similar to: busy signalling on PRI doesn't work...

hi I keep getting sound dropouts during SIP re-registration, and I can't find a remedy for it. I use SIP friends from MySQL. Below is SIP debug output for the re-registration Thanks in advance roy ------- *CLI> sip debug ip 80.202.161.221 SIP Debugging Enabled for IP: 80.202.161.221 *CLI> Sip read: REGISTER sip:sipgw1.briiz.no SIP/2.0 From:

Hello, I am very new to R and data analysis in general. I am trying to generate values to append to my data frame using conditional statements. I am playing with this simple example: a <- c(1:4) b <- c("meep", "foo", "meep", "foo") d <- cbind(a, b) now what I want to do is , each time there is a "meep" in column 2 of d, print

I've a call center using IAX softphones provided by a third party. We've observed problems where the IAX phones seem unable to use our PRI trunks. A sample anonymized call is provided below with the PRI debug calls embedded. Any thoughts, comments or suggestions would be welcome. In anonymizing it, I preseved the format and number of digits sent. -- Accepting AUTHENTICATED

hi all when I (using my D-Link DPH-100M MGCP phone) press #, I get told 'transfer'. I dial the new number and after that, * just tells me 'meep meep meep' [hangup] any ideas why? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to "no" and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce <bruceb444 at gmail.com> wrote: > Futher check

Hi Folks, I've Asterisk Bristuffed up and running behind an Auerswald Commander Basic ISDN PBX on the internal ISDN Bus (BRI/PTMP). The HFC Card works marvelleous for outgoing calls (as the parallely installed avm fritzcard with chan_capi does), but when I'm trying to call in, I get a short ring signal and then the connection is terminated. This does not happen with chan_capi and

handled. So....what do I do? Thanks, MD =1=================================================== !! Invalid Protocol Profile field 0x11 -- Accepting call from '2004000' to '111' on channel 0/23, span 1 -- Executing NoOp("Zap/23-1", "Incoming call from Meridian1") in new stack -- Executing NoOp("Zap/23-1", " From number: 2004000|

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (just one!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR some time

picture. I know the firmware on the Nortel is old, so I'm guessing that libpri is sending something that the Nortel does not know how to handle. Is there a way to dumb down what libpri sends? From everything I've read PRI is an evolving standard - and older devices may struggle with newer extensions/developments. (This might be very handy for users trying to talk to old pbx's.) Is

Hi all, I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on a openSUSE 10.2, i586 host. The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the destination FAX devices are in Germany too, but in different areas, so I have to use a city prefix. I did set the pri device in debug mode, below are two calls, to two different FAX numbers, the first is

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

Hi, Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls just rings and rings. We never get the "The number you are trying to reach...". If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get the message on the 1st ring. Here is the PRI debug of such a call that just rings and rings. Any ideas? PRI debug sur CPL: -- Executing

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

Hi all, I'm debugging a PRI problem, i can see the calling number but i get a busy all the time. From the output below, I guess asterisk hangs up immediately. Can anyone point out what the problem is? Thanks in advance. *CLI> < Protocol Discriminator: Q.931 (8) len=32 < Call Ref: len= 2 (reference 4865/0x1301) (Originator) < Message type: SETUP (5) < [04 03 80 90 a3] <

I have trouble getting ${EXTEN} or ${DID} when receiving incoming call from TE110P in Japan. Does anyone have idea how to fix this? Is this because TE110P does not support INS1500 as switchtype? pri debug does not show "Called Number". < Protocol Discriminator: Q.931 (8) len=32 < Call Ref: len= 2 (reference 68/0x44) (Originator) < Message type: SETUP (5) < [04 03 80 90 a2]

We have set up an Asterisk PBX managing a EuroPRI in Italy. We have conneccted to the asterisk PBX some Cisco IP Phones and a Panasonic PBX with 10 analogic phones. If we dial an unassigned telephone number we are not able to listen to PSTN Operator message telling that the subscriber does not exist both on IP phones and analogic ones. Asterisk simply hungs up. We have repeated the test using a

Salve, I'm somewhat stuck on how to get DTMF working with my setup and googling didn't yield anything similar. My setup consists of one CAPI-capable board (AVM Fritz!DSL) connected to a BRI (T-ISDN), one HFC-S board running in NT-mode connected to an internal S0 bus with some ISDN devices (DECT stations, TA) and, of course, some ethernet interfaces. ISDN standard used is Euro-ISDN.

I haven't seen anything about this is the archives, so here we go. Sorry, its a long one. My setup: Dual Xeon 2.4 GHz. One TE410P card. Span 1 populated with a PRI. Span's 2-4 empty. zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us I'm currently having a problem where one of our PRI channels seems to keep a call even after it is hung up. We have