search for: ua1

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user "ua1" and "33" are connected 5. Now when "ua1" presses...

...using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). SIP Software version: 1.6.3.0067 BootROM version: 2.6.2.0032 Reliably Transmitting (no NAT) to xxx.187.128.95:5060: NOTIFY sip:2944030@ua1.ipt.oneeighty.com SIP/2.0 Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport From: <sip:2944026@ua1.ipt.oneeighty.com>;tag=as6fd80d1b To: "Front Desk" <sip:2944030@ua1.ipt.oneeighty.com>;tag=3B576862-120A3007 Contact: <sip:2944026@xxx.187.142.203> Call-ID...

Hi All, I am stuck with an issue in the Openser+Asterisk Forking. In this solution we are using Openser as the Registrar. Hence it will 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 i...

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...

...ocurring for quite some time. We have about 35 phones and it's happening on most (also on the few running SIP software 1.6.6). > > SIP Software version: 1.6.3.0067 > BootROM version: 2.6.2.0032 > > Reliably Transmitting (no NAT) to xxx.187.128.95:5060: > NOTIFY sip:2944030@ua1.ipt.oneeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK4d777013;rport > From: <sip:2944026@ua1.ipt.oneeighty.com>;tag=as6fd80d1b > To: "Front Desk" <sip:2944030@ua1.ipt.oneeighty.com>;tag=3B576862-120A3007 > Contact: <sip:2944026@xxx.187...

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 ==&...

...ced in my testing.. When two or more Asterisk 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...

...UA-Local is able to hear UA-External, but UE-External 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...

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...

...and stops when the called peer answers, and the call is bridged successfully without problems. Im using asterisk 1.8.0 from sip.conf [Trunk-out] type=peer host=*.*.*.* context=from-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", &q...

...related agents) the queue application is running on that server and routed to correctly by it's peers. Enters DUNDi: Working scenario: 1) Configured 3 contexts, referenced by DUNDi, to manage which server is the primary, secondary, and tertiary server for each given queue. So: a. UA1, 2, and 3 register with Astbox1 as their primary server b. Their registration tables refer to Astbox2 as their secondary registration server and Astbox3 as their tertiary registration server c. Agents are logging into the queue1 via UA1, 2, and 3 d. Queue1's dial plan logi...

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...

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

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 recreat...

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 o...

...related agents) the queue application is running on that server and routed to correctly by it's peers. Enters DUNDi: Working scenario: 1) Configured 3 contexts, referenced by DUNDi, to manage which server is the primary, secondary, and tertiary server for each given queue. So: a. UA1, 2, and 3 register with Astbox1 as their primary server b. Their registration tables refer to Astbox2 as their secondary registration server and Astbox3 as their tertiary registration server c. Agents are logging into the queue1 via UA1, 2, and 3 d. Queue1's dial plan logi...

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.....

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

...load 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) |...