similar to: OT - Is sip.instance useful ?

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 is ringing -- SIP/snom-00797110 answered

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379

Hello List, following this thread: http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains I tried to configure on the pjsip.conf the same endpoint with different domains like: [1000 at sip.domain.com] type=endpoint [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console:

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

Hello Gentleman, I'm new to asterisk, this is my first instalation, first post... so I'd like to apologize if this question has already been made. I googled but I couldn't find nothing similar. Here's the thing. I'm migrating from ATA to Asterisk one of my client's office and I have a very simple setup. A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a

I''ve posted a lengthy description of an authorization plugin for Rails on my blog: http://www.billkatz.com/authorization It describes a proposed DSL for authorization, a pattern for use that describes conventions, and a reference implementation that lets you test the some of the ideas. I hope that some subset of Rails programmers gravitate toward a common DSL for authorization, which

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Has anyone else run into a problem with trying to use mock_model in spec/lib ? For some reason, I can take the same spec, put it in spec/models, have it run fine, but put it in spec/lib, and have it complain about not being able to find #mock_model Thanks, Edward

With a has_one/belongs_to relationship, what''s the best way to guarantee that the belongs_to object gets created and is accessible alongside the has_one object? I *think* the after_create callback is a good choice, but I discovered an oddity while trying it. F''rinstance, if every horse has a carriage: ============ ActiveRecord::Schema.define do create_table(:horses) {|t|

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

hello i was wondering what is the use of "rinstance" in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case "rinstance" parameter is missing. when i dial "SIP/username" audio is fine but when i dial SIP/x.x.x.x:port there is one way audion.

I've tried to put a call file to /var/spool/asterisk/outgoing/ to make an outgoing video call, but not succeeded. I could hear the audio, but no video. The asterisk version is 1.4.10, with videosupport=yes The client is eyebeam 1.5.7, with h263 support. Here are some debug messages. It shows the client and asterisk negotiated the video capabilities without problem. However, the 'show

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

On Aug 9, 2012, at 10:09 AM, Sergei Larin <slarin at codeaurora.org> wrote: > Yang, > > This might not be such a tough choice on engineering side - one of the > LLVM differentiators is the ground-up, early introduced support for VLIW > specific features… Actually, LLVM lacked support for VLIW until fairly recently, and it has relatively few VLIW-specific features. Dan

Hi all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set