similar to: Missing ring indications with DPH-100H/chan_h323

hi all I'm trying to setup a dlink dph-100h phone (actually a dph-100m but with the h.323 software) with asterisk and chan_h323. AFACS, the dph-100h software can only be configured to point to a gatekeeper. I know I don't need to do this, but it's a test before I setup my Symbol Netvision phones (I don't have an access point for them now). First, this is what I have, and what

hi IIRC, Jeremy once said that chan_h323 could be used as a gatekeeper but perhaps lacking a few features as compared to gnugk. Is this possible? I have some dlink DPH-100H phoes here for testing, but they require a gatekeeper, and if I can do it, I'd love to keep gnugk out of this. thanks roy

hi seems like things are closing in to something that might look like success. I have one problem left: I don't get ring indicator when I dial out from the h.323 phone... Sound is good, so it doesn't look like a codec problem. I'm using chan_capi with early B3. I also use gnugp to route the calls from the phones to asterisk, as the dlink dph-100h requires this. Debug output follows:

After 20 cups of strong coffee and wasteing most of tonight and obviously doing lots of googling and emailing many people, i've concluded that dlink voip phones specifically DPH-80 dosent work with asterisk. If anyone has had anymore luck with these phones then me please let me know -- regards Vikram

I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The

List, The D'Link phones are not reliable at this time. I am trying to get them fix their Firmware to my specifications. It is half done so far. However there are still hurdles. below email is self explanatory. At present if you want to use these phones, you need to buy D'Link's SIP Server and run this as one of your SIP servers in the blend to call to Asterisk. Seshu Kanuri "G

hi just after talking to Jeremy, I downloaded the source he's using and compiled. core dump! This was without gatekeeper: (gdb) bt #0 0x40141118 in mallopt () from /lib/libc.so.6 #1 0x401400fc in malloc () from /lib/libc.so.6 #2 0x405e981b in PAbstractArray::SetSize(int) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x405ed006 in PString::SetSize(int) () from

Hello Situation: I'm moving from a windows XP box to fedora 11. I used winXP to run skype with a dlink dph-50u USB adaptor so that I could talk on skype with my regular phone. Been working fine for months I'm now on fedora 11, able to run skype linux without problem, even the webcam work! Problem, cannot make the dlink dph-50u to work. After a lot of googling, cannot find anything. I

Hi, list! I'm playing on an Asterisk@home installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. >From this phones, I can make & receive calls with no trouble, but, when I try to use some "interactive" function (eg Directory or Voicemail), the phone seems

In my test box I've installed chan_h323 and I've been testing it with Micro$oft netmeeting and openphone with success. I alos have in my installation a Cisco 1700 series router with an FXS card on it. On the router I places the g711-ulaw codec and it worked but I experienced one bad thing. When I made up more than three calls, in the first three calls I was able to transmit and

hi all when I (using my D-Link DPH-100M MGCP phone) press #, I get told 'transfer'. I dial the new number and after that, * just tells me 'meep meep meep' [hangup] any ideas why? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

a) jerjers been doing a lot commendable work for * b) support is not mandatory, and i agree with royk it should not be withheld based on political viewpoints, that's pointlessly draconian c) choice is always good, so people should have the option of oh323 or h323, let them decide, and not limit them, unless astmaster chooses to limit them, and that too based on valid points d) jerjer gave a

Hi, I'm playing around with Asterisk and DPH-100M (Dlink mgcp phone) on my debian box. I've got stable version of Asterisk (packaged for debian) working with dlink phone and 7910 from cisco (minimalistic extensions.conf and chan_skinny for 7910) Everything works fine. Now I'm trying to get CVS HEAD working with MGCP. I want to test chan_sccp

We have an odd problem, where inbound H323 (chan_h323) calls will sometimes cause a Ringing Congestion that appears to keep the channels open and never release it until we kill and restart asterisk. These "Ringing Congestions" start to pile up, which eventually crashes Asterisk. H323 Gateway -> Asterisk (chan_h323) -> Tor2/PRI -> PSTN Has anyone ran into this problem or

Hello gentlemen, I am new here. I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying to make it work with Asterisk. I tried versions 1.0.7 and yesterday's CVS and the behavior is the same. The phone registers with no problem, and can accept calls. But when I try to make outgoing call, there is a series of invite requests from the phone, to which asterisk responds

hi I'm running SIP (msn messenger), MGCP (dlink hardphone), H.323 (soon - netvision 802.11 phone) and CAPI (AVM C2). Can I plot in the known extensions and use gastman to tranfer calls? I've tried, but no calls was visualized from the extensions I created ... and - no - I couldn't find it after RTFM roy -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323.c:790: error: dereferencing pointer to incomplete type chan_h323.c:791: error:

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

Hello all, I got the following error compiling h323 support in the latest cvs. Below the error is a diff to the file that I got to make it work. I took an example out of sip as far as the syntax for ast_rtp_new. Not sure if it is correct or not, but it seems to work. Please correct me if I am wrong in the additional 2 arguements. Regards, Scott cc -g -pg -c -o chan_h323.o -march=i686

Hi All, I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 10. When I compile asterisk, the build fails at chan_h323 with: ---------------------------------------------------------------------- chan_h323.c: In function `reload_config': chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this function) chan_h323.c:2863: error: (Each undeclared