similar to: snom 360 auto answer

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using

I was trying to get my polycom phone to auto answer. I added this to the dialplan. Used a different phone to call "22" and the phone rang it did not auto answer. Did I miss something? exten => 22,1,SipAddHeader(Call-Info:=\;answer-after=0) exten => 22,n,SipAddHeader(Alert-Info: Ring Answer) exten => 22,n,Set(__SIPADDHEADER=Call-Info:\;answer-after=0) exten =>

I am looking to setup paging using the auto answer feature on the Grandstream GXP2000. I am thinking I will follow the method as described here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page I will setup the 4th account on the phone to auto answer. Does anyone else have a method that works better? I also looked at the allpage AGI written on Voip-Info. But it seems

I have thsi code in a dial plan. The purpose of which is to set distinctive ring tones for internal and transferred calls. exten => _.,1,Noop(CALLERID_ALL=${CALLERID(all)}) exten => _.,n,Set(CallerIDNum=${CALLERID(num)}) ; This just shows a list of interesting variables and their values ; Comment it out when finished debugging ;include => macro-dumpvars ;exten =>

Hello, I have uploaded a custom ringtone to our SPA-922's for the purpose of sounding like a door bell chime when the doorbell is pressed. I am using __alert_info to set this ringtone. It appears that I can only set the ringtone via alert_info for the ringtones that come from Linksys. Has anyone else seen this issue? I am doing the following: exten => 100,1,SetVar(_ALERT_INFO=doorbell)

Steve Edwards <asterisk.org at sedwards.com> schrieb: > You can fiddle with the ring tone by phone specific configuration and > phone specific SIP headers (sipaddheader(Alert-Info: ...)). > > These seem relevant: > > http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the > discussion looks relevant as well). > >

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

Hi I am experiencing some weird behaviour. Calls get disconnected random. There is no error in the log files. Sometimes I can talk over 30minutes+ and it is fine. Just earlier I was only able to talk 2 minutes per session and get disconnected. All I hear when this happens is a fast busy. My set up is this: 8 * Grandstream Budge Tone 101. 4 * X100P cards. Compaq 1Ghz ML Server. I am running

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good evening, I have something strange, when I add an ALERT_INFO variable to a ring group, the invite generated contains 2 lines with Alert-Info and my phones return a 400 Bad Request... I've checked in my config files, there is only one line with Set(__ALERT_INFO..... Any idea?? PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4

I am dealing with a FreePBX generated dialplan. I have been following the processing traces attempting to make use of the advice I received here respecting setting a custom ring tone. I have discovered that the context I am using for incoming calls is not used at all during a blind transfer. Thus setting a third ring tone for that situation inside that context is an impossibility. I know now

> Hi Steve! > > Thank you very much! > It seems to run! > > I wrote that: > > exten => _00493513333333,n,Set(__ALERT_INFO=Bellcore-r3) > exten => _00493513333333,n,SIPAddHeader("Alert-Info:< http://www.notused.com > >\;info=alert-external\;x-line-id=0") > > and the phone rings with another melody. > Very curious is, that if I

Hello, I'm about to install SAMBA in a HA (High Avalibility) solution with Solaris 2.6 and First Watch. I have two SAMBA servers that normally runs on two machines, in case of a failure of one of the machine the other machine is supposed to take over the failed machine's services. Also the IP adress is moved (the machine will now have multiple IP adresses (on the same logical net)). So

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

Good Day I am having the following serious problem. I am running CentOS 4.1 with dovecot 0.99.11. The users are using POP3 login to retrieve mail. For the last 2 days we have experienced the following problems. I have recreated all the users and reconfigured the dovecot,but the same problems occur. Please see attached /var/log/maillog messages It appears so users login whereas others

We're using Samba 3.0.21a-1 on a CentOS 4.2 server (kernel 2.6.9-22.0.2.ELsmp); everything works more or less fine, it's functioning as the PDC with an OpenLDAP back end (setup essentially straight from the Idealx.org playbook), etc. We're trying to mount Windows shares to back them up, and getting "permission denied" errors when trying to read certain files from those

cheers, jerry --------------------------------------------------------------------- www.valinux.com VA Linux Systems gcarter_at_valinux.com www.samba.org SAMBA Team jerry_at_samba.org www.plainjoe.org jerry_at_plainjoe.org --"I never saved anything for the swim back." Ethan Hawk in Gattaca-- Return-Path: