search for: 5090

...ets UNAUTHORIZED messages, despite the fact that we set the following lines in its settings in sip.conf: subscribemwi=yes mailbox=21 at from-extensions We need help in understanding how this works and what we are doing wrong. This is the SIP debug we get: <--- SIP read from UDP:10.0.0.4:5090 ---> SUBSCRIBE sip:21 at 10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.4:5090;rport;branch=z9hG4bK3663664e35 From: <sip:21 at 10.0.0.10>;tag=6d8c6ac6 To: <sip:21 at 10.0.0.10> Call-ID: 055f7edd4081e1ec0f176e0a4b395a43 at 10.0.0.4 Contact: <sip:21 at 10.0.0.4:5090> CSeq: 1 SUBSCRIBE...

...is reported below. I never receive the call though, I have a catch all in my inbound routes and it doesn't hit my context at all or not sip invite comes in: FreePBX: Trunk Name: *Spikko* Peer Detail *username=MyUsername* *type=friend* *secret=MyPassword* *host=sip.spikko.com* *nat=no* *port=5090* *fromuser=MyUsername* *disallow=all* *allow=g729&gsm&ulaw&alaw* Register String: *MyUsername:MyPassword at sip.spikko.com:5090/MyUsername* Inbound Router: *Send Any DID and ANY CID to Music on Hold* Sip debug: *Really destroying SIP dialog ' 417b3c8f3a97a82d4629343a53b2feb6 a...

Hi Guys, I have tried every single rule I could into iptables but I can't register this VPS to a provider Spikko. Finally I did an iptable accept on INPUT, OUTPUT, and FORWARD, for ports 0:65000 just to test things and still I can't register to the provider. I can easily register to another provider which uses port 5060 and I can make calls. But sip show registry only show Request Sent

Trying to copy the (binary) header of a input file directly to an output file, I've had repeatable seg faults. The call: writeChar(hdr, outfh, nchars=6144) when hdr just contains one empty string seems to be the culprit. The stack traces weren't all that illuminating, with sig 11 in memory-related functions following this. But in src/main/connections.c it looks like do_writechar

Hi, I've got report about issue when dovecot fails to start and there is no error logged (error goes only to stderr) situation: 1) dovecot is running 2) dovecot is automatically updated to new version (by yum update daemon), after update, dovecot is restarted (it's part of update script) 3) new dovecot fails to start (for whatever reason) result: dovecot not running and no error

...ral] static=yes writeprotect=no LINES => 0 [E1] exten => 33006712,1,GotoIf($[${LINES} >= 10]?101:201) exten => 33006712,101,Dial(mISDN/g:E1/34507725) exten => 33006712,201,SetGlobalVar(LINES=$[ ${LINES} +1 ]) exten => 33006712,202,Dial(SIP/192.168.0.65:5080&SIP/192.168.0.65:5090) exten => 33006712-ANSWER,203,Answer() exten => 33006712-HANGUP,204,SetGlobalVar(LINES=$[ ${LINES} -1]) exten => 33006712,205,SetGlobalVar(LINES=$[ ${LINES} -1 ]) exten => 33006712,206,Hangup() My problem is that the increment works perfectly, but the decrement is not working. I added...

hello I am using ser with asterisk asterisk on 5070 (on back end) ser on 5060 (on front end) i am getting all requests at asterisk. i tried by changing asterisk port bindport=5090 but still getting all requests from sjphone at asterisk. can any one tell what is the reason regrads Kamran __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

Hello to all, Is there an "how-to" for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___________________________________________________________________________ Faites de Yahoo! votre page d'accueil sur le web pour ret...

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in Asterisk. Although I can't send FAXes to PSTN using T38. The problem, as far...

