similar to: Strange call transfer problem.

Hi all, I have a user who is having trouble transferring calls, using a Grandstream GXP2xxx. Here's the use case that I've seen: I call the user from phone A and he answers on phone B. Then, he hits the transfer button on his phone and dials an extension that is reachable by him, but not by me, based on administrative policy. However, the Asterisk logs indicate that the new call is

Hello, i?ve a problem i dont find the reason for. An incoming call coming over iax is connected to a Sip phone. Until the phone picks up the call i could hear moh without problems. Then the phone sets the call on hold and opens another call to another extension. The incoming call hears the Hold music and also the call to the other extension hears another moh. Everything works so far as it

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

This sure seems like it would be simple. Probably can't see the forest for the trees. I need to use the "native" MOH feature on my little WRT to save processor load. I normally don't use MOH but am playing with atxfer and would like to have something to play to the remote transferee. But when I comment out the "default" clause in musiconhold.conf, I get an error

Hi, my Problem ist he following situation, Caller A calls to my company. HE gets into my call queue and is then answered by caller B. Caller B answers and makes an attended transfer to caller C. Caller B hangs up bevor caller B has anwert the call. Caller A can hear MOH till caller B hangs up and gives the call to caller C. Then Caller A could only hear the free ringing sign.

I have a peculiar situation, and am hoping someone on the list can offer assistance. I am running CVS HEAD, and am using ITSPs for DIDs. The server has no Zap hardware, but is configured to use ztdummy. All incoming calls are via IAX2. Calls ring to SIP phones, voice mail, IVR, etc., with no trouble. I am also able to transfer calls among my SIP devices, voice mail, IVR, etc. All of my SIP

This message was just bounced back to me. I am not sure if it made it to the list originally or not, as I received no responses. Since this message was written, I have installed Zap hardware into this server. The Zap channels can be transferred to the Meetme conference. The IAX2 calls still cannot. Any suggestions will be greatly appreciated. Sincerely, Trevor Hammonds Trevor G.

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Hi all, I've got a new server up, but it's not staying up.... After a day or so, it segfaults with: [Mar 22 23:17:49] WARNING[12177]: res_odbc.c:1406 _ast_odbc_request_obj2: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.2(a) Driver]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use

The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is

Hi all, I've upgraded to Asterisk 13.14.0 and now I'm seeing that Asterisk is using 100% CPU. I have one AMI agent connected that is acting rationally. I've got a hand full of SIP (RT) registrations. There is no other call activity. I've tried to unload various modules; nothing resolved the issue. Any suggestions? -- Mike Diehl

Asterisk 1.4.6 at FreeBSD6.2-RELEASE Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Log says everything, MOH should stop after call pickup, not before Dial. -- Executing [113 at firma:1] Dial("SIP/zytek-08737000",

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked? ================================ ====== ====== ======== *CLI> It doesn't seem to be

On 2016-01-25, Always Learning <centos at u64.u22.net> wrote: [...] > As a C5 Gnome 2 user I dread G3 when I move my desktop to C6. Mate seems > an alternative. Anyone know more about the G2 folk ? C6 features G2. Nothing to dread. -- Liam

Hi all, I'm installing Hylafax on my Asterisk system. From what I've read, I can either use IAXModem or T38Modem to provide the virtual fax device. So at the risk of starting a religious war, which one should I use? I don't mind running IAX if I have to. I want as much flexibility and stability as I can get. So, what are your recommendations? Mike. -------------- next part

On Thu, 2005-01-13 at 12:42 -0500, Jean-Marc Valin wrote: > Le jeudi 13 janvier 2005 ? 10:59 -0500, Jared Whitby a ?crit : > > Interestingly enough.. I started playing around with preprocessing > > options in 1.1.6 and happened upon the denoise filter > > (SPEEX_PREPROCESS_SET_DENOISE). When i run the test tone using that > > option it is completely filtered out and I

Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the

Hi all, I've got a user who wants to receive voicemail notifications at two different email addresses. I could probably setup an alias in /etc/aliases, but then I'd have to manage that across multiple servers, which I don't want to do. Is there a way I can tell Asterisk to send to multiple addresses? Mike -------------- next part -------------- An HTML attachment was scrubbed...

I recently upgraded Asterisk from 1.8.x to 13.x and am now finding that music on hold isn't working like it used to. It seems that even though the correct MoH class is being set, the system still plays the "default" music. All of my call handling is done with an AGI script. When a call is made, the AGI script sets the MoH class before dialing. The log indicates that the correct

Hi all, I have a user who is reporting dropped calls at his site. We don't have any other users complaining of this. So far, this is what we know: 1. The manager bought all new Polycom phones. (POE) 2. They replaced the network switch with a POE version. 3. It's not just one or two of the phones that have problems. 4. It doesn't matter if they use the headset or the cordless