similar to: Asterisk as PSTN gateway

I'm trying out ASTCC. I set the card length to 10, and generated a test card. 10 digits. I set the extensions file to: exten => 9175954700,1,Answer exten => 9175954700,2,DeadAGI(astcc.agi) exten => 9175954700,3,Hangup I dial in and the prompt tells me to enter my 12 digit PIN, not 10 digits. How come it thinks it is 12 digits? I set both the Published number and DID in the Brand

I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. I also want the caller to not hear any audio until the DTMF has been sent. This gets the caller to the right place and he doesnt have to hear the welcome message from the PBX. Dial apparently isnt the application to use. Is

I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help,

We have been using Samba for many years. The company has just switched from an NT domain to an Active Directory domain. The new server is running Windows Server 2003. We are having trouble configuring our Solaris 8 server so it can join the domain as a server. Just getting Samba to compile and link was interesting enough. This included downloading and compiling a new version of the BerkeleyDB,

Hello , I have a PeriodicalUpdater executing every x seconds and sends a ajax request to update an element. Each time it comes back, the element is getting replaced with some html and and possibly <script>coolFunction()</script>. This seems to work fine when for prototype 1.5. However when I upgraded to prototype 1.6, it seems to execute coolFunction() as many times as the request has

Hello, As far as I'm aware, email is case insensitive ? Dovecot however, seems to be taking offence at the fact that someone is sending mail to user at SOMEDOMAIN.XYZ. Postfix has no issues handling it, it when it gets passed over to Dovecot LMTP that I get log lines such as : Sep 17 13:48:22 ukc-vm02-mx01 dovecot: auth: passwd-file(user at SOMEDOMAIN.XYZ): unknown user However user at

I'm getting an INVITE from the Cisco softswitch looking like this (from sip debug trace): Sip read: INVITE sip:9043940358@206.165.120.52;user=phone SIP/2.0 Via: SIP/2.0/UDP sia-ATLCA146.telefyne.com:5060;branch=z9hG4bK_1146_38l5 From: <sip:7278673170@sia-ATLCA146.telefyne.com;user=phone>;tag=1_1146_f153592_732 j To: <sip:9043940358@206.165.120.52;user=phone> Call-ID:

I'm looking to use the Digium E1 card in Iran. Does anyone out there have any experience with something like this? What are the odds it will work first time or ever? Bill

Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN --> E100P --> asterisk --> sipphones. Thanks Johan

I am using "maps". I am running the following code to get this list of all the counties: map('county', plot=FALSE)$names In the output, all counties have first the state listed and then, after a comma, the name of the county. However, county # 2395 (State = south dakota) has no county name. Anyone knows what this county is? Thank you! -- Dimitri Liakhovitski

Hello everyone. This is my first time attempting to use Xen and I am having some problems. I am trying to build a Xen system so that I can use the Ether program for malware analysis (http://ether.gtisc.gatech.edu/source.html). I''ve installed Debian 6, and downloaded the Xen 3.1.0 source. However when I try to compile the source using "make world" I receive the following errors:

We manage our own VOIP solution using Asterisk. Has anyone had success with an IP-PSTN provider? I'm looking for someone to terminate SIP calls to the PSTN in the Seattle, Washington area. (vice-versa as well if possible) Yes, I could do it myself via asterisk and digium cards but I would like to consider other options. Any opinions? Thanks, Chad -------------- next part

Hello list, I recently bought a Linksys SPA400 as a PSTN gateway. The gateway is connected to an * server and i have 10 users using this setup. I do have some problems in establishing a call to an outside location (call that goes through the SPA400). The first attempt doesn't get through. I suspect the spa400 being the source of the problem. The Linksys SPA400 has a lot of params on the

Hello Surprisingly, Google didn't return any thread in this ng about the Obi110 device, which is a new alternative to the 3102: www.nerdvittles.com/?p=720 I'd like some feedback from Asterisk users who have tried the Obi110 to connect it to a landline. Thank you.

Yo all, The PSTN access-number for the Dutch IAXTel <-> PSTN-gateway has changed. The new number is: +31 20 3987 567. Calling from IAXTel to Dutch toll-free PSTN-numbers is still done in the same way, by calling "31800<rest of number>". Mark: Could you please update your web-sites to reflect this change? The old number is mentioned on "http://www.gnophone.com/",

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my [bogon-calls] context. Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell me a great deal - I just see

I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] Can anyone help me locate why? (Config files are on the Wiki) I have done a packet sniff & decoded using Ethereal-0.10.0, but this doesn't tell

Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22 WARNING[1116941120]: chan_sip.c:2049 process_sdp:

Hello, I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway. My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to PSTN network (carrier). I need to configure * to

I'm looking for recommendations for UK-based VoIP-PSTN gateways. They should ideally offer: - IAX connection - Multiple simultaneous calls on a single account - Lower call rates than BT Business - Auto-top up or invoicing in arrears I can find several that offer one of these facilities, but none that offer all. Thanks! -- David Gurr Congruity Ltd. Hemel Hempstead, UK