similar to: Grandstream GXP2160 + SRTP

Hello, I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really destroying SIP dialog '78b0d1701d3694b1494a0c4b55344d57 at ip-sip-server:5060' Method:

Hello: I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls. Thanks in advance. These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN These are my

Hello, If the phone of my colleague rings, I can see this with BLF-lamps on my Snom IP-phone. I would also like to see *_who_* is calling. I would like to see the external number on my screen so I can choose whether to pickup the call with BLF. Therefore I have in sip.conf : notifycid = yes With this setting on, I see on my screen : 10 --> 10 10 is the internal extension of my colleague.

Background, Client - realtek rtl8111c tftpd version is 5.0 options on use -l -v Client: PXE-EX32 TFTP Open Timeout Server: Jun 12 10:48:38 damar in.tftpd[30132]: RRQ from 192.168.1.107 filename gpxelinux.0 Jun 12 10:48:48 damar in.tftpd[30133]: RRQ from 192.168.1.107 filename gpxelinux.0 Jun 12 10:49:24 damar in.tftpd[30134]: RRQ from 192.168.1.107 filename gpxelinux.0 Jun 12 10:50:36 damar

Using AMP, the configuration I have used to work fine with Broadvoice. Now it gets a busy signal every time. I've checked "sip show registry" and it says it's registered just fine. I've tried "sip debug" and it shows calls coming in, but they always get a busy signal & I can't tell why. Here's a SIP Debug output: Sip read: INVITE

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 --dport 5068 -j REDIRECT

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

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

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:

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

> Asterisk Project Security Advisory - ASA-2007-010 > > +------------------------------------------------------------------------+ > | Product | Asterisk | > |--------------------+---------------------------------------------------| > | Summary | Two stack buffer overflows in SIP

192.168.1.107 199 6bd3fb49505 00102/00000 ulaw No Tx: ACK 192.168.0.100 110 5c5a4953-65 00101/00005 ulaw Yes Rx: ACK Those channels are stuck talking to each other. The phones are disconnected yet that connection remains. I can clear w/ a restart obviously, but is there any way to tear down a call like that from the CLI? Bill -------------- next

<div><FONT size=2>Hello,</FONT></div> <div><FONT size=2>two questions: </FONT></div> <div><FONT size=2></FONT>&nbsp;</div> <div><STRONG><FONT size=2>1: How can I open/enable network connection to B?</FONT></STRONG></div> <div><FONT

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

I''ve compiled xen without any problems. but now i have to create an initrd file. When i use the command mkinitrd (without any options) the i''ve got some errors: # mkinitrd Root device: /dev/sda3 (mounted on / as reiserfs) Module list: ata_piix mptbase mptscsih qla2300 reiserfs Kernel image: /boot/vmlinuz-2.6.11.12-xen0 Initrd image: /boot/initrd-2.6.11.12-xen0 Shared

Hello All, Does anyone know why length(fit1$time) < length(fit2$n) in survfit.coxph output? Why is the predicted time length is not the same as the number of samples (n)? I tried: example(survfit.coxph). Thanks, parmee > fit2$n [1] 241 > fit2$time [1] 0 31 32 60 61 152 153 174 273 277 362 365 499 517 518 547 [17] 566 638 700 760 791

Hello to the list and thanks to Aaron for the cool software. I''ve been fooling around with Mechanize and Hpricot for a couple of days and from the docs I''ve read, the following code SHOULD work but doesn''t. I''ve tried the same code on a couple of different sites and I get the same exception for each. Any pointers or suggestions are appreciated.

Hi, I created a share with a dot. ["corman.pub"] comment = corman.pub path = /rsrv/data1/corman.pub read only = No inherit permissions = Yes inherit acls = Yes testparm says "no problem", but when i try to acces to this share, I have a window wessage : "could not find network name " and in the log : [2003/10/31 12:08:50, 0]

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">