similar to: RTP.C codec error 19

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

I am trying to get asterisk to dial this address : sip:723@216.52.153.207 Using a softphone on my PC (217.168.168.49) it dials immediately and I get a voice prompt .. I have configured an extension, 1303 on asterisk, modifying the demo configuration : exten => 1303,1,Dial(SIP/723@216.52.153.207) When from my softphone I dial sip:1303@217.168.168.51 on the console I get : -- Executing

Hello I have installed cepstral .... It works woderfull using an agi script but ..... when i try to use Swift("say this") is Dial plan .... I get the error [Aug 26 12:30:18] WARNING[7420]: pbx.c:3167 pbx_extension_helper: No application 'Swift' for extension (actdemo, 123, 2) Now i come to know to install app_swift Here is the issue... when i try to execute make command

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

Hi, is there a way to enable dtmf detection on zap channels? I am trying to pickup, play a ringtone and the dial out. I.e. exten => s,1,Wait,1 exten => s,1,Answer exten => s,2,Playtones(dial) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 exten => _X,1,StopPlaytones exten => _X,2,Dial,Zap/g8/BYEXTENSION|10

Hi ! I've fetched a spare AVM B1 card from the cellar, and installed it. After "modprobe b1pci" I did "capiinit" and capiinit moaned about a missing t1.b4. So I search the web and found one at http://www.avm.de/ftp/cardware/b1/x_misc/ddi/. When I now look at the controller, I finally see p2p-mode: # cat /proc/capi/controllers/1 name b1pciv4-a400 io

I went from asterisk 1.4.22 (which was working) to SVN and I am getting the message rejected because extension not found... How can I modify the print statement in chan_sip.c line 18388 to include not just the extension but the context its trying to find my extension in??? ast_log(LOG_NOTICE, "Call from '%s' to extension"

Dear, I am getting this in CLI on release candidate version of Asterisk. Any ideas, or points where to look? -- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi [Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe -- <SIP/916-fc001968>AGI Script rad-auth.agi completed, returning 0 Best regards, Josip

i'm getting Asterisk to dial an h323 call termination service .. right now getting this message: -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's' on channel 1, span 1 -- Executing Dial("Zap/1-1", "OH323/h323:723@216.52.153.206") in new stack 5:59.330 H323 Cleaner H323

Hi Shakeel, I had the same problem building app_swift (1.6..) myself and searched the web far-and-wide for a solution. I eventually contacted Darren Sessions -- who was maintaining that plug-in -- about a month ago. He was involved in another project and said he might be able get to it after a few weeks. But, since then, his website http://www.darrensessions.com/ has gone out of comission. I

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

just three "stupid" questions I need to ask .. 1. what's the sequence to press on a SIP phone to transfer a call to another extension. 2. what's the same thing if you want to hold an incoming call, speak to the other extension, then pass the call? 3. what's the extensions.conf syntax to dial two SIP extensions at once? many thanks Dave

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

i've just lost $2000 dollars or so on my first commercial asterisk installation .. i'm running a PIV class server, three Digium Wildcard FXO cards, and 10 Grandstream Budgettone SIP phones. The system was to be a PBX for a small company. After over 2 months of pissing about, the client has had his fill of asterisk problems, and asked me to take my equipment out of the building. Obviously,