search for: e65e

Hey folks, I am trying to get an E65 to connect to asterisk, and I would really appreciate a second set of eyes. The SIP dialog completes fine, but the phone subsequently says "Registration failed". I am in a network that has what seems to be a SIP-capable NAT gateway, but the asterisk is configured nat=yes anyway. Using a softphone (twinkle), I can connect just fine, SIP and RTP work.

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

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

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 --

Quoting "Fabio Pietrosanti (naif)" <lists at infosecurity.ch>: > Speex it's too cpu expensive for general S60 usage, it would require a > lot of ASM optimization. Did a quick search and saw ARM CPUs with speeds above 100 MHz. That should actually be enough for Speex, at least for narrowband. > If you are using CSD, like for a secure telephony solution >

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 ->

--- 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) {

Nik Engel wrote: > Hi list ! > > I implemented *8 to pickup any call on my asterisk system. But after the > pickup callerid is missing, so there is no way to see from where the > call originated. How can this callerid be passed on. > > Nik > Hi Nik, I'm after the same question as I would like to keep callerID data after pickuping up the call. Maybe using a

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,

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 All, Recently I added some Nokia N95 customers and it worked pretty good. Now the customers are complaining about the dialing rules... They are used to dialing +12486543210 and +4479XXXXXX for long distance calls. Is there anyway to create a "+" sign dial plan which will allow them to dial a number with "+" sign. Cheers, Nitesh

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

Adds a new menu item which launches a separate script. That separate script presents a list of log files to the user. Signed-off-by: Darryl L. Pierce <dpierce at redhat.com> --- Makefile.am | 1 + ovirt-node.spec.in | 3 +++ scripts/ovirt-config-view-logs | 37 +++++++++++++++++++++++++++++++++++++ 3 files changed, 41 insertions(+), 0 deletions(-)

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

Anyone using Nokia "E" Series handsets with Asterisk? I'm trying to deploy some e71's and am having an issue. I can get a single handset working, but when I try to create a SIP profile on the second phone, it won't allow me to save the profile, saying that devices in the same "realm" must have identical username and password. Anyone have a workaround for this

G'day. I hope this isn't off-topic for the list. I am looking at an Asterisk setup that includes cordless phones. The three choices I can see, at this stage, are: * wifi phones * an ATA and a cordless analog phone * a DECT to SIP basestation The various wifi phone options don't grab us as suitable -- they are costly, have poor battery life and even the best have pretty mixed

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,

...gt; > > [root at localhost network-scripts]# > -- |=- Red Hat, Engineering, Emerging Technologies, Boston -=| |=- Email: pmyers at redhat.com -=| |=- Office: +1 412 474 3552 Mobile: +1 703 362 9622 -=| |=- GnuPG: E65E4F3D 88F9 F1C9 C2F3 1303 01FE 817C C5D2 8B91 E65E 4F3D -=| -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://listman.redhat.com/archives/ovirt-devel/attachments/20081226/fef06f49/attachment-0001.htm>