similar to: Grandstream and setvar

Hi all, I've got a number of Grandstream phones and I'd like to be able to reboot them remotely, as I do my Polycoms... I've got this in my sip_notify.cfg: [grandstream-check-cfg] Event=>sys-control Doesn't seem to work. Any ideas? -- Take care and have fun, Mike Diehl.

Hi all, I used to configure each of my sip clients with a unique identifier via setvar. These clients were all configured as "friends." However, now that I've got some Polycom phones, which MUST be "peers," I am unable to define this variable. For example, this works: [friend-client] context = default accountcode = pcc type = friend username = username secret =

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: <Proxy_1_> _sip._udp.example.com </Proxy_1_> However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine.

Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar column, like this: callerid_internal=test <710>;did=5555551234 Again, this works

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

Hi all, I've cross-posted this to the -users and -biz groups. Hope that's OK. I have a customer who REALLY needs to be able to send/receive faxes reliably. I could probably get hylafax configured, but I'm not sure how reliable it is. If it is considered reliable, would someone let me know? Otherwise, is there a product/service they can buy that will allow them to fax to/from

Hi all, I have 2 servers running 1.6.2.9 and I'm about to build a third server. This suggests the possibility of doing a rolling upgrade of all of my servers. This brings up the question of what version to install and upgrade to. I don't have many upgrade opportunities, so I'd like to get as much bang for my buck. Since I've applied some custom patches to my 1.6, I'd

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

Hi all, I've got a device that seems to become unreachable for about 2 minutes, every hour. From what I can tell, it isn't due to network or server issues. Any ideas? TIA. -- Mike Diehl Diehlnet Communications, LLC. Voice: (505) 903-5700 Fax: (505) 903-5701

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

Hi all. I could have sworn this working at one time... But it doesn't look like any of the functions provided by features.so is working for me. (one-touch monitoring, attended/blind transfer, etc) I've (re)loaded features.so, as well as bridge_builtin_features.so. The config file looks sane. What else should I try? TIA, -- Take care and have fun, Mike Diehl.

Here is something I wrote years ago. I expect you can adjust it for your needs # cat remove_blank_vmail #!/bin/bash # remove_blank_vmail takes arguments as voicemail boxes and removes messages with audio files shorter then MINSIZE (in bytes) #---------------------------------------------------------------------- # Description: # Author: John Harragin Monroe-Woodbury CSD # Created at: Thu Nov 6

Hi all, Does anyone have any good/bad comments on the A1200P 12-port fxo/fxs card from OpenVox? I'll be using one to with 8-12 fxo interfaces.? The cards will be plugging into a cable-modem / phone adapter.? We weren't able to port the numbers, so we're going to use the existing PSTN connection and replace all of the office phones. With these short distances, will I need to worry

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example: