similar to: how to hung up a call immediately if it SIP response 486 "Busy Here" received

Hi. I am seeing the following in my logs. Happy to downgrade to an earlier version if you think this might be the problem, but dovecot.org is extremely slow (and has been for months) so I found it easier just to pull the latest from mercurial. Thanks. Laeeth Mar 12 20:48:39 indexer: Error: Indexer worker disconnected, discarding 1 reques ts for laeeth at laeeth.com Mar 12 20:48:39

Hi... Im problem is this, i have a asterisk server (FC8 - kernel 2.6.24) a the asterisk version is 1.4.18. If in the machine is all ok, i can stop start the asterisk service no prob, my problem is when in another server (in my case, debian etch 4) using the ssh the stop service is ok, but the start service dosend finalise. Like this: ssh etx9 'sudo /etc/init.d/asterisk stop' operacao at

If we get an incoming call I can edit the callerID provided to add the leading '90' and set the name so that sales calls can be identified according to the number called. If however the callerID is unavailable then setting the callerID name or number fails (it shows as unavailable on the phone). This is the call log from such an incoming call without callerID. == Spawn extension (voip,

Todd How do you have the context in the phones sip configs set? Bryant From: "Todd Reese" treese65 at gmail.com Hi all, I've been have problems with getting this system on line and would like to acquire some help with the extensions.conf. My current problem is that the phones won't dialout.on the VOIP lines listed as dialout1, dialout2, dialout3. This version of asterisk

Dear All I have a problem with a TDM400 card with 4 x FXS modules. The card carries extensions only and there are no incoming lines. I can make a call to the extension on this card with no problems. However, when I try and call out I just get a busy signal. I also get an error message (as shown at the bottom). Is this a problem? Configs below: [root at asterisk etc]# more zaptel.conf

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

On Tue, Jan 7, 2014 at 8:48 PM, Sean Silva <chisophugis at gmail.com> wrote: > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' | wc -l > 706 > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' ./test | wc -l > 317 Wouldn't this also catch commits to code generation that added tests as well? Diego.

Has anyone had any problems with the voicemail app in the last weeks cvs? I was using a checkout from 10-31 and everything worked fine, but the last few days I get a hangup right after it calls the voicemail app. No sound, just an immediate disconnect. Nothing has changed in my extensions.conf, etc. Here is what I have in extensions.conf: [macro-oneline] exten => s,1,Dial(${ARG1},20,tr)

Greetings, DIAX seems to work well placing calls, but I can't actually receive a call . Here, DIAX (x305) "registers", then I use a sip phone to place a call to DIAX (which definitely is not in use by me at debug time, but it is idle on my desktop.I think), and then * goes to vmail. Here's the debug output: affinity*CLI> iax debug IAX Debugging Enabled Rx-Frame Retry[N/A]

Hello all, I've got a T100P connected to an Adtran TA750 with a T1 crossover... This connects to a patch panel with phone ports. The Adtran is fully populated with FXS cards. All I get on any phone port is a fast clicking noise... No dialtone. Asterisk 'sees' the card, (the channels show up in /proc/zaptel). Incoming calls are routed to the zap/x channel, but no ring. I'm

On Wed, Jan 8, 2014 at 4:07 PM, Sean Silva <chisophugis at gmail.com> wrote: > > > > On Wed, Jan 8, 2014 at 7:58 AM, Diego Novillo <dnovillo at google.com> wrote: >> >> On Tue, Jan 7, 2014 at 8:48 PM, Sean Silva <chisophugis at gmail.com> wrote: >> >> > sean:~/pg/llvm/llvm % git log --oneline --since='1 month ago' | wc -l >>

I have a file in stata format, which I have read in, and I am trying to create a text file. I have exported the data using various delimiters, but I'm unable to read it back in. I originally read in the file with: library(foreign) myData <- read.dta("mydata.dta") I then exported it with write.table using comma, tab, and exclamation marks as a delimiter. When I was unable to

Still new to dovecot, I'm seeing a couple of errors in my log file indicating corruption of index files. Error: Corrupted index cache file/home/xxxxxx/mail/.imap/sent-mail/dovecot.index.cache: Broken virtual size for mail UID 67: 1 Time(s) Error: FETCH [] for mailbox ~/mail/sent-mail UID 67 got too little data: 6203 vs 7478: 1 Time(s) This account is used by multiple users, but it is not

Skipped content of type multipart/alternative-------------- next part -------------- ############ # amd BOX # ############ ## Step 1 ## Bob(ext. 6202) place a remote IAX2 call to the operator (ext. 6302) ## Reminder : _62XX are register on 'amd' and _63XX on 'dell' -- Executing SetGroup("SIP/6202-d193", "IAX") in new stack -- Executing

I'm a newbie with a TDM11B. I've read the FAQs about linking /usr/src/linux-2.6 to /usr/src/linux-2.6.8-1.521 and /lib/linux-2.6 to /lib/linux-2.6.8-1.521 but still get a million errors and eventual abort during compile. Could someone point me in the right direction? I do a yum update every day and I'm using the CVS from 9/14/04. I'm also using an ASUS CUSL2 system board.

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

As first discussed in the giant "New LLVM git repository conversion prototype" thread, I've pushed a git tag that allows git-describe to be used for generating human readable commit names on master. The tag is "llvmorg-10-init", to be interpreted as "the initial commit with major version number 10". The idea is to add similar tags along with future version bumps.

I'd like to revisit this. As a person who spends a fair amount of time monitoring our VS 2013 buildbots, I would say that I am ready to throw in the towel on MSVC 2013. Since this discussion, I have committed five (!) workarounds for MSVC 2013: # in llvm $ git log --author=rnk --grep=2013 --after='Aug 4 2016' --oneline 21a8ade Fix the MSVC 2013 build by using Elf_Word instead of