similar to: newbie question about PBX Call Pickup

Hi there asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs; here some info: *CLI> realtime load sipfriends name 104 Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104' Jan 13 11:52:21 DEBUG[8928]:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load => res_parking.so load => chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ddoes it feasible with * to capture a call? when arrives a call, floor bells ring and everyone can hear them in the company, then everyone can answer 'capturing' the call m. - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 IAXTel: (700) 350-1234 -----BEGIN PGP SIGNATURE----- Version:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, i'm in debian sid 3.1 with kernel 2.6.7, * last cvs & chan_capi 0.3.4b; nt1+ with 2 bri in ptmp (http://www.voip-info.org/tiki-index.php?page=DDI) i tried to install avm c4 following step by step http://www.voip-info.org/tiki-index.php?page=Asterisk%20How%20to%20connect%20with%20CAPI step 1. i compiled capi 2.0 support in kernel

Hi, Anyone knows why there's a limit of 32 callgroup/pickupgroup in * ? It is coded as unsigned int but there's an hardcoded "if( X > 31 )" like line. IMHO, 32 group is very low and I wonder what impact it would have to increase it to 2^16-1 . Anyone?

hi, is there anybody who was able to setup call pickup with a snom phone? searching through the web brought up this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+snom section "call pickup" but this doesen't seem to work with current releases of the snom firmware (and looking through the patch of easywe it never worked very good at all) current snom firmware

Hello all Sorry for the repost, but I believe this is essential for Aastra phones to use directed pickup with BLF in 1.2 (and it's still experimental in 1.4 I think). pickup-mgernoth-2006-07-28.patch.txt If anyone knows otherwise than that this is still required in latest 1.2 or 1.4 please reply with success/failure stories. Any info would be appreciated. We can get directed BLF pickup

Hey there list ! I'm receiving negative feedback when people try to pickup another ringing phone by pressing *8 on there own Grandstream device. These are my setting that should make pickup possible : all my sip-clients (Grandstream) have this in their config (sip.conf) : callgroup=1 pickupgroup=1 canreinvite=no qualify=yes So they are all in the same pickupgroup... This the

I'm trying direct pickup feature but it doesn't work. Wiki says: "Directed pickup, this allows you to pickup a call at a specific extension, maybe you're in another office and you hear a phone ringing and wonder if it's yours. You dial the pickup number and your extension, and the call will only transfer if it is your extension." Instead group call pickup (*8) works.

Hi, I'm banging my head over this. Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to enhance BLF with Directed Call Pickup : basically, SIP hardphone (here a Thomson ST2030S) is configured to send an INVITE message whenever a BLF is pressed while blinking. The INVITE is build with the extension number (attached to the BLF that was blinking and pressed)

Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040420/952ce2d4/attachment.htm

What's the proper syntax for implementing directed call pickup? Running cvs-head from today (9/24/05 including Mark's fixes), and tried: exten => *99,1,Pickup(${EXTEN:3}) but that does not seem to work, and there isn't an example in the configs directory. 'show application pickup' suggests the above should work with our sip phones, but apparently I'm missing

Hello Folks. I'm wondering if anyone has any helpful hints. I recently upgraded to 1.4.11, and I'm having problems with pickup, both directed, and the pickup feature. My server is on the public internet, and all phones are behind a NAT router, somewhere else on the public internet. When a ringing phone is picked up by another phone, you have audio for a few seconds, then the call is

I've tried the directed pickup like explained in http://www.voip-info.org/wiki-PBX+Call+Pickup but I've got no way doing it work. When I call the ringing phone it use the second line of that phone Group call pickup works perfectly. Is there some special configuration to do to make it works? Best regards, ?ric

Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? thx PJ exten => _*7.,1,Pickup(${EXTEN:2}) console log, when trying o pickup ringing line 324 (h323), from skinny phone (953) -- Executing

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

I am probably just being stupid again, but... I have some non-SIP phones which are set up for doing transfers by DTMF, by simply adding T or t to the appropriate Dial options. This works quite well in general. They can also do non-directed call pickup with *8. However, after a call pickup they can't transfer the call by DTMF -- there is no Dial command where I can add the t or T option. How

I'm having trouble with call pickups. Suppose ring group is 100 and has extensions 101 and 102. Someone calls 100, 101 rings and 102 wants to pick the call up. If 102 dials **100, call pickup works. If 102 dials **101, call pickup fails. In my dialplan I have: exten => **101,1,NoOp(pickup extension) exten => **101,n,Pickup(101) exten => **101,n,NoOp(pickup group) exten =>

Dear Sir, I'm trying to configure the SMS capabilities of my provider (Telecom Italia) with my asterisk. I tryed with a normal phone (SMSC number 42100) and all is working fine, incoming and outgoung SMS. Now I'm trying to configure my asterisk for support SMS, the incoming messages are working fine (incoming number 042111) dialplan match and SMS recived. I'm getting troubles in

Looks to me your server is hacked through the webserver of website. Stop apache, flush the postfix queue, are there new mails entering you postfix queue? If not, hunt down the leak, if it does, its not apache2 ;-/ Greetz, Louis > -----Oorspronkelijk bericht----- > Van: HomerWSmith at lightlink.com > [mailto:owner-postfix-users at postfix.org] Namens Homer Wilson Smith >