[asterisk-users] call pickup problem

2007-08-29 Thread aris skizas
i have TB instaled and i cant get call pickup when another phone rings
i tried ** , *8 , *8# , **+ext but nothing  seems to be ok.on extention menu
i put call pickup=1 and call group=1 but nothing look at my
features.conf;
; Sample Parking configuration
;

[general]
; do not manually enter parkinglot config information, use the parkinglot
module
;
; the parking_additional.inc file is auto-generated by the Parkinglot
Module, do
; not hand edit that file
#include parking_additional.inc
#include features_general_custom.conf

[applicationmap]
#include features_applicationmap_additional.conf

; *** IMPORTANT NOTE ***
; The original blindxfer was '#', and has been changed to '##' to avoid
; issues with sending DTMF '#' to remote parties.

[featuremap]
blindxfer => ##; Blind Transfer
disconnect => **; Disconnect Call
automon => *1; One Touch Record
;atxfer => *2; Attended Xfer


please tell the right steps for make it working

thank you
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[Asterisk-Users] Call Pickup Problem

2005-01-06 Thread Tim Leeland
I'm having a problem with the call pickup with the latest CVS.  Before I
updated to the latest CVS it was working fine.  Now, whenever anyone
tries to pickup a call using *8 it dumps all calls going on at the time
and hangs up on the incoming call.

Tim
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[Asterisk-Users] Call Pickup problem in Asterisk with SIP phones

2004-06-09 Thread Nik Martin
I'm having a tough time getting call pickup to work on *.  Here's my
configuration:

X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image

A call comes in, and * picks up and presents a menu. Caller chooses
extension, (in this case ext 103, SIP/wsmith)

Wsmith is sitting in my office, hears his phone ringing, picks up my phone,
gets dial tone, and presses *8.  He gets a reorder (fast busy) on my phone,
and his phone continues to ring (he then curses loudly, and goes racing down
the hall to try to catch the call)

In * , I get a 

Jun  9 15:45:14 NOTICE[229391]: chan_sip.c:5417 handle_request: Nothing to
pick up

I turned on SIP debugging, cleaned out all the Sip register messages that
were flying about while debugging, and present the logs here.  My version is
CVS-05/24/04 

My zapata.conf looks like:

group=1
callgroup=1
pickupgroup=1-4
context=NuFone-Outgoing
signalling = fxs_ks
callprogress=no
callerid="Radiance Technologies" <(251)-445-0045>
usecallerid=yes

My SIP.conf looks like:

sip.conf[]  0 L:[105+37 142/142] *(3505/3516b)= c  99 0x63
dtmfmode=inband
mailbox=102
context=Outgoing
callerid="Dean Li" <102>
username=dli
secret=rad1ance
pickupgroup=1

;the ringing SIP phone:
[wsmith]
type=friend
host=dynamic
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.108
dtmfmode=inband
mailbox=103
context=Outgoing
callerid="Walter Smith" <103>
username=wsmith
secret=**
pickupgroup=1-4

;The phone attempting the *8
[nmartin]
type=friend
host=dynamic
insecure=no
nat=yes
canreinvite=no
qualify=1000
;defaultip=192.168.30.100
dtmfmode=inband
mailbox=105
context=Outgoing
callerid="Nik Martin" <105>
username=nmartin
secret=**
pickupgroup=1-4
callgroup=1



The SIP debug:

pbxMobile*CLI> 
-- Starting simple switch on 'Zap/1-1'

pbxMobile*CLI> 
-- Executing Wait("Zap/1-1", "3") in new stack

pbxMobile*CLI> 
-- Executing Answer("Zap/1-1", "") in new stack

pbxMobile*CLI> 
-- Executing NoOp("Zap/1-1", ""MOBILE, AL" ") in new stack

pbxMobile*CLI> 
-- Executing Wait("Zap/1-1", "1") in new stack

pbxMobile*CLI> 
Jun  9 15:45:02 WARNING[2211866]: chan_zap.c:3073 zt_handle_event:
Ring/Off-hook in strange state 6 on channel 1

pbxMobile*CLI> 
-- Executing BackGround("Zap/1-1", "radiancewelcome") in new stack

pbxMobile*CLI> 
-- Playing 'radiancewelcome' (language 'en')

