similar to: siemens optipoint 400 SIP

The file "rtp.c" currently contains the following hack for special-case handling of some Cisco-specific protocol: } else if (payloadtype == 121) { /* CISCO proprietary DTMF bridge */ f = process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); As I noted in my earlier message, I'm

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

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

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

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

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

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

I have not seen anybody on the web to have found this so I thought I would check here. Anybody got this firmware? I've found firmware for the 400, but it doesn't seem to load in the 410.

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

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

hi .. when placing a SIP call to a sip host in the states every few seconds I get an RTP codec 19 error. I know this is related to comfort noise, and the call goes through OK ... how can I suppress the error message ? Also, many times I get "Invalid CSeq Number" back from 216.52.153.207 (which is the host i'm calling) and the call drops.. is there a solution for this ? cheers Dave

Hello, I work at SmarctJog.com, we have here some patches on IceCast for performance and reliability, these are mostly client/connection/source cleanups (a slave merge is underway, and some more good stuff (c)), but we'd like this to be merged in before the list gets any longer. Please find attached a list of our patches with a short desc: This one is actually not from us/me, it was found

Dear all, Anyone get the lastest spandsp with udptl.c and tpkt.c compile in Fedora 3? tpkt.c: In function `accept_thread': tpkt.c:140: error: `TCP_NODELAY' undeclared (first use in this function) tpkt.c:140: error: (Each undeclared identifier is reported only once tpkt.c:140: error: for each function it appears in.) tpkt.c:144: error: invalid application of `sizeof' to

The code to detect fragments in checksum_setup() was missing for IPv4 and too eager for IPv6. (It transpires that Windows seems to send IPv6 packets with a fragment header even if they are not a fragment - i.e. offset is zero, and M bit is not set). Signed-off-by: Paul Durrant <paul.durrant@citrix.com> Cc: Wei Liu <wei.liu2@citrix.com> Cc: Ian Campbell <ian.campbell@citrix.com>

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

Hi, This patch fixes a couple of segfaults in music-on-hold, frame smoother routines and channel allocation in Asterisk. Mark, feel free to apply it in CVS (if approved). Regards, Michael. -------------- next part -------------- Index: channel.c =================================================================== RCS file: /usr/cvsroot/asterisk/channel.c,v retrieving revision 1.25 diff -u

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

Hi... Here''s a working Perl script that I want to be able to do in Ruby: use WWW::Mechanize; my $url = "http://host/tt?name=chris"; my $mech = WWW::Mechanize->new(); $mech->add_header( Referer => "http://chrismcmahonsblog.blogspot.com" ); $mech->add_header( Cookie => "messageid=170118; memberid=1007"); $mech->get($url) so the header