...proper way. When this is tested and stable, work will continue to see if we can make Asterisk an T.38 endpoint. This is a very important addition to Asterisk. There is code for testing available. If you are interested, please check this URL in the bug tracker: http://bugs.digium.com/view.php?id=5090 I think this is a big step for Asterisk. Do you? If so, don't forget to say "thank you" to Steve Underwood - Coppice! Have a nice weekend! /Olle --- * Olle E. Johansson - oej@edvina.net * Asterisk Training http://edvina.net/training/

Here is the call flow: [ivr-incoming] exten => s,1,LookupCIDName exten => s,2,DigitTimeout(2) exten => s,3,ResponseTimeout(10) exten => s,4,Wait(1) exten => s,5,Background(custom/ivr-incoming) exten => 1,1,Background(pls-wait-connect-call) exten => 1,2,Dial(${RINGPHONENUMBERS},20,r) exten => 1,3,Voicemail,u${VMBOX} exten => 1,4,Hangup Running * 1.0.5. The calling party

...plicator { process_min_avail = 1 unix_listener replicator-doveadm { mode = 0666 } } ssl = no userdb { args = /etc/dovecot/userdb driver = passwd-file } -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 5090 bytes Desc: S/MIME Cryptographic Signature URL: <https://dovecot.org/pipermail/dovecot/attachments/20181126/be10f7b7/attachment.p7s>

...cernko at mpi-klsb.mpg.de> +49 681 9325 5815 Joint Administration: Information Services and Technology Max-Planck-Institute fuer Informatik & Softwaresysteme -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 5090 bytes Desc: S/MIME Cryptographic Signature URL: <https://dovecot.org/pipermail/dovecot/attachments/20181210/78042ff4/attachment.p7s>

Dear John Fox, these is still error in Rcmdr package when locale is cp936. > Loading required package: tcltk > Loading Tcl/Tk interface ... done > Error in parse(file, n, text, prompt) : syntax error on line 5090 > Error: unable to load R code in package 'Rcmdr' > Error: package/namespace load failed for 'Rcmdr' > version _ platform i386-pc-mingw32 arch i386 os mingw32 system i386, mingw32 status Patched major 2...

...cuting SetVar("Zap/3-1", "ALERT_INFO=Bellcore-r3") in new stack -- Executing Dial("Zap/3-1", "SIP/101@192.168.17.6:5060|20") in new stack -- Called 101@192.168.17.6:5060 -- SIP/192.168.17.6-67bd is ringing NOTICE[1142127920]: File chan_sip.c, Line 5090 (handle_request): Nothing to pick up == Spawn extension (house-admin, 101, 2) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1'

...ugs.digium.com/view.php?id=5374 - This is a patch that allows for the asychronous generation of outgoing frames. Very cool. Needs a decision by the Core Developers (Read Kram / KPFleming). BJ has been keeping this in synch w/ Trunk in his own branch. http://bugs.digium.com/view.php?id=5090 - T.38 Support anyone? http://bugs.digium.com/view.php?id=5574 - A nifty new app called Find Me, Follow Me that does exactly what it says. If you can, take a look at these and add your entropy to the mix! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company...

is there recomended source files for t.38 pass? latest cvs does not work for me. is it possible publish working src? ------------------------------------ turby -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060214/8dfb088c/attachment.htm

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything tha...

...progress support for T.38. > t38-bits.tgz, which can > be found in the same directory as spandsp, contains > UDPTL, TPKT, app_rxfax, > app_txfax and other code needed top bring T.38 > support to Asterisk. > > > and also here: > > http://bugs.digium.com/view.php?id=5090 > > > > > Harry > > --- Alexandr Dubovikov <surik@start4.info> a ?crit > : > > > > > On Tue, Apr 25, 2006 at 11:00:15PM +0200, > > > hgaillac-sip@yahoo.fr wrote: > > > > No, i ve just patched chan_sip.c > > > > >...

...efinition.... What am I doing wrong ? Anyone else tried Zanzibar (for those not familiar - it's a first open source based way of doing speech recognition and synthesis in convenient way) ? Thanks in advance, regards, Rob. Trunk definition : [Zanzibar] type=peer host=192.168.0.50 port=5090 dtmfmode=info canreinvite=no qualify=no Extension definition for calling IVR : exten => 510,1,Answer exten => 510,2,SIPAddHeader(x-channel:${CHANNEL}) exten => 510,3,SIPAddHeader(x-application:beanId|Parrot) exten => 510,4,Dial(SIP/Zanzibar)