similar to: Pass CallerID when call forwards to PSTN?

I've searched and tried several ideas (na.action. and other things), but I can't see to figure this out. I'm guessing this is so simple I'll feel foolish for asking, but here goes. Thanks, L.A. Dataset$Rcil=with(Dataset, ifelse(Rpr >= .95, Dataset[,"percentchgn"], NA)) Dataset$LLCI<-with(Dataset, ave(Rcil, LEAID, Property, FUN=function(x)max(x))) LEAID

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem

Just hit Slashdot: http://slashdot.org/article.pl?sid=06/06/08/1725215&threshold=1 http://money.cnn.com/2006/06/07/news/companies/pluggedin_fortune/index.htm >From TFA: "But what's really cool about what will happen in Cardiff - and eventually the rest of the U.K. - is that BT is creating an open, standards-based platform for which anyone can develop new applications. In other

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

Hi, This is really strange (but probably simple solution). The CALLERID(all) setting doesn't seem to work when the incomming callerid is 'unknown'. Dialplan looks like this: exten => _3072,1,Answer exten => _3072,n,Set(CALLERID(all)=DIRECT <0850553072>) exten => _3072,n,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004&SIP/2201&SIP/2202&SIP/2

Hi, I am running xen 2.0 for some days now on a local network and it''s running fine :) Every virtual server on my host gets it''s own IP address. But now, I am wondering if it''s possible to run xen on a hosted root server with only one ip address. Lets say I have a domain ''domain.org'' that points to my host, on which xen is running. Is there a

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

Is it possible using asterisk to allow someone to dial in and remotely change where their call is forwarded to? For example, I'm working from home so I want my calls to go to 555 1234, now I need to go out for a bit so I'd like to phone the office and using DTMF tell the asterisk PBX to now forward my calls to my cell phone 555 3456 Has anyone implimented anything like this? R.

Good Day All I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings. But on external network with PROXY setting ASTERISK DID NOT WORK. My question are 1 Can ASTERISK work in a PROXY setting . 2 If it can work how can i implement it . Expecting your reply Thanks Paul --------------------------------- Yahoo! Messenger

Hi, I''m trying to set up networking on a new machine. I''m not getting any networking from the domU''s I notice that I have two bridges being created: xenbr0 Link encap:Ethernet HWaddr FE:FF:FF:FF:FF:FF inet6 addr: fe80::200:ff:fe00:0/64 Scope:Link UP BROADCAST RUNNING NOARP MTU:1500 Metric:1 RX packets:3024 errors:0 dropped:0

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users

I appear to be seeing the performance of a local ZFS file system degrading over a short period of time. My system configuration: 32 bit Athlon 1800+ CPU 1 Gbyte of RAM Solaris 10 U6 SunOS filer 5.10 Generic_137138-09 i86pc i386 i86pc 2x250 GByte Western Digital WD2500JB IDE hard drives 1 zfs pool (striped with the two drives, 449 GBytes total) 1 hard drive has

Hi all, With a great help from Richard Alexander (thanks Richard!) I have now a functional CallerID on my X100P. This is what I have done: - update to the latest CVS (as today at 5:00pm GMT) - modify the callerid.c file in the asterisk source like that. original : /* MDMF */ /* Go through each element and process */

Hi, I cannot see cid for incomming call from PSTN (Quintum gateway) to IAX client (FireFly). Client displays blank but when I look into cdr's /var/log/asterisk/cdr-cvs/Master.cvs, the callerid is registered properly. Why it's not displaying? L.

Hello, I am trying to make my Asterisk to recognize CallerID from incoming PSTN calls. I am using a TDM400. CallerID in my country's PSTN is based on DTMF. Some information from my configuration files: /etc/zaptel.conf: loadzone=nl defaultzone=nl /etc/asterisk/zapata.conf usecallerid=yes cidsignalling=dtmf cidstart=polarity As displayed on the console, my Asterisk cannot detect the event

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server.