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From: Joaquin C. A. <e98...@fr...> - 2004-06-27 11:23:37
|
Hi! I've tried to compile iaxclient (HEAD) on windows with MSVC6 but unsuccessfully until now. My first problem come from the current project, as it tries to build a version with libiax 1, and this version doesn't compiles. So I've changed the libiax files for the libiax2 ones, but I'm still experiencing some problems. In the libiax2\src\iax.c file, line 670, I'm getting a: error C2229: struct '__unnamed' has an illegal zero-sized array It seems to complain due to the last member of iax_frame. Does anybody knows what should I do to make it work? Thank you for any comments! Cheers, |
From: Steven S. <ss...@so...> - 2004-06-25 22:55:52
|
Folks, I have a user who has alerted me to the fact that on occasion IAX Phone ignores incoming calls. I have turned on the tracing and it appears that we have situations where messages arrive out-of-order again. The last time we ran into this, it was a result of confusion over how to handle the registration. We were leaving it open, and that seemed to cause the phone to stop responding eventually. Well, this seems eerily similar. I have also noticed the Qualify option does not appear to work with iaxClient-based clients, but does with others (Asterisk and FireFly). If you try to qualify an iaxClient-based device, it fails to ring at all - it shows as unavailable or something when you do 'iax2 show peers' Are any of these conditions related? Anybody (Adam?) have any thoughts on this? I'll code it if you can come up with the answer. Here's the trace: libiax2/src/iax.c line 2020: Received iseqno 0 not within window 5->5 libiax2/src/iax.c line 2039: Packet arrived out of order (expecting 2, got 0) (frametype = 6, subclass = 1) libiax2/src/iax.c line 2045: Acking anyway libiax2/src/iax.c line 2020: Received iseqno 0 not within window 5->5 libiax2/src/iax.c line 2039: Packet arrived out of order (expecting 2, got 0) (frametype = 6, subclass = 1) libiax2/src/iax.c line 2045: Acking anyway libiax2/src/iax.c line 2020: Received iseqno 0 not within window 5->5 libiax2/src/iax.c line 2039: Packet arrived out of order (expecting 2, got 1) (frametype = 6, subclass = 5) libiax2/src/iax.c line 2045: Acking anyway libiax2/src/iax.c line 2008: Cancelling transmission of packet 0 libiax2/src/iax.c line 2008: Cancelling transmission of packet 1 Thanks, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web: http://www.sokol-associates.com |
From: Steve K. <st...@st...> - 2004-06-25 13:22:21
|
Ronan SALMON wrote: > Hi, > > It is possible to get iaxcomm (and I guess the iaxclient library) to > work with esd(esound) ? The idea is to be able to have an audio > application and iaxcomm that can work at the same time. At the moment, > if I have xmmx running, I won't be able to hear any incoming calls. > I tried to run iaxcomm with esddsp, but iaxcomm doesn't work. It should work with esddsp, at least it did some time ago when I tried it.. If it doesn't, it would be possible to write an esd driver for either iaxclient itself, or (probably a better idea) for portaudio. > An other way arround would probably be to create two pipes (in and > out) and I could use esdrec/esdcat to read/send the sound to the esd > deamon but I can't change the audio settings within the GUI. > > I'm using the latest CVS and linux 2.4.26, esound 0.2.34. > > Thanks, Ronan. > > > ------------------------------------------------------- > This SF.Net email sponsored by Black Hat Briefings & Training. > Attend Black Hat Briefings & Training, Las Vegas July 24-29 - digital > self defense, top technical experts, no vendor pitches, unmatched > networking opportunities. Visit www.blackhat.com > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > |
From: Ronan S. <ron...@wa...> - 2004-06-25 12:31:06
|
Hi, It is possible to get iaxcomm (and I guess the iaxclient library) to work with esd(esound) ? The idea is to be able to have an audio application and iaxcomm that can work at the same time. At the moment, if I have xmmx running, I won't be able to hear any incoming calls. I tried to run iaxcomm with esddsp, but iaxcomm doesn't work. An other way arround would probably be to create two pipes (in and out) and I could use esdrec/esdcat to read/send the sound to the esd deamon but I can't change the audio settings within the GUI. I'm using the latest CVS and linux 2.4.26, esound 0.2.34. Thanks, Ronan. |
From: Dmitry M. <ar...@od...> - 2004-06-22 14:17:48
|
