similar to: grandstream troubles

from extensions.conf: exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) I basically try to lookup the CLIP and attach a name for each inbound call. This works fine, except when I have just restarted asterisk - at which time I've more than once seen the message from the subject. As far as I can tell, with my Set(CALLERID), I should always have an argument in the DB

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Over the last two weeks, we have had at least two "incidents" where our asterisk server got flooded (a hundred or more per second) by SIP packets. Once from 114.31.50.10, second time from 173.212.200.146. We became aware of the problem when bandwidth started suffering because asterisk got very busy sending back replies or rejects (dunno which, I didn't investigate it any further).

Hello I'm about to order a GrandStream GXP-2000 and a Linksys SPA-921 I'd like to have some user feedback about how those phones perform, and whether their LCD screen displays both the caller ID name and number (The GrandStream BT-100 only displays numbers, which isn't very helpful). Thank you.

I just got spammed by TechnoCo - www.technoco.biz. Those guys must be a little dim if they believe they can openly go about borrowing email-addresses like this. /Per Jessen, Z?rich -- http://www.spamchek.com/ - managed email security.

> On Mon, Feb 8, 2010 at 2:20 AM, Olle E. Johansson <oej at edvina.net > wrote: >> >> 7 feb 2010 kl. 15.09 skrev Per Jessen: >> >>> Thomas Winter wrote: >>> >>>> Hi, >>>> >>>> my Asterisk on debian lenny died after 80 days. >>>> >>>> server kernel: [7572666.186852] asterisk[3673]:

I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [4439000@Business:3] Queue("mISDN/3-u0", "enidan|t|||20") in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable

This must have been asked before, but googling didn't help much. How do I define a callerid that contains non-USASCII characters? E.g. ?, ?, ?, ?, ?, ? etc. ? /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Does anyone have any suggestions for a decent receptionists phone? Aastra? Grandstream? Something with (potentially) lots of BLFs, large(ish) screen, headset and most importantly the ability to transfer calls? I've installed five Snom 370s that seemed ideal but my client is very very unhappy as the Snom 370 can't transfer a call correctly if there's another call coming in (details

I have the following dialplan: ; calls prefix by '8' are recorded exten = _8[01]./_251,1,Set(something=shortened) exten = _8[01]./_251,n,Set(WAV=filename) exten = _8[01]./_251,n,Monitor(wav,${WAV},mb) exten = _8[01]./_251,n,Dial(mISDN/2/${EXTEN:1},,g) exten = _8[01]./_251,n,System(send-recorded-conversation ${WAV}.wav ${EXTEN:1} emailaddr) exten = _8[01]./_251,n,Hangup() The idea is that

I'd like to be able to search the list archives when I'm reading someone's message to put what they say in context based on what they've said, and what others have said in conversation with them, in the past. It would help me figure out whether to trust some submitters on some issues, and just learn more from the community's collective/cumulative research and discussion. Is

Apart from religious grounds (!), is there any pros or cons why I should choose one over the other for a new install of asterisk ? Julian

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

Hi All, I'm interested in how people are "clustering" Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. I

Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to

Hi, Does anybody know how to properly execute a transfer (without using the |Tt option) from a GS100? Scenario: 1. I call from X-PRO (ext 1100) to Grandstream (1101). 2. Grandstream answers. Call is established. 3. Press [TRANSFER] on the Grandstream. X-PRO caller is put on hold. Grandstream gets dial tone. 4. Grandstream dials 1103 (the extension of another GS100). 5. Grandstream hangs

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at