search for: ua2

Hi I have the following configuration | UA1 --|------ asterisk1 -----------------------+ UA2 --|------ asterisk2 -----------------------+ DB UA3 --|------ asterisk3 -----------------------+ UA4 --|------ asterisk4 -----------------------+ | All UA is located in the same area. A seperated PC is used as a centralized DB for storing a common dial plan, user account and register inf...

...ill store all the contact bindings along with the q values for a given user, say ua1. The current setup is such that the INVITEs are sent to Asterisk by Openser and Asterisk sends out the INVITE. Now if ua1 is registered with two different contacts having different q values and i make a call from ua2 to ua1. Openser will recieve the INVITE check for the multiple contacts of ua1 in the database. and send out an INVITE for the first contact. On recieving a 486 busy it sends out an INVITE to the second contact. This is where the problems lies. Openser is sending Asterisk the second INVITE but no...

Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Call from UA1 to Asterisk (UA2) to UA3 UA3 sends RTP before SIP OK to Asterisk (UA2) Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to UA1. Instead I would like it to just send on the early audio, is this possible? Thanks in advance, Richard

[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to UA1. Instead I would like it to just send on the early audio, is this possible? Thanks in advance, Richard

...servers are connected by IAX2 trunks it does not make use of any "shortest path" type system.. (maybe this is still planned somwhere down the line, but may come in handy to those who have multi asterisk installations) Here is the setup.. UA1--- Asterisk1----[IAX2 Trunk]---Asterisk2---UA2 | UA3 If I have a call between UA1 and UA2 and I transfer the call from UA2 to UA3 the result is that the voice path is like this.. UA1---Asterisk1---Asterisk2---Asterisk1---UA3 Asterisk does not work out that UA1 and UA3 are local and so does not create a sh...

Hi, We are using asterisk version 1.0.5. We have registered two UA's with asterisk. (Registration was successful) UA1 <-------> * <--------> UA2 Now, UA1 subscribes for UA2 to asterisk. asterisk sends NOTIFY to UA1 with UA2's state as open. But if UA2 gets un-registered then, asterisk is not sending NOTIFY to UA1. But when there is state change from UA2, asterisk is supposed to send NOTIFY to UA1 (keeping state as closed). Why is...

Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to xxx.yyy.142.139:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP xxx.yyy.142.139;branch=z9hG4bKd22096a5A22CE654;received=xxx.yyy.142.139 From: &qu...

Hi, I am trying to integrate asterisk with openser for a simple call. I am facing some issues with Asterisk. Below is the explanation: I have a UA1 sending invite to UA2 through Openser and Asterisk with the below sequence. Sequence is UA1->OpenSER->Asterisk->Openser->UA2 When Asterisk gets the INVITE, the INVITE contains two Via headers, one of the UA1 and the other Openser's. As Asterisk acts as a B2BUA, it recreats the Dialog....

Hi everybody, I am trying to make up call flow diagrams for for a setup which include ser as a sip proxy/registrar and asteriks as a voicemail server. Is my sequence correct?: UA 1 send an invite to SER. SER forwards this invite to UA2. UA2 sends back a sends back a 100 trying and 180 ringing message. SER forwards these. However UA2 doesnt answer the phone,so what happens then?...is there a timeout message?...I know SER sends a notify message to asterisk at some stage but im not sure of the exact sequence or if asterisk contacts...

I have this configuration: UA1 ---> FW1 ---> Asterisk ----> FW2 --> Internet --> UA2 UA has provate address (192.168.x.x) Asterisk has public address I want to be reach somebody at the internet. My idea was that asterisk works as a Proxy. Then i would have a SIP/RTP connection between UA1 and Asterisk and an other SIP/RTP connection between Asterisk and UA2. (asterisk is bridgei...

