similar to: Re-INVITE on BYE

Hello List, I have been trying to configure a sip profile ( peer / friend ) for each of my customers behind a sip proxy for some time, but I have had no success, so I would appreciate your help. Customer -> OpenSIPS -> Asterisk -> PSTN The opensips is working as a sip proxy with record route, for billing, load balancing and authentication purposes. I would like to be able to define

On Tue, Apr 9, 2013 at 1:22 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>wrote: > ** > Hi Nick, > > The BYE is not properly formed and rejected by script - in the 200 OK of > the INVITE, you can see that your opensips is doing Record-Routing, but the > BYE does not contain the corresponding Route hdr, so SIP routing is > impossible. > > Regards, > >

Hi list, I am still a newbie and struggling with tweaking the dial plan to my requirements. I have tried googling for this specific problem, and apologies if I have overlooked the obvious answer already. If you could please be so kind as to point me in the right direction, that would be most appreciated. What I am trying to do, is get rid of the initial "+" in phone numbers coming in

Hello fellows VOIPer, If you want to share with the rest of the VoIP & RTC community some news, interesting or breaking through ideas, or even more, some experience you had in terms of designing, integrating or operating various solutions or platform based on Open Source Softwares, then you should consider submitting a paper for the OpenSIPS Summit 2020 in May, Amsterdam.

I currently have OpenSIPS set up with users and most of my call handling. OpenSIPS won't be able to handle things like Call Park, Hunt Groups, ACD, etc. So I want to send these types of requests to Asterisk. I also want to set Asterisk up as Multi Tenant. So my question is How can I send requests to Asterisk and have them funnel into the specific context for that specific Tenant? So if

Hi all how to send SIP HOLD Invite from asterisk to other sip client/server.? Thanks Nikhil

Hello, Which one of these ensures that SIP packets are sent and received in a secure format so that users using public wifi don't allow MITM type of attacks or others can't read the plaintext SIP packet info. VPN is not an option. Looking for 2nd most secure to VPN. P.S. Are both options part of the configs of Asterisk or need modules to be selected and installed before doing the

Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

Hi there noticed a strange thing in asterisk 1.6.2x 1.6.1x after one of the clients sends bye asterisk first sends invite to other side then after 200 ok it sends bye I am not sure but that could be some missconfiguration issue or a bug? so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK

Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with

(Probably OT) Has anyone else noticed these flaky (new?) behaviors in Evo since the 5.2 upgrade: - REALLY slow saving messages from inbox to another folder - search capability separated by folder instead of overall (with no option for control) - failure to autocomplete email addresses for known contacts I think that's it. I was wondering if it was just me, or if others had noticed this.

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hello Everyone! We wanted to let everyone coming to Astricon know that we will be holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast Casino & Spa. Suncoast is about 10 minutes away from Red Rock and we will be provide shuttle service to and from the Summit. For those of you that had to book at Suncoast it should be really easy to find us! Here are some things you can

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2.

Hello, I have 2 servers I'm synchronizing using rsync, I have a situation where I : 1. rsync from rnd-dev2 to rnd-dev1 2. change the rsynched file on rnd-dev1 3. rsync from rnd-dev2 to rnd-dev1 again 4. File gets overridden on rnd-dev1 over though it has newer change time then file on rnd-dev2. here is the bug(?) reproduction: [root@rnd-dev1 test_rsync]# rsync --version rsync version

Hi list i setup successfull asterisk version 1.4 + opensips, Opensips is the Registrar Server, Asterisk is the IVR server the call flow IP phone ---INVITE 1001----> opensips -----> ASterisk ----INVITE 5001--->opensips ---> Busy|cancel|404..--->asterisk---wait 10s to bye --->IP phone (5000) my case is: 1/ IP phone(5000) --->Opensips 2/ IVR number : 1001 3/ IP

Benchmark shows small performance improvement on fusion io device. Before: seq-read : io=1,024MB, bw=19,982KB/s, iops=39,964, runt= 52475msec seq-write: io=1,024MB, bw=20,321KB/s, iops=40,641, runt= 51601msec rnd-read : io=1,024MB, bw=15,404KB/s, iops=30,808, runt= 68070msec rnd-write: io=1,024MB, bw=14,776KB/s, iops=29,552, runt= 70963msec After: seq-read : io=1,024MB, bw=20,343KB/s,