similar to: SIP CANCEL problem

On Tue, 30 Aug 2016 10:39:14 -0400 Eric Wieling <ewieling at nyigc.com> wrote: > The dialplan below cannot go to voicemail, either something else is Of course not. It's the individual extensions that have voice mail. I have a similar problem when one of those destinations is a cell phone but I know that there is no Asterisk solution for that problem. If the cell phone answers and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all. I'm new to this so please forgive my ignorance if I missed something obvious. I've set-up asterisk 0.9_1 on a FreeBSD 4.10 server (I know it's not linux but that's all we have available at that stage). After some struggle to understand how everything works, I set up some SIP accounts for test purposes. I can log in,

Hi Sergio On 16.10.20 at 07:54 sergio wrote: > Sometimes, linphone shows missed calls as missed. Look like asterisk > replies with cause=487 that time, but I can't understand why. > > Grandstream always shows calls as missed ones. > > How should I investigate this? You could try to reproduce it while activating pcap traces and analyze it afterwards - or you could

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hi All, Have any way to asterisk forward the 487 Request Cancelled in SIP TO SIP call? In a SIP to SIP call when the called peer B send 487 to Asterik, Asterisk return to calling peer A 603 PEER A ASTERISK PEER B | INVITE ------------>| | |<------------TRYING| | |

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

Hello. Calls cancelled by caller during the dialing phase, are shown in Linphone as simply past calls, not missed ones. I thought this is an Linphone issue, but Sylvain says it's on my PBX side: https://github.com/BelledonneCommunications/linphone-android/issues/832#issuecomment-557474864 > The CANCEL message has a Reason header with Q.850 protocol and cause 0, which doesn't mean

On Mon, 29 May 2017 23:01:43 +0200 (CEST) Sébastien QUESSON via samba <samba at lists.samba.org> wrote: > samba-tool group addmembers domaingroup 'SAMDOM\user1' > ERROR(exception): Failed to add members "SAMDOM\user1" to group > "domaingroup" - Unable to find "SAMDOM\user1". Operation cancelled. > > samba-tool group addmembers

Hi everybody, got it from svn: dtmf_2833_1.pcap */asterisk/trunk/tests/rfc2833_dtmf_detect/configs/extensions.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/configs/sip.conf PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/run-test PRE-CREATION *>* /asterisk/trunk/tests/rfc2833_dtmf_detect/sipp/broken_dtmf.pcap UNKNOWN *>*

samba-tool group addmembers domaingroup 'SAMDOM\user1' ERROR(exception): Failed to add members "SAMDOM\user1" to group "domaingroup" - Unable to find "SAMDOM\user1". Operation cancelled. samba-tool group addmembers 'SAMDOM\domaingroup' 'SAMDOM\user1' ERROR(exception): Failed to add members "SAMDOM\user1" to group

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are

Hi Michal, By the way, I noticed ipv6 loopback IP addresses in your pcap. As I normally try to capture on nic where migration carried out, I thought of checking with you if your wireshark could dissect libvirt RPC in such pcap too (captured on a nic) ?. During migration, I do not see any traffic on loopback and I think it is expected, but thinking how you get those captured ?. Any

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample pcap/g711a.pcap file includes RTP from 10.1.3.1:5000 to 10.1.6.18:2006 1. How can you "forge" IPs

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

Thank you Michal. With your pcap, I could confirm that, libvirt dissector worked in my environment as well. Yes, it could be that, my pcap do not have libvirt rpc packets correctly though I would have expected. I am checking on it. Regards, Gowrishankar On Thursday 07 January 2016 03:51 PM, Michal Privoznik wrote: > On 07.01.2016 08:05, gowrishankar wrote: >> Hi Michal, >>

Hello Luca, We are still playing with visualization of your data, but I didn't want you to wait any longer for some results.  I think I blame both DT and the Pi :) First, a look at the phone side of your Banana Pi.  The first thing we noticed is there were a LOT more packets in one direction (north towards DT) than the other (towards the phone): jeff at

Hello: I have Samba 3.0.30 running on SCO Openserver 6. It seems to work fine, but I get this error in /var/adm/messages: Jul 17 08:15:03 smbd[5023]: [2013/07/17 08:15:03, 0] smbd/server.c:(527) Jul 17 08:15:03 smbd[5023]: open_sockets_smbd: accept: Protocol error Jul 17 08:16:22 smbd[5056]: [2013/07/17 08:16:22, 0] smbd/server.c:(527) Jul 17 08:16:22 smbd[5056]: open_sockets_smbd:

Thanks for the replies! On Thu, Nov 28, 2024, at 04:34, Rowland Penny via samba wrote: > I do not use the SMB3 Unix extensions, but perhaps you may not be > either, have you tried replacing 'server min protocol = SMB2' (which is > the default anyway) with 'server min protocol = SMB3' ? I took a packet capture and do see the the client making POSIX extension requests,

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):