similar to: Call Waiting + Call Transfer Problem

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

What's the importance of the impedance matching in a FXO interface ? Kind regards, Miguel

Hi all, I curious to know if other UIP200 users have this same issue: You flash (XFER button) to consult-transfer a caller to another extension. If the transfer target party is unavailable (ie: voicemail), there appears to be no way to get the original caller back. If it's a known limitation, has anyone come up with a functional work around? Thank -- ..................................

I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. When I start the IAX Client the CLI show me the message: -- Registered '2004' (AUTHENTICATED) at 192.168.199.69:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

Hi, I am a newbie to Asterisk; need help understanding three-way conferencing & call-transfer features implemented over standard extensions i.e. on a TDM800P card (4 FXO + 4FXS) In Asterisk I have observed that if an extension is already participating in an active call (e.g. Ext A & Ext B communicating): 1. An incoming call to one of these active extensions would be presented with

Dear all I am trying to work out how make call trasfer work the way I want is I am the called party I want to transfer a call so I press # and enter the ext but then it disconnects me this is a blind transfer how do I make it so its not a blind transfer so i can talk to the person before i transfer the call...and go backl to the orig caller if the transfered to ext doesnt answer.... also can

So far no one I have talked to has either had this issue or does not know a answer. I currently run asterisk 1.0.9. I have two issues to deal with. 1. The caller waiting caller ID does not show on the uniden hand held that is hooked to a sipura spa 1000 or my Polycom ip500. 2. When a caller is calling in and I hear the caller waiting beep on the line when talking with somone, is there no way

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I

I'm using asterisk blind/attended transfer feature on a queue (also tried with sip phones feature), and both type of transfers work fine. The problem is that attended trasfers doesn't get logged to queue_log, but blind transfers are logged just fine. Anyone knows if this is the correct behavior? -- Regards, Miguel Paolino -------------- next part -------------- An HTML attachment was

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

Hi folks, I have a simple question regarding attended transfers. I have some queues where agents take calls and I have configured attended transfers between queues. That is, the agent dials the attended transfer extension that routes it to the aproppiate transfer queue where the second agent answers and they both talk for a while. Finally the transferrer leaves the call with *, connecting

Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to transfer the call to this extension Currently they must pus "flash" for a new line speak to the

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael

hi All, We are working on Soft-PBX using Asterisk. This relates to CALL TRANSFERRING aspects of Asterisk. We were able to do one type of call transfering, ie, the called person can transfer the original call to another person. but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)