similar to: app_conference

Any C gurus out there that can tell me if this code compiled ok to be used in x86_64 (Pentium Dual Core). It's for the app_conference application. Im using Centos 4.3 x86_64 kernel: 2.6.9-34.ELsmp libgcc-3.4.5-2 gcc-3.4.5-2 after the compilation part is the makefile ************begin compilation******************* [root@centos app_conference]# make clean rm -f *.so *.o app_conference.o

Hi all, I've been trying to get meetme working for a while now (complie problems - will probably try again later on another machine) but have given up and started looking at alternatives. I've managed to get app_conference compiled and installed - show modules shows its there in asterisk, but I don't know how too actually use it in the dial plan... The info on voip-info

As stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MeetMe A Meetme room uses Ulaw as the audio codec, so if the other channels use different codecs, then * will transcode. Does the app_conference application works the same way? Or if i have SIP/g729 users and i create a conference with other users also at g729 asterisk will not transcode (when using app_conference)?

I have Asterisk running in Xen virtual machines. Unfortunately, this kind of virtualization makes a real time clock impossible, which in turn makes ztdummy or a Zaptel driver impossible to load, which also makes MeetMe conferences impossible. As an alternative, I have downloaded, patched, compiled and installed the app_conference source code against the headers in Asterisk CVS HEAD. I can load

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hi, i was successful in compiling app_conference and setting up an conference was quite easy. :-) Does anyone knows if it is possible to have an IVR accessable from inside the conference. So, if i dialed into an conference i want to be able to press '*' and then the actual discussion is muted for me and i and menu is read to me. Something like the ${MEETME_AGI_BACKGROUND} in MeetMe.

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. -------------- next part -------------- A non-text attachment was

jonathan blais wrote: > I'm using Linphone. I tested with Asterisk and Speex only, I created a > channel with echo and it worked. It seems to have problem when using > app_conference. If you just use app_echo, then asterisk won't be trying to decode your frames; it will just be sending them back to you. Therefore, if your client is using an incompatible packing of the

Jean-Marc Valin wrote: >Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, >Linphone just sends raw packets, as specified in the RTP draft. > > Asterisk expects speex frames to have a terminator. The phone I was referring to was the X-Ten/X-Lite phones, which seemed to be adding something _before_ the speex data to indicate the length of the frames.

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

I'm using Linphone. I tested with Asterisk and Speex only, I created a channel with echo and it worked. It seems to have problem when using app_conference. Jonathan 2006/1/31, Steve Kann <stevek@stevek.com>: > > jonathan blais wrote: > > > Hi, > > > > Does anyone ever used Speex with app_conference in Asterisk ? I'm > > having a hard time to figure

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hi, Does anyone ever used Speex with app_conference in Asterisk ? I'm having a hard time to figure why I always get this error "warning: Invalid mode encountered: corrupted stream?". Jonathan Blais -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060131/386141a8/attachment.htm

Just curious, how does Asterisk pack Speex frames in a packet. AFAIK, Linphone just sends raw packets, as specified in the RTP draft. Jean-Marc Le mardi 31 janvier 2006 ? 10:43 -0500, Steve Kann a ?crit : > jonathan blais wrote: > > I'm using Linphone. I tested with Asterisk and Speex only, I created > > a channel with echo and it worked. It seems to have problem when >

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

Do anybody could compile app_conference on x86_64??? I tryied with two versions of app_conference and got the same problem on compiling: relocation R_X86_64_32 against `a local symbol' can not be used when making a shared recompile with -fPIC app_conference.o: could not read symbols: Bad value" ENVIRONMENT:

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for