similar to: PSTN Gateway setup

Hi all, I have asterisk connected to my voice application server. Asterisk is connected and registering to a linksys spa400 box. I am running an application on a perticular extention (141). Here is a snip from my extensions.conf... exten => spa400,s,MyApp(/etc/asterisk/MyAppConfig.conf) exten => spa400,s+1,Hangup when an incoming call comes,It is accepted properly,And the application

Hi! I'm a new member in VoIP world. I want to implement a VoIP PBX using asterisk/tribox in the office but I have one doubt. Which is the best way to use actual PSTN lines in the VoIP PBX? Using a box as Linksys SPA400 or installing a PCI card (Rhino/Digium...) into the server? Which benefits/problems will I have with each option? Regards -------------- next part -------------- An HTML

Hi there, I was wondering if anyone has successfully used Asterisk as a dedicated Analog PSTN gateway to take the place of, for example, a Mediatrix 1204 or an 8 port model? Basically, I am thinking of using a Linksys SPA9000 as the PBX and just need an Analog PSTN gateway for 4 to 8 FXO lines. It does not sound like the Mediatrix 1204 does a very good job and I figure I can build a much more

Hello I successfully hooked up a Linksys 3102 SIP gateway (http://www.voip-info.org/users/683/21683/images/716/SPA3102_lrg.jpg) to an Asterisk server, but since it's connected to a PSTN line, I must make sure it cannot be used by unauthorized users from the Net. Actually, even legit users with an account on the Asterisk server shouldn't be able to use it (outgoing calls should go

Hello I'm using a Linksys 3102 as VoIP gateway to connect a POTS line to a PBX. I also have an IP phone in a remote network across the Net. The PBX + gateway, and the phone are both behind a NAT router. I was wondering: 1. When a customer calls us through the POTS line and I pick up the call with the remote IP phone, do RTP packets go directly from the VoIP gateway to the IP phone, or

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Hi Steven, I wish, I already have 2 spare TE410p and 1 TE405p. But customer wants to use Analogue and they already installed the lines. Yesterday, I ready about the FXO modules being replaced by Digium, this relaxed me a bit. But as you said, I will have to worry about the ring and hangup detection, cleaner lines, impedance matching echo problems specially that I have not heard of anyone using

Has anyone seen the following problem? Until recently, I couldn't understand why some extensions on my * system would have a "congestion tone" as soon as I picked up the handset. A little sleuthing through the logs and the source code led me to understand that * thought it had seen the extension go off-hook, send some DTMF tones, and then wait. * treated this situation as a

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

The TDM410P doesn't support 'hvac', only the obsolete TDM400P supports that option was for the old phones that have a neon light (or equivalent LED+ZENER ciruit). Are other phones off the TDM410P (other than the VTECH) working, or is the Vtech the only model with VMWI available to you. I'm not able to check at the moment, I have copied the asterisk-users list, someone else may

Basically I've found that the 2.0.3 version of Samba is not supporting shadowed password files on SCO Unix 3.2v4.2. It appears to be doing it on SCO Unix OSR5 however. This shows up both in smbd and swat. We found this out by playing with the newest version on a SCO Unix OSR5 box. I recompiled the software on that box for a SCO 3.2v4.2 box but was unable to get the resulting binary to work

I have a Samba-LDAP PDC at an office and 5 BDC's at other offices. At corporate HQ I have a W2k Server and domain. I have properly configured an interdomain trust and Users in the Samba domain can get to sections on the W2k machine regardless of location. However, members in the W2K domain can only access shares on the PDC. Attempts to access shares on a BDC cause a user name

Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

After purchasing the Uniden TRU9480 and then the Panansonic 5672, both of which do not have "phone company" compatible FSK/stutter MWI, I finally got smart and found out just which Panasonic phones have this feature. Only the following 5.8G models in their current line have FXO compatible MWI. I purchased the 5771 unit and one remote. I have confimed it does in fact work with Asterisk

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

Hello I've been googling for a couple of days now, but still can't figure out what to put in zapata.conf to get it to report CID. Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202 as CID FSK Standard: http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg Does Zaptel support those on Digium

Hi, I'm currently having problems detecting FSK BT (UK) caller id in our API (Pika boards). I have a recording to test on but it is giving me checksum errors. I'm wondering if someone from UK using BT lines could send me a recording with the FSK signal so I can have more data to work on? If you have something, please send it to paulo.astuser at gmail.com. I appreciate all help about