On Tuesday 22 June 2004 17:01, Steve Kann wrote: > Thanks Dmitry. > > It was also wrong in the IAX1 case, although I don't even know if the > IAX1 stuff even compiles anymore. > As I know its not really supported in Asterisk. So is there a sense to keep it in iaxclient lib? > Just out of curiosity, what are you doing that actually causes this case > to happen? It would only happen, if you were using frames > 20ms long. > I have the following setup: asterisk1 -- iax2 ilbc (1)-- asterisk2 -- iax2 gsm (2)-- iaxclient On the leg #2 packets goes from asterisk to iaxclient with 20,40,20,40ms... timestamps. Packets sizes are 33,66,33,66...etc. Dmitry. > I'm actually thinking of making an option for iaxclient to use larger > frames, because it uses much less bandwidth, but the tradeoff is > slightly longer latency, and also more perceptible glitches when there's > a frame dropped, etc. > > > -SteveK > > Dmitry Mishchenko wrote: > >Hello, > > > >In iaxclient_lib.c in handle_audio_event sub there is a code: > > > >#ifdef IAXC_IAX2 > > while(total_consumed < e->datalen) { > > cur = decode_audio(call, fr, > > e->data,e->datalen-total_consumed, > > iEncodeType); > >#else > > > >It should be replaced with: > > > >#ifdef IAXC_IAX2 > > while(total_consumed < e->datalen) { > > cur = decode_audio(call, fr, > > e->data+total_consumed,e->datalen-total_consumed, > > iEncodeType); > >#else > > > >It gives more clear audio when "while" construction actually working. > > > >Dmitry > > > > > > > >------------------------------------------------------- > >This SF.Net email sponsored by Black Hat Briefings & Training. > >Attend Black Hat Briefings & Training, Las Vegas July 24-29 - > >digital self defense, top technical experts, no vendor pitches, > >unmatched networking opportunities. Visit www.blackhat.com > >_______________________________________________ > >Iaxclient-devel mailing list > >Iax...@li... > >https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > ------------------------------------------------------- > This SF.Net email sponsored by Black Hat Briefings & Training. > Attend Black Hat Briefings & Training, Las Vegas July 24-29 - > digital self defense, top technical experts, no vendor pitches, > unmatched networking opportunities. Visit www.blackhat.com > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
From: Steve K. <st...@st...> - 2004-06-22 14:01:38
|
Thanks Dmitry. It was also wrong in the IAX1 case, although I don't even know if the IAX1 stuff even compiles anymore. Just out of curiosity, what are you doing that actually causes this case to happen? It would only happen, if you were using frames > 20ms long. I'm actually thinking of making an option for iaxclient to use larger frames, because it uses much less bandwidth, but the tradeoff is slightly longer latency, and also more perceptible glitches when there's a frame dropped, etc. -SteveK Dmitry Mishchenko wrote: >Hello, > >In iaxclient_lib.c in handle_audio_event sub there is a code: > >#ifdef IAXC_IAX2 > while(total_consumed < e->datalen) { > cur = decode_audio(call, fr, > e->data,e->datalen-total_consumed, > iEncodeType); >#else > >It should be replaced with: > >#ifdef IAXC_IAX2 > while(total_consumed < e->datalen) { > cur = decode_audio(call, fr, > e->data+total_consumed,e->datalen-total_consumed, > iEncodeType); >#else > >It gives more clear audio when "while" construction actually working. > >Dmitry > > > >------------------------------------------------------- >This SF.Net email sponsored by Black Hat Briefings & Training. >Attend Black Hat Briefings & Training, Las Vegas July 24-29 - >digital self defense, top technical experts, no vendor pitches, >unmatched networking opportunities. Visit www.blackhat.com >_______________________________________________ >Iaxclient-devel mailing list >Iax...@li... >https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > |
From: Dmitry M. <ar...@od...> - 2004-06-22 13:51:09
|
Hello, In iaxclient_lib.c in handle_audio_event sub there is a code: #ifdef IAXC_IAX2 while(total_consumed < e->datalen) { cur = decode_audio(call, fr, e->data,e->datalen-total_consumed, iEncodeType); #else It should be replaced with: #ifdef IAXC_IAX2 while(total_consumed < e->datalen) { cur = decode_audio(call, fr, e->data+total_consumed,e->datalen-total_consumed, iEncodeType); #else It gives more clear audio when "while" construction actually working. Dmitry |
From: <ale...@ya...> - 2004-06-16 00:32:46
|
Hello to everyone on the list, can someone please help me to figure out what I'm doing wrong? I have built wiax.dll correctly (thanks Dan). Then I use it this way: typedef int (*IAX_Initialize)(int audType, int nCalls); IAX_Initialize IAXInitialize; HINSTANCE hinstLib = LoadLibrary("wiax"); if (hinstLib == NULL) { AfxMessageBox("wiax.dll not found", MB_OK, 0); return; } IAXInitialize = (IAX_Initialize) GetProcAddress(hinstLib, "IAX_Initialize"); if (IAXInitialize == NULL) AfxMessageBox("IAXInitialize == NULL", MB_OK, 0); else (IAXInitialize)(AUDIO_INTERNAL_PA, 1); This last call to IAXInitialize causes a memory violation and the app ends. I modified the DLL not to call iax_initialize but to print a message with MessageBox and it works fine. Also are there any suggestions about where in the MFC code I should start the IAX initializations. It is just a CDialog. Thanks in advance for any help Alessandro --------------------------------- Yahoo! Companion - Scarica gratis la toolbar di Ricerca di Yahoo! |
From: Masakazu N. <n-...@md...> - 2004-06-13 18:17:26
|
Hi all I was tryed to connect to a latest cvs with iaxclientocx. and I got an error following like that. Jun 14 02:22:13 NOTICE[24600]: chan_iax.c:3882 socket_read: Rejected connect attempt from 192.168.xx.xx, requested/capability 0x2/0x2 incompatible with our capability 0x4. what does it reason? Best Regards mack |
From: ePyron F. D. <fd...@eP...> - 2004-06-10 16:33:29
|
Hello, is there any support for alaw or ulaw planned? It would be great to use iaxclients without codec translations. Thanks Felix Deierlein |
From: Steven S. <ss...@so...> - 2004-06-03 17:45:53
|
Here's my take on the process: Personal Conferencing for IAX2 For every IAX2 user that has the feature conf enabled (a new entry in the iax.conf and/or the database) chan_iax2 will support the ability to conference two or more parties using the MeetMe application, or a derivation of the MeetMe app. A new IAX2 command for Conference (CONF) will be created. The command will require one or more IEs that indicates whom to conference. [Creating The Initial Conference] --------------------------------- The client (UA) should recognize that a press of the CONF button with a single active call to mean "put this call on hold while I establish another call for a conference". The client will then place the active call on hold and select a different appearance which can be used to establish the seconds call. The client will likewise recognize the second click of the CONF button to mean "Ok, conference these calls together" and will send a CONF command to Asterisk. Asterisk will move the held party into the conference room, then move the two parties to the active call into the meeting room. The appearance used to conference the additional party will be freed and return to idle (i.e. the call will be destroyed). [Adding Parties To An Existing Conference] ------------------------------------------ If the user chooses to add an additional party to the conference by again pressing the CONF key, he/she will put the conference on hold and access another appearance for dialing. When the call to the new party is active, the user can again press CONF to add the new active party. Somewhere along the line, the system will need to recognize that the user's personal conference room _is_ the held party and will simply move the new remote party into the conference. The client should be smart enough to re-select the held conference channel (taking it off hold) so that all parties are again active. [Simulating This Scenario] -------------------------- I managed to simulate this by using static conferences with extension numbers and the Manager API. I am able to merge the calls into the static conference without any real issue. I simply redirect the held channel to the extension for the MeetMe, then redirect the two active channels to the same extension. All parties wind up in the bridge. Works fine. [Rules/Thoughts] ---------------- The conference application should be silent and pin-less by default. The conference should collapse when the conference count reaches 1 instead of zero. The conference should collapse when the "host" (i.e. the IAX user) drops out of the bridge. Some similar mechanism for dropping (kicking) conference members should be included in the package. [Programming The Conference Mechanism] -------------------------------------- The question is, what is the best way to programmatically affect this same set of circumstances? I want to make it work from IAX2 semi-automagically. Adding the command to IAX2 looks pretty easy. Where I get lost is in the moving process. I don't want to have to create an extension for each user's conference number. I want to activate the app_meetme code, passing in the user's ID. (i.e. [iaxuser@192.168.1.194:5469]) as the name of the conference to either create or join. I believe this requires some small changes to MeetMe to support the named dynamic conference. But where I really get lost is in the process of redirecting the channels to the MeetMe app in the first place. Is there any code similar to pbx_async_goto_by_name that takes an application as a destination, rather than an extension/priority? Anybody want to kick me in the right direction here? Thanks, Steve |
From: Steve K. <st...@st...> - 2004-06-02 23:56:05
|