...print "Done\n"; my $filename=$res->content; return($filename); } # If defined, the return value is used for title streaming (metadata) sub ices_get_metadata { print "Getting meta data from \ http://tunes.3dcrm.co.uk/stream/meta.php\n"; my $ua2=new LWP::UserAgent; my $req2=HTTP::Request->new(GET => \ "http://tunes.3dcrm.co.uk/stream/meta.php"); print "Making request..."; my $res2=$ua2->simple_request($req2); if(!$res2->is_success) { print "Failed: "...

I've got the following situation where a UA is trying to call another UA via Asterisk and SER according to UA1 -> * -> SER -> UA2. Now in the event that SER generates a 404 Not Found for UA2 I would like Asterisk to return or relay or forward or whatever the 404 to UA1. Anyone know this might be able to be done (or maybe not possible at all?) Craig

Hello, everyone. I'm interested in one thing: as I know asterisk retranslates the media stream with the next way 1. Gets the frame with the UA1's codec 2. Retranslates it to slan 3. Ratranslates slan to UA2's codec 4. Send the frame It seems to me, that it follows these steps anyway, the question is: Will Asterisk retranslate the frame ua1->slin->au2, if the codecs of the 1-st user and the 2-nd are the same? I need him do not touch the frames, just retransmit them as is. -- Best Regards Al...

...rnal can't hear UA-Local. It all works perfectly fine, if UA-External were to call UA-Local. Then I get full two-way media. The problem is only when Asterisk calls out a non-locally subscribed user. Brief Setup Background: ---------------------- UA1 at mydomain.com: user subscribed in sip.conf UA2 at mydomain.com: user subscribed in sip.conf UAE at external.com: some user actively registered with some domain external.com. I am using OpenSER as my external proxy for external.com and I have my DNS setup all right. Following scenario is working fine in my setup: UA1 <---> NAT <---&g...

Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the "Request: INVITE mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy

Hi, It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem.. My setup.. UA1--[AST1]--{IAX}--[AST2]--UA2 | | PSTN1 PSTN2 I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent.. I have wildcard extensions that define which PSTN line to use when dialing out.. For example I have the follo...

Hi, I am designing a asterisk load balancing model as follow. There are 3 asterisks connected to a single DB and a single server storing all the configuration file and voicemail. Round Robin DNS will distribute the request to asterisks. DNS round robin ---+ asterisk1--------------------------+ DB and file server +---asterisk2-----------------------+

....berlios.de/projects/b2bua/ And I also download them(two components) and try to test it. But I have not enough knowledge about asterisk. It seems a Software PBX. Does asterisk_b2bua work? Does anybody ever try it? I have questions about my scenario. |======================> UA2 (Internet) | UA1 ===> SER ===> Asterisk B2BUA ===> Trunking A (PSTN) | | CDR + Prepaid + Handle Calls(Tear-down when call during limited)...

Dear ALL: My scenario lists below: Assume: UA1 with sip id "1011" And dial number to PSTN is "0939749xxx" There is no modification rule at my CISCO. (It will not change any dialed number) UA1 ==> SER ==> UA2 (SIP to SIP) UA1 ==> SER ==> Asterisk ==> CISCO 5300 ==> PSTN (SIP to PSTN) port:5060 port:5065 port:5060 IP:xxx.xxx.190.248 IP:xxx.xxx.190.243 (On the same server) (On another se...

...om-Trunk-out insecure=port,invite qualify=yes disallow=all allow=ulaw directmedia=no canreinvite=no dtmfmode=rfc2833 [ua1] type=friend secret=* host=dynamic nat=yes qualify=60000 disallow=all allow=ulaw allow=gsm allow=ilbc allow=alaw qualifyfreq=9 context=sswtrunks directmedia=no canreinvite=no [ua2] ... == Using SIP RTP CoS mark 5 -- Executing [***********@sswtrunks:1] Dial("SIP/ua1-00000776", "SIP/Trunk-out/***************,180,tTr") in new stack == Using SIP RTP CoS mark 5 -- Called Trunk-out/****************** -- SIP/Trunk-out-00000777 is ringing -...