similar to: * SIP dial out with multiple sip users

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Hi, I finally got Asterisk installed using the German installation CD from http://www.asterisk.de.ms. I got two SIP phones working (SIPPS) asterisk*CLI> sip show inuse Username incoming Limit outgoing Limit 5678 0 N/A 0 N/A 1234 0 N/A 0 N/A

I am having a couple of (unrelated) problems with my polycom 501. 1. The buddy watch is just not working. It tells me that everyone is online, whether or not they are. Here is an example directory entry for one of the peers (whose phone is not registered). <item> <ln>F</ln> <fn>J</fn>

Hi all I have members in my application, and every member can assign other members as buddies. class Member < ActiveRecord::Base # Hat folgende Buddies has_and_belongs_to_many :buddies, :class_name => ''Member'', :join_table => ''members_have_buddies'',

Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag -------------- next part

A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs

I am running CVS Head i686 running Linux on 2005-06-30 22:55:14. I have SIP Buddies installed using MySQL. If I try to set up a ATA that has 2 two phone lines (resulting in 2 lines on 1 IP address), my second line can never authenticate to dial out. I ran ethereal and found that Asterisk is "looking at the IP the request came from" and then, apparently looking up the IP address in

Hello! I have 2 table: users and buddies User: id, name, ... Buddy: id, user_id, user_buddy_id, ... So if you have 2 users 1,jack and 2,fred and fred is a buddy of jack, there is a Buddy object: id=1, user_id=1, user_buddy_id=2 I can declare only one belongs_to in Buddy and one has_many in User. And there is conflict if I had the second one (the first one is discarded) class User has_many

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 11.18.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.18.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip

>From a quick look at the source code, it looks like onComplete is only called when the whole thing is done - in the case of Updater, right after the update, in the case of PeriodicalUpdater, right after it has been stopped. It looks like you need to pass your callback method as an ''insertion'' method in the options. This takes two parameters, the first will probably be the

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the

Hi buddies, I encountered DTMF issue when I tried to place call from x-lite to a sip conference serice,here is the diagram. X-lite---->Asterisk--->Cisco SIP proxy---->SIP Conference service The Call can be established,and I can hear from x-lite the prompt of the conference,but when I input any digits,nothing happened,the conference service did not recognize my input.At the same time,in

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"

Hello all, I know it''s a Prototype question, but I hope someone can tell me what I''m doing wrong (I hope Prototype gets documented soon ;-( Problem: using a onComplete callback with Ajax.PeriodicalUpdater (using scriptaculous 1.5 with Prototype 1.4) Works: function fooBar() { Element.hide(''foobar''); } new

Hello: I'm newbie in asterisk, please help me. My context is as follows: 192.168.4.2 --> Asterisk 11.13.1 complied from source 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway When I call from a GSM cell phone, my TG100 GSM gateway answers and dials extension 7777 (configured as a hotline on TG100) to asterisk server, but asterisk server sends me "SIP/2.0 401

2014-11-12 2:45 GMT-02:00 Luis Eduardo Cortes <luedcortes at gmail.com>: > Hello: > > I'm newbie in asterisk, please help me. > > My context is as follows: > > 192.168.4.2 --> Asterisk 11.13.1 complied from source > > 192.168.4.4 --> Yeastar NeoGate TG100 GSM gateway > > When I call from a GSM cell phone, my TG100 GSM gateway answers and > dials

The only call that PeriodicalUpdater will make periodically to you is the insertion call. I think of onComplete as being ''I''ve done everything I was going to do'', which for Updater is right after it''s done a single update, but for PeriodicalUpdater, it''s after it''s finished all of its updates, so you''ll only get it once for any call

My girlfriend and I have been dating for two years, and she just told me she has an STI... Actually, I currently have three different models, like: Dog, Whale, Monkey They all have some similar attributes, but, they are unique enough to break into their own models. I could use STI, but I think the table would just have too many columns. Now, I wanted to let the Dog''s, Whale''s