I don't know how you'd implement exactly what you want to do, but while most office phones have "conf" buttons, the conferencing is most definitely done in the PBX, and not in the phones themselves. -SteveK On Jun 2, 2004, at 3:08 PM, Steven Sokol wrote: > Asterisk's MeetMe is powerful and flexible, but sometimes you just > want to > do an old-fashioned basic conference from the client, without having to > transfer everybody into a conference room. This is pretty common > functionality on most office phone systems. > > Scenario: > > 1. You (party A) are on an active call with party B. > 2. You decide to add in another party (C). > 3. You key the CONF button on your business set. > - Party B gets MOH or silence. > - Party B's appearance on your phone goes to the HELD state. > - You get new dial-tone on a new appearance > 4. You dial the number for party C. > 5. Party C answers and you communicate briefly. > 6. You press the CONF key again to establish the conference > - The bridging takes place in the PBX/KSU > - Party B is pulled out of MOH-land and added to the new call > - Party B's appearance returns to the ACTIVE state > > Now you are conferenced. Both call appearances show active. You can > even > follow the same set of steps to conference in another user if you have > additional appearances on your set. > > In Asterisk using most VoIP phones (soft or hard) this scenario does > not > play out. You wind up transferring party A into the MeetMe, calling > party B > and transferring them into the MeetMe, then dialing into the MeetMe > yourself. > > This is further complicated by the fact that any given static MeetMe > may or > may not already be occupied by others doing the same thing. Dynamic > MeetMes > make this a bit more flexible, but there's little documentation to the > dynamic feature. > > Is there an elegant way to execute dynamic conferencing in Asterisk? > Can > this be done without building conferencing logic into the client UA? > If so, > how? > > If not (which I think is the case), how do we add this in? I have a > number > of clients using my IAX Phone who are not really happy about having to > jump > through hoops to handle spontaneous conferences. > > I would be happy to _try_ to code the solution into IAX2 (both > chan_iax2 and > libiax2) if somebody can tell me where to begin. I would think there > has to > be some way to add in an IAX2 command to merge two calls or to command > Asterisk to bridge the calls at the server. > > Also, is there a way execute such a conference from SIP, MGCP, H323, or > other protocols? If so, who wants to implement it? > > Any thoughts would be appreciated. Apologies in advance if I missed > some > really simple way of doing this. > > Thanks, > > Steve > > Steven Sokol > Owner/Manager > Sokol & Associates, LLC > > Phone: 816.822.1807 > IaxTel: 700.613.9004 > Web: http://www.sokol-associates.com > > > > > > ------------------------------------------------------- > This SF.Net email is sponsored by the new InstallShield X. > From Windows to Linux, servers to mobile, InstallShield X is the one > installation-authoring solution that does it all. Learn more and > evaluate today! http://www.installshield.com/Dev2Dev/0504 > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > |
From: Steven S. <ss...@so...> - 2004-06-02 19:08:52
|
Asterisk's MeetMe is powerful and flexible, but sometimes you just want to do an old-fashioned basic conference from the client, without having to transfer everybody into a conference room. This is pretty common functionality on most office phone systems. Scenario: 1. You (party A) are on an active call with party B. 2. You decide to add in another party (C). 3. You key the CONF button on your business set. - Party B gets MOH or silence. - Party B's appearance on your phone goes to the HELD state. - You get new dial-tone on a new appearance 4. You dial the number for party C. 5. Party C answers and you communicate briefly. 6. You press the CONF key again to establish the conference - The bridging takes place in the PBX/KSU - Party B is pulled out of MOH-land and added to the new call - Party B's appearance returns to the ACTIVE state Now you are conferenced. Both call appearances show active. You can even follow the same set of steps to conference in another user if you have additional appearances on your set. In Asterisk using most VoIP phones (soft or hard) this scenario does not play out. You wind up transferring party A into the MeetMe, calling party B and transferring them into the MeetMe, then dialing into the MeetMe yourself. This is further complicated by the fact that any given static MeetMe may or may not already be occupied by others doing the same thing. Dynamic MeetMes make this a bit more flexible, but there's little documentation to the dynamic feature. Is there an elegant way to execute dynamic conferencing in Asterisk? Can this be done without building conferencing logic into the client UA? If so, how? If not (which I think is the case), how do we add this in? I have a number of clients using my IAX Phone who are not really happy about having to jump through hoops to handle spontaneous conferences. I would be happy to _try_ to code the solution into IAX2 (both chan_iax2 and libiax2) if somebody can tell me where to begin. I would think there has to be some way to add in an IAX2 command to merge two calls or to command Asterisk to bridge the calls at the server. Also, is there a way execute such a conference from SIP, MGCP, H323, or other protocols? If so, who wants to implement it? Any thoughts would be appreciated. Apologies in advance if I missed some really simple way of doing this. Thanks, Steve Steven Sokol Owner/Manager Sokol & Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web: http://www.sokol-associates.com |
