similar to: Outbound Call & SIP Results

Hi folks, I've been using * for quite a few years and everyday it surprises me more. I was recently analysing some captures with ethereal/wireshark and found out that * was doing DNS A queries for domain names like channel.mydomain.comwhere channel is the typical string of the dstchannel or channel field in the CDR entries. Obviously those queries returned with negative answer because it

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

Hi all I share my Asterisk Server with a few friends. It is connected to PSTN, and various SIP Providers. I offer Free Calls to my friends, but myself I would like to be able to make calls to non free destinations via my PSTN Line. Now I do this in my dialplan: ----------------------- [myself] ; National Destinations exten => _0z.,1,Dial(SIP/someisp/${EXTEN}); exten =>

we tested asterisk 1.6.2.0, found that when call from one sip_channel to another sip_channel , ------------------------------------------------------------------ exten => _X.,1,Noop() exten => _X.,n,Dial(SIP/${EXTEN},50,TtgM) ------------------------------------------------------------------ in asterisk 1.6.2.0 ,when sip user config to use dtmfmode=rfc2833 , it's ok, but when both

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi all, Is there a way to get the "called id" (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $callerid = $input{'callerid'}; $AGI->say_digits($callerid); } Thanks in advance, Angel.

I've just been doing some tests using the manager API to originate an outgoing call via a X100P and connect the call to an extension: Action: Originate Channel: Zap/1/01234567890 Context: local-extensions Exten: 6000 Priority: 1 I've noticed that extension is getting called as soon as the outgoing call has been placed, rather than when it is answered. Is the X100P capable of detecting

I've taken the liberty of starting a new thread to discuss the design of the Asterisk CDR mechanism. The discussion has been kindly initiated by murf putting together a proposal: http://svn.digium.com/svn/asterisk/team/murf/RFCs. After reading the proposal I still don't think it's the right way to go. To my mind adding more channel variables increases the complexity in a situation

hi. could somebody explain how exactly the following parameters in zapata.conf work: pridialplan prilocaldialplan internationalprefix nationalprefix localprefix privateprefix unknownprefix the wiki & comments doesn't quite explain them. and phone companies are absolutely no help. i've setup systems in the US & China with trial & error until it works. now i'm setting up a

Hello list, I got a couple of those "wouldn't it be great questions", as following: 1. Is it possible, with asterisk to hold a central phonebook directory of callers?, so that when this party calls a "textual" caller ID will be displayed on the phone display. 2. How can this be configured with Trixbox, I've looked at the configuration options - I assume it plays no

I've having a big problem after having upgraded to 1.2.4 and chan_capi-cm 0.6.4 When making outgoing calls I don't seem to have any control over the CLI that is presented to the called party -- it can be any one of the MSNs allocated to the line, allocated on what seems to be a random basis. This is on a BT Business Highway line (which is essentially an ISDN2e line with two built-in

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

I'm having a bit of trouble setting up my sip.conf entries to accept calls from a particular provider, and the problem really is that I am unclear exactly what parts of the INVITE are supposed to match what parts of sip.conf. I couldn't find this info on the wiki, so if someone here can shed some light, I would be very grateful! Here are the relevant lines from the INVITE (from sip

Hi eveybody again! I don't want to be annoying, but if nobody can help me with this, I'll have to desist of working with SIP.I have some questions about SIP, as I wrote in another mail. I have a SIP Gateway and I have two phones (an analog one and a DECT one) conected to it.Also, I have two Dlink dg102s with four phones conected to them. The main problems are two. Calls between the

Hello, I can't find the information on the wiki :( When is the quota recalculated after a mail deletion ? For instance, I am running low of storage and I use Thunderbird to delete large mail. I only notice a recalculation when I quit Thunderbirdb and I relaunch it. Even, with doveadm CLI, as long as Thunderbird is not disconnected on the client side, the server didn't recalculate the

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi- Having a little trouble with the SMS application in the UK. I've read the wiki topic and also the BT documentation SIN 413, which describes the service technically. I've been able to successfully send messages from my customer's ISDN landline to various mobiles, but cannot get the mobiles to be able to send messages back to the landline. I've tried sending from both Cellnet