similar to: DEFAULT_USERAGENT

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

I am looking for a good analog desktop phone to use with asterisk and my sipura ATAs. I know I want Caller ID, MWI, a few programmable buttons (for asterisk features), and no external power supply (so my users can dial 911 through the SPA-3000 when the power is out). I spent some time looking at the phones at Fry's today, without finding exactly what I need. Do any of you have any

When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If

Hello All, I am having an odd problem with Armin's chan-capi_cm on builds higher than 7955. It would seem that this happens on anything higher than 7955. What is happening is the isdn is ringing, then asterisk does a goto-if and just hangs. Asterisk itself is ok, but the isdn then rings out or busys out on the other side. Outgoing works fine, this only seems to effect incoming. I

Hi, Can someone help as how to use R to program google search in the R code? I know that other languages can allow or have the google search API If someone can give me some links or sample code I would greatly appreciate. Thanks. -- Waverley @ Palo Alto

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

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is public with no NAT... everything works on the Asterisk end just fine EXCEPT that I can never check voice mail After about 30 seconds the call drops with these messagess: [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum retries exceeded on transmission 320893f1-50c13ba3-78c26164 at 192.168.1.54 for seqno 2

Hi RUsers, Suppose I want to see the data on the website url <- "http://www.nseindia.com/content/indices/ind_histvalues.htm" for the index "S&P CNX NIFTY" for dates "FromDate"="01-11-2010","ToDate"="02-11-2010" then read the html table from the page using readHTMLtable() I am using this code webpage <-

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 >> }}} >> >>

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

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

Consider this a Ticket for Icecast 2.4 ******************************************************************************** If you look at {{{ admin/stats.xml }}} on a Icecast-KH server (default setup) and an Icecast 2.4 server (default setup) the following is one of the things that the KH branch has as extra info. {{{ <listener id="3581"> <ID>3581</ID>

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

Hi RUsers, Suppose I want to see the data on the website url <- "http://www.nseindia.com/content/indices/ind_histvalues.htm" for the index "S&P CNX NIFTY" for dates "FromDate"="01-11-2010","ToDate"="02-11-2010" then read the html table from the page using readHTMLtable() I am using this code webpage <-

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

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',

You can use the following: switch3*CLI> show function SIPCHANINFO switch3*CLI> -= Info about function 'SIPCHANINFO' =- [Syntax] SIPCHANINFO(item) [Synopsis] Gets the specified SIP parameter from the current channel [Description] Valid items are: - peerip The IP address of the peer. - recvip The source IP address of the peer. - from