search for: 5068

I have a host 192.168.1.3 that wants to run SIP on 5068 (long story). My host is 192.168.10.201. My host needs to stay on 5060 because of all the other devices I have connected. I tried putting port=5068 in my SIP extension definition but that did not work. So I thought about using iptables to accomplish this: iptables -t nat -A PREROUTING -p tcp --...

Hi all, I am trying to get iptables to work for me... I am running asterisk (11.23.0) on a C5 machine. Working fine on port 5060 udp. I have need to tcpenable=yes SIP and run that on port 5068. Since port 5060 is already running I was going to redirect 5068 to 5060. So I thought I could use iptables to do that - but does not seem to be working. 192.168.10.201 is my machine, 192.168.1.3 is the other machine. 1.3 should connect to 10.201 on port 5068. so I did: iptables -t nat -A PREROU...

...invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk replied to all the INVITE's it received before it says Ringing. Really need help on this badly, anyone has an idea. Thank you in advance. Regards Ron U 172.16.0.6:5068 -> 172.16.0.1:5060 INVITE sip:1234567 at 172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: ""<sip:unknown at 172.16.0.1>;tag=57677d009236 ed33..To: <sip:1234567 at 172.16.0.1>..Contact: <sip:172.16.0.6:5068>..Supported...

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP tcp on port 5068? telnet localhost 5068 Trying 127.0.0.1... telnet: connect to address 127.0.0.1: Connection refused telnet:...

...NVITE which is anwered with a 401. There should follow a new INVITE with a nonce, but this does not happen. Any idea why ? Is it the Grandstream IP-phone ?? <--- SIP read from TLS:my.pub.lic.ip:53416 ---> INVITE sip:0123123123 at ast.ser.ver.ip:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias From: <sip:testacc77005 at ast.ser.ver.ip:5061>;tag=263162018 To: <sip:0123123123 at ast.ser.ver.ip:5061> Call-ID: 1695864968-5068-8 at BJC.BGI.B.BAE CSeq: 50 INVITE Contact: <sips:testacc77005 at 192.168.1.104:5068;transport=tls> X-Grandstream-P...

...| | | > | |Via: SIP/2.0/UDP | > | | | > | |10.0.0.123:5068;branch=z9hG4bKfe06f452-2dd6-db11-6d02-000b7d0dc672;rport| > | | | > | |User-Agent: NGS/2.0 | > | |...

...| | | > | |Via: SIP/2.0/UDP | > | | | > | |10.0.0.123:5068;branch=z9hG4bKfe06f452-2dd6-db11-6d02-000b7d0dc672;rport| > | | | > | |User-Agent: NGS/2.0 | > | |...

Hi I am trying to call from Linphone on extention 109 to Xlite on extention 108 and I get this error ---------------------- to 216.75.167.18:5068 WARNING[98315]: File pbx.c, Line 1133 (pbx_extension_helper): No application 'Dial ' for extension (sip, 108, 1) == Spawn extension (sip, 108, 1) exited non-zero on 'SIP/sergeXlite-be43' --------------------- Can you tell me what might be wrong with my setup? Thanks Serge __...

...dencies of module mptscsih. Is modules.dep up to date? Cannot determine dependencies of module qla2300. Is modules.dep up to date? Cannot determine dependencies of module reiserfs. Is modules.dep up to date? Driver modules: none Filesystem modules: Including: klibc initramfs udev fsck.reiserfs 5068 blocks What do i wrong? Stephan _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

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See atttachment. I have already submit this bug via "report a bug" in Nov last year, but no reply, and the problem still exist in stable-6 now. Any one who knows how to solve this problem ? Thanks... -------------- next part -------------- <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">

Hi, when i use the *8 for the call pickup the call i fetch is directly connected and i can't see the callers number. What i want is that the call in the first rings at my phone and in the second i can see the callers number. I am using a polycom 500 ip phone. Is this a special polycom problem? Regards, Kib

...Port Status 2011/2011 10.1.1.10 5071 UNREACHABLE 2010/2010 10.1.1.10 5070 UNREACHABLE 2009/2009 10.1.1.10 5069 UNREACHABLE 2008/2008 10.1.1.10 5068 UNREACHABLE 2007/2007 10.1.1.10 5067 UNREACHABLE 2006/2006 10.1.1.10 5066 UNREACHABLE 2005/2005 10.1.1.10 5065 UNREACHABLE 2004/2004 10.1.1.10...

...member => SIP/1070@peername It works OK. But, after restaring I see in show queue that Members: SIP/1070@peername (Invalid) ... What does it mean? Why is it Invalid? BTW, reload command fixes it, so the member receives queue calls. Thanks! PS. 1.2.9.1 -- DSS5-RIPE DSS-RIPN 2:550/5068@fidonet 2:550/5069@fidonet xmpp:dsh@vlink.ru mailto:dsh@vlink.ru http://neva.vlink.ru/~dsh/

We have a script which executes "asterisk -n -r -x ....." periodically against the running server, to check the status of a few things, and pipe the output to a file. With prior versions of Asterisk this worked fine, but having just upgraded to 1.2.9, we are finding that if the output is lengthy, then Asterisk seems to terminate before fully flushing stdout. Is this a known bug, is

...chan_sip.c: Insufficient information for SDP (m = '', c = '') Jun 26 16:59:01 WARNING[62792] chan_sip.c: Insufficient information for SDP (m = '', c = '') And it seems that at this time I can't hear my peer correspondent. Thanks! -- DSS5-RIPE DSS-RIPN 2:550/5068@fidonet 2:550/5069@fidonet xmpp:dsh@vlink.ru mailto:dsh@vlink.ru http://neva.vlink.ru/~dsh/

...types for 'OffsetRect' winuser.h:4894: error: conflicting types for 'PtInRect' winuser.h:4963: error: conflicting types for 'SetCursor' winuser.h:4994: error: conflicting types for 'SetRect' winuser.h:5036: error: conflicting types for 'ShowCursor' winuser.h:5068: error: conflicting types for 'UnionRect'

I'm sorry if it is off-topic, maybe not. I have CentOS 6, webmin/virtualmin panel and I just install Horde 3/imp 4. In my virtualmin panel I have many email configured, but when I acces Horde and I try to login with email/password or user/password, Horde tell me login or password not good. What I have to do to synchronize Horde and virtualmin mailbox? Thanks you Ernesto

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Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXXXXXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and...