similar to: asterisk not sending ACK after reinvite

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

I'm taking the liberty to announce this event on the Asterisk mailing list, as Asterisk and OpenSER form a valuable combination in SIP architectures. The second edition of OpenSER Summit will take place in San Jose, USA ,on the 17th of March, 2008, during VonX Spring 2008 pre-conference events. This is the first US edition of the OpenSER Summit - to learn more about the agenda and layout of

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

Trying to find a solution to a sticky problem here. We have 3 OpenSER systems. Phones register with the OpenSER systems, and after they authenticate the user, pass the registration info using OpenSER's send() command to all Asterisk boxes sitting behind them. Each asterisk system then knows about every phone. For this to work, I had to turn off authentication in Asterisk for both

Hi, Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. Recently I have been getting errors from Asterisk due to corrupted From: headers, which appear to be caused by uac_replace. Here is a section of the debug log: Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: removing <From: <sip:lenc_domain.com@sip.domain.com>;tag=635c3ce6 > Mar

Hi, does anybody know if asterisk can authenticate on a NOTIFY send to a peer. I use OpenSER as SIP-Proxy and asterisk as voicemail system with ODBC Support for voicemail-messages, voicemail-users and sip-peers/users. My SIP-Users register with OpenSER. Asterisk has two views on the OpenSER database for voicemail-users and sip-peers. Call-Routing, leaving voicemail-messages in database an MWI

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER

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

Sorry if this hit the list twice, sent out yesterday, but didn't see it show up. Hi All, Can Asterisk be used as a SIP proxy, blah, blah, blah??? I've glanced over questions like this through the years, with a good idea on what a SIP proxy is and what Asterisk is and IS NOT. I never really took the time to lab-up SER and test drive it to see what advantages might be gained from using

Suggestions on how to use Asterisk to collect CDRs from a OpenSER-based proxy / call routing setup? I need to get simple CDRs; not for detailed settlement/rating, but just for reconciliation with an ultimate TDM carrier just to make sure we only get billed for what we're actually using. I'd use the often-heralded approach of dumping a call from OpenSER into Asterisk and having it

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC ----> openSER - - - ->

Hi! Meanwhile also the location is fixed: it is happening at metalab (http://metalab.at/) - a place for geeks. Thus, we meet there at Thursday, 20.9.2007, 19:00 CEST (=local Vienna time). Metalab is located next to the city hall: http://metalab.at/wiki/Lage Metalab is no pub/restaurant. Thus, don't come hungry! Nevertheless liquid food (drinks) is available. We meet in the library (in

I've just finish setting up OPENSER with Asterisk 1.2.2 In OPENSER, i have set extension 400 to push to asterisk, which in turn run apps VoicemailMain() I noticed after the INVITE came to asterisk, it reply to OPENSER with " We're at 203.125.68.66 port 16520 ". Right after that , it will keep on " Retransmitting #1 (no NAT) to 203.125.68.66:5060: " , all the way until

Works for me.... Thanks, Hristo Benev -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alexandru Oniciuc Sent: Monday, May 17, 2010 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: new way of asterisk and kamailio(openser) realtime integration

Hi Users, I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER, After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk, In openser.cfg ........... is not hiiting the Asterisk server ............. ... any one help me ........ .... .... modparam("tm","fr_timer",6) modparam("tm","fr_inv_timer",24)

Hello Users , I Posted to mailing list, No one is replying My issues, My Issue is No Audio when Openser and Asterik integrated in Same LAN , When UAC are Behind the NAT, With out the Asterisk integration Behind the NAT is working Fine. SIP port and RTP ports are forwarded into router to OpenSER System only. openser.cfg listen=192.168.2.11 alias=sip.hyperion.com # Invite Section if ( method==

Hi, I need some help, getting to work asterisk MWI. I set up Asterisk as voicemail server for Openser as this tutorial shows : http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3 . My voicemail system is working but, I can't get to work the message waiting indicator. It doesn't seems to send the Asterisk any NOTIFY message to the Openser box. How can I

One of the above is frequently used to front-end Asterisk. I used OpenSER to front-end a farm of Asterisk servers and was very happy with it. The ability to take a box out of service or to route a specific DNIS to a box for testing rocks. Since OpenSER has died (I don't care about the politics/personalities/trademarks), Kamailio and OpenSIPS have risen from the ashes. What are you using?