similar to: sip calls not going through

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

Hello everyone, I have created a patch that allows an iPod touch or iPhone with "CoreMedia" to receive Icecast streams directly from the server. I did modify the original patch from the mailing list to include user agent switching for CoreMedia vs. other browsers. Please let me know how it works. I have been streaming a local weather radio on Icecast with the iPhone patch (attached)

On 2014-10-23 09:02, "Thomas B. R?cker" wrote: > Thanks for taking the time to report this. > > On 10/23/2014 06:38 AM, Roger H?gensen wrote: >> Consider this a Ticket for Icecast 2.4 >> >> ******************************************************************************** >> If you look at >> {{{ >> admin/stats.xml >> }}} >> >>

I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While everything seems to be working fine I keep getting this error on my log files: [2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not execute 'UPDATE "sip_buddies" SET "ipaddr" = '192.168.2.23', "port" = '5060', "regseconds" = '1392692118',

I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. This is probably just me not understanding what is going on, but I was playing around last night and I used the sip unregister <extension> command on the CLI. I thought the boxes would re-register when their registration interval was up. This is not what is happening. Now the devices are failing to register, even my

when media player classic is usually bundled in vary codec packs which are widely used by most of the windows user that insode code pack, so it would be nice if icecast support it instead of just fixing media player classic. and btw it looks like media player classic is not maintained anymore, except there is a mpc-hc the problem is that MPC/MPC-HC is too strict that it require the server

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Hi all! Once again, Apple has messed with their trailers/website such that I can no longer play their trailers on Centos 6 (note that it still works fine on my two Fedora machines, F17 and F19, without any special settings. on Centos, I've long ago found that by setting the user agent to certain values I could make it work, but now that no longer helps. when I try to view a trailer all I get

Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 => SPA-1 and 5061=> SPA-2 on the internet router * SPA-3, we have a pat 5062 => SPA-3 * SPA-4, we have a pat 5063 =>

Greetings, I have one PSTN line connected to my Asterisk@ Home box with call waiting. I also have an SPA-2002 connected to an analog phone. When I am calling on the PSTN and a call waiting beep comes through, I can hear it, but when I press the flash key, nothing happens. It is as if the Sipura is not passing the flash through. I monitor the asterisk box with the verbosity turned up, but

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent

I am sending this problem for 2nd time. Please help. Thanks _____ From: Omar McKenzie [mailto:omckenzie@trenetinc.com] Sent: Thursday, September 08, 2005 9:57 AM To: 'asterisk-users@lists.digium.com' Subject: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" I am not able to get softphone registered (active) with * . new installation , new user

Hi R Users, I want to get the data from the url given from 10/09/2012 to 15/10/2012. I don't know how to pass the parameters . ....................................................................................................................................... library(RHTMLForms) > > ff = getHTMLFormDescription("

Hello, I'm running an Asterisk 1.4.14 on a linux machine. Serving SIP Snom users. I've noticed that each time Asterisk is restarted, for the first 5-10 minutes, the SIP users can dial but cannot be dialed until each phone re-registers itself against the server. So only after the "Saved useragent...for peer 111" line appears on the Asterisk console, then the 111 user can be

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

Somone have tested the last scriptaculous version that ships with last prototype? I you make an Effect.Pulsate, the element stays hidden after the effect finish if the element don''t have opacity stablished. This is for the changes on the setStyle method on prototype. The original code is:

Hello all, I have a new Sipura SPA-2000 that I am trying to configure beind a NAT. The SPA is able to register to the asterisk server without a problem and the SPA can make calls to other extension that are not behind a NAT. However, when I try to call the SPA from another extension, the extension connected to the SPA rings, the user at the SPA answers, and there is no audio in either

Hi, I followed the TLS/SRTP tutorial on the wiki [0] using Asterisk 11.8.1 on CentOS 6.5 x86_64 and CSipSimple on a Nexus with Android 4.4.x local wifi. The phone seems to register but directly after that things fall apart (turning SELinux off made no difference): *CLI> -- Registered SIP 'encrypted' at 10.0.0.137:58079 > Saved useragent

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk