similar to: INVITE/REFER with only 2 ends on asterisk

https://bugzilla.mindrot.org/show_bug.cgi?id=1489 Summary: ssh should normalize IP addresses before comparison Classification: Unclassified Product: Portable OpenSSH Version: 5.0p1 Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 because of strange reasons my notebook sometimes crashes short after startup (but that's not ext3's fault, maybe mem?, when i wait several minutes it works without problems) the problem is that after 3 crashes at startup, when my notebook finally worked i got the msg: Sep 23 23:29:17 blackbox kernel: EXT3-fs warning (device ide0(3,3)):

I've set up an architecture in which OpenSER acts as a registrar and load balancing server for Asterisk machines. I currently have only one Asterisk machine serving as a Media Gateway. My problem is that when A calls B, and then A makes a blind transfer to C, everything works: REFER goes to Asterisk, which processes it, replies with 202 Accepted, and then it generates a valid INVITE to C.

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:3254102@216.186.142.203 SIP/2.0. Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1. From:

I just realised my problem seems to be related to bug 0003710 - "0003710: [patch] Consultative transfers between asterisk servers". It's unclear from the bug info if this problem has been resolved yet. Anyone know? Doug. > -----Original Message----- > From: Douglas Garstang > Sent: Monday, March 27, 2006 4:41 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Asterisk 1.4.12 I wish to extract some custom headers from a SIP REFER message but am unable to do so. However I can extract them from an INVITE. The code is: exten => _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ; exten => _.,n,Set(custom-valid=${SIP_HEADER(custom-valid)}) ; Examples of the INVITE (works) and REFER (doesn't) messages are below. U 147.202.001.001:5060 ->

Hello, I'm currently using OpenSER as REGISTER server and Asterisk for the call routing. Do i need the OpenSER nathelper module if i want to offer (mostly) automatic NAT traversal to my users or does Asterisk have the same functionality? It seems that the nathelper module should be able to automatically traverse any NAT as long as the User-Agents use symmetric RTP. Further it is possible (in

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

Sorry Rowland, didn't read that part. Yes, the 'Domain Admins' group has the gidNumber attribute the value "512", and 'BUILTIN\Server Operators' value "549". Regards, Pablo Sanz Fern?ndez -----Mensaje original----- On 11/05/2020 11:09, Pablo Sanz Fern?ndez wrote: > Hi Rowland. > > It's CentOS 6.10 with Python 2.6.6. > > I guess then we

I can get this game to install and launch, but it does not run properly or at all. I have: Ubuntu Jaunty Jackalope AMD64 Alternate Wine 1.1.18 Cedega 7.1.1 Sid Meier?s Alpha Centauri and Alien Crossfire CDs (Sold Out version) The game is best played with the expansion pack, Alien Crossfire. Alien Crossfire and Alpha Centauri are both available in linux versions, if you can find them. There

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hi, Hello, I have been investigating and I am afraid that our case is the same as this one: https://lists.samba.org/archive/samba/2017-September/210724.html As you said, we have a problem with the gidNumber inherited from a migration from samba 3.x NT4 to samba 4.x AD. I have followed your prompts, removing the gidNumber from all AD 'BUILTIN' groups, in addition to the

Hi, We have solved the problem and the command 'samba-tool ntacl sysvolreset' is working correctly again. We have been able to reset the SYSVOL permissions and the AD GPOs are working again. The problem is that if we have the audit options active in the smb.conf, that command stops working. We don't know why. If we temporarily remove them if it works. I know that we have an old

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on

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 found out that Syslinux has a problem with Asus P4B Board. Even if I use the -s compability option i get always 'Boot failed'. Do you plan to fix this? If you need further information or a tester for this problem please contact me. Greetings Markus Tavenrath

Hello all, Very recently, djm added compability patch so that aes/rijndael encryption problems could be avoided when talking to broken server/client; and you wouldn't have to toggle off the protocols yourself. Might this be a candidate for 2.5.2p2 or the like? This would be helpful when there are a lot of broken, 2.3.0 and like, systems. -- Pekka Savola "Tell me of

> JC> After doing a lot of automated seek testing on files, it looks like > JC> the length of the sync code (currently 9 bits) is not long enough to > JC> enable a really robust AND efficient seek algorithm. Lengthening the > JC> sync code will require a format change (i.e. FLAC <0.8 streams won't > JC> play back on FLAC >>0.9 decoders), so I'm