similar to: voipstunt: "Forbidden - wrong password ..."

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

Hi, does any know why? i can make call out with my asterisk and voipstunt but i can't get call in on my voip in number i get rejected. if i use Sipura without asterisk i get in calls here is my sip.conf ---------------------------------------------- [general] useragent=nedi port=5060 context=default ;tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 language=de

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

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 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

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

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

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

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

[Jan 23 19:56:44] -- Got SIP response 300 "Multiple choice" back from 194.120.0.201 [Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to 'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:5060@default' (thanks to SIP/voipstunt-5c8c) [Jan 23 19:56:44] NOTICE[3439]: chan_local.c:483 local_alloc: No such extension/context

I have setup an Ast 11.6 host and I want to login via AJAM. I setup manager.conf, http.conf described in the docs. When I login via the AMI it works fine (see below), but when I login via AJAM the same credentials fail (see further down) Can someone tell me how to fix this? ----------- Connection closed by foreign host. root at pbx:/tmp# telnet localhost 5038 Trying 127.0.0.1... Connected to

Dear R users, I am using spatstat to analyze point patterns (tree locations). I would like to import the shapefile with the study area polygons (six total) into R and use it to create the window for the spatstat analysis. I do not simply want to use a rectangle because the study areas spread out over 40000 ha. Any suggestions would be greatly appreciated. Thanks, Charlotte Reemts Charlotte

Hello I keep getting the following error when trying to connect to the Asterisk server using AMI : $socket = fsockopen("tls://11.22.33.44","5039", $errno, $errstr, 5); Erorr on CLI : [Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Problem setting up ssl connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca [Oct 26 14:38:19]

CentOS Errata and Bugfix Advisory 2020:5039 Upstream details at : https://access.redhat.com/errata/RHBA-2020:5039 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: d70d1392ce3021f70e51c4dc35c2fa251da70e2e0b7384fc3ca8003c9c2f59bf ibus-1.5.17-12.el7_9.x86_64.rpm

I've recently come across the following results reported from the lm() function when applied to a particular type of admittedly difficult data. When working with small data sets (for instance 3 points) with the same response for different predicting variable, the resulting slope estimate is a reasonable approximation of the expected 0.0, but the p-value of that slope estimate is a surprising

Pan Bříza, to se stane, když Custom image - Pick a file from your drives(s) ... nouveau 0000:02:00.0: fifo: DMA_PUSHER - ch 5 [mediawriter[20975]] get 0020171c34 put 00201746ec ib_get 00000017 ib_put 00000018 state 8000a32c (err: INVALID_CMD) push 00406040 nouveau 0000:02:00.0: gr: DATA_ERROR 00000004 [INVALID_VALUE] nouveau 0000:02:00.0: gr: 00100000 [] ch 5 [001f7bd000 mediawriter[20975]] subc 5

The Wine development release 1.7.20 is now available. What's new in this release (see below for details): - X11 drag & drop fixes. - A few more C/C++ runtime functions. - Fixes for various memory issues found by Valgrind. - Some OLE storage fixes. - Various bug fixes. The source is available from the following locations:

Hello, First of all, sorry for my poor english level. I pass to solr indexing. All is OK, except for the inbox of one of the mailboxes. For this inbox (other folders of the same mailbox have no problems), when i do text search, there is always no response. Others mailboxes have no problems. If i do a : "doveadm -Dv fts rescan -u mybox at domain.tld", i can do one and only one