search for: sipclient

...ly 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 -------------------------------- REGISTER sip:192.168.100.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.13;rport CSeq: 4399 REGISTER To: "sjphone2" <sip:sjphone2@192.168.100.3> Expires: 900 From: "sjphone2" <sip:sjphone2@192.168.100.3> Call-ID: 797316...

...logs on its condole that it just "-- Playing 'demo-congrats' (language 'en')", nothing else. On the other hand, kphone finishes their log with that: ===================================================================== ... res_search: NO result ! res_search: NO result ! SipClient: Sending to '127.0.0.1:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 127.0.0.1:13998 ERROR: Open Failed ** audioIn: openDevice Failed. CallAudio: Creating OSS->RTP Diverter dtmfsenderTimeout DspAudio: Broken pipe (b)(b)(b)(...

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 ( <https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+stru cture> https://wiki.asterisk.org/wiki/display/ ......

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

...ying 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 sip:500@192.168.0.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport CSeq: 6121 ACK To: <sip:500@192.168.0.3>;tag=as12aab0bf From: "ivan2" <sip:ivan2@192.168.0.3>;tag=7F6911ED Call-ID: 155660827@192.168.0.2 Content-Length:...

...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 sip:500@192.168.0.3 SIP/2.0 >Via: SIP/2.0/UDP 192.168.0.2;rport >CSeq: 6121 ACK >To: <sip:500@192.168.0.3>;tag=as12aab0bf >From: "ivan2" <sip:ivan2@192.168.0.3>;tag=7F6911ED >Call-ID: 155660827@...

...rver 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 me the lowdown on what I need to do to set this up in extensions.conf and sip.conf? I have my softphones working fine but the asterisk server doesn't appear to handle SIP URIs. Is it necessary to also modify a DNS record to make this...

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: SIP/2.0/UDP 213.2X...

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

...F1000 (WLAN settings, and sip.conf settings, just to go sure?) Would chances of a working setup increase with asterisk on the LAN (which would make those phones worthless for me...)? My sip.conf relevant parts are [sip505] mailbox=05 callerid=505 type=friend username=sip505 secret=abcd123 context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm allow=ulaw Thanks for all input, Anselm

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

Hi After going from AST_DATA (RES_DATA) to realtime with mysql-driver my console is jam'ed with SIP Seeding peers from Astdb '000b8201XXXX' at 000b8201XXXX@81.146.XX.XX:35273 for 120 I got arround 4000 sipclients registered at that server and all the sip-client re-register every 120 sec. so the console is totaly fill'ed with SIP Seeding messages. Is it posible to not see that messages ? Asterisk is started with safe_asterisk and console with asterisk -r /HH

Hello! Is it legal to use RedirectAction to redirect a call that is waiting in a queue? The idea is to have an external application manage a queue via manager API. The queue would merely collect calls and play moh. I've tryed this already but asterisk sends SIP/Forbidden to the channel in queue, after the channel has been redirected by RedirectAction, even though the response to

Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as "asterisk" on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,