similar to: Display on OptiPoint400std SIP

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

Hi there, I tried to get a few "Optipoint 400 SIP" working with *, but it refused to work properly. In my testing-network i have two Sjphones (they are working really fine) and three optipoints. I?m able to dial the number of a Sjphone on all of the optipoints. The call is signalled at the Sjphone with the right number of the optipoint and an connection can be established. But when I

Hi everybody, I am testing Optipoint 400 Standard SIP (Firmware 2.3.14) with Asterisk. It is posible to dial from another Phone (x-lite) to the Optipoint, but when I try to dial from the Optipoint there is no dialtone and there is only a short beep when I dial Numbers. The Optipoint shows "no Server..." (Registrar?) in Display. Sip debug shows no unusual (to me) Messages. Sip show

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

Hello, I?m looking for Optipoint 420 Standard SIP Firmware to make my first tests with Asterisk and SIP, but I?m unable to find it. Could someone help me? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070415/c2da9cc0/attachment.htm

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Anybody using Siemens optiPoint 300 H.323 phones? I saw a few references to them in the archives of this list, and the Wiki seems to be down. I have a chance to pick up a bunch of these, cheap. Questions: * Asterisk support? * What sort of power supplies will they need? The bunch I am looking at are surplus and have no supplies. Thanks, </edg> Ed Greenberg San Jose, CA

Not strictly a problem with Asterisk but one of my phones. A couple of days ago I decided to update the firmware in my Optipoint 600 Office which looked as though it went swimmingly until that is, it rebooted. Since then the phone just boots up and displays the following: Can't Boot!! Development shell active. It doesn't try to request a DHCP address, in fact it does seem to do

Hi all, I have an optipoint 400 Sip phone with 2.46 sip image which stores its music on hold file localy (on the phone) rather than indicate to the server the call is on hold. i'd like to have Moh play from the server, and i'm not sure if there is a work around to achieve this anyone else has the same phone and came across similar problems ?? thanks fam

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Dear, Is there somebody who have experience with the Siemens Optipoint 400 standard SIP ? And where can we buy it (i'm from belgium) We are using for the moment Cisco 7960, 7905, snom 200, Mitel 5055 and in my opinion the Mitel does his work the best combined with the 7905, the 7960 is realy anoying to transfer, On the snom 200 you can't dial a number veryfast due to the

Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi! I'm working an a asterisk test project at college at the moment. right now we're experiencing two problems. calling our sipphone (optipoint400) from a firefly client leaves us with no audio (no noise...nothing at all...) [the phone is ringing however and the connection seems to be set up] other way round works just fine!! firefly2firefly (stun enabled) also works

I have a question which I'm sure has been asked before but my research has yet to find it. I have Asterisk running on a Linux server but in order to get it to connect I needed to punch a hole in my firewall on port 5060 for it to receive the registration responses from broadvoice. If I run sjphone as a softclient on my home PC I do not need to punch that same hole and it works just fine.

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

For all who are interested: A quick review of CeBIT 2005. :-) CeBIT was a very successfull event. Most of the time, the asterisk-booth was crowded with more people than we could talk to. We had with us a demo-installation including different IP-phones, digital and analog phones as well as a Siemens HiPATH PBX to which our Asterisk-server served as a VoIP-gateway, and many people were impressed

hello: i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729. I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2