[Asterisk-Dev] IAXClient
TechniqueCHK
equipedev at chaka.sn
Tue Jul 6 03:39:35 MST 2004
Helo,
I search a delphi code source witch use the dll wiax.dll to develop a
iax client.
Thank!
asterisk-dev-request at lists.digium.com wrote:
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> Today's Topics:
>
> 1. Re: MeetMe Improvement (dorian logan)
> 2. Re: MeetMe Improvement (dking at pimpsoft.com)
> 3. Half installed on Sparc: Drivers for FXO card refusing to install. (dking at pimpsoft.com)
> 4. Re: Module unloading (Rich Adamson)
> 5. Re: Module unloading (Michael Sandee)
> 6. Help on LeadTek's Latest VideoPhone (Dean Huang)
>
> --__--__--
>
> Message: 1
> From: dorian logan <dorian at tuxstar.com>
> Subject: Re: [Asterisk-Dev] MeetMe Improvement
> Date: Mon, 5 Jul 2004 11:11:31 +0100
> To: asterisk-dev at lists.digium.com
> Reply-To: asterisk-dev at lists.digium.com
>
>
> --Apple-Mail-3--954058126
> Content-Transfer-Encoding: 7bit
> Content-Type: text/plain;
> charset=US-ASCII;
> format=flowed
>
> I think the functionality that you describe will prove to be very
> useful - providing Asterisk with functionality to rival other
> conference systems. One additional item that I would like to see is
> integrate some streaming bridges e.g. allow conferences to be attended
> by people via a stream rather than a phone or SIP connection. I know
> the Ices project is working on this too.
>
> Asterisk is well positioned to demonstrate what can be done when
> Internet and telephony are seamlessly integrated.
>
> Another nice application would be web based management for a conference
> - alowing the manager to see who is attending and control the
> conference eg mute speakers etc.
>
> Please let me know how you progress - I am happy to assist you with
> this project too.
>
> D.
>
>
> On 5 Jul 2004, at 03:42, Marc Olivier Chouinard wrote:
>
>
>>Hi everyone,
>>
>>I've start working to add a few new feature on meetme, I'll list them
>>here :
>>(bug 1973)
>>- Hold everyone until Admin login
>>- When admin join, he get a prompt about how many users will join the
>>conference(was on hold)
>>- Prompt to the user about waiting for the admin
>>- Prompt when the admin join the conference
>>- Prompt when the admin leave the conference
>>- Allow retry when asked for pin
>>- Added a adminpin in meetme.conf
>>- Beable to login as admin when current a user(or waiting for admin)
>>(option * 9)
>>
>>I've talked alittle bit to mark about maybe the need for a more
>>advanced
>>meetme application, maybe just called Conference. MeetMe was made at
>>first
>>as a basic conference system, and if we want to keep backward
>>compability if
>>we want to add new feature, we should look at a new framework that
>>base more
>>it config on the meetme.conf or conference.conf depending on what we
>>want to
>>do.
>>
>>So those feature I showed he seem to be more willing to add them after
>>the
>>head goes out, but if we can work on building on a new conference
>>platform,
>>it could just be put into that one instead.
>>
>>I wanted to know if it was only me that was interested in making the
>>conference system more advanced. I don't even use it really, but after
>>using other conference system at work, I feel that * shouldn't be
>>without
>>them. If not adding more.
>>
>>Here a list of new thing the new conference system could be included,
>>please
>>add to the list so it help build a new framework for it:
>>- Better management interface, maybe via the Manager API
>>- Better VideoConference support(Maybe support ISDN Videoconferencing,
>>and
>>make * also a Videoconference Bridge)
>>- Multiple layer of users, maybe 1 PIN per user predefined in a
>>database
>>somewhere
>>- Have the posibility for the chairman to record the conference
>>
>>Hey maybe even use a speech reconition and convert to text, pass it to
>>a
>>translator and put it back with festival to another language... I
>>donno, but
>>the idea is to make it more open and easy to add feature, we keep
>>adding
>>flag, but someday will be out, and maybe we want to specify flag for a
>>specific user, that we cannot make rightnow.
>>
>>So basicly I'l dlike to know what you guys think about that
>>
>>Hope to hear from you all soon
>>
>>Marc O. Chouinard
>>PS. Sorry for my bad english
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> --Apple-Mail-3--954058126
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> --Apple-Mail-3--954058126--
>
>
> --__--__--
>
> Message: 2
> From: dking at pimpsoft.com
> Organization: pimpsoft.com
> To: asterisk-dev at lists.digium.com
> Date: Mon, 05 Jul 2004 03:44:39 -0700
> Subject: Re: [Asterisk-Dev] MeetMe Improvement
> Reply-To: asterisk-dev at lists.digium.com
>
> It would be VERY interesting if IAX could tap into this and allow
> multiple systems to work on the exact same conference. Better yet
> (and it probably can do this anyway; I'm ignorant) if it could allow
> us to create a decentralized system of our own:
>
> Given the following is true:
>
> The POTS phone network is composed of interconnected systems working
> in cooperation.
> The POTS system works due to its large scale.
> The POTS system while dynamic in design is mostly static in nature
> due to the needs of its users.
> The internet is worldwide.
> Open source telephone systems software and cheap hardware exists.
> Many people like creating things.
> Many people hate centralized power over communication.
> Many people exists who have a love for the telephone system and learn
> all they can about it.
> Many People hate paying for what they can get for free.
>
> Then the following is also true:
> One person can connect the internet to there local phone system.
> A group of like minded individuals could create there own phone
> system attached to the worldwide POTS system.
> This system could be used for local calls if the local community was
> large enough in one area.
> If this group was large enough then both long distance and local
> telephone bills would end for the entire world.
> This group of people exists yet has not been tapped into yet.
> This would hurt the phone companies revenue exceptionally depending
> on the number of people involved.
> This in turn would drive down the costs to the consumer.
>
> SO.. Whats to stop people from giving asterisk as the opensource gpl
> product it is this functionality?
>
> - Belgarath
> On 5 Jul 2004 at 11:11, dorian logan wrote:
>
>
>>One additional item that I would like to see is
>>integrate some streaming bridges e.g. allow conferences to be attended
>>by people via a stream rather than a phone or SIP connection. I know
>>the Ices project is working on this too.
>>
>>Asterisk is well positioned to demonstrate what can be done when
>>Internet and telephony are seamlessly integrated.
>>
>
>
>
>
>
> --__--__--
>
> Message: 3
> From: dking at pimpsoft.com
> Organization: pimpsoft.com
> To: asterisk-dev at lists.digium.com
> Date: Mon, 05 Jul 2004 03:19:08 -0700
> Subject: [Asterisk-Dev] Half installed on Sparc: Drivers for FXO card refusing to install.
> Reply-To: asterisk-dev at lists.digium.com
>
> I was just able to install the Asterisk application on a Sparc Ultra
> 5 after allot of work. The make files seem to take into account
> every arch except for anything based on sparc. After I checked out a
> copy from cvs earlier tonight and modified a number of makefiles it
> DID compile and says that it is installed.
>
> I'm running debian 3.0r2 stable on the Ultra Sparc 5.
>
> The biggest problems I had was that its gcc states that the '-march=3D'
> system being used is invalid. I suspect that is because it is
> considered obsolete but I think if we split up the makefile variables
> for this option into 2 instead of one with one being a compound of
> the two, the system can be configured to work well on all
> architectures.
>
> Another thing that gave me trouble was the gsm code and the k6opts
> files; This can be fixed by a few additions to the make file I
> believe, I just took the references to this file out of the
> gsm/Makefile.
>
> I just tried to get my FXO card working and it seems that part of the
> system is munged; depmod refuses to proc the modules on a =91make
> install=92 of the zaptel files and modprobe says the files are not
> there when I know damn well they are; this however is the only thing
> that is refusing to install after my tweaking of the makefiles.
>
> The errors I am getting from depmod for each file in the zaptel
> codebase is:
>
> depmod: ELF file /lib/modiles/2.4.19/misc/[file name] not for this
> architecture
>
> Yet I KNOW they are compiled correct and are 32 bit SPARC binaries.
>
> So much for my dreams of a Sparc Based PBX, unless of course any of
> you have a fix so I can get the drivers to load? Hopefully I'm asking
> this in the right spot.
>
> Has anyone else had any experience getting this working on a sparc,
> or am I the first crazy enough to try with the spare time?
>
> - Duane
>
>
>
>
> --__--__--
>
> Message: 4
> Date: Mon, 5 Jul 2004 06:58:21 -0600
> From: Rich Adamson <radamson at routers.com>
> Subject: Re: [Asterisk-Dev] Module unloading
> To: asterisk-dev at lists.digium.com
> Reply-To: asterisk-dev at lists.digium.com
>
>
>>I still don't see how this is a sollution which can be used by module
>>developers in short term... This is the whole point of the problem...
>>ofcourse it can be fixed in the core... but that doesn't help anyone at
>>this point, it will also need good testing, some icky stuff will happen.
>>However it's a good thing that it will be fixed in the future...
>>
>>Other than that you are pushing this into a discussion between atexit()
>>or unload_module()... which it isn't... I am pointing out that the
>>"documentation" is incorrect regarding the current implementation. Which
>>is *very* annoying for module developers. All I wanted to do is raise
>>some attention on this subject so that we could see if it could be fixed
>>in the long term... and propose a short term sollution to module
>>developers (which already exists in existing * installations, in
>>contrary to your sollution).
>
>
> I'm not a developer, but this list/system is highly oriented around
> placing your thoughts/code in the bug tracker even if its relatively
> simple text changes. If the text is reasonable, someone will move it
> into cvs sooner or later. So, open a bug and others can add comments
> to it.
>
>
>
>
> --__--__--
>
> Message: 5
> Date: Mon, 05 Jul 2004 12:14:26 +0200
> From: Michael Sandee <ms at zeelandnet.nl>
> To: asterisk-dev at lists.digium.com
> Subject: Re: [Asterisk-Dev] Module unloading
> Reply-To: asterisk-dev at lists.digium.com
>
> Hi Rich,
>
> I do not have any code to contribute, and if I have I do not want to
> disclaim it... (but that is a different discussion)
>
> The point is, that module authors should be aware of this fact, and I
> can imagine that all module authors read this list. Most don't follow
> the bugtracker... What Asterisk core developers do with this? I cannot
> care less basicly... It works the way it is with the patch for proper
> atexit handling I made...
>
> According to Paul it is a planned feature, to make it right in version
> 1.2......
> So I am not going to clutter the bugtracker with this and have another
> place to monitor for replies... no thanks. Since noone seems to get the
> point I was making...
>
> I wrote the information I had down, and you can use it or not... I do
> not care. People who understood the problem were happy I found out, the
> others will keep discussing it instead of fixing the problem...
>
> Michael
>
> Rich Adamson wrote:
>
>
>>>I still don't see how this is a sollution which can be used by module
>>>developers in short term... This is the whole point of the problem...
>>>ofcourse it can be fixed in the core... but that doesn't help anyone at
>>>this point, it will also need good testing, some icky stuff will happen.
>>>However it's a good thing that it will be fixed in the future...
>>>
>>>Other than that you are pushing this into a discussion between atexit()
>>>or unload_module()... which it isn't... I am pointing out that the
>>>"documentation" is incorrect regarding the current implementation. Which
>>>is *very* annoying for module developers. All I wanted to do is raise
>>>some attention on this subject so that we could see if it could be fixed
>>>in the long term... and propose a short term sollution to module
>>>developers (which already exists in existing * installations, in
>>>contrary to your sollution).
>>>
>>>
>>
>>I'm not a developer, but this list/system is highly oriented around
>>placing your thoughts/code in the bug tracker even if its relatively
>>simple text changes. If the text is reasonable, someone will move it
>>into cvs sooner or later. So, open a bug and others can add comments
>>to it.
>>
>>
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>>
>>
>
>
>
> --__--__--
>
> Message: 6
> Date: Mon, 05 Jul 2004 10:21:46 -0500
> From: Dean Huang <ubinh at yahoo.com>
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] Help on LeadTek's Latest VideoPhone
> Reply-To: asterisk-dev at lists.digium.com
>
> Hello All,
>
> I need some helps on LeadTek's latest VideoPhone.
>
> 1) Voice functions of the phone works perfect
> 2) Asterisk always dropped the video part in SDP from one phone and
> invited the another phone with only voice capacity
>
> The following is the log from *.
>
> Regards,
> D.H.
>
> ======================================================================
> Jul 5 09:08:29 VERBOSE[1133742896]: 8 headers, 0 lines
> Jul 5 09:08:29 DEBUG[1133742896]: Stopping retransmission on
> '50fab160-40794-c0a81dd8 at sipusers.com' of Response 100: Found
> Jul 5 09:08:29 VERBOSE[1133742896]:
>
> Sip read:
> INVITE sip:9722352288 at sipusers.com:5060;user=phone SIP/2.0
>
> Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com
>
> From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1
>
> To: <sip:9722352288 at sipusers.com;user=phone>
>
> CSeq: 101 INVITE
>
> Via: SIP/2.0/UDP 192.168.10.19:5061
>
> Contact: <sip:9722352287 at 192.168.10.19:5061;user=phone>
>
> Max-Forwards: 70
>
> Supported: timer
>
> Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS
>
> Expires: 90
>
> Session-Expires: 180
>
> Proxy-Authorization: Digest
> username="9722352287",realm="asterisk",uri="sip:9722352288 at sipusers.com",response="961a05c280b881d57abc6e05043bd523",nonce="58b80f5a"
>
> Content-Type: application/sdp
>
> Content-Length: 246
>
>
>
> v=0
>
> o=9722352287 0 0 IN IP4 192.168.10.19
>
> s=-
>
> c=IN IP4 192.168.10.19
>
> t=0 0
>
> m=audio 9838 RTP/AVP 4 0 8 101
>
> a=ptime:30
>
> a=rtpmap:101 telephone-events/8000
>
> m=video 9840 RTP/AVP 34 31
>
> b=AS:128
>
> a=rtpmap:34 H263/90000
>
> a=rtpmap:31 H261/90000
>
>
> Jul 5 09:08:29 VERBOSE[1133742896]: 15 headers, 12 lines
> Jul 5 09:08:29 VERBOSE[1133742896]: Using latest request as basis request
> Jul 5 09:08:29 VERBOSE[1133742896]: Sending to 192.168.10.19 : 5061
> (non-NAT)
> Jul 5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 4
> Jul 5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 0
> Jul 5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 8
> Jul 5 09:08:29 VERBOSE[1133742896]: Found RTP audio format 101
> Jul 5 09:08:29 VERBOSE[1133742896]: Found video format UNKN
> Jul 5 09:08:29 VERBOSE[1133742896]: Found video format UNKN
> Jul 5 09:08:29 VERBOSE[1133742896]: Peer RTP is at port 192.168.10.19:9840
> Jul 5 09:08:29 VERBOSE[1133742896]: Found description format
> telephone-events
> Jul 5 09:08:29 VERBOSE[1133742896]: Found description format H263
> Jul 5 09:08:29 VERBOSE[1133742896]: Found description format H261
> Jul 5 09:08:29 VERBOSE[1133742896]: Capabilities: us - 0x1(G723), peer
> - audio=0xd(G723|ULAW|ALAW)/video=0xc0000(H261|H263), combined - 0x1(G723)
> Jul 5 09:08:29 VERBOSE[1133742896]: Non-codec capabilities: us -
> 0x1(G723), peer - 0x1(G723), combined - 0x1(G723)
> Jul 5 09:08:29 VERBOSE[1133742896]: Found peer '9722352287'
> Jul 5 09:08:29 DEBUG[1133742896]: Setting NAT on RTP to 0
> Jul 5 09:08:29 DEBUG[1133742896]: Setting NAT on VRTP to 0
> Jul 5 09:08:29 DEBUG[1133742896]: Check for res for 9722352287
> Jul 5 09:08:29 DEBUG[1133742896]: 9722352287 is not a local user
> Jul 5 09:08:29 VERBOSE[1133742896]: Looking for 9722352288 in sip
> Jul 5 09:08:30 DEBUG[1133742896]: build_route: Contact hop:
> <sip:9722352287 at 192.168.10.19:5061;user=phone>
> Jul 5 09:08:30 VERBOSE[1133742896]: list_route: hop:
> <sip:9722352287 at 192.168.10.19:5061;user=phone>
> Jul 5 09:08:30 VERBOSE[1133742896]: Transmitting (no NAT):
> SIP/2.0 100 Trying
>
> Via: SIP/2.0/UDP 192.168.10.19:5061
>
> From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1
>
> To: <sip:9722352288 at sipusers.com;user=phone>;tag=as6eeffd24
>
> Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com
>
> CSeq: 101 INVITE
>
> User-Agent: Asterisk PBX
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>
> Contact: <sip:9722352288 at 192.168.10.158>
>
> Content-Length: 0
>
>
>
>
> to 192.168.10.19:5061
> Jul 5 09:08:30 VERBOSE[1209277232]: -- Executing
> [1;36;40mDial[0;37;40m("[1;35;40mSIP/9722352287-32ef[0;37;40m",
> "[1;35;40mSIP/9722352288|30|r[0;37;40m") in new stack
> Jul 5 09:08:30 DEBUG[1209277232]: SIMPLE DIAL (NO URL)
> Jul 5 09:08:30 DEBUG[1209277232]: Setting NAT on RTP to 0
> Jul 5 09:08:30 DEBUG[1209277232]: Setting NAT on VRTP to 0
> Jul 5 09:08:30 DEBUG[1209277232]: Outgoing Call for 9722352288
> Jul 5 09:08:30 DEBUG[1209277232]: 9722352288 is not a local user
> Jul 5 09:08:30 VERBOSE[1209277232]: We're at 192.168.10.158 port 13818
> Jul 5 09:08:30 VERBOSE[1209277232]: Video is at 192.168.10.158 port 31668
> Jul 5 09:08:30 VERBOSE[1209277232]: Answering/Requesting with root
> capability 1
> Jul 5 09:08:30 VERBOSE[1209277232]: Answering with non-codec capability
> 0x1(G723)
> Jul 5 09:08:30 VERBOSE[1209277232]: 12 headers, 10 lines
> Jul 5 09:08:30 VERBOSE[1209277232]: Reliably Transmitting:
> INVITE sip:9722352288 at 192.168.10.20:5061 SIP/2.0
>
> Via: SIP/2.0/UDP 192.168.10.158:5060;branch=z9hG4bK339687e8
>
> From: "9722352287" <sip:9722352287 at 192.168.10.158>;tag=as50ef7b7a
>
> To: <sip:9722352288 at 192.168.10.20:5061>
>
> Contact: <sip:9722352287 at 192.168.10.158>
>
> Call-ID: 4df028c320569b3b6c7a62b875bb35f8 at 192.168.10.158
>
> CSeq: 102 INVITE
>
> User-Agent: Asterisk PBX
>
> Date: Mon, 05 Jul 2004 14:08:30 GMT
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>
> Content-Type: application/sdp
>
> Content-Length: 218
>
>
>
> v=0
>
> o=root 9587 9587 IN IP4 192.168.10.158
>
> s=session
>
> c=IN IP4 192.168.10.158
>
> t=0 0
>
> m=audio 13818 RTP/AVP 4 101
>
> a=rtpmap:4 G723/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=silenceSupp:off - - - -
>
> (no NAT) to 192.168.10.20:5061
> Jul 5 09:08:30 VERBOSE[1209277232]: -- Called 9722352288
> Jul 5 09:08:30 VERBOSE[1209277232]: Transmitting (no NAT):
> SIP/2.0 180 Ringing
>
> Via: SIP/2.0/UDP 192.168.10.19:5061
>
> From: <sip:9722352287 at sipusers.com;user=phone>;tag=20f8c-176cb1
>
> To: <sip:9722352288 at sipusers.com;user=phone>;tag=as6eeffd24
>
> Call-ID: 50fab160-40794-c0a81dd8 at sipusers.com
>
> CSeq: 101 INVITE
>
> User-Agent: Asterisk PBX
>
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>
> Contact: <sip:9722352288 at 192.168.10.158>
>
> Content-Length: 0
>
>
>
>
> to 192.168.10.19:5061
> Jul 5 09:08:30 VERBOSE[1133742896]:
>
> Sip read:
> REGISTER sip:sipusers.com SIP/2.0
>
> Call-ID: 5cd8-40a18-c0a81dd8 at sipusers.com
>
> From: <sip:9722352287 at sipusers.com;user=phone>;tag=21210-1b31
>
> To: <sip:9722352287 at sipusers.com;user=phone>
>
> CSeq: 222 REGISTER
>
> Via: SIP/2.0/UDP 192.168.10.19:5061
>
> Contact: <sip:9722352287 at 192.168.10.19:5061;user=phone>
>
> Max-Forwards: 70
>
> Expires: 75
>
> Supported: timer
>
> Authorization: Digest
> username="9722352287",realm="asterisk",uri="sip:sipusers.com",response="d062dc737803fcf362f74516a3db6f78",nonce="127dd9f4"
>
> Content-Length: 0
> ==================================================
>
>
>
> --__--__--
>
> _______________________________________________
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>
>
> End of Asterisk-Dev Digest
>
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