similar to: Sipura SPA-3000 and PSTN dtmf

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hi, I attached the changes I made to a vanilla libguestfs-1.22.6 in order to make it work in mingw/win32. Added is also the patch required to make QEMU compatible (add a command to QMP that lists the supported devices (the regilat way you do it print it to stderr, which is difficult to redirect in win32)). This is done on behalf of Intel Corp. Thanks, Or (oberon in irc)

Although RAW sockets can be used when specifying the source address of packets (defeating one of the aspects of the jail) some people may find it usefull to use utilities like ping(8) or traceroute(8) from inside jails. Enclosed is a patch I have written which gives you the option of allowing prison-root to create raw sockets inside the prison, so

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hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

OK, I have removed the autoconf test for s6_addr32 (which is no longer used), and wrote a new test for IN6_IS_ADDR_V4MAPPED. The latter is probably portable, but since we're already in the business of testing, it does not hurt to do so. Arjen, I wonder about server/access.c, line 60-61: return (IN6_IS_ADDR_V4MAPPED(ip6) && ((((const u_int32_t *)ip6)[3] &

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

nmbd has been dying on me occasionally. I'm running mandrake 9.2 with samba3-server-3.0.0-2mdk. We've got our users in ldap but I'd seen this symptom previously when I was using an earlier version of samba that didn't support ldap. There is nothing useful in the logs and a PS shows a nmb process still running. This last time around, I had a ptrace running on both nmbd

On 7.3.2019 23.37, A. Schulze via dovecot wrote: > > Am 07.03.19 um 17:33 schrieb Aki Tuomi via dovecot: > >>> test-http-client-errors.c:2989: Assert failed: FALSE >>> connection timed out ................................................. : FAILED > Hello Aki, > >> Are you running with valgrind or on really slow system? > I'm not aware my buildsystem

In the current implementation, ssh always uses the hostname supplied by the user directly for the SSHFP DNS record lookup. This causes problems when using the domain search path, e.g. I have "search example.com" in my resolv.conf and then do a "ssh host", I will connect to host.example.com, but ssh will query the DNS for an SSHFP record of "host.", not

Hi Jean-Marc, I'm implementing a VoIP client and I want to use your codec init. I need to ask you some questions about Speex because I can't get too much information about it on the web. I hope you can help me. 1.- First of all, I want to know if Speex provides a buffer for Jitter..If it does, what can I do to use it in my softphone? 2.- How does Speex manage the packet loss? (In a

Hello, I work at SmarctJog.com, we have here some patches on IceCast for performance and reliability, these are mostly client/connection/source cleanups (a slave merge is underway, and some more good stuff (c)), but we'd like this to be merged in before the list gets any longer. Please find attached a list of our patches with a short desc: This one is actually not from us/me, it was found

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

Currently, o2net_accept_one() is allowed to accept a connection from listening node itself, such a fake connection will not be successfully established due to no handshake detected afterwards, and later end up with triggering connecting worker in a loop. We're going to fix this by treating such connection request as 'invalid', since we've got no chance of requesting connection

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

We just updated our configuratiosn to have multiple NIS servers, when we initiated a test of client failover, we were disapointed. It seemed that the only way to get a filaover was to /etc/init.d/ypbind restart. It behaves as indicated in http://bugs.opensolaris.org/bugdatabase/view_bug.do?bug_id=5084845 using ypbind-1.17.2-13 on Centos 4.5 / Linux xxxxxxxxxxxx 2.6.9-55.0.12.ELsmp #1 SMP Fri Nov

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

- perform dns lookup on hostname, - randomize local tunnel port - simple autobuild script - bump rpm spec version --- autobuild.sh | 41 +++++++++++++++++++++++++++++++++++++++++ main.c | 14 +++++--------- ovirt-viewer.spec | 9 ++++++++- tunnel.c | 29 ++++++++++++++++++++--------- 4 files changed, 74 insertions(+), 19 deletions(-) create mode 100755