similar to: call pickup with Sipura-3000

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

How to configure Digium S101i adapter to work with "pickupexten *8" ? I have few Sipura adapters and "*8" work OK but my new Digium S101i refuses to cooperate. -- #Joseph

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

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add

Daniel Heckl <daniel.heckl at gmail.com> schrieb: > You are searching for ?Call Pickup?. It is implemented in Asterisk by > default. > > https://wiki.asterisk.org/wiki/display/AST/Call+Pickup > <https://wiki.asterisk.org/wiki/display/AST/Call+Pickup> Take a look under > section ?Configuration Options?. Hi, Daniel! Thanks for your answer... I'm using Asterisk

Hi there, it appears that call pick-up only works _within_ a technolgoy, i.e. with a SIP phone when another SIP phone is ringing. Is that correct, or is my configuration faulty? * Case 1: SIP phone 1 ringing - SIP phone 2 can pick the call up with *8 We are happy! :-) * Case 2: IAX phone ringing - SIP phone can't pick the call up: NOTICE[10250]: Nothing to pick up * Case 3: SIP phone

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf

Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I've looked at a lot of posts and the wiki, which states that you need to define the pickup extension in features.conf and the pickup groups in sip.conf. I've done this, however there is no definition for *8 in extensions.conf. Is there supposed to be and it has been removed?

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to