From: The S. <kc...@al...> - 2004-06-02 07:13:53
|
Hai All, Does anybody know exactly how the cmds for * should work, i've been trying to get the following done but it would not work, is there a special key to touch, or something else special ?? I used different clients also, my own (build on libiax1 & 2) but got stuck when I want to communicate with the pbx. I setup a line between A and B, where I use iaxtransfers=no in the iax.conf, then when I try for example to park the call by dialing 700 or #700, I can see the the digits come from client A to the * and he forwards them to client B, instead of acting on them .. in the extensions.cnonf I used the rules like this [default] include => parkedcalls exten => 100,2,Dial,IAX/100/s|30|t exten => 200,2,Dial,IAX/200/s|30|t also the transfer shows the same effect .. the only thing that works is a blind transfer .. Should I use an special func to tell * the cmd's ??? I thought it was all handled by the iax2 (ies). Anybody any idea ? greetz KC -- ========================================================== Please avoid mailing Word or Power Point attachments. See http://www.fsf.org/philosophy/no-word-attachments.html |
From: Saad F. <pro...@ya...> - 2004-06-01 13:27:29
|
how do use the lib file generated by IAXcomm project so that i can use it in my Microsoft Visual C++ project? Any ideas guys? I read in the archive that we can generate a dll instead of the lib can you guys guide me how i should go about this. I need to develop applications for the windows platform and i need to use both VC++ as well as Visual Basic. Any help regarding this will be greatly appreciated. --------------------------------- Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger |
From: <ben...@id...> - 2004-05-22 12:04:43
|
Dear Open Source developer I am doing a research project on "Fun and Software Development" in which I kindly invite you to participate. You will find the online survey under http://fasd.ethz.ch/qsf/. The questionnaire consists of 53 questions and you will need about 15 minutes to complete it. With the FASD project (Fun and Software Development) we want to define the motivational significance of fun when software developers decide to engage in Open Source projects. What is special about our research project is that a similar survey is planned with software developers in commercial firms. This procedure allows the immediate comparison between the involved individuals and the conditions of production of these two development models. Thus we hope to obtain substantial new insights to the phenomenon of Open Source Development. With many thanks for your participation, Benno Luthiger PS: The results of the survey will be published under http://www.isu.unizh.ch/fuehrung/blprojects/FASD/. We have set up the mailing list fa...@we... for this study. Please see http://fasd.ethz.ch/qsf/mailinglist_en.html for registration to this mailing list. _______________________________________________________________________ Benno Luthiger Swiss Federal Institute of Technology Zurich 8092 Zurich Mail: benno.luthiger(at)id.ethz.ch _______________________________________________________________________ |
From: Chris G. N. <cg...@gl...> - 2004-05-21 00:04:18
|
so, after a first pass on the sources, perhaps add a new interface/beef = up lib/libiax2/src/iax.c to have: int iax_init(int preferredportno, int ipflavor) { .... netfd =3D socket(AF_INET, ip_flavor, IPPROTO_IP); } and the existing routine would become a wrapper... int iax_init(int preferredportno) {=20 return(iax_init(preferredportno, SOCK_DGRAM))=20 } ???? Chris -----Original Message----- From: Steve Kann [mailto:st...@st...] Sent: Thu 5/20/2004 10:53 AM To: Chris G. Nicholas Cc: iax...@li... Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? Chris G. Nicholas wrote: >If this is already done, packaged, and reasonably priced, and can still = work with the other free iax clients and servers, sounds like a way to = go... > >Can you provide more info on this setup? > =20 > Unfortunately, our solution is not for sale, but you can probably find=20 other similar solutions with your friend google: I think zebedee does this; it can even encrypt/compress things=20 [compression probably not too helpful]. http://www.winton.org.uk/zebedee/ Also This might work: http://tunnel.mrq3.com/ >thanks! >Chris =20 > > > >-----Original Message----- >From: Steve Kann [mailto:st...