similar to: blocked caller id

Users withing the office can dial a 3 digit extension and that will ring a phone. The problem I'm running into is you have to press xxx then press 'send or 'dial'. The pbx doesn't recognize a 3 digit number as an internal extension and automatically dial it the user has to initiate that call. Asterisk automatically initiates calls w/ 9+7 digits and LD calls,

I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked.... Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten => 107,1,Dial(SIP/102&SIP/107,40|r) exten => 107,2,Voicemail(u102@pstn) exten => 107,3,Hangup exten => 107,102,Voicemail(b102@pstn)

I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx-xxxx for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx-xxxx Currently my dialplan looks like so <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/>

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

I recently had a bear of a time getting a Polycom Soundpoint 500IP up and registered.. Now that its registered I ran into a problem w/ the dialplan. Needing to dial x101 I'd dial 10 - then get a fast buzy.. Also making a local call - dialing 95551212- would give me a fast busy after the 7th digit - so 9555121.. Same w/ LD calls... This dialplan really got me down as I didn't find

I'd like to propose a change to the voicemail remove script found in the contrib directory of the asterisk source Currently the find command looks like so system('find '.$dir.'/'.$context.' -name msg????.??? -mtime +'.$age.' -exec rm {} \; -exec echo Deleted {} \;'); I'd suggest it be changed to system('find

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

http://bugzilla.mindrot.org/show_bug.cgi?id=83 Summary: fork() fails when there are PAM limits set Product: Portable OpenSSH Version: 3.0.2p1 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: openssh-unix-dev at mindrot.org ReportedBy:

Hi Folks, I am trying to get my wireless network card work under dom0 of XEN. I could successfully install the ndiswrapper drivers and could see it being associated to AP. However I amn''t able to transmit any packets outbound. I could''nt even ping the AP. Interestingly when I run a tcpdump I can see few broadcast packets on the interface. There no rules in iptables for nat or

I tried it w/ mine as well and it hung up on me because I just have Voicemail running not Voicemail2. It seems as though you have Voicemail2 because it's trying to play the Unavialable message. Just a thought though. Does it do the samething w/ [qout-phillyq] exten => 0,1,Voicemail(u1) exten => 0,2,Goto(default,s,1) Tim Thompson http://www.amatechtel.com (806) 722-2227

Hi All Just been looking at stats for one of my sites, and I'm conserned about the number of error cause codes being returned from the telco for example 12000 calls processed 131 are cause code 31* normal. unspecified.* 139 are cause code 28 * invalid number format (address incomplete).* 112 are cause code 1 *Unallocated (unassigned) number. *this adds up to about 3% of calls not

Does anyone have a SEP<MAC>.cnf.xml file that works with asterisk? I have the SIP firmware loaded on my Cisco 7970 but the status log shows errors parsing the config. I copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- ______________________________________________________________ Rock River Internet

We wrote and submitted a patch to do this. Just modify app_directory.c and recompile. It adds a new flag "b" to the directory( ) app where you can have it use both first and last name. -= Info about application 'Directory' =- [Synopsis] Provide directory of voicemail extensions [Description] Directory(vm-context[|dial-context[|options]]): This application will present

hi, I use automounter to mount to many hosts /net/<hostname>/dir When the hostname is unavaliable it seems NFS gets stuck. I would like to set a NFS timeout (say 60 secs). If not available then error. I need to use hardmount but is there such a thing as NFS client timeout? -- --- Get your facts first, then you can distort them as you please.--

Hi, I have an odd problem where when a call is made to any landline phone the callerid number shows up on the caller id device with no problems. However when I call a cell phone from Sprint or Verizon it always shows up as unavaliable. I've tried multiple cell phones from each carrier. Though today I called a t-mobile phone and the callerid came through fine. Is this an * level problem

Has anyone tried to compile Samba on Windows 2000? I know it's a weird idea, but I don't like the fact that W2K shares c$, d$, RPC$, etc. With Cygwin, I didn't get past configure, it complained that locking was unavaliable and bombed out. Is it technically feasable to get Samba running on Cygwin? (I disabled Server&Co in my Services already :} ) Thanks! --Brandon Bergren (Please

I have an icecast server (B) which relays another icecast server (A). Server B is configured to play a local fallback file when server A is unavaliable. Both are version 2.3.3 When I listen to server B (the relay), while server A is turned off, all works fine ie I can listen to the fallback track without problem. However if I listen to Server B while server A is running then I quickly get: INFO

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

Hello, I have the exact same problem as mentioned in this post,(http://lists.xensource.com/archives/html/xen-users/2005-06/msg00221.html) (or see the bottom of this post) Yet the conversation was continued elsewhere, is this a bug or somthing stupid.. or more importantly is there a solution? Thanks! ~Rob ---(Quote From Archives)--- "Hi Folks, I am trying to get my wireless network card