similar to: Trying to get *8 call pickup to work

Compiling a simple automaton created by GNU bison with -O1 or -O2 resulted in the following machine code: 1300 /*-----------------------------. 1301 | yyreduce -- Do a reduction. | 1302 `-----------------------------*/ 1303 yyreduce: 1304 /* yyn is the number of a rule to reduce with. */ 1305 yylen = yyr2[yyn]; 0x0000000000400c14 <rpcalc_parse+628>: mov

Hi, What is cast telling me when it says the following? Aggregation requires fun.aggregate: length used as default What is 'length'? I've taken a small subset of data and wondered what EnTime vs ExTime might look like. cast is kind enough to give me a table but I don't understand the values in the table. They seem to sum up ro be the same as the total dimension of the data

On Sat, Mar 19, 2011 at 1:44 AM, Csaba Raduly <rcsaba at gmail.com> wrote: > Compiling a simple automaton created by GNU bison with -O1 or -O2 > resulted in the following machine code: > > 1300    /*-----------------------------. > 1301    | yyreduce -- Do a reduction.  | > 1302    `-----------------------------*/ > 1303    yyreduce: > 1304      /* yyn is the number

Hi everyone! I just upgraded my Samba PDC to a active directory (I followed the migration instruction of samba-wiki). Without any error message or something. *happy* My PDC was running with a bind9 and slapd->openLDAP. I just turned both services off and want to use the samba-internal ones. My problem now is that I can't login with my domain members (just tried it on my server ->

Hi, I've tried to capture the basics of this problem I'm having. Been working on this for a couple of days and just cannot get past it. As a test of this list software I've attached is a small text file zipped up. I hope it gets through but if it doesn't I'll post the actual text which is only 26 lines. Put it somewhere sensible and change the first line in the code to point

Chris Lattner wrote: > On Thu, 2 Mar 2006, Chris Lattner wrote: >>> Any ideas what could be wrong? >> >> Sorry for the delay, please try this tarball: >> http://nondot.org/sabre/2006-03-02-llvm-gcc-4.tar.gz > > Actually, do to a recent change in CVS, this tarball will probably not > work anymore. Please apply the attached (small) patch on top of it in >

Hello. I am in a strange situation. I have two asterisk. Asterisk "A" makes a call for asterisk "B" by IAX. Asterisk "B" recives the call and delivers it to Openser by SIP. The problem is openser printing this in the screen: ERROR: parse_to : unexpected char ["] in status 5: <<"David" <sip:>> . ERROR:parse_from_header: bad from header

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

hi, I met some problems in R, plz help me. 1. How to do a intersect operation among several groups in one list, without a loop statement? (I think It may be a list) create data: myData <- data.frame(product = c(1,2,3,1,2,3,1,2,2), year=c(2009,2009,2009,2010,2010,2010,2011,2011,2011),value=c(1104,608,606,1504,508,1312,900,1100,800)) mySplit<- split(myData,myData$year)

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information:

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

On Mon, 6 Mar 2006, Vladimir Prus wrote: > ../../2006-03-02-llvm-gcc-4/gcc/llvm-convert.cpp:1305: error: parse error > before `__attribute__' Can you send me the preprocessed .i file for this file? To work around the above error, just remove "ATTRIBUTE_UNUSED" from that line. > ../../2006-03-02-llvm-gcc-4/gcc/llvm-convert.cpp: In function `tree_node* >

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

SDP question if * recieves "a=rtpmap:103 telephone-event/8000" it shouldn't it send out the same "a=rtpmap:103 telephone-event/8000" to the other side of the connection? and not something like "a=rtpmap:101 telephone-event/8000"? Thanks