search for: drings

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

Hi Guys, I am making another module for Voicemail. I have three fields in a POST form that have to be connected together to make it a single 10 digit number but there is something wrong in my syntax probably. $npaa = "('$_POST[anpa]')"; $nxxa = "('$_POST[anxx]')"; $blocka = "('$_POST[ablock]')"; *$grplist = $npaa.$nxxa.$blocka;*

Hi Everyone, I have done some php coding to come up with my own FollowME module for FreePBX. The need for this has some security considerations behind it. This is what my code does at core: $sql="REPLACE INTO findmefollow(grpnum, strategy, grptime, grppre, grplist, annmsg_id,postdest, dring, needsconf, remotealert_id, toolate_id, ringing, pre_ring) VALUES

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called number clears. At the moment, if I try to do this, my IVR autoattendant picks up the ring...

Godd evening! What happens if someone calls and asterisk doesn't "Answer()" itself, but another analog phone does? Can I somehow catch this situation in my dialplan. I have an ISDN line, but with it I got a box with an adapter for good old analog phones. This doesn't seems to be directly connected to the ISDN line asterisk sees. But somehow, it must know, that the call

I am trying to get distinctive ring to work on my PSTN with no luck. I can get 2 different ring codes but it skips the context assigned... here is my complete zapata.conf: [channels] signalling=fxs_ks usecallerid=yes rxgain=1.0 txgain=1.0 language=en context=default usedistinctiveringdetection=yes dring1=134,0,0 dring2=137,0,0 dring1context=internal2 dring2context=default

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.

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,

This is an Australian situation. I have a PSTN connection that has CLID presentation enabled and has two numbers assigned to it, the primary number with the standard ring cadence: 400,200,400,2000 and the secondary number with the alternative cadence: 200,400,200,400,200,1600 CLID presentation is working fine and in zapata.conf I have: usecallerid = yes usedistinctiveringdetection = yes I am

Hi, Any of you have any experience with SIP softphone dialler that capable of local recording? (recording to files in harddrive) So far I only know eyeBeam and Express talk. eyebeam fine but there are known error with recording. Express talk recording looks ok, but sometime it doesn't have incoming voice with *. Cheers Benni-

Hello, I'm having some trouble getting callerid to work through a distinctive ring line. Specifically, I'm having a problem getting the callerid to pass from an AT&T POTS line connected to a TDM400P to SIP phones, both hard phones and soft phones. What leads me to believe that it's a problem with the distinctive ring line is that I have another AT&T POTS line connected

Using distinctive ring detection with bell202 cid, is there any way to tell DAHDI to sometimes expect the cid after the 2nd ring, other times after the 1st? I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P FXO line. No problem setting dringcontext for the 2nd DID. The 1st DID works normally, but I get no CallerID on the 2nd because the call is picked up before the

Hi, i''m sort of testing a configuration and things are not working sa i planned. i have the following network diagram: PC1 to 7 cnneced on the same ethernet hub. PC1 PC2 PC 3 PC4 PC5 PC6 on network 192.168.5.0 PC6 and PC7 on network 192.168.1.0 so PC6 work as a router. in addition, PC6 is connected to both network on the same device eth0. now on PC6, put a tbf on dev eth0 root

Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search & read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to,

Hi All, I've recently got a "second" number installed on my PSTN line, trusting the Asterisk distinctive ring detection would work as expected. It appeared to work fine at the start, as the second number generated a different ring pattern to 0,0,0 (in the console) only to realise that almost every phone call to this "second" number generated a different ring pattern.

Hello all! We have a PSTN line with four numbers calling into it. There is distinctive ring on these lines. They are are follows: 1. standard ring 2. short ring 3. long ring 4. short ring, long ring, short ring Based on the information I have been able to find, I have created the following entries in my zapata.conf file, to try and weed out some of the timings: dring1=95,0,0

I've followed the recent thread on caller id with UK British Telecom networks (where the caller id data is delivered before the first ring). My understanding is that if I use a recent CVS head (e.g. CVS-HEAD-09/18/04-17:45:52) and a TDM400 with FXO modules, all I need to do is include the line: usecallerid=uk In my zapata.conf (in the [channels] section) I've done this, but I get: Sep

.../block/sunvdc.c @@ -592,7 +592,7 @@ static int generic_request(struct vdc_port *port, u8 op, void *buf, int len) return err; } -static int __devinit vdc_alloc_tx_ring(struct vdc_port *port) +static int vdc_alloc_tx_ring(struct vdc_port *port) { struct vio_dring_state *dr = &port->vio.drings[VIO_DRIVER_TX_RING]; unsigned long len, entry_size; @@ -725,7 +725,7 @@ static struct vio_driver_ops vdc_vio_ops = { .handshake_complete = vdc_handshake_complete, }; -static void __devinit print_version(void) +static void print_version(void) { static int version_printed; @@ -733,7 +733,...

.../block/sunvdc.c @@ -592,7 +592,7 @@ static int generic_request(struct vdc_port *port, u8 op, void *buf, int len) return err; } -static int __devinit vdc_alloc_tx_ring(struct vdc_port *port) +static int vdc_alloc_tx_ring(struct vdc_port *port) { struct vio_dring_state *dr = &port->vio.drings[VIO_DRIVER_TX_RING]; unsigned long len, entry_size; @@ -725,7 +725,7 @@ static struct vio_driver_ops vdc_vio_ops = { .handshake_complete = vdc_handshake_complete, }; -static void __devinit print_version(void) +static void print_version(void) { static int version_printed; @@ -733,7 +733,...