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Is there a way to combine these lines into one? exten => s,2,GotoIf($["${CALLERIDNUM:0:3}" = "800"]?s|108) exten => s,3,GotoIf($["${CALLERIDNUM:0:3}" = "866"]?s|108) exten => s,4,GotoIf($["${CALLERIDNUM:0:3}" = "877"]?s|108) exten => s,5,GotoIf($["${CALLERIDNUM:0:3}" = "888"]?s|108) Thanks --John

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

When I call to VoicemailMain it just sits. ; Retrieve Voice Mail exten => 2500,1,Wait(2) exten => 2500,2,VoicemailMain(s100) exten => 2500,3,Macro(endcall) 1.4.3 latest SVN. voicemail(100) works and the mwi systems works. I am not using ODBC or SQL. Voice mail to email works ok. I just cannot retrieve it by the application. I'm not sure when this quite we get little voice mail

Hi ,all : as the subject says: I'm stuck with ldap-auth. Distribution: CentOS 5.3 i386 Davical/awl: Latest *.rpm (davical_0.9.7.6-noach.rpm) So far, I've got davical up and running using the Postgresql database . And I want to use the LDAP to auth the accounts and their password My *.conf: ,----[ /etc/davica*l*/myserver.conf ] | <?php | | /********************************/ | /*********** LDAP hook **********/ | /******...

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this

I can part a call (dial #700 it is parked on 701) but if I dial 701 I am told it is not a valid extension? I have include => parkedcalls in my local extension context. I have Ttr on all extensions and the incoming pots line. It parks, plays MOH but I can't retrieve it. --john -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am getting pounded by junk faxes. I have noticed that all of them have no callerid. Can I set up a dial plan to ignore a fax if no callerid is sent? I am using RxFax (spandsp) on an x100p card. Thanks John Hill

I have a rather dumb question, perhaps I over look the obvious? What is the recommended best practice for obtaining, compiling and installing a new cvs asterisk source. At present I do a cvs update and look for changes, to see if I need or want to update. Then because of the spandsp patch I delete the asterisk source and co the cvs. I then patch it, make clean, make, stop asterisk, make

I get this error message in my syslog. I have searched the list but I can't seem to find a answer that solves the problem. chan_sip.c: Registration from '<sip:2238@192.168.100.1;user=phone>' failed for '192.168.100.2' The asterisk server is running dev head. The server is the 100.1 ip. I have 3 ata186's numbered 110.2-4. In sip.conf I have 6 entries all the same

The 1.2 beta1 works fine. When I install the current cvshead it gives me different errors: I have seen checksum errors, Got event ring 18, etc. all give empty callerid. I have an x100p. Thanks --john

I have 1.2.0 beta 1 running and it works fine. My x100p returns caller id with no problems. When I test the CVSHEAD callerid fails with checksum and len < 0 errors. I can run with the cvshead of zaptel and libpri with beta1 but only the beta1 source works for caller id. Any source after beta1 fails. Thanks --john

I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the