ex. 101). It is possible to dial directly 700101 and asterisk to dial PBX prefix, wait for PBX to

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

I have a SIP trunk built between a Cisco CallManager version 8. I can dial the phones registered to the Asterisk PBX from a phone registered to the Call Manager. I've tried to keep the config as small as possible to help the troubleshooting process. Attached is he most recent debug. My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251 SIP.CONF [6001] type=friend

Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange "Unrecognized prilocaldialplan" error with the SIP username when a SIP call is dialed to a PRI trunk. The error shows up like this: Unrecognized prilocaldialplan TON modifier: a Unrecognized prilocaldialplan TON modifier: b Unrecognized prilocaldialplan TON modifier: c Where abc is the SIP username. Is this a bug

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

Hi all, I am trying to perform the following outgoing call: exten => _49.,1,Log(NOTICE,Dialing German number: ${EXTEN}) same => n,Set(route=DAHDI/g1/00${EXTEN}) same => n,Dial(${route}) exten => _0049.,1,Goto(${EXTEN:2},1) exten => _01149.,1,Goto(${EXTEN:3},1) exten => _+49.,1,Goto(${EXTEN:1},1) But this is not working. I have also tried changing the

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

Hi all, I am trying to setup a ZAP interface to do MF signaling for a handoff to a 911 tandem. The signaling I need to perform on the T1 is this: 9-1-1 Tandem: Wink CLEC end office: KP (Keypulse) NPA ST (Start) 9-1-1 Tandem: Wink CLEC end office: KP I (Info Digit) NXX XXXX ST As I'm not as familiar with the Zaptel configurabliity, I'm not really sure how to do this. Do I dial twice

Dear sir, I have to set up a mail gateway which will be explored to Internet and a secure mail server in the Intranet. I need a smart imap proxy in the mail gateway which will fetch the mail from server and present to user through either a stand alone mail client or a web mail client. All authentication is through ldap server. I followed the instructions given in

Hello, I have problem after killall -9 asterisk and asterisk -f Asterisk stops to send to DNS resolving of trunks Regards -- We never did too much talking anyway So don't think twice, it's all right -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081218/07751113/attachment.htm

I was wondering if anyone has an easy way to emulate dialing in a round robin fashion like when you use Zap/r1 for Zap trunks. At the moment what I do is simply make a macro that will dial the sip trunks in order so if the first one fails it goes to the second and so on. The problem with this approach is that the first few SIP trunks will always be busy because of outgoing traffic. Is there an

TRUNKMSD1=1 ; MSD digits to strip (usually 1 or 0) TRUNKMSD2=2 ; MSD digits to strip (usually 1 or 0) ; logn distance calls exten => _91NXXNXXXXXX,1,NoOp("Dialing: "${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD1}}) exten => _91NXXNXXXXXX,3,Congestion When I dial

I have set up a * box with two hfc ISDN pci cards using mISDN both in TE mode with PmP mode. (using $MODPROBE hfcpci protocol=0x2,0x2 layermask=0xf,0xf) I have no problem dialing out by explicitly naming the mISDN port, ex: Dial(mISND/1/${EXTEN},60) or Dial(mISDN/2/${EXTEN},60) But it does NOT work when specifying the mISDN group: exten => _(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)

Hi everyone, I am having installation problems, but this is how it all started: I had some errors running the bioconductor package affyPLM that uses LAPACK/Blas > Pset <- fitPLM(Data) Background correcting PM Normalizing PM Fitting models /usr/local/lib/R/bin/exec/R: relocation error: /usr/local/lib/R/lib/libRlapack.so: undefined symbol: s_copy # thrown out of R .... I was using R

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP