similar to: Call-leg stays on MusicOnHold forever

Hi, I am a system admin and I am evaluating using dovecot as our email server. In my test, I found that if I telneted to 993 port and did not do anything or I telneted to 143 port, sent starttls command and then did not do anything, the connection stayed forever without timeout. This will make our mail server vulnerable to DOS attack. I dig into dovecot Wiki and did not find any solution. This

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.

There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload" works only once with a bigger dialplan. If I issue "dialplan reload" again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

Hi, Here is my confing: [out] Exten => _X.,1,Noop() Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1)) Exten => _X.,3,Playback(tt-monkeys) Exten => _X.,4,Playback(tt-monkeys) Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call] Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60) Exten

Hello folks, i upgraded to the actual CVS head from yesterday (27.5.) but can not get incoming SIP calls from my provider (sipgate). If someone calls my number, my asterisk responds with the following error: May 27 21:30:21 NOTICE[1114606512]: chan_sip.c:6351 handle_request: Failed to authenticate user "<CallerID>" <sip:<CallerID>@217.10.66.11>;tag=as38e9693c I

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

Hello, I have a problem with MoH at attended transfers. - Mobile A dials into Asterisk - Asterisk dials another Mobile B - Mobile B presses *1 for attended transfer and for example 20 to dial extension 20 - Asterisk sends "Remote hold" message to Mobile A, so the carrier of Mobile A starts playing it's own music-on-hold - Mobile B hang up, so Mobile A should be connected to

Greetings. I have next feature in features.conf : send => *9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl What it does is parsing CALLERID and DNID from AGI input, performing some actions in MySQL with these values, and then running application for peer (for example, PlayBack) Sounds simple, and it really is. When my user is receiving a call (we are the B leg) and presses *9, everything

> OK, make sure that the two idmap.ldb files match and then run > 'samba-tool ntacl sysvolreset' on both machines and see if this cured > this problem. I did: root at dc1:~$ service sernet-samba-ad stop root at dc2:~$ service sernet-samba-ad stop root at dc2:~$ mv /var/lib/samba/private/idmap.ldb /root/idmap.ldb.bak root at dc1:~$ scp /var/lib/samba/private/idmap.ldb

Dear R-Helpers In R 2.5.1, the command library(Rgraphviz) fails on my Windows (XP SP2) system with error popup "The procedure entry point Rf_allocString could not be located in the dynamic link library R.dll". Thanks in advance for any suggestion in solving the error. My D. Coyne Imagination is more important than knowledge... (Albert Einstein) mcoyne@boninc.com

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXXXXXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

I am sending this problem for 2nd time. Please help. Thanks _____ From: Omar McKenzie [mailto:omckenzie@trenetinc.com] Sent: Thursday, September 08, 2005 9:57 AM To: 'asterisk-users@lists.digium.com' Subject: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" I am not able to get softphone registered (active) with * . new installation , new user

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you