similar to: kphone & Asterisk CVS HEAD: no audio

Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 -------------------------------- ACK

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

I am clearly doing something ridiculously wrong. Running Asterisk 0.7.2 on FreeBSD 5.1, I have SIP soft phones which are unable to register. They keep trying and then time out. With the sip debug on in Asterisk nothing is logged. Here is the trace from one of the phones (kphone): (192.168.100.13 is kphone, 192.168.100.3 is Asterisk) sipclient: sending: 21:47:45.454

Hello there I'd like to use sip users and peers realtime. I think I done all I need to get asterisk works fine in realtime: res_odbc.conf configuration. extconfig.conf sippeers => odbc,asterisk,sipclient sipusers => odbc,asterisk,sipclient sip.conf [general] rtcachefriends=yes The sipclient table as suggest in this article: SIP Realtime, MySQL table structure (

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username

Hai,   Im wondering, im missing the  _ldaps._tcp. INTERNAL.DOMAIN.TLD entries in my dns. Now, before the updates ( badlock ) etc. this wasnt notice i think. But now since im setting up that everything is doing ldaps i noticed this in my squid setup   ( squid mailing subject : [squid-users] ext_kerberos_ldap_group_acl problem )   My question is...   did someone resently setup a new AD

On 8/24/2016 11:00 AM, L.P.H. van Belle via samba wrote: > Hai, > > > > Im wondering, im missing the _ldaps._tcp. INTERNAL.DOMAIN.TLD entries in my dns. > > Now, before the updates ( badlock ) etc. this wasnt notice i think. > > But now since im setting up that everything is doing ldaps i noticed this in my squid setup > > > > ( squid mailing subject :

I would like to setup my Asterisk server to process an incoming SIP URI and redirect all requests to a specific context. Example: (1) using a sip phone I'd like to be able to call: sip:somedomain.com *or* sip:someone@somedomain.com (2) i'd like my asterisk server to answer the call and route it to the context=in-from-sipclient which would play thru some DP actions Can anyone give

Hi, I have * running on a SuSE Email Server II, which uses an older libc-version. Calls to res_ninit et al. from enum.c prevent asterisk from building on that system. Could that possibly be changed to use res_init and res_search in order to provide backward compatibility with these systems ? Thanks, Holger von Ameln

Hi I have some trouble getting asterisk to compile on my system. I get unresolved external symbol in enum.c et srv.c on res_ninit, res_nsearch and res_nquery. I've looked through my /usr/include/resolv.h file, and endeed I didn't found any declaration of such functions. It seems to be some res_init, res_search, etc.. but the declarations are the same the res_nxxx ones. Does asterisk

Hello Folks! in my sip-logs i see that asterisk uses the User-Agent ID "Asterisk PBX": SipClient: Received: 16:34:03.023 --------------------------------- BYE sip:102141@131.130.XXX.XXX:44343;transport=udp SIP/2.0 Max-Forwards: 10 Record-Route: <sip:213.2XX.XXX.XX8;ftag=as2eb3c466;lr=on> Via: SIP/2.0/UDP 213.2XX.XXX.XX8;branch=z9hG4bK539a.47e6e8a7.0 #this is SER Via:

Hello, dear Asterisk experts. When I run Asterisk (CVS HEAD version), I'm not able to play music anymore -- asterisk seems to capture sound device. Is it not a bug, but a feature? That's unlike stable (1.0.7 and 1.0.9) versions, when I can, say, run an IP telephone on the *same* machine and listen what Asterisk' autoattendant says. Now I can't do that, I need Asterisk and client

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

Dear friends, please excuse me if my question will be trivial. I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD I experienced just the same problem), and changed a bit sip.conf: [general] ; ... dtmfmode = inband disallow = all allow = ulaw allow = alaw allow = gsm run kphone, and call the 1235 extension. According to sample extensions.conf, Asterisk would

Hello all, When I try to compile the mod_webauth module on CentOS 5, the dependencies for Kerberos fail. I have the Kerberos libs installed, which is what I assume it's complaining about. Ideas? Is there an RPM missing? Here's some of what I found: [root at localhost webauth-3.5.4]# ./configure checking for gcc... gcc checking for C compiler default output file name... a.out

Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hi everyone, here is a new version of xcb-proto for you to enjoy. Highlights are lots of improvements especially to the xinput extension, support for RandR 1.5 and an automatic alignment checker. Changes: Christian Linhart (98): xinput: ChangeFeedbackControl: add missing pad xinput: SetDeviceModifierMapping: fix length of pad

Hello guys. Did anybody use voicemail ODBC storage feature? All "voicemessages" fields name are clear except for "dir", which is assigned to "/var/spool/asterisk/voicemail/default/1234/INBOX". Why do we need any directory when we store voicemessages in the database? Noreover, that field is choosen as the KEY or INDEX (marked as MUL in README.odbcsrorage, so I

Dear Asterisk experts and users. Let's assume I purchased the call number, let it be 2200 for simplicity. So when we call 2200, our call will be processed with Asterisk, and come in, say, [incoming] context. Well, then I'd like Asterisk to process all calls that a *begin* with 2200, i.e. 2200101. My dialplan would include something like: ;; extensions.conf [incoming] exten =>