similar to: Crash without "make valgrind"

I am trying to find the app ChanSpy for Asterisk v1.0.7. I have tried looking on VOIP-info.org's ChanSpy page (http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ChanSpy)and also referred to the link regarding bug 3836 (http://bugs.digium.com/bug_view_page.php?bug_id=0003836). I downloaded the attachments and tried to use the patch and compile the source. However, it seems that

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

IAX2/guest@switch-3.asterlink.com/996 at 1pm CDT on Thurday the 30th. If you have any topics that need to be covered please direct them to me. Thanks, /b --- Anakin: ?You?re either with me, or you?re my enemy.? Obi-Wan: ?Only a Sith could be an absolutist.?

Here's something I haven't been able to discover as of yet - I need to set up a "direct link" from my Asterisk box to an external line... basically I need to be able to pick up an internal extension and have it call a local phone number. This is call forwarding, I know - the question that I have is how do I set it up so that the extension always forwards. There will never be a

Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. "top" shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R 88.4 1.9 1:02 0 asterisk

I have an incoming 800-number over IAX from Teliax and I'm experiencing the large packet loss on connection. When a call comes in there is no ring tone and the first few words of the welcome message are cut off, regardless of the delay I set. Standard call (not 800-number) coming over IAX with the same provider works just fine only the tall free number. So it seems there are some packet loss

>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

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

I've got a friend who's spending 6 months on the other side of the world. So before he left I configured him a softphone on his laptop to connect to my asterisk so he can call home free of charge. Unfortunately, he just found out he has horrible internet connection. Bandwith and latency is ok, the problem is the stop almost all connections. He has to connect to a proxy server for his web

Hello all, I've been playing around with early audio, and I'm able to get some things working We have PSTN calls coming in to asterisk in SIP from a Cisco AS5300. If I do the following: Exten => i,1,Playback(ss-noservice,noanswer) Exten => i,2,Congestion(15) Exten => i,3,Hangup() The PSTN caller does not get an answered call (doesn't get billed) but hears the ss-noservice

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts

Hello everyone, I have a couple of bugs I'm trying to debug compiling asterisk with valgrind. But of course when compiled like that the bugs don't occur. What are the exact effects of Valgrind? Would there be a hit on performance running asterisk compiled with valgrind ? Thanks for you insight. -- Benjamin

https://bugs.freedesktop.org/show_bug.cgi?id=74316 Priority: medium Bug ID: 74316 Assignee: nouveau at lists.freedesktop.org Summary: [NV92] Graphical corruption on KDE shutdown/restart/log out screen QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux

We are currently in the process of replacing one of our customer mail systems with a dovecot solution. However, one of the sticking points right now is how to get the old mail to the new system. On the dovecot side, we are using mdbox storage. On the old system we are using qpopper/mbox mailboxes in the following setup. All user inboxes are in /mnt/mail/mail_spool. Only select, privileged,

https://bugzilla.mindrot.org/show_bug.cgi?id=1818 Summary: SSH2_MSG_CHANNEL_FAILURE on closed channel Product: Portable OpenSSH Version: 5.1p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

Hi, I wish to use Asterisk as a SIP server. How do I use Asterisk in a redundent network? So, if one Asterisk server fails, how does failover work? James

https://bugs.freedesktop.org/show_bug.cgi?id=64323 Priority: medium Bug ID: 64323 Assignee: nouveau at lists.freedesktop.org Summary: Severe misrendering in Left 4 Dead 2 Severity: normal Classification: Unclassified OS: Linux (All) Reporter: bryancain3+fdo at gmail.com Hardware: x86 (IA32)

I'm looking for a way to transmit a user specific variable to my dialplan If we use the example of the hair color, I was thinking of having something like: [bob] context=users host=dynamic secret=password type=friend username=bob hair=brown [lary] context=users host=dynamic secret=password type=friend username=lary hair=black And be able to access from the dialplan: [users] Exten =>

https://bugzilla.mindrot.org/show_bug.cgi?id=2390 Bug ID: 2390 Summary: PROTOCOL.key mis-describes private section Product: Portable OpenSSH Version: 6.8p1 Hardware: All OS: All Status: NEW Severity: normal Priority: P5 Component: Documentation Assignee: unassigned-bugs at

I'm using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]: