similar to: Grandstream RS-232 config (slightly off-topic)

Displaying 20 results from an estimated 2000 matches similar to: "Grandstream RS-232 config (slightly off-topic)"

2009 Jan 16
1
ATA gateway with 2 ethernet interfaces
Hi, I'm looking for an 8+ FXS ATA gateway (at least 8 ports but preferably at most 24 ports) with 2 ethernet interfaces for network/switch redundancy. So far I've only found the Grandstream GXW4008. I've searched similar brands such as Linksys and higher-end brands such as Quintum, but they all seem to have just one NIC. So, if the switch the ATA is connected to fails then I'm
2007 Aug 06
2
ATA phones ring when they register
Hi, I have an 8-port Grandstream GXW-4008 V1.2A ATA converter with analog phones connected to it. They work fine except for just one "feature" I would like to modify. Somehow, each time the ATA re-registers the SIP clients or each time the device has to be rebooted for maintenance, the phones ring once. This feature can be useful as it notifies the user of the re-registration.
2009 Jan 10
3
Asterisk/GXW410x IP Analog Gateway
Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...
2007 Dec 02
4
get SIP extension status without calling it
Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.
2009 Jan 29
1
early dial: asterisk and ATA
Hi, I have a set of Grandstream GXW4008 (units of 8 FXS ATAs) and another set of Linksys SPA8000 (8 FXS ATAs). The GXW4008 has a "neat feature" called "early dial" which allows me to define a "dial pattern" as generic as {*X+,#+,X+} (or something similar; the idea is to "match all digits") and send those digits >>immediately<< as they are
2008 Mar 13
2
queue log vs. cdr
Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql> select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 20080308000000 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql> select * from cdr where dst = 4010 and calldate between 20080308000000
2008 Mar 08
3
replace astdb with a cluster-capable sql database engine
I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a
2008 Mar 27
2
callers in queue passed to agents who accept only one call at a time
I have a queue I configured as "strict" and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy).
2007 Aug 30
1
dialed peer number
I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get
2007 Jul 30
6
outbound caller ID
Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your
2008 May 06
3
asterisk queue cluster
I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced
2009 Jan 17
1
compare Linksys SPA8000 and Grandstream GXW4008
Hi, Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and hardware stability (the feature sets are apparently similar)? Vieri
2006 Nov 30
6
200+ analog phones connected to FXS modules
I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating
2008 Jan 07
3
asterisk CLI and no such command "stop"
Hi, I'm probably missing something trivial but I don't understand what. Asterisk is loading fine but when I connect to the console (asterisk -vr) and type "stop" I get a no such command reply: *CLI> help (...) skinny show lines Show defined Skinny lines per device soft hangup Request a hangup on a given channel unload Unload a
2011 Feb 08
3
fail-over server
Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to
2014 Mar 04
2
Cannot chain to another PXE server on the same subnet
Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network
2009 Nov 18
3
asterisk 1.4.26.3 makes kernel panic
Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:
2012 Feb 08
4
SIP hardware phones
I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when
2008 Jan 01
4
zaptel 1.2.22.1 on kernel 2.6.22: wctdm24xxp.ko needs unknown symbol pci_module_init
Hi, Before I report a bug on http://bugs.digium.com, I would like to know if someone is seeing the same error message. Personally I am not using wctdm24xxp but other modules such as wcte12xp and wctdm. The latter modules load fine and are compiled with pci_register_driver as expected. The only module that seems to require the deprecated function pci_module_init is wctdm24xxp. Is this normal?
2009 Feb 25
4
switchtype QSIG and Asterisk implementation
Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by