similar to: Generating 484 "Address Incomplete"

I'm loosing hair at cosmic speed now for the past 10 days. Welltech's Wellgate 38xx/35xx FXO/FXS SIP hardware versions seem to have very buggy firmware possibly due to hastely done porting from H.323 firmware. Is there anyone on this mailing list who was able to: 1. setup a 35xxA FXS with all ports authenticating properly with *? or 2. setup a 38xx FXO to work as dial-in from pstn to

Anyone else has problems with CVS HEAD's from today with voicemail hanging up silently without any debug/error messages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same

Replying to myself, I've just read in 1.6.1 announcement that a new Incomplete dialplan application is the one that provides what I'm looking for ... 2008/10/3 Olivier <oza-4h07 at myamail.com> > Hi, > > If my memory serves me right, there was thread (in dev mailing list ?) > explaining how we could implement 2 stages dialing with SIP endpoints: > user dials 1234

Hi, If my memory serves me right, there was thread (in dev mailing list ?) explaining how we could implement 2 stages dialing with SIP endpoints: user dials 1234 then asterisk replies 484 Address Incomplete, then user dials 5678 then asterisk begins to treat extension 12345678 as if it had been dialed as a whole. With compliant hardphones, you could get you phone to display a short text invite

Hi, I would like to know if someone can suggest me an efficient way of writing a dialplan to match "variable-length" international calls when using SIP clients with the "early dial" or 484 feature. What I usually do for clients that do NOT "early dial" is define something like this in my outbound context: For local calls (they fortunately have a fixed length):

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

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

Hi, I was wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64? I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection. Any ideas? Vahan

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card.

Hi, I have the Cisco PIX 506 firewall right in front of the asterisk and I am getting a one-way audio. I need your help/guidance to resolve this problem. I have the "fixups" disabled for SIP in the Cisco PIX 506. Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the

Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of

http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

Hello, InstAlias does not allow tied operands (repeated operands) in the asm string to be matched. It seems this situation is explicitly prevented in AsmMatcherEmitter.cpp: if (!Hack) PrintFatalError(TheDef->getLoc(), "ERROR: matchable with tied operand '" + Tok + "' can never be matched!");

On Fri, Aug 24, 2012 at 4:07 PM, Villmow, Micah <Micah.Villmow at amd.com> wrote: > > >> -----Original Message----- >> From: Villmow, Micah >> Sent: Friday, August 24, 2012 2:56 PM >> To: 'Eli Friedman' >> Cc: LLVM Developers Mailing List >> Subject: RE: [LLVMdev] RFC: Supporting different sized address space >> arithmetic >>

hello list, i have asterisk 11.15.0 and i have some trunks sip from my provider we have some ip phone astra 6731i each Ip-phone is configured with trunk and we call no ihave configured another trunk from the same provider in my asterisk i can call all numbers just the numbers are configured in thses ip phones. but when i configured the same trunk in x-lite i can call theses ip-phones without

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

I had MCGP working to a ADIT 600 fine with debain sarge stable / asterisk stable - wanted to try red hat and got the below message - then I re-installed debian and am still getting the same message below - any comments are greatly appreciated - I did play with the config files with no prevail - the Adit seems to be doing its job per tech support at CAC. I listed my conigs below I go off hook

Hi, On Instructions you can use checkEarlyTargetMatchPredicate() to check that the operands are the same. There's an example of that in MipsAsmParser.cpp for DATI and DAHI. I can't think of a reason TableGen couldn't be made to allow this for InstAlias too. > On 15 Dec 2017, at 02:12, via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > Hello, > > InstAlias