@st...] >Sent: Thu 5/20/2004 9:23 AM >To: Chris G. Nicholas >Cc: iax...@li... >Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? >Chris G. Nicholas wrote: > > =20 > >>just learning about Asterix; cool! >> >>Apologies in advance if this has been answered before. =20 >> >>Even though I fully realize TCP/IP connections suck compared to UDP = for streaming media, unfortunately the only port one can generally count = on, particularly at industry conferences, is outbound port 80 using TCP = connections from behind a NAT. (hence the famous John Walker = 'speakfree' end-of-life lament...) >> >>Just wondering if anyone has thought about the feasibility of a TCP = connection option in iaxclient, initially just connecting to an = external/simple/stupid tunnel proxy that turns around and connects to an = Asterix PBX, or even just building in port 80/TCP support into Asterix = itself... >>=20 >> >> =20 >> > >Actually, we do IAX over either straight TCP, or actual rfc-compliant=20 >HTTP.=20 > >Unfortunately, we do this with a proprietary program [external to=20 >iaxclient], and can't release it, but I think there are similar=20 >solutions available. They don't need to be built into iaxclient to use = >them. The same reasons that make IAX work well with NATs make it work=20 >fine with tunnels as well. > > >So basically, what you need to do is this: > > Client machine =20 >Network Server >[Iaxclient <-UDP-> tunnel ] [<- HTTP or TCP -> ] [ tunnel<-UDP-> = asterisk ] > >The tunnel programs at either side are what you need, and it needs to=20 >support UDP tunneling. > >-SteveK > > > > > =20 > |
From: Chris G. N. <cg...@gl...> - 2004-05-20 18:13:20
|
Bummer :-( So back to the original thought here, I would just assume have the TCP = proxy code just 'baked' into an IAX client as an optional connect = method, since the process of installing/configuring anything additional, = particularly if it involved Java, will lose > 95% of potential users, = and defeats the purpose of a ~100Kbyte download. Need to dive into the sources... Chris -----Original Message----- From: Steve Kann [mailto:st...@st...] Sent: Thu 5/20/2004 10:53 AM To: Chris G. Nicholas Cc: iax...@li... Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? Chris G. Nicholas wrote: >If this is already done, packaged, and reasonably priced, and can still = work with the other free iax clients and servers, sounds like a way to = go... > >Can you provide more info on this setup? > =20 > Unfortunately, our solution is not for sale, but you can probably find=20 other similar solutions with your friend google: I think zebedee does this; it can even encrypt/compress things=20 [compression probably not too helpful]. http://www.winton.org.uk/zebedee/ Also This might work: http://tunnel.mrq3.com/ >thanks! >Chris =20 > > > >-----Original Message----- >From: Steve Kann [mailto:st...@st...] >Sent: Thu 5/20/2004 9:23 AM >To: Chris G. Nicholas >Cc: iax...@li... >Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? >Chris G. Nicholas wrote: > > =20 > >>just learning about Asterix; cool! >> >>Apologies in advance if this has been answered before. =20 >> >>Even though I fully realize TCP/IP connections suck compared to UDP = for streaming media, unfortunately the only port one can generally count = on, particularly at industry conferences, is outbound port 80 using TCP = connections from behind a NAT. (hence the famous John Walker = 'speakfree' end-of-life lament...) >> >>Just wondering if anyone has thought about the feasibility of a TCP = connection option in iaxclient, initially just connecting to an = external/simple/stupid tunnel proxy that turns around and connects to an = Asterix PBX, or even just building in port 80/TCP support into Asterix = itself... >>=20 >> >> =20 >> > >Actually, we do IAX over either straight TCP, or actual rfc-compliant=20 >HTTP.=20 > >Unfortunately, we do this with a proprietary program [external to=20 >iaxclient], and can't release it, but I think there are similar=20 >solutions available. They don't need to be built into iaxclient to use = >them. The same reasons that make IAX work well with NATs make it work=20 >fine with tunnels as well. > > >So basically, what you need to do is this: > > Client machine =20 >Network Server >[Iaxclient <-UDP-> tunnel ] [<- HTTP or TCP -> ] [ tunnel<-UDP-> = asterisk ] > >The tunnel programs at either side are what you need, and it needs to=20 >support UDP tunneling. > >-SteveK > > > > > =20 > |
From: Steve K. <st...@st...> - 2004-05-20 17:53:10
|
