similar to: Different Caller ID for each Zap Interface

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 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

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

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

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

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

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,

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

Hi Folks, I've been looking around and found some references of some Caller ID patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden and UK. It's been quite hard to understand what has finally been incorporated to the distribution (if anything) or which patches must be applied in witch snapshot of the repository. I've tried some different approaches but nothing worked

Hello list, I have an X100P clone card, and I've been trying to get it to receive Telstra Caller-ID without any luck. Hence these two questions:- 1) Will a genuine X100P work? 2) If so, where can I get one in Australia? Cheers, Jon

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

You require no changes to detect caller id with a x100p card in Au, however most carriers only provide cid as an extra cost option. Your most likely problem is CID is not being sent to you. If you believe you have the caller id being sent to you, Find a caller id box/phone and confirm it is there. In 99% of cases where I have seen people in Au complaining of no caller id, it was because they

Hi there, I have a x100p card in an asterisk box. Does anyone know if it's possible to do a hook flash / recall on an active zap channel? On what I'm trying to do... >From an ordinary analogue pstn telephone I can call someone, press recall, call someone else, press recall 3 & presto we're on a three way chat, with me only using one line - using the telephone company's

Hello I was wondering whether anyone has managed to get an X100P card to receive Caller ID information in the UK (connected to a BT line)? I guess that it probably won't work at present because BT sends the CID information before the first ring - hence there is no event to wake up the channel. I can hear it using ztmonitor, but it is far too soon for anything to receive it. According to the

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel