similar to: Asterisk to BabyTel VoIP SIP Provider

Hi, I'm new to *. I just installed my X101P card with * from the source on Mandrake 10.0 and I test it. Everything seems to work fine. When I call at my home office all the demo ivr seem to work. But I have one question regarding * using /dev/dsp0. I only have one sound card on my system and it has to be use by my personnal PVR called MythTV. I though that * did not need a sound card to work.

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

I am having problems burning DVD's from commandline. I was wondering if anyone else has had any luck using cdrecord-2.01 and cdrtools that is supposed to add DVD support? Thanks! -- Dexter -- Dexter Fitzgerald Stowers Systems Programmer I Systems Administrator Unix/Linux Systems 142 Freeman Hall College of Engineering and Sciences GSEC, GCIH, GCIA, RHCT, RHCE -------------- next part

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

Turn on sip debugging for this peer and watch for the options sending and response. If you are getting a response to your options asterisk shouldn't be marking the peer as unavailable. is your asterisk behind a firewall? On 20 March 2015 at 13:42, thufir <hawat.thufir at gmail.com> wrote: > I wasn't able to get much out of babytel, beyond the fact that I was, > apparently,

It looks as if that is more of a question/issue with your router, rather than Asterisk. I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut" You SHOULD be able to communicate between devices on the

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

In the Asterisk log I see: --- [Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29] <--- SIP read from UDP:198.38.7.34:5065 ---> SIP/2.0 200 OK To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5 Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport Call-ID:

Hello Everyone, I am using this bash script to pull resource id http://fpaste.org/84173/51169913/ this whole macro http://fpaste.org/84174/51176013/ that what I see when dialplan ran Executing [s at macro-missed-call-in:3] Set("SIP/babytel-00000022", "RES=9c32ecc4 -- ") in new stack -- Executing [s at macro-missed-call-in:4] GotoIf("SIP/babytel-00000022",

On Wed, 08 Apr 2015 16:10:30 -0700 thufir <hawat.thufir at gmail.com> wrote: > I want to do something like: > > > exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) > exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) > exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) > exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) > exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN})

I want to do something like: exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _011.,1,Dial(Dial({TOLL}/${EXTEN}) exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN}) exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN}) exten =>

>I called Telus before Christmas requesting some sort of VOIP connection. >We are going with babytel. I'll advise how that works when it is up and >running, hopefully next week. [plug] www.thinktel.ca I know the guys they are competent they will sell IAX. Peered thru GT in Downtown Edmonton.

Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default

Hi, As I understand, composite primary keys aren''t supported in ActiveRecord yet. May I ask if there are plans for this feature? Better yet, if this feature is under development, how''s the progress going? I''m not trying to use legacy databases. I tend to think that using multiple integer column id''s (composite primary key) are often natural way to