search for: ngreps

ngrep is a great network packet capture. will it be included in epel? -- Peng Yong

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

...ndstream way') to a collegue. Most of the times this goes ok, but sometimes, when the receptionist puts the call on hold, and tries te reconnect to the caller there's only one-way-audio. The receptionist can hear the caller, but the caller cannot hear the receptionist! I've done several ngreps etc. and I can see that traffic is going from asterisk to the receptionist phone, and vice versa. I can predict when this is going to happen: when the receptionist places the call on hold, the caller doesn't hear musiconhold. If the caller does hear musiconhold then everythings goes well. Ast...

Hi there I am wondering if anybody can help me illuminate a problem I am having with my asterisk installation. I am using: - IP phone (Siemens gigaset S685IP) behind a modem/router that has ports udp 5060 and 10000:10100 forwarded to the static ip of the IP phone (192.168.0.3). This has to go to: - modem that operates in half bridge mode (no nat) to a linux firewall (does natting ip is

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

Hi, I managed to get a remote extension to work through a router which can now call all the other local extensions in asterisk. For some reason, nobody can call me back. They get failed upon trying. Keep thinking there must be some caller group to which I need be added. Or perhaps I need to add the IP address of this phone to the sip.conf file? Please let me know. Thanks. Nick

We're running Asterisk 1.6.1.17 for our campus voicemail server and Juniper M120s as our SBC. Unanswered calls, which arrive via the SBC, are diverted to voicemail using a 302 redirect when the called party doesn't answer. In this case the caller is able to hear the greetings and begin to leave a message only to have Asterisk terminate the call mid-recording. We're uncertain why

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

hi all, I keep asking the question and getting no replies, so i'll keep asking :-) In asterisk 1.09, with autocreatepeer=yes, if i send asterisk a SIP request from SER, specifically rewritehostport("myIP:5070"); (asterisk running on port 5070) asterisk picks up the request and matches it to the dialplan, i.e. if in ser i was sending to 151@myServer, it will make it

Hi, I have been going around in circles trying to find the solution. Many others appear to have the same problem, but never a solution or explanation. I am running Dovecot 2.0.21 under Fedora 16. All components are running on the same server, whose IP address is shown as '192.168.x.y'. The dovecot -n output is: <quote> /SSH Secure Shell 3.2.0 (Build 267)/ /Copyright

Hi I tried to get some logs: https://pastebin.com/Z8xVzpzW As you can see the process isn't shutdown and still in transaction as long dovecot is running. It destroyed the transaction when I stopped Dovecot. And this behavior only happens when the mailbox of user is full... Any Ideas how to debug this correctly? > So far, I haven't been able to reproduce anything weird at this end.

While trying to debug a goofy XML loading issue in IE, I''ve found that Mongrel (latest) returns Content-Type: 0 with every request on a particular (CentOS 4) server, yet not on my local (OS X) box. These both access identical Rails apps. This seems like a clue, but thought I''d ask here if for some reason this is expected behavior. Both running Ruby 1.8.4. Both return

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty

I've build this nightly snapshot with my installed openssl, which is 0.9.8b (Fedora 6) and the latest openssl development tree. The 20070605 version of dovecot starts up doing ssl protocol perfect, then after clicking a few directories, then when clicking back to INBOX, dovecot hangs. There is no logging info, even with --enable-debug on and all .conf file directives I can find. Doing a

I'd like to grab the SIP response code that comes back from an INVITE. The HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get the SIP response code instead? Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten <-> PSTN, Xten <-> Analog, IAX <-> IAX, but exten => 3733,1,Dial(SIP/fred@somewhere.com) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- Dave Cotton <dcotton@linuxautrement.com>

Hi, I face a problem which look like the same as David with his thread "Asterisk not receiving call from VPN address". I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10.x I updated lately to 11.7.0 version and now no one of my phones can register anymore to the asterisk. Ngrep as well as

I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood, and I was hoping someone on this list might have managed to do this. I have configured some numbers to route to a SIP endpoint %e164 at customer's server and convinced the customer to open up UDP ports 5060 and 10000 - 20000. Calling the number gets a SIP request from Simwood. The customer's machine