similar to: SIP Error 401 Problem

Dear All, Sorry to be posting again. I have solved my problem. The problem is that when exiting from the macro, the priority number is still in effect. For example, priority 1 is at the start before entering macro after the macro the priorty will be 2. Since there isn't any other dialplan command, the switch statement would be search for a priority 2 in the Realtime extensions table. One

Dear All, I am currently have this problem in which I am sending call out from the Zaptel TE405 to a VoIP gateway. But the problem that the call over to the VoIP Gateway will always have a fake ring tone. Can you please give some pointer how to fix this problem? This problem is existing in my Asterisk 1.2.1 box. Also when compiling Asterisk 1.2.5 and tried to run it with the Asterisk-addon,

Dear All, I was wondering will there be any problems or changes that I will need to do to compile the current Asterisk(1.2.7)/Zaptel(1.2.5)/Libpri(1.2.2) source from www.asterisk.org into a 64-bit binaries? I am currently using the following hardware for my new server. CPU: Pentium D 930 3.0 GHz Mobo: Intel D945PSN Motherboard RAM: 512MB 533MHz DDR-2 Drive: SATA II Seagate 160GB Card: TE406

Dear All, It seems that there is a bug on the ooh323 while using registering with gatekeeper. The gatekeeper is GnuGK and the problem is when the Asterisk recieves a call from the Gatekeeper and routes it back out to an SIP Phone. The call would be connected and immediately dropped after 1-2 seconds connection time. This doesn't happen when ooh323 module isn't registered to a

Dear All, I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all. Asterisk Version is 1.2.1 Asterisk RTP Range is 10000 to 20000 UA Listen RTP Port is 15000 Below is the the

Hi All, I am having some problems with Asterisk on static IP and Sipura-1001 on dynamic IP. Is there any solutions to in the Asterisk configuration or Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. Regards, Kengie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I have a problem with the Marco and the Realtime Extensions in my extensions.conf. The problem is that when I exit from my Marco, I should return to my calling context, which is default but the next step for it should be switch statement which will use realtime extension. Somehow I am getting the following error below with autofallthrough=yes : -- Executing

Hi, I have already set canreinvite=no in the sip.conf and also used the NAT=yes. But the funny thing that was in one case the user call and it wasn't working (one way audio as described) using an online dialer and then tried again using X-lite it was working. Then hanged up and tried X-lite again, it was not working. The second call was only a few seconds apart. Moving back to the online

Dear All, I have a very strange situation here and wondering if anyone can assist me. I am trying to connect an H323 call from an GnuGK to Asterisk 1.2.1 which routes the call to an SIP Hard Phone. The funny thing that I can collect the connect but the call always drop about 1 second or 2 seconds after it is connect. I am not sure if this will help but I do see some 'Trapped RCF' in

Hi All, I am having some issues with Asterisk 1.8.3 extensions with a SIP Phone and an gateway. My setup is that I have my SIP Phone setup to register with the gateway. Then the gateway should sent calls to the Asterisk as a type of friend. This works fine if the SIP Phone configuration username and password isn't already set into the asterisk. The configuration of the SIP Phone username

the contexts are as follows, looking forward to your reply, thanks [1]enviromention : xen3.4.2+CentOS-5.5 [2]config file: xp.hvm, as follows _s_name = 'xp-101' _s_vnc_display = 5310 _s_vm_id = 101 _s_vm_mac = '00:16:3e:eb:ca:65' _s_memory = 512 _s_allow_destroy = False _s_use_vnc = True # -*- mode: python; -*-

Hello, unfortunately I got some trouble using ext3 - the system hangs when working in a certain directory. Not completely, I can switch between consoles and reboot with SysRQ, but cannot do anything which requires disk IO: not start any applications, not umount, nothing. I could send the strace output, but I don't think this would help anyone. Sorry, but I don't have a clue where to start

Hi, Use ?split() #dat1 is the dataset: lst1<- split(dat1,dat1$BaseProd) lst1 #$`2231` ?# BaseProd? CF OSA #1???? 2231 0.5 0.7 #2???? 2231 0.8 0.6 #3???? 2231 0.4 0.8 # #$`2232` ?# BaseProd CF OSA #4???? 2232? 1?? 2 #5???? 2232? 3?? 1 # #$`2233` ?# BaseProd? CF OSA #6???? 2233 0.9 0.5 #7???? 2233 0.7 0.5 #8???? 2233 4.0 5.0 #9???? 2233 5.0 7.0 lst1[[1]] #? BaseProd? CF OSA #1???? 2231 0.5 0.7

https://bugzilla.mindrot.org/show_bug.cgi?id=2232 Bug ID: 2232 Summary: curve25519-sha256 at libssh.org Signature Failures When 'ssh' Used with Dropbear, libssh Servers Product: Portable OpenSSH Version: 6.6p1 Hardware: All OS: All Status: NEW Severity: major

https://bugzilla.mindrot.org/show_bug.cgi?id=3639 Bug ID: 3639 Summary: server thread aborts during client login after receiving SSH2_MSG_KEXINIT Product: Portable OpenSSH Version: 9.2p1 Hardware: ARM OS: Linux Status: NEW Severity: critical Priority: P5 Component:

Greetings - In my logwatch report this morning I noticed reference to an attempt to connect to rsync from an external IP address. It doesn't appear that the connection was successful based on correlating information between /var/log/secure and /var/log/messages. But I am looking for some suggestions for implementing more preventative measures, if necessary. The log information from

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx

Hello. I've got a network of FreeBSD servers which traditionally hosted a classic domain. I upgraded some months ago, removing the old PDC and BDC and migrating to an AD DC controller in a jail. This is working fine with Samba 4.4.13. Now I'm trying to add a second DC, so I created a new jail on another physical server and went on with the setup, following: >

We've had another random imap process crash. This is with the original 1.2.5 imap (I haven't applied the patch for two processes creating an index simultaneously): > Oct 03 13:24:56 imap-login: Info: Login: user=<xxxxxxxx>, method=PLAIN, rip=134.225.1.46, lip=134.225.16.6 > Oct 03 13:25:59 IMAP 6067 xxxxxxxx 134.225.1.46 : Info: delete: uid=483, msgid=<xxxxxxxx> > Oct

mbox messages gets header corruption caused by an extra linefeed after Content-Length Users sees their mails in Sent mbox folder without the from and to fields, without attachments and with the date of 1/1/1970 Diego. --- Here is an anonymized header: >From xxxxxxxx at xxxxxx.xxxxxx.xxxxx.xx.xx Tue Jun 03 09:14:33 2008 Message-ID: <xxxxxxxx.xxxxxxx at xxxxxx.xxxxx.xx.xx> X-UID: 3913