[Asterisk-Users] Sip client
Hello! If I want to build a Sip client application in Java . What kind of Java Api would I use to connect to the server and to implement the sip signaling? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager Api
Hi One question for you! Which operation are allows by Asterisk Manager API? Can I connect to Asterisk server, create a new Channel,add channel on Asterisk server,specify a Voip protocol like SIP and generate Sip signaling? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Manager Api
Hi One question for you! Which operation are allows by Asterisk Manager API? Can I connect to Asterisk server, create a new Channel,add channel on Asterisk server,specify a Voip protocol like SIP and generate Sip signaling? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-api
Hi I try to create a sip client with asterisk-api package, I've a question: I can create a channel sip that generate sip signaling with Class Channel or with another class ? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] gnugk
Hi
I've a problem with a gnugkv2.0.7
I've compiled gnugk successfully
I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully
When i run gnugk i have this error:
error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot
open shared object file No such file or directory
I try to use command export:
export LD_LIBRARY_PATH=${HOME}/openh323/lib:${HOME}/pwlib/lib
in directory where i have this libraries
Have you suggestions?
Thanks
Ale
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[Asterisk-Users] H.323
Hi I've a problem with the H.323 calls The Asterisk's version is 1.0.7 The version of pwlib library is 1.6.6 The openh323 library is 1.13.5 Oh323 driver channel's version 0.6.6 I use a gatekeeper openh323gk registered successfuly to asterisk when i do a call from client h323 connected to gatekeeper to sip client I've this signalling: Asteriskclient h.323=ohphone+phonejack setup Q.931 -- Ack Call proceeding Q.931 -- Ack -- At this point I must have a RAS message ARQ(Admission request) from Asterisk to gatekeeper,but I've Release complete Q.931 Have you suggestions? I think the problem is with libraries! Another question how can debug h323 channel? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:how do I register my Asterisk with oh323 on gatekeeper?
Hi My oh323.conf is: listenaddress=0.0.0.0 listenport=1720 tcpStart=1 tcpend=2 udpStart=1 udpEnd=2 faststart=no h245Tunnelling=no h245inSetup=no gatekeeper=DISCOVER accountCode=H323 context= (ex.default) alias= (ex.gw) Codec= (ex.G711A) Frame= (ex.20) Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk-h323
Hi
I've this plattaform
sip sjphone - - - asterisk- - -gatekeeper- - -ohphone- - -phonejack card-
- analog phone
Asterisk is registered with Gatekeeper
Ohphone is registered with Gatekeeper
Phonejack is installed successfully and get a dial tone
When i try to call phonejack with sip phone i have this message :
call with mysjphone ('alias/IP' of asterisk) completed duration 0:00
and analog phone don't ring
when i try to call mysjphone with analog phone i've this message:
Speed Dial 3231 not defined ,trying gatekeeper..
phonejack is calling host 3231
'alias' 'ip' of asterisk is busy duration 0:01
Have you suggestions?
Thanks Ale
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[Asterisk-Users] problem with H323:Gatekeeper could not find user
Hi I've an Openh323 gatekeeper and on the same pc phonejack PCI card and Ohphone. The Gatekeeper is registrered with Asterisk The Ohphone is registered with gatekeeper when i do the following command: ohphone -g IP -q /dev/phone0 --callerid phonejack -l The analog phone connected to phonejack get dial tone The problem is: when i call with sip phone the h323 client I've this message H.323 call 'ip$localhost/19931' cleared reson 11 (gatekeeper could not find user) in extensions.conf : exten=5552,1,Dial(OH323/phonejack) Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:oh323 compile error
Hi Try the step descibed at this link: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg86875.html and make attention to edit correctly Makefile. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323
Hi I've installed successfully: - PWlib v1.6.7 library -Openh323 v1.13.5 library -asterisk-oh323 v0.6.5 and so the modules chan_oh323 is installed successfully Now I try to install chan_h323 First question: is this necessary? I edit the Makefile in the directory /usr/src/asterisk-1.0.7/channels/h323 to point to the right includes directories I do makeand I've the following error: make:***[ast_h323.o] Error 1 Have you some suggestions? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phonejack PCI-card
Hi I am using Phonejack PCI card connected to analog phone. I've installed this card succesfully but i get no dial tone. Have you suggestions? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323
Hi I've successfully installed Asterisk-1.0.7, I've successfully installed Openh323 gatekeeper but not registered to Asterisk and so I've install PWlib v1.5.2 and Openh323 v1.12.2 libraries Now i try to install asterisk-oh323-0.5.10 : -edit Makefile inside the asterisk-oh323-0.5.10 directory and set the paths optionsaccording to system - Then i do make to build the oh323wrap library and the ASTERISK OH323 channel driver but i've this error: -I/usr/src/asterisk-1.0.7/include -I.../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:260: '__use_AST_MUTEX_DEFINE_STATIC_rather_then_AST_MUTEX_INITIALIZER..' :Undeclerad here (not in function) ... make:***[chan_oh323.o] Error 1 make :Leaving Directory /lib/asterisk-oh323-0.5.1/asterisk-driver make :***[Subdirs_all] Error 1 Have suggestions? Thanks Ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-h.323
Hi I've a problem with the registration of the openh323gatekeeper. First I've downloaded and installed the pwlib and openh323 libraries successfully. Then I've downloaded the package openh323gk.tar.gz,executed the binary file, but the gatekeeper is not registred on asterisk! Then I've also downloaded and installed the pwlib and openh323 over the Asterisk's pc, and launch the make command in the directory /../asterisk/channels/h323, as suggested by README file, in order to compile h323. I've several compilation errors related on ast_h323.o. Can someone help me about it?Are the installation steps correct? Thanks for all ale ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bottlenecks
Hi I must to estimate the* performance. I am try to understand which can be the eventual bottlenecks. Have you some suggestion? Can you to signal to me some problems? Thanks Alessandra ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk
I?m a telecommunication engineering student. I?m working on my degree thesis, it?s about Astrerisk . My goal is to estimate the performance of a hybrid platform for the Volp. I?m looking for documentation about: ? Architecture ? Tools for the performances? analysis (to analyse performances) ? Informations about the scheduler ? Informations about the transcoding, to understand how the Volp Protocol (Sip,H.323,IAX) interact If you can help me, please, send me some informations to understand how to start to analyse performances; what I found on internet is not enough!!! Thank you so mutch ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