pbxMobile*CLI> 
11 headers, 2 lines
  
8 headers, 0 lines

pbxMobile*CLI> 
  == CDR updated on Zap/1-1

pbxMobile*CLI> 
-- Executing Dial("Zap/1-1", "SIP/wsmith|20|tT") in new stack

pbxMobile*CLI> 
We're at 172.31.30.3 port 15418

pbxMobile*CLI> 
Answering with preferred capability 4

pbxMobile*CLI> 
Answering with preferred capability 2

pbxMobile*CLI> 
12 headers, 9 lines

pbxMobile*CLI> 
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.3:5060;branch=z9hG4bK5cc08566 From: "MOBILE, AL"
;tag=as05f4b37a To: 
Contact:  Call-ID:
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:
Asterisk PBX Date: Wed, 09 Jun 2004 20:45:09 GMT Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 181  v=0
o=root 20260 20260 IN IP4 172.31.30.3 s=session c=IN IP4 172.31.30.3 t=0 0
m=audio 15418 RTP/AVP 0 3 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -  (NAT) to 172.31.30.11:5060

pbxMobile*CLI> 
-- Called wsmith

pbxMobile*CLI>  
Sip read: 
SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" ;tag=as05f4b37a To:
 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact: 
Content-Length: 0  
10 headers, 0 lines

pbxMobile*CLI>  
Sip read: 
SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.31.30.3:5060;branch=z9hG4bK5cc08566
From: "MOBILE, AL" ;tag=as05f4b37a To:
;tag=000b46e9ae7e485f2abff4bc-43940b23 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004 20:48:00
GMT CSeq: 102 INVITE Server: CSCO/6 Contact: 
Content-Length: 0  
10 headers, 0 lines

pbxMobile*CLI> 
-- SIP/wsmith-7e27 is ringing

pbxMobile*CLI>  
 

pbxMobile*CLI>  
Sip read: 
INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP
172.31.30.7:5060;branch=z9hG4bK2408959c From: "105 - Nik Martin"
;tag=003094c4481f49565aff56ad-226e8953 To:
 Call-ID:
[EMAIL PROTECTED] Date: Wed, 09 Jun 2004
20:47:30 GMT CSeq: 101 INVITE User-Agent: CSCO/6 Contact:
 Expires: 180 Content-Type: application/sdp
Content-Length: 244 Accept: application/sdp Remote-Party-ID: "105 - Nik
Martin"
;party=calling;id-type=subscriber;privacy=off;scree
n=no  v=0 o=Cisco-SIPUA 24482 2915 IN IP4 172.31.30.7 s=SIP Call c=IN IP4
172.31.30.7 t=0 0 m=audio 26676 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
14 headers, 11 lines
Using latest request as basis request
Sending to 172.31.30.7 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found descripti

Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)

2003-09-23 Thread James Sizemore
Yes this is a known bug.

Manuel Marín García wrote:

Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked up but the other phone continues ringing. Is
there any problem with call pickup in SIP.
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Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)

2003-09-22 Thread Brian West
Here's one thats way out in left field... don't use call pickup! :P
Problem solved sorta!

bkw

On Mon, 22 Sep 2003, Jared Smith wrote:

> On Mon, 2003-09-22 at 15:42, Manuel Marín García wrote:
> > Please help! When I try to place a call pickup from a cisco phone 7960
> > using *8 the call is picked up but the other phone continues ringing. Is
> > there any problem with call pickup in SIP.
>
> It's a known problem... I wish someone would hurry up and fix it.
>
> Jared Smith
>
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Re: [Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)

2003-09-22 Thread Jared Smith
On Mon, 2003-09-22 at 15:42, Manuel Marín García wrote:
>   Please help! When I try to place a call pickup from a cisco phone 7960
> using *8 the call is picked up but the other phone continues ringing. Is
> there any problem with call pickup in SIP.

It's a known problem... I wish someone would hurry up and fix it.

Jared Smith

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[Asterisk-Users] Call Pickup problem with cisco 7960 (SIP)

2003-09-22 Thread Manuel Marín García

Please help! When I try to place a call pickup from a cisco phone 7960
using *8 the call is picked up but the other phone continues ringing. Is
there any problem with call pickup in SIP.

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