similar to: UNREACHABLE peer

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,

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

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:

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 =>

On 15-04-09 12:06 PM, Chad Wallace wrote: >> but don't know where to put those lines. I have BABY defined as >> >channel variable: >> > >> >BABY = SIP/babytel_out >> > >> >but that seems circular, somehow. > You put them in the context for your clients... From what you show > below, I'd say they go in the "local_200"

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"

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

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

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, 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 Fri, 20 Feb 2015 15:07:35 -0500, Andres wrote: > This is showing nothing so I don't think your test message even made it > here. I think it looped in the 'doge' server. I was wondering the same thing :) in tleilax, I looked in /var/log/asterisk/messages and see: [Feb 20 15:13:19] VERBOSE[3661] chan_sip.c: [Feb 20 15:13:19] <--- SIP read from UDP:192.168.1.3:38154

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

Hi, Does anyone has configured Asterisk to connect to BabyTel (a SIP Provider in Canada) ? Here is my sip.conf (I'm behind a firewall and I already opened port 5060 and 5065 (udp and tcp) to my Asterisk server): [general] port = 5065 context = Test insecure = very register => 1514XXXXXXX:password@sip.babytel.ca When starting Asterisk, the sip registration failed after 5 connecting

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})

Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \

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

Looking at: http://appdb.winehq.org/objectManager.php?sClass=application&iId=10591 What are the steps to install this application? Yes, it's a garbage application, but I'd like to at least give it a go. Looks like msiexec apparently isn't the right approach. Should that be through wcmd instead? thufir at ARRAKIS:~/.wine/drive_c$ thufir at ARRAKIS:~/.wine/drive_c$ msiexec

I ran dovecot -a and the blizzard of data seemed ok to my limited knowledge. Is there another log I should look into to trace this error down? Dovecot and system info: thufir at dur:~$ thufir at dur:~$ dovecot --version 2.0.19 thufir at dur:~$ thufir at dur:~$ cat /etc/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=12.04 DISTRIB_CODENAME=precise DISTRIB_DESCRIPTION="Ubuntu 12.04.1

I want to iterate ?nodes? and ?leafs? for a yaml document: thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user.rb user.rb:6: undefined method `[]'' for nil:NilClass (NoMethodError) from user.rb:5:in `each_key'' from user.rb:5 thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user2.rb user2.rb:5: undefined method `[]'' for

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works