similar to: Queue members, URI.

Does anyone know why when calling out with an ATCOM AG-188 registered with IAX (haven't tried SIP), there is no ring. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070222/a4f29a97/attachment.htm

I know I must be being daft, but is there a way to set which context the queuing system uses when it dials the operators/agents? By default it appears to use the default context. I've looked through voip-info.org and can't find anything, someone please put me out of my misery.

I have an asterisk 1.2.9.1 machine with zaptel 1.2.6 running. On the TDM400P, I have 1 FXS port and 3 FXO ports. dmesg reveals: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.2.6 Echo Canceller: KB1 PCI: Found IRQ 10 for device 01:01.0 PCI: Sharing IRQ 10 with 01:05.0 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1:

I have been seeing this problem for a long time and it occurs in 1.4.0b2 (as well as 1.2.0-1.2.12.1). If the internet connection is lost and I have SIP services that require me to register, any SIP devices attached to the system stop working. I have an IAX phone connected to one of my servers that I've been having this problem with which will work fine (and filover to the PSTN) the

I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11,

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) Which should allow me to listen to conversations that hit the first (Set

Hi, in the menu of voicemailmain, appear a lot of options, there is a way to leave only some of them? Also I want to know if there is a option that erase all message in a user box. Best REgards Ever Zalazar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060512/98a6f962/attachment.htm

Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

I'm trying to isolate the IP address of inbound calls to my switch over IAX2. Is the proper way to get that information as follows: ${IAXPEER(IP)} If the caller was inbound via SIP, this works: ${SIPCHANINFO(PEERIP)} So I'm looking to return the IP address of the caller via IAX2. Thanks Lee -------------- next part -------------- An HTML attachment was

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get

I am inquiring about setup of the SWAT utility I have installed Red Hat 7.0 with samba installed during the initial setup of Redhat. I have two network cards installed in my Server and I am connected to the internet via a Cable Modem. When I try to start SWAT netscape displays the message that it cannot find the local host on port 90. I have downloaded the book over samba and I have also tried to

Hi I have been using the CHANNEL variable as a way of checking if a user is allowed to make outgoing calls, and what their source caller ID should be (these values are in a database). This works all of the time with SIP and most of the time with IAX, however sometimes with IAX the channel variable seems to be wrong. I have been using Zoiper as my IAX client and Asterisk 1.6.2.0-rc6. For the sake

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as