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<snip> Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102 <snip> This Replaces header refers to RFC3891 which is not yet supported in Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA) This INVITE fails with : <snip> chan_sip.c: Trying to pick up 7792 at subs <snip> app_directed_pickup.c: No target channel found for 7792. If I'm dialing *87792 instead

r> <h2>Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010= ) </h2>With SIP 3.2.X firmware (available on the Polycom download site)=20 and Asterisk 1.6.1, Polycom phones now support a full featured BLF=20 showing statuses of Ringing, Inuse and Online and one touch directed=20 call pickup. <br>On the asterisk side all that needs to be done is to add a hint

Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With SIP 3.2.X firmware (available on the Polycom download site) and Asterisk 1.6.1, Polycom phones now support a full featured BLF showing statuses of Ringing, Inuse and Online and one touch directed call pickup. On the asterisk side all that needs to be done is to add a hint to the extension and enable directed pickup.

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

it work with just the main ringing phone and not have it pickup the rest of the phones in your group. That's more what you are looking for. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of mitcheloc Sent: Saturday, June 23, 2007 10:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

charge, but instead will be raised a few seconds before the power will be cut by the UPS, either because of a low battery event or a user initiated shutdown (with delay) command. The thing I don't like about that, is that by that time the shutdown sequence has apparently already started by the UPS, which means that we have no control over when the power will be cut. Also, if it indeed only

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Email eller vedhæftet fil afsendt fra din adresse (eller med din adresse som afsender) er blevet afvist fra Allerød Kommune. Spam og virus bliver typisk sendt under dække af andre afsendere og den blokerede email behøver derfor ikke oprinde direkte fra dig. (Husk dog altid at have et opdateret antivirusprogram på din computer.) Du kan evt. scanne din computer med det gratis' værktøj

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 =>

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 >

Unfortunatly it is not :(* -----Original Message----- From: Thomas Bork [mailto:tombork@web.de] Sent: Tuesday, July 01, 2003 11:42 AM To: Hindman, Mike J Cc: samba@lists.samba.org Subject: Re: RE: [Samba] Re: Winpopup message to all user on a PDC Hi Mike, > Except.. Most of my net sends don't end up working.. They fail unless I > explicitly do a nmblookup and specify the broadcast

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 =>

FreeBSD 4.1.1-RELEASE #1: Tue Oct 10 18:09:51 CDT 2000 root@test-drive.tamu.edu:/usr/src/sys/compile/TEST-DRIVE Timecounter "i8254" frequency 1193182 Hz CPU: Pentium III/Pentium III Xeon/Celeron (501.14-MHz 686-class CPU) Origin = "GenuineIntel" Id = 0x683 Stepping = 3 Features=0x383fbff<FPU,VME,DE,PSE,TSC,MSR,PAE,MCE,CX8,APIC,SEP,MTRR,PGE,MCA,CM

>Added: >>"OPTIONS=-4 -named" to /etc/sysconfig/named, restarted named. >disabled ipv6 per interface, >>Odds are you didn't actually do this, just configured them so that they >>won't pickup a GUA. U were right :D >>created /etc/modprobe.d/ipv6_disable.conf file with >>alias net-pf-10 off >>alias ipv6 off >>options ipv6

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-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 using the Debian Sarge package of Asterisk - 1.0.7 + bristuff. When I call from a zap channel or a SIP phone to another SIP phone, then dial *8 from a third SIP phone, I get 503 Service Unavailable on the third phone and I get this at the Asterisk console: Jun 28 09:01:24 DEBUG[16774]: res_features.c:1709 ast_pickup_call: No call pickup possible... Jun 28 09:01:24 NOTICE[16774]:

Getting module_request errors from SELinux. Errors being thrown by metacity sendmail.postfix cleanup trivial-rewarite local postdrop pickup All errors are essentially the same System was working well until I began to apply some basic security hardening configuration. Postfix started complaining when I made /tmp noexec, nodev, nosuid, and then did a mount --bind of /var/tmp under

Hi, according to the description of Pickup() on page http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup I can use this command to pickup a call at a certain extensions. When I try this with e.g. exten => *8200,1,Pickup(200) Asterisk tells me that the highest value for the Pickup command is 63. Wenn I enter the number of a callgroup instead of an extension, I can pickup the call.