similar to: NOKIA E series Phone for SIP-VOIP calling

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

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 --

Hi Friends, Recently, I bought NOKIA E70 mobile. I have configured my mobile to connect with my Asterisk server depends on the information available in Internet. But, its telling that "Registration failed". If anybody configured this mobile for Asterisk server, please tell me the step by step configuration or please tell me a good website link to do this. Looking forward to your

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 >

I had asked a few months ago if anyone knew if mondo is compatible with CentOS4. One person said it was still working, but never mentioed if he had done a restore. I had the stable version installed and whenever I ran it there were a few messages about command not found. Well, I had a HD go out and tried booting from the CD and it failed. It was just my desktop PC and my home dir was backed up a

Hi, While creating identification header in the function * theora_encode_heade*r in *encoder_toplevel.c*, it assigns bits not mentioned in the current theora spec released on Octomber 29, 2007 (page 40 &41). But this implementation in function* theora_encode_heade*r is correct according to the *Figure 6.2 (page 42)*. But not according to the *table mentioned in pages 40 &

I've made a start writing a Theora player for Series 60 phones (mostly Nokia phones, but some Samsung and Sony Ericsson ones too). Download here: http://symbianoggplay.sourceforge.net/OgvPlay/OgvPlay_010.zip I've been using Big Buck Bunny, Elephants Dream and a few other ogv files for testing. I've uploaded 320x180 versions of those here:

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 have the following symptoms: Call X-lite / Nokia E51 X-lite press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH Call X-lite / SCCP phone MOH works as supposed Call SCCP phone / Nokia E51 SCCP press hold: Nokia DOES hear MOH Nokia press hold: X-lite does NOT hear MOH In addition, the BLF on the SCCP phones does NOT show the hinted SIP extension on hold. With 1.4

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

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'm trying to make ooh323 works with one asterisk box running 1.2.15 version. I can ring from a h.323 to SIP and SIP to H.323, but when the call is finished when the phone is answered. This is the log when I call from the H.323 device to a SIP device: Feb 23 10:57:32 VERBOSE[6096] logger.c: -- Executing Dial("OOH323/Telconet Mantaer-c5f8", "SIP/666|30|TtrwWC")

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

I'm getting this messages trying to call with one sip trunk: Feb 3 16:43:09 DEBUG[3389] channel.c: Avoiding initial deadlock for 'SIP/usa-e2ea' Feb 3 16:43:09 VERBOSE[3491] logger.c: -- SIP/usa-e2ea answered Zap/1-1 Feb 3 16:43:09 WARNING[3491] channel.c: No path to translate from Zap/1-1(68) to SIP/usa-e2ea(256) Feb 3 16:43:09 WARNING[3491] app_dial.c: Had to drop call

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I got one quick question on A2Billing. Specs: - - A2Billing v1.3 - OS CentOS 4.5 - Asterisk 1.2 - Zaptel 1.2 Did the installation and everything is working as it suppose to... Using the A2Billing documentation, I created the RateCard, SIP Trunks, and SIP Customers. I was also able to login using XLite Dialer and was able to call out to my SIP Trunk also. Now how can I remove the

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

Dear Sirs, we ... 1) buy minutes from other providers 2) sell minutes to out clients some calls terminate to our equipment, others - to h323 proxies. we want calls to be routed according to costs (a route is chosen from many by lowest cost). at the end of it, we'd like to bill our clients and see how much have we earned (money we receive from client on one side, money we pay to proxies on

Hi, Already posted once but I need more feedback. What kind of servers is everyone using for asterisk and what problems have you ran in to ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050630/dd52bf35/attachment.htm