similar to: Security tests

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing with a Grandstream BT200 telephone and, according to I read, it has a LED that blinks if for that extension messages were left. In "Voice Mail UserID", under the "ACCOUNT" tab, I put *100 that is the extension in which my Asterisk answer the voicemail service and if then I press MESSAGE button, the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

I have just upgraded my Polycom 501's from 1.6.2.0041 to 2.1.0.2708 to get the microbrowser. Almost everything is fine except when receiving calls from a BT200 (1.1.14 and earlier) the Polycom rings but when answered, drops out and the BT200 gets a busy tone. I have many PAP2T's and SPA3000's etc and they all cal call the Polycom without problem. Does anyone know what could be going

Hi everybody and sorry by the insistence. Nobody has working Tinc Server over a Raspberry in an environment in production? Best regards and sorry again, Ramses De: Ramses II [mailto:ramses.sevilla at gmail.com] Enviado el: martes, 08 de octubre de 2013 17:59 Para: tinc at tinc-vpn.org Asunto: Tinc Server and Raspberry PI (Rev. B). Dear gentlemen, I need configure a VPN

If anyone can help im trying to get my jabra bt200 or bt250 headset working with chan_bluetooth. They seem to pair ok but they will not come out of "Negotiating" state. I get this on first start of *: [HS] jabra > AT^SPTT=? [HS] jabra < ERROR If anyone can be of help please advise, im pulling my hair out on this one. Thanks Jason Price NOTES: JABRA BT200/250

Hi All, I have working asterisk version 1.4.24. I have a blind transfer issue with grandstream bt200. I have updated the latest firmware to the phone. The phone is sending the *refer* to asterisk but asterisk is not able to connect with the another call that i have checked in sip debug. I am using transfer button of the grandstream phone. Can anybody provide help for this issue? Thanks in

https://bugs.freedesktop.org/show_bug.cgi?id=57278 Priority: medium Bug ID: 57278 Assignee: nouveau at lists.freedesktop.org Summary: [xf86-video-nouveau] flightgear crash when loading scenary Severity: critical Classification: Unclassified OS: Linux (All) Reporter: king.infet at gmail.com

Hi I have a server with the last version of asterisk branches, zaptel branches, 2 Digium Card with TDM800P 16 HPEC channels (Echo Cancelation), 40 Grandstream BT200 and 10 Grandstream GXP2000. zapata.conf echocancel=64 rxgain=0 txgain=0 when i place a call o receive a call, I finish a sentence i hear a ssssssss, AND when the both side talks at the same time i have choppy audio. Any

Hi All Sorry to be a bother again but seems like I just cant get away from the problems. This time my problem is that *sometimes* a user cant transfer a call from one extension to another, I have narrowed down the problem to it only happening to calls from outside the internal system. The wierd thing about the problem is that it comes and goes one moment the user can transfer, and the next

Hi! I was searching for info about this issue and found almost nothing. So, let's go directly to the matters... - Problem: AutoCAD says "You do not have permission to save to this location." when trying to save the file in the samba share dir. (This problem only occur with AutoCAD.) - Scenary: Running AutoCAD in a WinXP/Win7 PC, opening a DWG AutoCAD file from samba share dir in

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

These is my scenario. Asterisk 1.4.16 Zaptel 1.4.8 Grandstream BT200 Grandstream GXP2020 Grandstream GXP2000 For some reason the end user ask to configurate son direct access like *01,*02,*03 thru *78. After they began to use these direct access, I cant place a 3 way CONFERENCE. I remove the direct access, but i dont know if one of them block the CONFERNCE. Do you know if i can make

Hello, I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is obscure to me. In particular, in a day I got the "Deletion of invalid timer" message almost ten times from one phone, which has some call problems. Can someone point me to a resource on BT200 error codes? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l.

Hello, I have a problem with a grandstream IP Phone. The SIP autentication is OK, but when try to call someone I get the message --> WARNING[14281] chan_sip.c: Error in codec string '=audio 5004 RTP/SAVP 3' I tried to change the CODECs (ulaw, alaw, GSM, etc), the result is always the same. Tried to change the RTP port but the result is the same. The grandstream IPhone is behind a

Hi Guys: I'm using Asterisk with Addpac 2620 as gateway, internally I'm using Grandstream BT200, unfortunately when I called to external phones (PSTN), and I have to choose some extensions, the Phone don't dial the extensions, I believe that DTMF relay in ADDPAC is not working well. I'm using RFC 2833 and ALaw for SIP Channel (Between ASterisk and ADDPAC). Someone have any

Scenary: Fedora Core 5 + VirtualimPro Ruber RPM''s : ruby-libs-1.8.4-8.fc5 ruby-rdoc-1.8.4-8.fc5 ruby-mode-1.8.4-8.fc5 ruby-libs-1.8.4-8 ruby-debuginfo-1.8.4-8 ruby-ri-1.8.4-8 ruby-1.8.4-8 ruby-rdoc-1.8.4-8 ruby-devel-1.8.4-8 ruby-mode-1.8.4-8 ruby-tcltk-1.8.4-8 ruby-1.8.4-8.fc5 ruby-irb-1.8.4-8.fc5 ruby-ri-1.8.4-8.fc5 ruby-devel-1.8.4-8.fc5 ruby-irb-1.8.4-8 ruby-docs-1.8.4-8

Just by the way, do anybody knows if call pickup of a call to a sip extension from a mgcp phone is supposed to work (even if sip keeps ringing). The scenary is as follows: 3@mgcp02 (ext 136) calls sip/julia (ext 133) and after It starts ringing 2@mgcp02 (ext 135) dials *8. Nothing happens, only 135 gets congestion tone, 133 keeps ringing and in the asterisk console I get: --

Hi This bug is really crazy, please help me In the follow scenary ATA-186 -> SIP -> Asterisk -> SIP -> ATA 186 : No DTMF gets through * in outbound mode, Sip conf [204] type=friend username=204 secret=somesecretpassword host=dynamic canreinvite=no ; The follow line don't work dtmfmode=rfc2833 nat=1

Hi for all I'm making some test and I can see an incorrect behaviour. I have two asterisk with an IAX2 trunk. In asterisk 1 I have a queue and an agent and, in Asterisk 2 I have three clients. When the clients make calls to an asterisk 1, its calls entry in the queue. While they are waiting, an agent login into the queue. I'm waiting and waiting and waiting but the clients never contact

Hello all, Please send a Cc: to me in addition to the list. I am not subscribed and this will make it easier for me to follow the thread and reply. Thanks. My scenary is as follows: I have a Linux server with OpenSSH 3.7.1p2 installed and this server acting as an LTSP. Some PCs (all Pentium+ classes) acting as X-Terminals connects to this server via PXE boot --> DHCP --> booting