similar to: voipstunt can't get call in asterisk

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an idea, but the line does not show up in CLI) [Apr 5 09:22:36] -- Executing

Hi, all I am reallty pissed with their service. I wonder if this is common problem. Firstly, all of my calls are terminated after 30s. And termination happens in a strange way. My local asterisk server does not see the disconnection, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20

Hello everyone! I am using wine 0.9.30 with openSUSE 10.2 I've tried to install and run VoipStunt, and program installs with no error, but fails to start with the following output: dodo@Locutus:~> wine "C:\Program Files\VoipStunt.com\VoipStunt \VoipStunt.exe" preloader: Warning: failed to reserve range 00000000-60000000 err:module:import_dll Library gdiplus.dll (which is needed

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial

Hi all, I've built llvm + clang on OS X 10.4 : Darwin 8.11.0 Darwin Kernel Version 8.11.0: Wed Oct 10 18:26:00 PDT 2007; root:xnu-792.24.17~1/RELEASE_PPC Power Macintosh powerpc However, make check-all fails immediately with: Traceback (most recent call last): File "/Users/useruser/LLVM/llvm/utils/lit/lit.py", line 4, in ? import lit File

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

Hi Everyone, I'm looking for rpms of nedi for centos 5. Does anyone know of a repo that has them? Regards, Ranbir -- Kanwar Ranbir Sandhu Linux 2.6.22.2-42.fc6 i686 GNU/Linux 12:58:56 up 11 days, 11:24, 1 user, load average: 0.88, 0.92, 0.84

hi, friends! i have samba on mandrake. i want to set encrypted passwords for win98 winNT clients, and also i want to set passwords synchronization to automatically update a user's regular Unix password when the encrypted samba password is changed on the system. i can change user's passwords for samba but synchronization doesn't work. here are some lines from my smb.conf and

Hi, Could somebody help me this SIP trasport? I'm receiving SIP "invite" with CLI of calling party from the SIP gateway, aster that my IVR has to answer the call. sip.conf: ========= [general] port = 5060 bindaddr = 0.0.0.0 context = incomingsip videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs allow=g729

ok! i did everything as John said and it works! " %o " is not necessary. so there must be a mistake in the book "using samba". thanks for helping slawek ----- Original Message ----- From: "John Benedetto" <jbenedet@unm.edu> To: "Rasmus Reinholdt Nielsen" <rasmus@narani.dk>; "Slawek W" <to-slawek@wp.pl>;

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

I am trying to connect to my icecast server with the following php script and I keep getting an error. Script and errot follow: $server = "www.dallypost.com:8000"; $user = "admin"; $passw = "my_pass_word"; $mountpoint = "/DallyPostRadio.ogg"; $fp = fopen("http://$user:$passw@$server/admin/stats","r") or die("Error reading

Hello All I am trying to answer incoming SIP calls, first, by dialing an extension, thence into voicemail, which works; and secondly by going straight into voice mail which does not. The extension.conf that works is like this; [incomingSIP] exten=>_.,1,Dial,Zap/2|1 exten=>_.,2,Voicemail,u5152 exten=>_.,3,Hangup the extension.conf which does not is like this; [incomingSIP]

https://bugzilla.mindrot.org/show_bug.cgi?id=3210 Bug ID: 3210 Summary: Confusing errors when pam_acct_mgmt() fails Product: Portable OpenSSH Version: 8.3p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component: PAM support Assignee:

I've setup samba to use ldap. I've propogated the directory. I've setup the kerberos realm. I can authen to samba & browse shares via uid/passw held in ldap. I cannot seem to get samba to accept kerb authen instead of uid/passw. Help...... Thanks. Read the #$@^(!*&$!* manual, and about 200 webpages. Scanning news groups, recompiling..... Grrrrr!

I got the following script from this mailing list but when i try using it doesnt really seem to work and gives me the following error. Warning: fopen(http://...@64.157.204.179:9095/admin/stats): failed to open stream: Bad file descriptor in E:\Public ftp\epakimusic\test.php on line 9 Error reading Icecast data from 64.157.204.179:9095. So could someone please help me out with this.. or if

Hello Everyone,   This is my first attempt to do something in R. As a precursor to a Willingness to Pay analysis, I want to conduct a Mixed Conditional Logit analysis but am unsure how to proceed because of some nesting within my data.   Below is some data and code that illustrate what I’m trying to do. The data are based on responses to a conjoint survey obtained during pilot testing. In the

Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i

ok well i uploaded at a webhosting and no errors are there but it aint displaying any stats.... listeners are currently connected. Currently playing: nothing is displayed with it... wht could possibly be wrong now? must be some settings of the encoder or icecast.. btw i am using winamp and sam plugin also tried it with sam broadcaster.. the one with encoder nd player ... ----- Original Message

I have seen a lot of references to SER. Currently, I have: 1 PRI to Telco 1 PRI to old PBX Several SIP phones with the intention of having approx. 200. I do have people that travel with softphones (currently X-Lite, but will be testing EyeBeam for better codec and echo cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I