similar to: problem transferring calls some of the times

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I just set up a new Samba4 AD controller, created users, etc. When I join a test workstation from our old, currently active domain to the new AD server (separate network) the join succeeds, and the user can log in the first time to be prompted with the "change your password" prompt. Immediately after changing the password, the

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten => s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten => s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten => s,3,GotoIf($["${temp}" = ""]?5) ; If not existing, goto priority 5 exten =>

-----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

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 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

Is there an ATA type device out there that has low level audio out for connecting speakers? My asterisk server is in one building, I wish to have speakers in another building and connect them up to a low level audio device that I can call into and speak. Can I connect speakers into the FXS or FXO of a grandstream HT 503? Does that work? I'd prefer not to have a PC setting there (sound

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

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 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 folks, Am new to asterisk pbx systems. I am trying to figure out what to do, I'll list and folks feel free to give feedback and advice. MAIN purpose for usage: 1.exposure to setup an asterisk box 2.get home phone service via VOIP/internet connection. tasks so far ------------------ 1. setup and install asterisk (1.4.x) --> DONE -currently configuring sip.conf

On 09/04/15 15:52, John E.P. Hynes wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi List, > > I just set up a new Samba4 AD controller, created users, etc. When I > join a test workstation from our old, currently active domain to the > new AD server (separate network) the join succeeds, and the user can > log in the first time to be prompted with the

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

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

Hi All; I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise. Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected? Regards Bilal

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle attack. In this scenary, the attacker is a virtual machine that it tries to see the SIP traffic between a PC with a softphone and a Grandstream BT200 telephone. But it draws attention to me between