Re: [Asterisk-Users] Under which project , auto-dial feature comes

2006-05-02 Thread Eric \ManxPower\ Wieling
John Joseph wrote: Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Define auto-dial. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery.

Re: [Asterisk-Users] How does asterisk behave when multiple phones are logged in on a single SIP/account?

2006-05-02 Thread Eric \ManxPower\ Wieling
Arne Morten Johansen wrote: Hi. How does this work? What if this SIP/account was a member (agent) of a queue? Ex: dial(SIP/account,20,tT). Would the dialstatus be set as busy when one of the phones is actively talking, or will the other phones continue to ring? You may have seen my other

Re: [Asterisk-Users] Sangoma Card Question

2006-05-02 Thread Matt
By the system you mean the phone company? Or asterisk? So what you are saying is I hang up... the sangoma hangups... but the phone company sees it as a flash... then says.. HEY DUDE! YOU JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*. ? On 5/2/06, Melcon Moraes [EMAIL

Re: [Asterisk-Users] Sangoma Card Question

2006-05-02 Thread stevanus
Hi Matt, I guess this is the problem within asterisk which wrongly assume the hangup as on-hold call. Do you/your staffs/your customer hang up the phone so quickly that asterisk mistakenly belief that the act is for call waiting? As we know to do some call waiting we just flash the hook

[Asterisk-Users] Grandstream GXP-2000 call end

2006-05-02 Thread J Shaun Hofer
Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use

[Asterisk-Users] asterisk hung again

2006-05-02 Thread stevanus
Hi, Yesterday, one of my asterisk servers was hung... On the log, I found these: May 2 09:38:26 DEBUG[28201] rtp.c: RTP Transmission error of packet 50596 to ip address:16480: Network is unreachable May 2 09:38:29 WARNING[17120] chan_sip.c: sip_xmit of 0x81f60b8 (len 396) to ip address:5060

[Asterisk-Users] Queue reporting seems broken.

2006-05-02 Thread Thermal Wetland
I am trying to figure out which one of our agents is answering the calls.According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff logon.There is the connect completeagent message, but it doesn't show which

[Asterisk-Users] Asterisk Imposter binary

2006-05-02 Thread Bill Michaelson
I found a bogus binary in my (obviously) hacked system in /usr/sbin. I am still investigating. FWIW, it was 608828 bytes big. It appears to have arrived recently, but I haven't determined how. Here is some more info... sum /usr/sbin/asterisk.suspect 15139 595 I'm just posting this in

[Asterisk-Users] OT - but relevant

2006-05-02 Thread Dean Collins
This is OT but relevant to the tech depth of this list. The Always-On Top 100 of 2006 were announced this evening a streaming webcast of some of the CEOs will be online tomorrow at http://www.alwayson-network.com/comments.php?id=14349_0_1_0_C If you want to see a full list of the

[Asterisk-Users] Need help in asterisk fax

2006-05-02 Thread Gidean Chan
Can anyone tell me how to make it work? I have asterisk 1.10.006 and hylafax in the same linux server. 2 x100p on PCI slots connected with 2 PSTN lines. I was using hylafax on one line with an external modem before. Now I have already removed the external modem and want to use asterisk to

Re: [Asterisk-Users] PAP2/Sipura XML Provisioning File

2006-05-02 Thread Ed
Gonzalo Servat wrote: I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use

Re: [Asterisk-Users] PAP2/Sipura XML Provisioning File

2006-05-02 Thread Ed
Ed wrote: you can try spaconf utility. it can backup/restore config for sipura devices, for linksys we need small patch. oops... this patch is included already ;) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

2006-05-02 Thread Ed
[EMAIL PROTECTED] wrote: I'm in the process of writing an autoprovisioner which can handle fresh out-of-the-box linksys, snom, and grandstream with 0-config (other than entering the mac into a textfile). You never have to touch the phone. Just plug it in. any result?

RE : [Asterisk-Users] IAX Configuration

2006-05-02 Thread f6hqz-m
exten = 19,1,Dial(SIP/19,20,tr) Must be : exten = 19,1,Dial(IAX2/19,20,tr) Because you are using IAX IP-Phones... Best Regards, Francois BERGERET, France -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Olivier Saulnier Envoyé : mardi 2 mai 2006 16:09

[Asterisk-Users] Unicall MFC/R2 B3,B4 and clear back

2006-05-02 Thread Dennis Nacino
Hi All, I have an R2 installation still undergoing testings, during the test I notice that the Unicall always respond B6 to a II-1 (from a forward switch). Except, for a DNIS that can't be found in the dial plan, in this case it respond with B5. My real problem is, the call will be terminate

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