[Asterisk-Users] Speeding up UK BT incoming call detection
Hi, I am running Asterisk v1.2.7.1 with a Digium TE110P. My dialplan is very simple, when a call comes in on my analogue BT PSTN line, it rings the other ZAP interface (my house phone). Slightly pointless (having a 1x1 switch) I know, but I am planning on doing more with internal SIP extensions, and outgoing SIP services etc At the moment it's just simple though whilst I try to fix this problem. My problem is that it normally takes 2-3 rings on an incoming call before Asterisk detects the line ringing and then a further 2 rings before it starts ringing my phone. I know this because I have plugged a normal analogue phone into another extension of the BT line as my Asterisk connected phone starts riniging about 4 or so rings later! Can anyone tell me what the optimal settings for zaptel.conf and zapata.conf for a BT line or have any other suggestions to try and speed things up? Cheers Richard Dutton ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *
Hi, I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these particular model and would like to use them in an Asterisk server. Cheers Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dialogic D/300SC-1E1 and D/600SC-2E1 with *
Hi, I've seen from the Asterisk Hardware list that the Dialogic D/300JCT-1E1 and D/600JCT-2E1 cards are supported by Asterisk, can anyone tell me if the D/300SC-1E1 and D/600SC-2E1 cards are as a client has quite a few of these particular model and would like to use them in an Asterisk server. Cheers Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite and * SIP Problem
Hi, I am playing around with SIP extensions on my local lan using X-Lite but I am having a bit of difficulty, I have set up X-Lite and my sip.conf accordingly, but when I start it I get the following message: "Login failed! Contact Network Admin" I am still able to dial local extensions on my * with x-lite even though it is in this state, although trying to dial my sip extension from a real extension results in a busy tone. Every 30 seconds or so, my asterisk console shows the following message: "Mar 12 09:48:00 NOTICE[802]: chan sip.c 8448 handle request: Registration from 'richard ' failed for '192.168.0.100'" My sip.conf is as follows: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes [200] type=friend username=richard secret=password host=dynamic reinvite=no canreinvite=no dissallow=all context=sip allow=gsm And my X-lite Default SIP Proxy config is as follows: Enabled: Yes Display name: richard Username: richard Authorisation User: richard Password: password Domain/Realm: 192.168.0.102 (my asterisk server's IP) Sip Proxy: 192.168.0.102 rest left as default Can anyone tell me what I'm doing wrong? This is all through a local lan, no nat or anything. Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP with Asterix
Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get two TDM400P cards in two machines attached to separate adsl connections (in two different physical locations). I'd then like to be able to plug POTS telephones into each of them (I understand I will need FXS interfaces in both) and be able to make calls between them. Is this hard to set up? Are there any guides that you guys would recommend? I would then like to attach a POTS line to one side and be able to dial into my asterix, and out to the other side across the adsl. What codecs do you guys use for VoIP and what sort of quality can you get over regular 512mb adsl? Any help would be most appreciated, Asterix looks like an excellent system and I can't wait to get started with it! Cheers Rich -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.8.2 - Release Date: 28/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
