similar to: Confusion about construction of RURIs from contact headers for BYEs generated by *

Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Hi, I'm trying to port our configuration form sip to pjsip channel and have following issue. Sip.conf has a host parameter that sets the RURI to a given value. This functionality is needed in some of our scenarios where we need to send requests to specific IP address with specific domain in RURI. I did not found an equivalent to the host parameter in pjsip configuration. Did I

Exchange 12 will support "OVA" Outlook Voice Access. It will do this using VOIP. I was wondering if anyone has given any thought as to how Asterisk might interface with Exchange 12. It will do this using a VOIP gateway. If this could happen it seems to me this could be a really big win for Asterisk. I am going to setup a test server as soon as I can and start digging into this.

It will be interesting to see how many standards get broken, and how many proprietary hooks get thrown into the pot. The bean counters smell some money, and their OS franchise is waning: http://www.nytimes.com/2006/06/26/technology/26soft.html -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

Does anyone have any ideas why my recently updated * 1.2.5 system should spawn multiple * process at seemingly random intervals? Regards L:ee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ -------------- next part -------------- An HTML attachment was scrubbed...

On 1/2/20 1:00 AM, L.P.H. van Belle via samba wrote > Yes, you can use : > > install -d /var/lib/samba/bind-dns/ -o root -g bind -m 770 > And rerun it. > > Just dont forget to adjust the bind config files also. So after moving everything, fixing the bind config and restarting there is still an issue. While the forward zone myhome.nurdog.com is working fine,? the reverse zone

Hai, > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens Paul > R. Ganci via samba > Verzonden: woensdag 1 januari 2020 23:49 > Aan: samba at lists.samba.org > Onderwerp: Re: [Samba] Cannot remove old NS record > > On 12/22/19 12:07 PM, Rowland penny via samba wrote: > >> > >> Are there any suggestions to

Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0 Record-Route: <sip:192.168.129.38:7160;lr=on> ... for instance when I use such extension:

Hello, Is there any way to identify an incoming session based on the Diversion header? In my scenario, I have some unregistered endpoints (mobile phones) that make calls through our Asterisk, which controls the external call rights based on the endpoint's context. In a normal call, their number will be in the From header and the destination in the To an RURI, but when they make a call

Minor update - use the following: > if (strcasecmp(data, > "x-Asterisk-Request-URI-pseudo-header")==0) > { > ast_copy_string(buf, p->initreq.rlPart2, len); > -----Original Message----- > From: Steve Langstaff > Sent: 23 October 2006 09:58 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [asterisk-users]

Hello All. I am having some trouble with call transfers when asterisk is the 2nd party called and I hope to benefit from your experience. I want to use asterisk for call park/pickup and have configured openser to relay calls made to ruri 700-720 to asterisk running on localhost:5069 Call flow: phone A calls phone B (both phones are polycom) Phone B answers then phone b

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

We seem to be having a good bit of issues sending faxes and can't pinpoint the issue. I'm hoping someone here may have a different outlook on the issue that leads to a resolution. Our call path is Sip trunk from MAX TNT - Asterisk - T1 - Adtran endpoint converting sip trunk to copper line for house wiring. Users at the endpoint can receive faxes without a problem. However, sending

I have two datasets, the first has this shape (each word is a column) Name address phone .. .. The second one has the following shape Name request I need a contingency table with for example phone and request. The people registered in these datasets are present in both datasets, BUT in the first every record is a person, so every person is counted once and is 1 row, in the second every row is

On 10/09/2016 12:14 PM, Rowland Penny via samba wrote: > On Sun, 9 Oct 2016 11:50:42 -0600 > "Paul R. Ganci via samba"<samba at lists.samba.org> wrote: > >> >On 10/09/2016 02:51 AM, Rowland Penny via samba wrote: >>> > >Have you by any chance got another 3001108 'xidNumber' in >>> > >idmap.ldb ? If you give a user a

I am, from a front end, calling functions like LLVMModuleCreateWithName, found in Core.h, ultimately calling LLVMWriteBitcodeToFile, found in BitWriter.h. Do I correctly presume, from the existence of LLVMDisposeModule, that I am responsible for calling it when I'm done? Will I need to do deeper disposing myself? I presume at least I will need to free strings I allocated myself, such as the

Before metadata was separated from values, I could create a debug info forward declaration and eventually resolve it using LLVMReplaceAllUsesWith in core.h. Now, I can't figure out how to resolve it. I can find no function that seems to do this. My one wild guess that giving the forward decl and the resolving decl the same UniqueId might do it is not working. I am currently using 3.6.1, but

I am generating llvm code for nested procedures. Each one gets a static link, which I am passing as an internally added, first parameter. However, it needs to be stored in memory too, so a debugger can find it. So, when I need to use the value, is it better to use the ssl parameter or load the memory copy? I can imagine the ssl value might avoid unnecessary fetches, but then, it might also