[asterisk-users] better timing source for an asterisk gateway
Hi, I have to make an asterisk gateway in front of several other asterisk. This gateway will essentialy be used for outbound call. This gateway will be connected to other asterisk by IAX trunk, outbound call will use SIP trunk (voip provider or patton isdn). I have a TE220BF available than i can use for dahdi timing source. Is a good idea, or this will give me zero benefit for timerfd timing source (will host this gateway on debian squeeze or centos 6.2) ? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi and digium debian package
Hi I'm trying to install dahdi. I just need the dahdi timer for conference. I currently using digium debian package for asterisk 1.8.8.1. When i install asterisk-dahdi , i've got several dependencies which came for official debian repository (including the dahdi package) and are outdated. Is it normal than dahdi is not include into digium packages ? Do i have to compil it before install asterisk-dahdi ? Thanks for your help. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] smsq, Zaptel in UK
Hi all, I've been trying to get SMS operational on my Asterisk box, which has a TDM400P card with a pair of FXO interfaces configured (ZAP/1 ZAP/2). I've not had luck with either of my lines, after issuing the command smsq --motx-channel=ZAP/1/1709400X 0 register. I see the following output in my Asterisk console: -- Attempting call on ZAP/1/17094009 for application SMS(0) (Retry 1) -- Hungup 'Zap/1-1' [Dec 26 18:17:07] NOTICE[526]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (1) Hangup It keeps retrying with the same message as above until giving up. It doesn't seem to make any difference whether I specify ZAP/1 or ZAP/2. When I look in /var/spool/asterisk/outgoing, I see a file with the following content: Channel: ZAP/1/17094009 Callerid: SMS 0 Application: SMS Data: 0 MaxRetries: 10 RetryTime: 1 WaitTime: 10 In /var/spool/asterisk/sms/motx I see a corresponding file with the following contents: da=0 ud=register I'm probably missing something really obvious, but I've not found anything via Google that suggests what I'm doing wrong. I'm running Asterisk 1.4.14 Zaptel 1.4.6 on Ubuntu 7.10. Any help would be appreciated. Cheers, Chris -- Chris Notley [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceConduits - Notice
Hello, This is David Deutsch, and Im the owner of VoiceConduits. There seems to be some confusion related to our company, regarding the past few posts. VoiceConduits is currently NOT open for public business, we have never to date advertised or attempted to attract business. It appears that a few people heard about our company via a mention in a SineApps article and found our beta system that is under development. We apologize that a few people managed to sign up via this interface, and we will happily refund anyone who did so immediately, additionally we will supply them with free credit to be used once we are in fact live. It was certainly never our intention to defraud individuals of the asterisk or voip community, our understanding is that only 5 people have managed to signup thru this automated system, and we will be contacting each of them individually to insure they are refunded and happy with the resolution. Thank you, David Deutsch, President Tris Telecommunications, LLC (800) 547-4057 x1001 ___ 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
