similar to: Need configuration sample for VoIP(SIP) -> PSTN Gateway

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

Hi, I'm a new user of Aterisk, and I have to configure a VoIP Gateway. I have an Alcatel PBX with an E1 card, connected, for the moment, to a local carrier. I would like work with a french VoIP provider, but, for this, I need to use a VoIP Gateway for connect my E1. Thus, I want to build my own voip gateway. It very simple, I want to connect my PBX to the gateway (E1 link) for both call

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

OK, I think I may have found the problem for myself at least. Actually, a friend of mine suggested it. Apparently, Asterisk is a little too fast for the card. Placing a "w" in front of the number to insert a pause looks like it did the trick! Dial(ZAP/1/w5555555) Looks like it gives the card a chance to come online? So, at least in this case, it was not that Asterisk was keeping

Sure would like to hear experiences using various FXO to VoIP gateways with *. It seems that any thread that has anything to do with problematic FXO interfaces goes on forever with speculation about everything under the sun. Unless there is someone out there with the engineering experience to build a better one it is a waste of time, let Digium deal with it. If the TDM400P can ever be made 99.99%

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

hi I am trying to stop the application running in the VM from the host machine.....that means by typing some command in the host machine, (script or using some API''s or sending some signal to VM from the host), i want stop application running in the VM.......is there any way to do this.....if anybody know this please help me....... I want this because......I want to take VM consistent

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

Hello, on 9.1-R, I highly appreciate the new jail(8) and jail.conf capabilities. Thanks for that extension! But I have one problem: If I want to stop a jail with 'jaill -r jailname', I get "umount: unmount of /.jail.jailname failed: Device busy" It seems to me that the order of fstab.jailname entries are not reverted by jail(8) when shutting down/umounting. My C skills

I have the following code. When I call the extension, it either ignores the first "Hello there everyone", or says "hello" and moves on sometime stoping before it finishes hello. The rest of the text reads fine. Anyone else have this issue?? Thanks! require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi->answer();

Hey guys, I have a problem: I created a silly for loop without saving the results on each step. After a while I realised that it will take days to finish the loop until I get the results. Is there a way to get the data R saves in working memory or in a temporary file while runing the loop? So that stoping the loop will not result in complete data loss? Thank you very much! Greetings,

Hi, I am facing issue stoping NetworkManager to update resolv.cfg, I am using below configuration for eth0 interface: TYPE=Ethernet BOOTPROTO=dhcp DEFROUTE=yes IPV4_FAILURE_FATAL=no IPV6INIT=yes IPV6_AUTOCONF=yes IPV6_DEFROUTE=yes IPV6_FAILURE_FATAL=no IPV6_ADDR_GEN_MODE=stable-privacy NAME=eth0 UUID=93b90a46-dab5-4a67-8fd0-fefe8874a8b9 DEVICE=eth0 ONBOOT=no PEERDNS=no PEERROUTES=yes

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