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Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

Hi, I've got a fairly working Asterisk setup, with a few minor glitches, one of which is very very irritating. Sometimes, during a call, the remote end just drops off. We're using software SIP phones (SJPhone) connecting to * then out through analogue lines with X100P cards. There is nothing in the logs and nothing on the console, the call just seems to 'go away'! Can anyone

Hi there, I'm wondering if there is a way to assign a different Caller ID to each Zap interface. I have 3 Digium X100P cards, and I'm sure there must be some way of configuring zapata.conf to allow each line to identify itself with a different Caller ID string. Many thanks, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

Hi there, I'm having some trouble with getting Asterisk to make a call, I think it should be quite easy, but anyway... Using the following file contents: ## Channel: Zap/3/<TEL NUMBER HERE> MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: phones Extension: 502 Priority: 1 ## Extension 502 is simply one that plays a sound back. When I dump this file into

Is it true that every time we make a change in the configuration file we need to restart the asterisk server. This will not be practical in the production environment. Thanks,

This strikes me as something that should be really very simple to do, but I can't figure it out. Is there a way of logging all SIP debuging info to a file somewhere? It would help me greatly! Cheers, Steve -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077 7338

-----Original Message----- From: woody+asterisk@solutionsfirst.com.au [mailto:woody+asterisk@solutionsfirst.com.au] Sent: January 12, 2004 11:25 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]

Hi there, Just wondering if there is a way to speak to the person you are transferring a call to before actually connecting them to the incoming call. E.g. "Hi, Colleague, I've got Bill from Microsoft on the line here... putting you through now" Then actually transfer the call. Does that make sense!? -- Steve Foy | http://www.unite.net UNITE Solutions | Tel: 028 9077

Our VoIP provider wants to use a restricted set of ports for H323. All their current users have hardware at the customer premises which does this successfully for them. We are trying to set up Asterisk to do the same (with their co-operation). I am not familiar with the H323 terms for these connections, so bear with me... The (Connect/Listen) Port is 1720 (set in (o)h323.conf) Tcp Range

Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----

Hi, I am getting occasional rejected packets like so: Jul 31 09:52:03 firewall kernel: Shorewall:all2all:REJECT:IN=eth2 OUT=eth0 SRC=192.168.10.91 DST=132.147.22.6 LEN=48 TOS=0x00 PREC=0x00 TTL=127 ID=55364 DF PROTO=TCP SPT=1147 DPT=23 WINDOW=16384 RES=0x00 SYN URGP=0 Jul 31 09:52:46 firewall kernel: Shorewall:all2all:REJECT:IN=eth2 OUT=eth0 SRC=192.168.10.26 DST=10.9.100.30 LEN=48 TOS=0x00

I am having issues with Privacy Manager and Zapateller. If I set callerid="" on a sip user zapateller sends the tones If I set callerid="Anonymous" <8475551212> zapateller doesn't send the tones If I call from a phone after dialing *67 zapateller doesn't send the tones In the last 2 cases, the display on the phone shows -Blocked Call- PrivacyManager always gives

Hi All, In theory if I do this; exten => s,1,Zapateller(nocallerid) exten => s,2,Privacymanager exten => s,3,Dial(a bunch of SIP extensions) My callers should only hear the anti-telemarketing tones if they call from a line that has no caller*ID and then get offered an opportunity to enter it, right? What I'm finding is that in the event of no CID the caller gets dumped into the

Hello there fellow R-users, I have received some data which comes in the following format: example1<-"200301" The first 4 digits correspond to the year and the remaining 2 digits correspond to the week of the year. I have tried to convert this to a date by using strptime as follows: strptime(example1,format="%Y%U") where U (looking up strptime) is the week of the

Hi there... I have a simple * setup with about 11 Soft phones (SJ Phone). The clients don't support a consultive or supervised transfer (I believe that's what it is called). Tris is a feature much desired by the powers that be and they want me to "make it work" :) I was wondering if there was a way to do this with and AGI script or the like so that when Staff 1 gets an external

Been tinkering and found Zapateller appears to be answering when I didn't expect it to. I have something like so: [incoming] exten => s,1,NoOp ;Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Dial(${RING},20) ... I have a 1x1 analog * installation with a couple IP phones too. I've got the FXO interface connected to the home phone line. When I get an

Here's the latest batch of words to get shipped out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all

Hi, I have been using Zapateller with a TDM400 no problems at all, but recently I have ported our BT number to a VoIP provider, and have a strange problem. When I phone our number I first get the BT unavailable three tone sound, and then it actually connects the call via IAX2. So, I disabled zapateller in the dialplan and tried again. Would you believe it worked fine. Has anybody else come

This message is in response to Flash operator problem. My op_server.pl seems to be same. I also created the variable.txt to the /var/www/html/panel folder and when I run htt://192.168.0.0/panel it just says at the bottom transferring data. I don't see anything on the screen. I also checked my manager.conf file. I was able to telnet into the manager interface and it's running fine. So I am

Hi All I'm tracing a very strange problem which I could reproduce with different versions up to 1.2.5 (sorry, didn't update to a newer one yet). Scenario 1: Problem does not occure. ============================= Sip Phone registered directly to the Asterisk. exten => i,1,Zapateller() exten => i,n,Playback(invalid,noanswer) exten => i,n,Hangup() Works like expected. I dial an