similar to: early dial: asterisk and ATA

Hi, Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and hardware stability (the feature sets are apparently similar)? Vieri

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

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

Hi, I'd like to know what's the most "popular" method for automatic fax/voice detection for incoming calls on mISDN cards such as the B410P (hfcmulti). I'm running: kernel 2.6.17 misdn 1.1.3 asterisk 1.4.21.2 B410P card I'm using iaxmodem and hylafax with asterisk (the setup works for zap channels). I've used the following options in /etc/asterisk/misdn.conf:

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.

Hi, I am testing some IP phones (eg. GXP2000) and noticed that the "early dial" feature works fine with Asterisk 1.4 but not with 1.2. "early dial" is when digits are sent immediately, one by one, and Asterisk replies with a "484 Address Incomplete" and waits for the next digit until a match is found. This is a very useful feature where no dial patterns have to be

According to http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS when a caller hangs up before the callee has time to pick the phone up then DIALSTATUS should be CANCEL. And it is. However, the disposition field in the CDR table is "NO ANSWER". So if I analyze the CDR data I won't be able to discriminate calls cancelled by the caller and calls not answered by the callee

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

Hi, I'm trying to configure a Linksys/Cisco SPA8000 talking SIP to both a local Asterisk server and also with a trunk directly to a VOIP provider. Everything works great, except I'm having a problem setting the outbound caller ID to a value different from the SIP username/authname. The SPA8000 has SIP setting for Display Name, User ID, Password, and Auth ID, as well as a "Use Auth

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

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

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

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:

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

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

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?

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

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a SIP trunk (so the problem is on my side, definitely, and it doesn't seem to be related to

Hi, I'm trying to find a way to implement the following: I have 1 media source (IceS or MPD) and 1 Icecast stream (say, LAN radio). Once in a while I'd like this stream to be interrupted by short announcements (PA system). Input for these announcements can be from another source (IceS, MPD, Asterisk call). Anyway, to make things simple: I'd have dir1 with ogg music files for