similar to: Calls dropping off

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

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

Has anyone from digium looked at why there is a 30 min to 3 hour lag on messages on this list? I.e looking at the last 50 messages I've received, the lag is about 90 minutes between the time sent and the time received. Sometimes this drops to as little as 4 minutes. Is this problem worse for me because my email address starts with "w" and my copies of the emails get sent after

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

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

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

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 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

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, Have posted to this list a couple of times and have always received great responses and help. I have a basic * system setup Using 3 X100P cards with 6 Snom200 IP phones. It was a bit of a struggle getting everything up and running but have been pretty happy with the flexibility and ease of *. My major problem is one that has been discussed on this list many times before. The echo

I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait("Zap/1-1","1") in new stack -- Executing Answer("Zap/1-1","") in new stack -- Executing DigitTimeout("Zap/1-1"."5") in new stack -- Set digit timeout to 5 --

I have a * box running * version 1.0.3 with two X100P line cards in it and Cisco 7960 IP phones. Everything seems to work pretty well with the exeption that the system hangs up on phone calls for no apparent reason. It does this on both incoming and outgoing calls through the POTS line (currently only have one). The only thing in the asterisk console with maximum verbousity is " -- Hungup

Hi everyone, I just installed asterisk on my system with the purpose of rerouting calls on sip channels. I don't think i need any hardware for that. I am using LIPZ4(zultys) and sjphone as softphones. I tried setting up both of them and to call one from the other on the same machine, however could not. I 1-) I could connect sjphone in isolation to freeworld dialup howver i got no sounds

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

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

Hi ho, is there anyone out here that is making use of the regcontext and regexten settings in sip.conf? I've tried this on two Asterisk boxes (1.2.10 and 1.2.12.1) and in both cases I don't see the Noop priority 1 being created upon SIP client registration, "show dialplan xxx" reveals no change. And yes, I have also read and checked bug 7144; if I go down that route and no

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

Dear Friends, I installed CENTOS 4.4 on server. I need DROP MSN Messenger using IPTABLES, I created the rule below. $IPTABLES -A INPUT -p tcp -m string --string "x-msn-messenger" -j DROP But, When I run IPTABLES, I have received follow error: DROP -> MSN Messenger iptables v1.2.11: Couldn't load match `string':/lib/iptables/libipt_string.so: cannot open shared object

Some of my end users have reported to me that occasionally they'll be in the middle of a conversation and the call will be dropped. I have yet to catch anything unusual when debugging the channels. Has anybody had this problem before, if so, how did you solve it? My hardware: 2 X100P 1 TDM40B Thanks for your time _________________________________________________________________ STOP MORE