similar to: No ring when doing blind transfer.

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

Hi; I don't know what these things are called so that I can look for an OS solution. I want to install an email server interface like Hotmail where people can check their email TTW. What is this called? Any recommendations? TIA, Susan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi; I have an email form that worked fine until now. For some reason, if I send an email to an email address at a domain that I control, I can receive the email TTW no problem. However, if I try and push it to, for example, this gmail account, I never get it. It's not even in the spam filter. What could this be? TIA, Susan -------------- next part -------------- An HTML attachment was

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

An Asterisk extension calls an Alcatel extension via a PRI link which rings 4 times for about 10-15 seconds and then drops. So if the Alcatel user doesn't answer within 10-15 seconds the call is aborted. (A timeout is *not* specified in the Asterisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What

Good morning gentlemen, is my first post in the list, now I'm starting asterisk wanted to have your help for some questions. Well the first function is as follow me. Here I will demonstrate how this configuration follow me on my extensions.conf but it is not working, and do not know why, but something is missing? You must set up followme.conf ? What I want is that the follow-me is

Hi, I have a working Asterisk 1.2 server on kernel 2.6.22 with the OSLEC echo canceller on a Digium PRI card. I recently switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing

Any body have an idea why my first attempt to use "observe_field" isn''t working? <%= start_form_tag(:name => ''my_form'', :controller => ''boo'' , :action => ''ya'') %> <%= radio_button_tag(''foo'', ''true'', checked = true) %> <%= radio_button_tag(''foo'',

Previously, by mistake I added request in wish list for slide like MSN. Actully it was Blind like msn. However it was my mistake so I tried to created script by my slef - and able to do it. But I am new to this, so my script is not in good condition. Effect.Scale2 = Class.create(); Object.extend(Object.extend(Effect.Scale2.prototype, Effect.Base.prototype), { initialize: function(element,

Greetings, I am trying to add a row to a table using the blind down effect. I have this working for lists (aka <li>) but not tables. With tables it seems to display the newly inserted row then blind. I basically want to start with the row hidden and then blind down. For example this works with lists: <ol style="" id="list1"> <li

I wrote a simple toggle function similar to Element.toggle that will toggle a section between being up and down using the Blind effect. I''m relatively new to the code, so I''m not sure if it''s implemented 100% correctly or if it can be done better. I am using it though and it seems to be working fine. Anyway, here''s the code and if you think it''s

HI callee side blind transfer is failed in 1.8 but caller side blind transfer is succes,Transfer doing by refer method,please help me on this Nikhil

Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

Hi I'm having 2 strange problems (or rather I am doing it wrongly!) ..need some help.I am working on trixbox 2.6 with Asterisk 1.4.22 1. I've a Queue with dynamic agents (ampextensions=deviceanduser mode in amportal.conf), some of agents when they punch in the wrong UserID - they still get logged onto the Queue as a result many of them complain that they don't receive calls on

Hi all, when doing a blind transfer using the keys defined in features.conf, we hear a confirmation of the attempt to blindly transfer, followed by an invalid extension message. The console says this: [Jun 4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/570-00000006 [Jun 4 22:30:31] VERBOSE[11301] file.c: --

My SIP users need to transmit the "#" key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. Anyone have a handy answer? Tim McKee -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef

I've been unable to figure out how to make blind xfer transfer a call on a zap interface by using hook flash instead of a native bridge. Is this even possible to change how a blind xfer works on the 7940s? zapata.conf: threewaycalling=yes cancallforward=yes transfer=yes