similar to: Double dial.

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070126/a49e3bdb/attachment.htm

Hello r helpers! Below is the whole coding for my programme. Before proceed more further, let me explain for you. First of all, I need to compute trimmed mean. Till that step is ok. Then I need to compute ssdw which is sum of square deviation. If I do equal trimming at both tail of distribution that I chose, I will use the first ssd formulae which is "a". But if I am doing unequal

I'm trying to link up Cisco MGCP-enabled router (residential gateway) with Asterisk, and it looks like some sort of protocol mismatch, could it be MGCP 0.1 vs 1.0? Look at this (x.x.x.99 is the router, x.x.x.98 is Asterisk): MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427MGCP read: NTFY 2 aaln/0@voip-gw1 MGCP 0.1 X: 0 O: hd from 192.168.154.99:2427Verb:

Hello, I''m having a very strange problem concerning traceroute and routing and didn''t know if lartc or netfilter would be the correct choice for asking. (so sorry if my question is misplaced) I have the following setup: public ip -- gw1 -- 172.16.0.1 --- 172.16.0.2/and public ip''s --- gw2 --- switch --users (public and private ip addresses; ip-user-pub) from the

Hello, I've got an analog phone which is currently receiving unsollicited FAX calls from PSTN. For learning purpose, I'm preparing an Asterisk/SPA3102 setup that would let voice calls come in and out and translate incoming FAX calls to TIF files (forwarded through email)). My target setup is : PSTN <-- analog--> SPA3102 Line Port <-- SIP --> Asterisk <-- SIP -->

Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicemail at mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested

Hy all, Recently a have a little problem with a Cisco device, SPA3102. I use this device with asterisk to dial out with outbound trunk. (SPA3102 has 1 FXO port) It working ok , but the device SPA3102 do this : when a call is placed for outgoing in asterisk and send to SPA3102 , this device "answer and dial the number in the same time" , in my CLI I see the channel is open , but on

Hi list, After failing to get a Sipura/Linksys SPA3000, which I've configured as a PSTN gateway, to pass on the Caller ID, I decided to try my luck with a Linksys SPA3102 after hearing some promising stories. Unfortunately, I've run into a completely new problem: it seems Asterisk won't let this device register. I went about configuring the SPA3102 in much the same way as I

Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else.... I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN

Hi Who concern, I setup TINC VPN follow these. 192.168.1.x / 24 (Client groups) | 192.168.1.1 (eth1) (GW1) 202.44.34.206 (eth0) || Internet || 202.44.45.14 (eth0) (GW2) 192.168.2.1 (eth1)

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Hi, I've searched the archive but can't seem to find the answer to my problem. i have two gateways running with asterisk , my question is : is there any possibility to do failover with gateways with asterisk ? i mean that if one gateway is down , asterisk switch automatically to other gateway . i have succefully used failover with limit number off calls (if gw1 have max calls ,asterisk

I'm missing something simple I think: I have an spa3102 for which I want asterisk to use the incoming pstn callerid when it sends the call to a local extension (207). callerid works fine for the internal phones (between each other) The spa3102 is picking up the PSTN callerid and displays it in its own status pages Asterisk however, doesnt see the callerid at all. The spa3102 is set to:

Hello all. I am just getting back into Asterisk and I am setting up my Linksys SPA3102. I have incoming calls working fine, as is the phone plugged into the unit. My problem is I cannot get the SPA3102 to dial a phone number automatically. I can call the extention of the PSTN and I get a second dialtone, and I can then manually dial. I'd like to be able to have Asterisk pass the

Hi I am trying to setup asterisk at home, I have 1 in bound VSP (I have a register cmd setup for that in asterisk). At home I have a cordless phone with 2 line capability - I currently have 2 spa3102's in place to handle the 2 lines ( I am in the process of buying tdm410 to handle to handle this and the backup pstn line). I also have 2 laptops setup with soft sip phones. What I would like