Chris G. Nicholas wrote: >If this is already done, packaged, and reasonably priced, and can still work with the other free iax clients and servers, sounds like a way to go... > >Can you provide more info on this setup? > > Unfortunately, our solution is not for sale, but you can probably find other similar solutions with your friend google: I think zebedee does this; it can even encrypt/compress things [compression probably not too helpful]. http://www.winton.org.uk/zebedee/ Also This might work: http://tunnel.mrq3.com/ >thanks! >Chris > > > >-----Original Message----- >From: Steve Kann [mailto:st...@st...] >Sent: Thu 5/20/2004 9:23 AM >To: Chris G. Nicholas >Cc: iax...@li... >Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? >Chris G. Nicholas wrote: > > > >>just learning about Asterix; cool! >> >>Apologies in advance if this has been answered before. >> >>Even though I fully realize TCP/IP connections suck compared to UDP for streaming media, unfortunately the only port one can generally count on, particularly at industry conferences, is outbound port 80 using TCP connections from behind a NAT. (hence the famous John Walker 'speakfree' end-of-life lament...) >> >>Just wondering if anyone has thought about the feasibility of a TCP connection option in iaxclient, initially just connecting to an external/simple/stupid tunnel proxy that turns around and connects to an Asterix PBX, or even just building in port 80/TCP support into Asterix itself... >> >> >> >> > >Actually, we do IAX over either straight TCP, or actual rfc-compliant >HTTP. > >Unfortunately, we do this with a proprietary program [external to >iaxclient], and can't release it, but I think there are similar >solutions available. They don't need to be built into iaxclient to use >them. The same reasons that make IAX work well with NATs make it work >fine with tunnels as well. > > >So basically, what you need to do is this: > > Client machine >Network Server >[Iaxclient <-UDP-> tunnel ] [<- HTTP or TCP -> ] [ tunnel<-UDP-> asterisk ] > >The tunnel programs at either side are what you need, and it needs to >support UDP tunneling. > >-SteveK > > > > > > |
From: Chris G. N. <cg...@gl...> - 2004-05-20 17:34:12
|
If this is already done, packaged, and reasonably priced, and can still = work with the other free iax clients and servers, sounds like a way to = go... Can you provide more info on this setup? thanks! Chris =20 -----Original Message----- From: Steve Kann [mailto:st...@st...] Sent: Thu 5/20/2004 9:23 AM To: Chris G. Nicholas Cc: iax...@li... Subject: Re: [Iaxclient-devel] firewalls and TCP connections ... ? Chris G. Nicholas wrote: >just learning about Asterix; cool! > >Apologies in advance if this has been answered before. =20 > >Even though I fully realize TCP/IP connections suck compared to UDP for = streaming media, unfortunately the only port one can generally count on, = particularly at industry conferences, is outbound port 80 using TCP = connections from behind a NAT. (hence the famous John Walker = 'speakfree' end-of-life lament...) > >Just wondering if anyone has thought about the feasibility of a TCP = connection option in iaxclient, initially just connecting to an = external/simple/stupid tunnel proxy that turns around and connects to an = Asterix PBX, or even just building in port 80/TCP support into Asterix = itself... > =20 > Actually, we do IAX over either straight TCP, or actual rfc-compliant=20 HTTP.=20 Unfortunately, we do this with a proprietary program [external to=20 iaxclient], and can't release it, but I think there are similar=20 solutions available. They don't need to be built into iaxclient to use=20 them. The same reasons that make IAX work well with NATs make it work=20 fine with tunnels as well. So basically, what you need to do is this: Client machine =20 Network Server [Iaxclient <-UDP-> tunnel ] [<- HTTP or TCP -> ] [ tunnel<-UDP-> = asterisk ] The tunnel programs at either side are what you need, and it needs to=20 support UDP tunneling. -SteveK |
From: Steve K. <st...@st...> - 2004-05-20 16:23:12
|
Chris G. Nicholas wrote: >just learning about Asterix; cool! > >Apologies in advance if this has been answered before. > >Even though I fully realize TCP/IP connections suck compared to UDP for streaming media, unfortunately the only port one can generally count on, particularly at industry conferences, is outbound port 80 using TCP connections from behind a NAT. (hence the famous John Walker 'speakfree' end-of-life lament...) > >Just wondering if anyone has thought about the feasibility of a TCP connection option in iaxclient, initially just connecting to an external/simple/stupid tunnel proxy that turns around and connects to an Asterix PBX, or even just building in port 80/TCP support into Asterix itself... > > Actually, we do IAX over either straight TCP, or actual rfc-compliant HTTP. Unfortunately, we do this with a proprietary program [external to iaxclient], and can't release it, but I think there are similar solutions available. They don't need to be built into iaxclient to use them. The same reasons that make IAX work well with NATs make it work fine with tunnels as well. So basically, what you need to do is this: Client machine Network Server [Iaxclient <-UDP-> tunnel ] [<- HTTP or TCP -> ] [ tunnel<-UDP-> asterisk ] The tunnel programs at either side are what you need, and it needs to support UDP tunneling. -SteveK |
