similar to: Unable to use 'valid users' from Active Directory

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

I'm trying to add a second extension to my setup. The second device is able to successfully connect to the Asterisk server. I am unable to contact extension 101 from 102 and vise-versa. Also are my context setup logically or is there a better fashion to organize them? My error is at the bottom. Here is the extension.conf [default] ; ; By default we include the demo. In a production system,

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

My latency is kind of high and the voice delay is noticeable. The Asterisk server is on a dedicated host outside of the network. I am performing PAT/NAT using a Cisco router. ns1*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.116

I was thinking about complicating my Voip setup by adding CME. I found this example here: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration and here: http://www.pasewaldt.com/cme/cme_index.htm Would anyone like to comment on their experiences using CME with Asterisk... I would like one of my Cisco phones to remain SIP connected directly to my Asterisk system. The

Is there a way to include: _NXXNXXXXXX _NXXXXXX _011. _911 into my current plan:

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing

I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide

Dear Help, After loading the pd.Citrus library and checking the DataFrame, I ran > the R code for: > > 1) 'oligo' > > > > {> library(pd.citrus) > Loading required package: RSQLite > Loading required package: DBI > > data(pmSequence) > > > show(pmSequence) > DataFrame with 341730 rows and 2 columns > fid sequence > <integer>

I would like to create a simple menu that would allow a caller to decide whether they want to leave a message or be forwarded to another number (i.e cell phone). Thanks in advance for any insight. Here's my current extension.conf [general] static=yes writeprotect=yes [globals] [default] exten => 101,1,Dial(SIP/101,20) exten => 101,n,Voicemail(101 at default) ;This automatically

I had this problem some last year and never got it figured out. Now it is bugging me. It seems that sometimes when a student writes his/her file to a directory, it will not keep the correct group. It puts his/her main group as the group owner and that fouls things up. Here is what I have. Unix Permissions /school 3777 admin.teacher /school/bhs 3777 admin.teacher /school/bhs/reese

Hi, I'm trying to read in a bunch of CSV files into R where many columns are coded like $111.11. When reading them in they are treated as factors. I'm wondering if there is an easy way to convert them into numeric in R (as I don't want to modify the source data)? I've done some searches and can't seem to find an easy way to do this. I apologise if this is a trivial

Hi, I''ve configured mongrel_clusters behind an Apache 2.2 proxy using named virtual host. Session are saved as ActiveRecordSession. But the cookies created on client side doesn''t correspond to session data saved in database (keys are different). The RoR app react just like it doesn''t have a session at all. If I don''t use Apache as a proxy/load balancer

I have two identical disks /dev/sdb1 == source /dev/sdc1 == dest When I rsync the source -> dest rsync -a /tree2/ /tree2m the destination size id considerably larger. /dev/sdb1 17654736 6513064 10244848 39% /tree2 /dev/sdc1 17654736 7691192 9066720 46% /tree2m When I look down into the directories I find the following /tree2 ... SOURCE [root]# du -sk

Hi All, I'm stumped on this and I looking for some clues to fix this. This is a new install of Slackware 12.1 onto an IBM x330 Server. Asterisk 1.4.20.1 plays the wav files and the Cepstral_Allison Swift just fine, but when I play the gsm files the audio quite choppy. And, the files produced from the MixMonitor don't even record any audio other than noise. I have a hard drive from

Hello All, I am having an issue with AAF and sorting results of a search. Right now, I have results being split onto pages of 10. The results are being sorted alphabetically, but not across multiple pages - it''s just sorting the 10 it pulls down on each page. I noticed another post from April regarding this same issue (http://www.ruby-forum.com/topic/62993#66934) where the issue was

Hello everyone, I did some initial tests and it seems that mongrel does not support byte-range requests. Is this correct? The reason I ask is that the iPhone requires byte-range requests to work in order to stream audio or video from a web server. Thanks in advance, alan

While I was helping a SAS-using friend with an analysis I noticed some differences in the multivariate test statistics, approximate F statistics, and p-values in the manova function using R and proc GLM using SAS. The univariate coefficients are identical. Is there a reason to expect R and SAS to give different results? Thanks, Bill Kristan.