similar to: Random disconnect of calls

On 2016-09-27 16:40, Rowland Penny via samba wrote: > On Tue, 27 Sep 2016 16:17:10 -0500 > Bob of Donelson Trophy via samba <samba at lists.samba.org> wrote: > > On 2016-09-27 13:31, Marc Muehlfeld via samba wrote: > > Hello Elias, > > Am 27.09.2016 um 18:16 schrieb Elias Pereira via samba: > > We have samba 4.1.17 installed via debian package and would

Hi, I'm testing paging using snom 360. Can someone correct my dialplan? Regards, Jason. ================================================== ;exten => _99XXXX,1,SIPAddHeader(Call-Info: Answer-After=0) ;exten => _99XXXX,n,SIPAddHeader(Call-Info: <sip:192.168.1.113>\;answer-after=0) ;exten => _99XXXX,n,Dial(SIP/${EXTEN:2}) exten => _99XXXX,1,Set(__SIPADDHEADER=Call-Info:

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

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

Thank you and sorry...There is something going wrong with the system I only sent one mail... _____ From: Kerry Garrison [mailto:kerryg@techdatapros.com] Sent: Wednesday, May 11, 2005 5:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [SPAM] - RE: [Asterisk-Users] Grandstream-Budge tone - Email found in subject This is usualy a problem with either

Hi; Have two grandstream Budge tone...Connected them to the network and able to make call to/from them. But when the coming call answered, I can not hear any voice and also my voice is not heart... I am able to hear voice only if I pressed the hold button and take the call again....This problem also Occurs in calls from x-lite to cisco7940... Does anybody has any idea or documentation

We have 3 Grandstream Budge Tone 100 phones which are being very fluid on incoming calls. They are set up as extensions 2501, 2518, and 2536. When calling out to another phone, they always identify themselves correctly. But sometimes they will respond to the wrong incoming calls. (By respond, I mean that the phone rings and if someone picks up the receiver, the call then goes thru.) For

Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages

Try as I may, I cannot get Dovecot to allow anything but blank passwords listed in a custom file I made for the purpose. It will not recognise anybody in passwd. Only using POP3 on an internal network so little danger of incursions. Once I created a password file with user names and no password, one user could login (from Outlook 2003) but another could not. Telnet test also fails to

Hi, I am looking for hardware sip phone with very good sound quality. Can anyone recommend ? I use to have Grandstream Budge-Tone 100 but I feel that the sound is not very satisfactory and volume too soft Regards ASLAY

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Greetings I have been running * for about a month now. Configuration. (5) Cisco 79xx IP phones (1) XP100P Pentium III (300mhz) 192meg memory Redat 8.0 (updated) It seems to run for about 3-6 hours, then the process stops. I have noticed, that * does not stop, if I do NOT have it register to other sip servers. (FWD and PCH). Here is are the last few lines in the /var/log/asterisk/messages

I have asterisk 1.6.2.11 talking to a cisco 1760 running ipvoicek9-mz.124-25b. whenever a call goes through the 1760's FXO or FXS (in or out) there is a 915 second maximum call time due to asterisk hanging up the call because of a "critical packet" being missed. I read doc/sip-retransmit.txt and I don't see anything there that is helpful to my situation - the asterisk box is

Hi everyone, This is off topic and is for GS technical support really but it seems that there are a lot of Budge Tone 100/101/102 users out there. I've got a Budge Tone-100 (101 - without the extra 10base ethernet connetion?) here. I changed the configuration through its web based interface and I clicked the reboot link. But then something went wrong and ever since then it doesn't

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled

Hi, I search How To transfer call between my SIP phone. I have an PSTN line (X100P) and 10 grandstream budge tone phone. For example I want : - Reveive an external call and send it to SIP/phone1. At this point no problem. - After my receptionnist want transfert extern call at SIP/phone2... I don't known how to properly transfert call.... Thanks

I finally got my Asterisk all setup and everything seems to be working except for menu interaction between my Grandstream Budge Tone 100 and my Asterisk. I have the SIP phone setup to properly connect when pressing the 'Message' button and that's working perfectly. When the menu starts, it says press 1 to read your messages, but pressing 1 (or any number) fails to send. Does anyone

I'm not sure if this is possible, but I was hoping to find an address book that runs on Windows XP that will allow me to select a phone number and send that to my Asterisk. The Asterisk system would make the call and connect the call to a SIP phone (Grandstream Budge Tone-100). Is there anything out there that can do that? Thanks, Dave -------------- next part -------------- An HTML

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.