similar to: DTMF received twice

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

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

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar column, like this: callerid_internal=test <710>;did=5555551234 Again, this works

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 --

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 am working with a customer and their SIP provider is IPitimi. The customer needs to sometimes provide various caller id number for the calls going to IPitimi. They are processing calls for multiple businesses who want their caller id to show up. When no caller id is provided, the From must be the DID at ipitimi ip address and caller id is DID at customer IP address. When caller id is

Hey all. My boss asked me to implement the following When DID 713xxxxxxx is dialed send an email to mmosier at xxx.com. with the time date and CID included in the email. I know how to code some but am looking for the best way to do this. Sorry I might have asked this a couple months back. I forgot. Mmosier Houston Respectfully Michael D Mosier Ftoc Certified -------------- next part

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

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 >

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:

OK, here's another problem I've run into with Asterisk. In the musiconhold.conf file, I can set the music on hold mode to files and the directory to the place where I have my MP3s stored and they play. If I set the music on hold mode to any other setting, instead of getting my MP3s, I get something that sounds like a motorcycle being cranked up and driving off. It sounds really weird,

Hello. I'd like to know if the two following functionalities are available in Asterisk. -1- A stop/wait/halt functionality in the Dialplan. Like: exten => myexten, n, Halt where execution of the dialplan would wait indefinitely. I guess a Wait would be OK, but I'd like this wait to wait indefinitely. -2- A Goto functionality from the AMI: You give the channel, and you can ask

--- 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,

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

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

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;