From: Chris G. N. <cg...@gl...> - 2004-05-20 15:57:07
|
just learning about Asterix; cool! Apologies in advance if this has been answered before. =20 Even though I fully realize TCP/IP connections suck compared to UDP for = streaming media, unfortunately the only port one can generally count on, = particularly at industry conferences, is outbound port 80 using TCP = connections from behind a NAT. (hence the famous John Walker = 'speakfree' end-of-life lament...) Just wondering if anyone has thought about the feasibility of a TCP = connection option in iaxclient, initially just connecting to an = external/simple/stupid tunnel proxy that turns around and connects to an = Asterix PBX, or even just building in port 80/TCP support into Asterix = itself... thanks in advance -=20 Chris Nicholas=20 |
From: Chris G. N. <cg...@gl...> - 2004-05-20 15:56:45
|
just learning about Asterix; cool! Apologies in advance if this has been answered before. =20 Even though I fully realize TCP/IP connections suck compared to UDP for = streaming media, unfortunately the only port one can generally count on, = particularly at industry conferences, is outbound port 80 using TCP = connections from behind a NAT. (hence the famous John Walker = 'speekfree' end-of-life lament...) Just wondering if anyone has thought about the feasibility of a TCP = connection option in iaxclient, initially just connecting to an = external/simple/stupid tunnel proxy that turns around and connects to an = Asterix PBX, or even just building in port 80/TCP support into Asterix = itself... thanks in advance -=20 Chris Nicholas=20 |
From: <ade...@so...> - 2004-05-15 16:46:27
|
Are there somebody planning to add recording features to IaxClient ? I would like to have this functionality as soon as possible.=20 I could allocate a C Programmer to do that, if somebody show me the Road=20 Map ... Tks Adelino Baena Brazil=20 Citando Steve Kann <st...@st...>: >=20 > On May 14, 2004, at 9:53 PM, Michael Van Donselaar wrote: >=20 > > Since everyone but me is named Steve, I knew that would get everyone'= s > > attention. > > > > It's been so quiet around here lately. Last I heard there *might* be= =20 > > Some work > > going on to add other codecs to the library. > > > > Is anyone in fact working on this ? >=20 > I think Steve Sokol started working on this, but I don't believe he has= =20 > finished; not sure how far he got. >=20 > I've been working on both iaxclient and app_conference, and have=20 > committed a bunch of changes to make things more reliable. >=20 > -SteveK >=20 >=20 >=20 > ------------------------------------------------------- > This SF.Net email is sponsored by: SourceForge.net Broadband > Sign-up now for SourceForge Broadband and get the fastest > 6.0/768 connection for only $19.95/mo for the first 3 months! > http://ads.osdn.com/?ad_id=3D2562&alloc_id=3D6184&op=3Dclick > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel >=20 "Estas informa=E7=F5es s=E3o destinadas apenas para a pessoa ou entidade = para a qual =E9 endere=E7ada e pode conter material restrito ou confidenc= ial. Qualquer revis=E3o, retransmiss=E3o, dissemina=E7=E3o ou outro uso d= estas informa=E7=F5es, ou tomada de qualquer a=E7=E3o em rela=E7=E3o a es= tas informa=E7=F5es por pessoas ou entidades que n=E3o sejam o destinat=E1= rio pretendido =E9 expressamente proibida. Se voc=EA recebeu esta mensage= m por engano, por favor contate o remetente e apague este material de tod= o e qualquer computador." "This information is intended only for the person or entity to which it i= s addressed and may contain confidential or privileged material. Any revi= ew, retransmission, dissemination, or other use of, or taking of any acti= on in reliance upon, this information by persons or entities other than t= he intended recipient is prohibited. If you received this in error, pleas= e contact the sender and delete the material from any computer." |
From: Steve K. <st...@st...> - 2004-05-15 13:00:58
|
On May 14, 2004, at 9:53 PM, Michael Van Donselaar wrote: > Since everyone but me is named Steve, I knew that would get everyone's > attention. > > It's been so quiet around here lately. Last I heard there *might* be > Some work > going on to add other codecs to the library. > > Is anyone in fact working on this ? I think Steve Sokol started working on this, but I don't believe he has finished; not sure how far he got. I've been working on both iaxclient and app_conference, and have committed a bunch of changes to make things more reliable. -SteveK |