similar to: SIP ACCOUNT CODE not included in CDR when SIP Status is "Unknown"

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Hi, The iax.conf is below and the trace. Any ideas please? disallow=all allow=g729 trunk=yes qualify=yes qualifysmoothing=yes nat=yes canreinvite=yes context=OutboundWS transfer=mediaonly Executing [082449627 at private:1] Dial("SIP/919-094d6e60", "IAX2/ECom-iax/2782449627|60|") in new stack -- Called ECom-iax/2782449627 -- Call accepted by xxx.xxx.xxx.x (format

Hi, I have a problem with IAX accounts... I set up a few months ago an Asterisk server, with mysql support to load iax accounts. Settings seems fine because apparently the system works as expected. Yesterday I tried to add another iax account in the iax.conf directly. And I have a problem with this new account (named 444). I can authenticate from 444 to the server, and I can receive calls from

Greetings list, I have a scenario where half a dozen phones at a site appear to be dropping offline for a few seconds every few hours, but the connection between them and the asterisk server remains up. It's been suggested to me that the problem might be to do with qualify - which is enabled in this case. However, I don't really want to disable it if at all possible - it's a very

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: >>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>>> trunk between them. > Carlos, > > Had caller-id ever worked between these two systems? > > Doug > -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161

Hi, I am puzzeled with a differing result of princomp in R and FACTOR in SPSS. Regarding the amount of explained Variance, the two results are the same. However, the loadings differ substantially, in the unrotated as well as in the rotated form. In both cases correlation matrices are analyzed. The sums of the squared components is one in both programs. Maybe there is an obvious reason, but I

ok, i'm puzzeled here I just recently reinstalled windows 2k, and i've been useing EAC to rip my CDs for a while now. i wanted to test it out (i had to reconfigure it) so i put in a burned CD with a few pretty bad scratches (used a hunting knife to make em) When i rip it i don't get any errors (with CDex to) but of course when i listen to it i hear them. This is strange bceause

hi netters, I'm a newbie to R and there are some very simple problems puzzeled me for two days. I've a dataframe here with several columns different in modes. Two of the columns are special for me: column 1 has the mode "factor" and column 2 has the mode "numeric vectors". The values for column 1 are either "T" or "F". I wanna do two things:

Dear ExpeRts,t I am trying to read tab delimted data produced by somewhat brain dead software that seems to think it's a good idea to have an extra tab character after the last column - except for the header line. As explained in the help page, read.delim now assumes that the first column contains the row.names (which is not even wrong) but now and all col.names get shiftet by one column.

My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse & repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 00000

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will work because port 5060 on the private address will still be port 5060 on the public address. With PAT the port could be anything over 1024, but usually much higher, and the originator will send to port 5060, which your NAT router will drop.

I'm attempting to connect to a Metaswitch, inbound only (at this time). The Metaswitch is the only "connection" (at this time). All I'm getting so far is a bunch of "OPTION" messages which my Asterisk box replies to but I don't get inbound calls. Here's my sip.conf. As you can see I've been trying a bunch of different options without success :(

hi, I have 2 asterisks with the following configuration. asterisk server 1 (S1) has an user 9002 asterisk server 2 (S2) has an user 9003 Both users can make call to each other without problem. Now I add both users to both servers, i.e. asterisk server 1 (S1) has users 9002,9003 asterisk server 2 (S2) has users 9002,9003 When 9002 dials 9003, Dial(SIP/9003@S2) or visa versa. Both processes

Hi all, I am writing here because the JRI mailing list seems abandoned... maybe some of you guys know an answer or has a pointer into the right direction. I am quite confused by the jars that exist between rServe, JRI and rJava. What I want to do is simply using a local R instance from Java and I thought (and still think) JRI is the way forward. I did the following steps: 1) Installed

Hi, i have context in a dialplan, I want to "execute" this context without insert the Answer Application (s? ..without call any ext). Example : [sistema-allarmi-principale] exten => s,1,Set(GRUPPO=${DIAL:-2:1}) exten => s,2,Set(ALLARME=${DIAL:1:1}) exten => s,3,AGI(checkgroup.php|${GRUPPO}) ;rest of... I tried with a Call Data File.. i create a CallDataFile like this :

I am trying to tweak my dial plan and I am running into a problem. Sometimes my VoIP out bound calls do not complete on overseas calls(busy or just a hang-up). Is there a way in the dial plan to automatically dial out of my PRI when something like this happens. Either by time limit by a failure event? Any point in the right direction would be great Thanks, CLI output (cleansed to protect the

Hi, I recently installed asterisk, i used sterisk-1.4.20.1, i i set directrtpsetup to yes, no whow would i know if the rtp/media is not passing to asterisk. any tool> or can u just sniff? regards, ron

Hi, I want to be able to bridge two sip channels using direct RTP between my endpoints (Audio IP : not local) but without using reinvites. So I set up my asterisk sip endpoints as follows: [test1] type=friend host=dynamic username=test1 dtmfmode=info context=test_rtp allow=all canreinvite=no directrtpsetup=yes [test2] type=friend host=dynamic username=test2 dtmfmode=info context=test_rtp