similar to: SIP to SIP redirect while ringing

I didn't get any response on that question, so i supose this feature is possible but there isn't an implementation of it. I'm ready to sponsor this feature in the manager interface (i tried the redirect command but it doesn't work) can somebody help me ?? this feature would make it possbile to use drag & drop features ... Kind Regards Michael Devenijn

I have an asterisk server and I am trying to set the server up as a redirect server of all my internet SIP phones. My Asterisk server as his own internet IP address. At this moment I can make international calls to a IAX provider but I am now trying to setup a SIP provider as well And I get the following error -- Executing Dial("SIP/t10002-4666",

I found this tool, but didn't have the time to test it... http://www.dylogic.com/sito/ArticlesDMD/mirial.html -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of shabanip Sent: donderdag 24 juni 2004 13:59 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Video/H323/SIP Is there any software based

I am using AMI to call a phone and play a wave file. That works fine to SIP/401. Now I am trying to "redirect" that call that is ringing to another phone (SIP/404). When I do it the other phone rings but the first phone continues to ring also. Then when I answer on SIP/404, I get a ring not the wave file. Action: Setvar Channel: SIP/401-00000004 Variable: SMVOICE_CALLAT

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

Has anyone ever worked opn TAPI stuff to make asterisk work with it ? I'm a Windoze C++ developer dig'n into asterisk (and linux at the same time) since a few months and i'm quite interested in creating a TAPI driver for asterisk. so if anybody did any research in that way please inform me. Also i've you think it's quite impossible to do it we can discuss our idea's

Hi I want to connect a Windows machine over TAPI with the Asterisk PBX. So is it possible to connect the Windows machine directly to Asterisk (Zaptel card)? Thanks Harry Baron

Hi all, Has anyone have an idea why, if you capture the files on a Asterisk network (ex with Ethereal) you always see the communication between the two sip phones( hard or soft) passing through the asterisk server (on UDP layer) Isn't SIP a protocol that (after that it has established the call) , he connects the two users with each other? Maybe a stupid question, but I'm not a SIP

We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

I try to use a SIP trunk from a VOIP provider to make land to mobile calls. If I do these from a ZAP channel, using an analogue phone, after few seconds of silence (I don't like to generate fake [r]inging) I ear the ringing tone from the mobile operator along with any message the mobile operator decide to say me. If I try to use a SIP phone (or a IAX phone) attached to my asterisk box, I

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

>All, >I have a TDm04b card an 8 SIP extensions. Calls come into the TDM and >are answerd by the auto attendant. When an extensions is entered I see >the Dial(SIP/100) on the console but the phone never rings... >I can pick up any extension and call 100 and it rings just fin>e and they >answer and everthing is fine. All 8 extensions can dial 9 for an analog line >and call

Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B,

Justy to let you all know that i tested 7905G phone with a SIP image (latest download) and it works great ! for a reasonable price but with a good quality and a brand ... which inspires trust and helps selling better The only minus : Missing a microphone to work handsfree (or i didn't find it.) but strange enough their is a speaker ... Michael Devenijn IT DKMA -------------- next

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

Hi all, I have 12 SIP phones at a particular site all connected to a local asterisk server. It's in turn connected to 2 ISDN BRIs to provide up to 4 incoming calls. An IAX gateway is used for outbound calls. At the moment, when an incoming call comes in, asterisk dials every SIP phone like so: Dial (SIP/1&SIP/2&etc.) This has worked fine for some months, but I noticed a few days ago

Hi there, I am seeing some very interesting thing with the latest Zaptel 1.2.X, hope may be someone can shed some light on this. Normally, to dial via your Zaptel T1 card, you would do something like: ;Dial to PSTN exten => _9.,1,Dial(Zap/g1d/{EXTEN:1}) by not adding any option after the extension e.g. no "r" and no "m", Asterisk will pass thru the session media from the

Hello all! I have a problem with ringing indication when calling from h323 (oh323+open phone client) to sip users. The phone rings and users can talk to each other with no problems but the calling h323 user hear silence unless sip user picks up the phone. Calling to pstn no problems. Calling from sip to that open phone client also no problems. I tried latest ooh323 and oh323... no difference

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing -- SIP/0699073201-dc61 answered