similar to: CallWaiting CallerID

Hi all, With a great help from Richard Alexander (thanks Richard!) I have now a functional CallerID on my X100P. This is what I have done: - update to the latest CVS (as today at 5:00pm GMT) - modify the callerid.c file in the asterisk source like that. original : /* MDMF */ /* Go through each element and process */

Hi Gurus. I searched the lists, wiki and the rest of the web but I still do not understand this. My Setup is as follows: [ISDN via chan_capi or IAX2 DiD Provider] => [* PBX] => [IAX2 Clients (Atcom AT-320ED)] I want to get callgroup/pickupgroup and callwaiting working on the IAX phones. Some web sources told me that this was not implemented, others say that the phone has to handle

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

Some of you have been following our progress on http://farfon.convergence.com.pk as we blundered our way through the development of a low-cost ethernet IP phone that does IAX and augments the client options currently available for the kick-assterisk server. With help from the denizens of #asterisk and kind words of advice from Mr. Spencer and the rest of the gang ... we're proud to have

I have recently set to * to play with and I'm getting the hang of it.. I've seen various posts about problems where the SIP clients are behind NAT but I anted to ask a slightly different question. At the current CVS can the * SIP client register with a SIP provider (such as ICH) when the * box itself is behind a NAT. I have an ICH account which I can configure fine on an ATA 186 using

Hi, I''m trying to configure IMQ to work on the same machine with frottle (http://frottle.sourceforge.net). The problem is both feed themselves packets through netfilter queueing mechanism, but currently there can only be one netfilter queue per protocol family. To explain why I need IMQ in the first place I have to explain what frottle does. It is a deamon that tweaks the behaviour of a

Hi folks, Any Brits on the list got experience of hooking devices with standard BT plugs to the TDM400P? The TDM400P has the RJ11/RJ45 socket but I'm not sure if a standard adapter will do the trick or if we need to consider ring capacitors etc. Any links to buy the stuff would also be appreciated. Cheers, Simon -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is it possible to send callwaiting callerid to a channel without actually having a call waiting, I'm thinking if messaging a handset through the manger interface for example? Robb

Hi, now I've got caller id working on my BT line in the UK, I'd like to play a different message to those pesky sort who with hold their outgoing number. How can I do this in my extensions.conf for my [incoming-analog] context? I realise some people may call who are unable to change the way that their system withholds the outbound number, so I'll give them chance to leave a voice

Windows-32 has a time structure called FILETIME, a 64-bit value representing the number of 100-nanosecond intervals since January 1, 1601 (UTC). That is not a typo, the year is 1601. Does anyone have a clue(or algorhithm)for how this is converted to something a little more POSIX-like ? Thank you, Derek -- Derek N. Eder Gothenburg University VINKLA - Vigilance and Neurocognition

Hi my Digium FXO card isn't picking up the callerid I get from the PSTN. I have verified with a deskphone that can display the callerid that the facility works. So, it's definitely the FXO card not picking it up. As I am in Japan, I guess that NTT uses a different method to provide the callerid and so I guess that it is just a matter of configuring the FXO card so that it uses the

Hey all...I can't seem to figure out *exactly* what needs to be done when I can't flash over to an incoming callwaiting call on FXO from an FXS card. Right now if I get a callwaiting call from the FXO and hit flash nothing happens. I've been over the archives and google and it appears that this is possible and I guess it can be done with *0 but I haven't found the config on

Don't know what else to call this. Googling and some time on the IRC channel haven't gotten me anywhere. Here's the sitch, which is a bit complicated but is something my customers are in fact encountering on an everyday basis: 1. Bob is on a Zap channel talking through the PSTN to Carol. Both have the misfortune, like so many of us, of having LECs who do not offer disconnect

I have a 7960 with P0S3-08-2-00 firmware. Call waiting is enabled on the phone: ... call_waiting : 1 ... When I call from the 7960 to the 7960, I get a 486 busy here on the asterisk console. I can call from the 7960 to a soft phone no problem and vice-verse. I can also call from soft phone to soft phone with no issues either. Anyone know offhand what's going on? bug in the

Hello List, I am looking at implementing auto callwait on my asterisk box ( has single FXS to which my phone is connected). Is there a way to terminate the current active call (soft-hangup) and auto accept the new incoming call ? (I don't care for call indication). Please advice, -Akshay -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am wondering if anyone could help me figure out how to turn down the volume on all the dial tones, indications, etc.. and especially the call-waiting CHIRP! I don't want to change the txgain and rxgain because they are working at levels that I would like. However, when voice conversations and voicemail recordings are at good levels then the dial tones, busy tones, etc are way too loud.

I have a Cisco ATA-188 with two POTS phones and latest stable cvs. In any situations with call waiting (existing connection and calling again) the second call cause both calls to drop. This is the same for "internal" extensions and from external (ZAP and SIP). It seems to be a "502 - The transaction could not be executed, because the endpoint does not have sufficient

Hi, Is it possible to host call waiting service on Asterisk for a SIP device? What i am trying to achieve is that while a SIP user is busy on a call and a new call for that user comes in, asterisk should play the call waiting tone to that user. I have a vague idea that if i can get hold of the existing bridged channel when a subsequent call is received, i can then redirect that channel to

Hello All, I have new TDM04B installed and working fine with Asterisk 1.0.5 built on RedHat 9. All is working fine except CallerID that bothers me big time. I have several Panasonic and Sony phones and CallerID works fine with it (when I plug in the line into phone instead into Asterisk I get CallerID) but fails with Asterisk. I am based in California (San Francisco) and my Telco is SBC

Hallo, is there a Freepbx mailinglist? or can this be posted here? Best Regards,