similar to: E65 fails registration, soft phone works

Hi all, I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so I can use the GW. For this I use: exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling office) ;callerID is the one of the calling mobile phone exten => s,n,Background(silence/1) ; Nokia E65 send digits in

It would be very nice to know about some good success without too much cpu issues on Nokia devices, i tried pjsip.org project on S60 FP1 E65 +200mhz ARM cpu with speex but don't had much luck in using it without having 100% cpu usage :( Please let us to know about your testing E65 CPU: http://www.nokia-tuning.net/index.php?s=processor Fabio Pietrosanti Jordan Dimov wrote: > Thank

Hi i want to Buy Nokia E series Phone which have InBulit SIP-VOIP Calling client so i can make VOIP calls thru that phone. Aslo that Phone easly able to register with Asterisk Pbx to recive inter-office calls. i try to search from web & also from Nokia site but they only mention this features as "VOIP call from wifi" they mentioed only this much info. they not mentioed info about

--- openssh-2.3.0p1/sshd.c Sat Oct 14 01:23:13 2000 +++ openssh-2.3.0p1.new/sshd.c Tue Dec 19 11:26:51 2000 @@ -506,7 +506,7 @@ initialize_server_options(&options); /* Parse command-line arguments. */ - while ((opt = getopt(ac, av, "f:p:b:k:h:g:V:u:diqQ46")) != EOF) { + while ((opt = getopt(ac, av, "f:p:b:k:h:H:g:V:u:diqQ46")) != EOF) {

Hi, I've two wifi-phones 1. Nokia e65 2. HP Ipaq I've configure two sip exten in my asterisk and using these exten in my phones. But my Nokia phone is keep on loosing the connectivity very soon life 1-2 min the qualify packet will be double of my HP. So, when I try to call my Nokia SIP exten it takes very long, but HP works fine. I tested one more phone also that works fine. so,

On Thu, Jan 14, 2016 at 02:25:50PM +0100, Jerome Marchand wrote: > Hi Michael, > > KASan detected a use-after-free error in virtio-pci remove code. In > virtio_pci_remove(), vp_dev is still used after being freed in > unregister_virtio_device() (in virtio_pci_release_dev() more > precisely). I don't know the proper way to fix this. Thanks a lot for the report, I posted a

Someone can help me? I tried several things and always don't converge # Model library(sem) dados40.cov <- cov(dados40,method="spearman") model.dados40 <- specify.model() F1 -> Item11, lam11, NA F1 -> Item31, lam31, NA F1 -> Item36, lam36, NA F1 -> Item54, lam54, NA F1 -> Item63, lam63, NA F1 -> Item65, lam55, NA F1 -> Item67, lam67, NA F1 ->

I'm trying to build a SEM using the sem package. I'll attach my model specification below. When I turn debug=TRUE, it seems as if I'm getting to convergence because I get this message: Successive iterates within tolerance. Current iterate is probably solution. However, at the end of the process I get this message: Warning message: In sem.default(ram = ram, S = S, N = N,

Hi Michael, KASan detected a use-after-free error in virtio-pci remove code. In virtio_pci_remove(), vp_dev is still used after being freed in unregister_virtio_device() (in virtio_pci_release_dev() more precisely). I don't know the proper way to fix this. Here is the KASan output: [ 467.987227] ================================================================== [ 467.990023] BUG: KASAN:

Hi Michael, KASan detected a use-after-free error in virtio-pci remove code. In virtio_pci_remove(), vp_dev is still used after being freed in unregister_virtio_device() (in virtio_pci_release_dev() more precisely). I don't know the proper way to fix this. Here is the KASan output: [ 467.987227] ================================================================== [ 467.990023] BUG: KASAN:

Hello, is anyone on the list using a normal cell/mobile phone which is able to act as a SIP client over WLAN? Or has anyone heard of a SIP client for cell/mobile phones running windows mobile 6.x? The phone should use SIP, when the asterisk server is reachable and should automatically switch to a German telco if it is not reachable. Thanks for any hints, Stefan --

Hello, I am trying to test-call my own asterisk server to see if I can receive SIP calls properly. I use a softphone to call the SIP address, and because twinkle doesn't support SRV records, I go via a proxy. When the call comes in, asterisk says: handle_request_invite: Sending fake auth rejection for user "martin f. krafft" <sip:xxxxxxxxxxxxxxxx at

At the risk of self promotion, I thought I'd share this wise thought: 21:16 * madduck loves dovecot because a dove is a bird and the german word for animal shit is kot, so there... 21:16 < madduck> your mail is served by bird shit! -- martin; (greetings from the heart of the sun.) \____ echo mailto: !#^."<*>"|tr "<*> mailto:"

From the digest: On Aug 14, 2007, at 1:04 PM, dovecot-request at dovecot.org wrote: other messages cut out? > > > Message: 9 > Date: Tue, 14 Aug 2007 19:03:58 +0200 > From: martin f krafft <madduck at madduck.net> > Subject: Re: [Dovecot] use of deliver from procmail advisable? > To: dovecot at dovecot.org > Message-ID: <20070814170358.GA17390 at

Hi, we are using dovecot's deliver to deliver mails to a virtual mailbox tree owned by the vmail user, by piping the message to the following command spawned by vmail: /usr/bin/env HOME=/srv/vmail/mydomain.ch/myaccount /usr/lib/dovecot/deliver When a message is delivered to an account that has reached its quotum, deliver issues a failure message saying: From: Mail Delivery Subsystem

Dear list, I am experimenting with a new mail handling setup and it involves a single IMAP folder with just under 70'000 messages. When OfflineIMAP connects to the server, the imap process starts to eat up a lot of memory: PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 15607 madduck 35 19 283m 244m 239m D 16.9 49.3 0:09.96 imap On the contrary, when

https://bugzilla.mindrot.org/show_bug.cgi?id=1256 martin f. krafft <bugzilla.mindrot.org at pobox.madduck.net> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |bugzilla.mindrot.org at pobox. | |madduck.net

Hi, I am on amd64 Linux and not really too happy with twinkle, linphone and ekiga. Unfortunately, X-Lite and Zoiper, even though they provide Linux versions (w00t!) have only x86 versions for download. Do you guys know of amd64 versions of those, or can you recommend other softphones that will run on amd64, or which come with source code? Thanks, -- martin | http://madduck.net/ |

Hey folks, I can park a call with #70 after enabling that feature in features.conf. However, once I retrieve the call from the parking lot, #70 cannot be used to park it again. Worse yet, none of the keys defined in the featuremap work anymore, include blindxfer or automon. Any ideas what may be the problem? -- martin | http://madduck.net/ | http://two.sentenc.es/ "man sagt nicht

also sprach Steve Edwards <asterisk.org at sedwards.com> [2015-05-17 08:31 +0200]: > While preprocessing could be called 'templating,' this may be > confusing because Asterisk already as a configuration file feature > called 'templates.' Fair point. Preprocessing it shall be. > And you find preprocessing/templating complex? Hehe. The difference is that e.g.