search for: qops

Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does Asterisk support

...26a81 100644 --- a/src/gallium/drivers/nouveau/codegen/nv50_ir_lowering_gm107.cpp +++ b/src/gallium/drivers/nouveau/codegen/nv50_ir_lowering_gm107.cpp @@ -95,18 +95,15 @@ GM107LegalizeSSA::visit(Instruction *i) bool GM107LoweringPass::handleManualTXD(TexInstruction *i) { - static const uint8_t qOps[4][2] = - { - { QUADOP(MOV2, ADD, MOV2, ADD), QUADOP(MOV2, MOV2, ADD, ADD) }, // l0 - { QUADOP(SUBR, MOV2, SUBR, MOV2), QUADOP(MOV2, MOV2, ADD, ADD) }, // l1 - { QUADOP(MOV2, ADD, MOV2, ADD), QUADOP(SUBR, SUBR, MOV2, MOV2) }, // l2 - { QUADOP(SUBR, MOV2, SUBR, MOV2), QUA...

.../drivers/nouveau/codegen/nv50_ir_lowering_gm107.cpp > +++ b/src/gallium/drivers/nouveau/codegen/nv50_ir_lowering_gm107.cpp > @@ -95,18 +95,15 @@ GM107LegalizeSSA::visit(Instruction *i) > bool > GM107LoweringPass::handleManualTXD(TexInstruction *i) > { > - static const uint8_t qOps[4][2] = > - { > - { QUADOP(MOV2, ADD, MOV2, ADD), QUADOP(MOV2, MOV2, ADD, ADD) }, // l0 > - { QUADOP(SUBR, MOV2, SUBR, MOV2), QUADOP(MOV2, MOV2, ADD, ADD) }, // l1 > - { QUADOP(MOV2, ADD, MOV2, ADD), QUADOP(SUBR, SUBR, MOV2, MOV2) }, // l2 > - { QUADOP(SUB...

Hello All, I've been trying for some time to get Asterisk to register with a remote SIP gateway. I?ve recently managed to configure an SJ Phone to work with W2000 so know the configuration parameters work correctly. Asterisk doesn't authenticate properly and I notice that the authentication request appears different to SJPhone's. Do any tools exist to enable me to check these

Again rc, just in case I broke something. Or maybe I should just start creating nightly CVS snapshots. Anyway, fixes the few bugs people have reported lately: - Mails with nested MIME parts might have caused incorrect BODY and BODYSTRUCTURE fetches and sometimes might have crashed dovecot (assert at imap-bodystructure.c). If client had already successfully done the BODY fetching a

Since I so miserably misspelled packaging - a new thread specific to the issue at hand. I found a "workaround". In short, xlc does not accept arrays of nested array - the bottom line (best message) is: 1506-995 (S) An aggregate containing a flexible array member cannot be used as a member of a structure or as an array element. At the core - all the other messages come because this

Hi, my dovecot installation works since months and clients authenticate using CRAM-MD5. But today I got the first chance to test an client that supports DIGEST-MD5 - and it doesn't work. Because of lack of other supporting clients and servers I'm now at the point I don't know which side is to blame. The error I get after the client answers the servers challenge is "-ERR

Hi! I'm using rc9 and DIGEST-MD5 passwords. Sometimes it works, sometimes it's not... When failed, I get this in the logs: 12:45:11 Info: auth(default): client in: AUTH 1 DIGEST-MD5 \ service=POP3 secured lip=<lip> rip=<rip> resp=<something_long> 12:45:11 Info: auth(default): client out: CONT 1 <something_long> 12:45:11 Info: auth(default): client in: CONT 1

Hi, I need to create something like this: <?xml version="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml`

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. The registration works though, right? You might want to compare

http:/www.theregister.co.uk/2017/09/01/asterisk_admin_patch/ -- Dave Topping e: info at dntopping.uk t: 03445 888 888 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170901/ae060564/attachment.html>

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

George, I have the detailed log below. (Resending after trimming the email to 40KB.) The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? Thanks! --------------------- Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 Via: SIP/2.0/UDP 18.18.19.123:5060

Surely there must be someone here who can help me with this problem. I have spent weeks trying to get this damned service to work with no luck. I have incoming calls working, but no outgoing. If get outgoing working then incoming don't work. I have sent this problem to this list a couple of times with little or no response, and I _really_ need some help to sort it out. I have an asterisk

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

Several bugfixes since 0.99.6rc3. Here's again the change summary since 0.99.5: v0.99.6 2003-01-13 Timo Sirainen <tss at iki.fi> + THREAD=REFERENCES extension support. ORDEREDSUBJECT would be easy to add, but I think it's pretty useless. + SORT is much faster now. + mbox: If ~/mail directory isn't found, create it. + Log login usernames * Some coding style changes

Calling all SIP gurus-- I'm trying to register my asterisk to an ISP's SIP gateway. I'm getting authentification errors. Here's the results of SIP DEBUG against it's IP. [I've tweaked all confidential fields so as to protect the innocent (namely, me).] --- (9 headers 0 lines)--- Responding to challenge, registration to domain/host name myfavoriteisp 12 headers, 0

Well, "release candidate" wasn't really right name for last two versions. Maybe not for this one either, it has a few changes that could potentially break things badly (new hash table code especially). Anyway, seems to work with me. If I don't see any problems with it for a few days it'll be the final. The really great news is that we support threading. A lot of people have

Hi list, We have a network with some XP and some Windows 7 computer, we use samba 3.6.6 on debian 6.0.6 from debian-backports. It's a pdc with passdb backend = ldapsam. In our logs there are lots of: ARCServer slapd[1263]: SASL [conn=46778] Failure: realm changed: authentication aborted I found out that at that time this emerges the tcpdump says: 12:59:54.656399 IP client.49551 >