search for: 18185

...ce provider my asterisk switch seems to be bridging the two calls. As a result I loose all accounting information. All I get is the call setup time (15 or 20 seconds). How can I either make asterisk not bridge the call, or keep correct tabs on the call accounting for me? Mar 23 21:55:24 VERBOSE[18185] logger.c: -- IAX2/calleveryone-10 stopped sounds Mar 23 21:55:24 VERBOSE[18185] logger.c: -- IAX2/calleveryone-10 answered IAX2/gnutech-6 Mar 23 21:55:24 VERBOSE[18185] logger.c: -- Attempting native bridge of IAX2/gnutech-6 and IAX2/calleveryone-10 Mar 23 21:55:24 VERBOSE[18185] logge...

Hi, I just purchased a 9331i phone. It's great! However, having a slight issue with line 2 and 3. line 1 works fine. But when I try to make a call with line 2 or 3 I get a fast busy from the phone. Asterisk shows this: Mar 23 15:19:12 NOTICE[18185] chan_sip.c: Failed to authenticate user No User <sip:No%20User@mysipserver.myhost.com:5060>;tag=34c1cd2bf55cb8d Can anyone with a 9331i comment on this? What do I need to do to get line 2 and 3 to authenticate outgoing?

...010%252F10%252F21%252Fasterisk-users%252Fblind-transfer-failed-sip- refer-method.html> http://asteriskfaqs.org/2010/10/21/asterisk-users/blind-transfer-failed-sip- refer-method.html <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185> https://issues.asterisk.org/view.php?id=18185 <http://www.odesk.com/leaving_odesk.php?ref=https%253A%252F%252Fissues.aster isk.org%252Fview.php%253Fid%253D18185%2523c128539> https://issues.asterisk.org/view.php?id=18185#c128539 How to do this: We make a call from number to DDI, as...

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Can someone show me how to patch the new asterisk 1.8.1-rc1 code with the sip transfer fix described @ https://issues.asterisk.org/view.php?id=18185 ? I tried 'svn co -r 295866 http://svn.asterisk.org/svn/asterisk/trunk/[apps|include|main]', but the files I checked out didnt seem to be ones i was looking for. jr

...for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix reloading of peer when a user is requested. Prevent peer reloading from...

...for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix reloading of peer when a user is requested. Prevent peer reloading from...

Hi, In my blog i display a post. I want to display a link tot he previous post if there is 1 and a link tot he next post if there is one. So i have in my controller def post @post = Post.find(params[:id]) @post_pages, @posts = paginate :posts, :per_page => 1 render(:layout => false) end And the following code in post.rhtml <% if @post_pages.current.next %> <%=

...en possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix bugs in saying numbers using the Swedish language syntax (Closes issue...

...for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Linux and *BSD disagree on the elements within the ucred structure. Detect...

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

...en possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Fix bugs in saying numbers using the Swedish language syntax (Closes issue...

...for multiple servers. (Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) * Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call. (Closes issue #18171. Reported by: SantaFox) (Closes issue #18185. Reported by: kwemheuer) (Closes issue #18211. Reported by: zahir_koradia) (Closes issue #18230. Reported by: vmarrone) (Closes issue #18299. Reported by: mbrevda) (Closes issue #18322. Reported by: nerbos) * Linux and *BSD disagree on the elements within the ucred structure. Detect...

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

Hi, Alan Thanks for making it clear. But I was more confused now :( I tested on a simple program and used both gcov and lcov to get branch coverage. The code and build commands as below: *Example simple.cc* #include <string> // If not comment this line, the branch coverage won't reach to 100% // #include <iostream> int main(int argc, const char* argv[]) { std::string str =