search for: ngrep

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. As...

...g ip is 192.168.0.20) that has the ports above forwarded to the staitc ip of the asterisk box (192.168.0.21 packaged version for ubuntu hardy). This phone works fine with a commercial provider of viop (via asterisk), but I can't get it to work with my install of asterisk in my remote network! ngrep-ing traffic on the firewall shows asterisk continually sending invites to the public ip of the ip phone. I would be very grateful for any pointers of where to start. If you need sip debug or ngrep info let me know and i'll reply with it. I've been beating my head against this for some time...

...isn't from the VPN then forwarding it to the 172.x address doesn't work. So basically the problem is going between the real network and the VPN. The question is, how can we make this work when calls are received on either network on the Kamailio server and are forwarded to Asterisk? Using ngrep on the Asterisk server we see that it does receive the INVITE, but Asterisk's logging shows no sign it at all. We guess it's a Linux networking issue rather than Asterisk's fault, but don't know where to fix it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk se...

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

...ler is any set on the PSTN. They call one of our IP phones which no one answers. Our proxy, SER, responds to the SBC with a 302 redirect and the call is diverted to Asterisk. The caller hears the unavailable greeting for 6-4050, begins to leave a message and is cut-off after about 10 seconds. In an ngrep trace we see Asterisk receive an ACK from the SBC and it immediately responds with a BYE message for that call. Has anyone else experienced this type of issue? --- ISC Networking & Telecommunications 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 215-573-8396 215-898-9348 (fax) ----...

...7805861ab2dcaa54b52a97@GATEWAY. 8. CSeq: 102 INVITE. 9. User-Agent: Asterisk PBX. 10. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER. 11. Contact: <sip:*voicemail-busy-USERNAME@FEATURE>. 12. Content-Type: application/sdp. 13. Content-Length: 270. See links these links for full sip dialog. Ngrep from PROXY's point of view http://pastebin.ca/23469 Ngrep from GATEWAY's point of view http://pastebin.ca/23470 Ngrep from FEATURE's point of view http://pastebin.ca/23471 _______________________________________________ Serusers mailing list Serusers@iptel.org http://mail.iptel.org/mai...

...ng 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 151@myIP:5070, and asterisk will match it to 151 in the dialplan. In asterisk 1.2 asterisk completely ignores the request (even at most verbose level) and an ngrep shows a "not found" returned to SER. anyone have any idea why this is happening, bug/feature, or how to get it to work the way it did in 1.09? I want to upgrade but I don't want to lose this functionality. thanks for any help, yair -------------- next part -------------- An HTML at...

...}/ </quote> On attempting to connect to the managesieve port from Squirrelmail, using the 'Filter' button i get the following error message: /*Could not log on to timsieved daemon on your IMAP server localhost.*/ /*Please contact your administrator*/ Running/*# /usr/sbin/ngrep -d lo port 4190 */produces the following trace: <quote> /[root at nsi-server root]# /usr/sbin/ngrep -d lo port 4190/ /interface: lo (127.0.0.0/255.0.0.0)/ /filter: (ip or ip6) and ( port 4190 )/ /####/ /T 127.0.0.1:4190 -> 127.0.0.1:35495 [AP]/ / "IMPLEMENTAT...

...> So far, I haven't been able to reproduce anything weird at this end. > Can you provide: > > - Output from `dovecot -n' > - Protocol logs from the connections between Exim and Dovecot > director/proxy and between Dovecot director/proxy and Dovecot backend > (e.g. using ngrep when connections are plaintext or using the rawlog > facility). > - Dovecot debug logs produced with `log_debug=category:lmtp' for both > director/proxy and backend > > Regards, > > Stephan. > >

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

...3250072-ed28", "60") in new stack -- Started music on hold, class 'stream2', on channel 'SIP/3250072-ed28' -- Stopped music on hold on SIP/3250072-ed28 If I replace SetMusiconHold(stream2) with SetMusiconHold(default), I get the default music on hold. Running ngrep on port 80 shows me that the Asterisk system is not sending or receiving ANY data on port 80. What am I doing wrong? Yes, it has network and DNS connectivity. Can't believe it's this hard! Doug.

...atest 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 'ngrep port 993' shows no SSL dialog. If I go through my same compile sequence using the dovecot-1.0.0 source all is well with both the system openssl and the latest openssl snapshot, so it appears something is wrong in the current nightly snapshot. My primary question is howto debug this hang, shor...

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

...tually 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 anything? > > > >The sip message I included in my last message is what I see when I ngrep on 5061, but >asterisk doesn't see it. When I tell Kamailio to send the message to 5060 chan_sip shows >the invite in the CLI. > > My setup has chan_sip running on 5060 and pjsip (tcp and udp on 5061). > > I'll get PJSIP running on 5060 and see if that makes any difference...

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>

...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 wireshark shows the traffic going in on eth0 from VM, but inside Asterisk, nothing, sip set debug ip <ip from phones> shows nothing. This asterisk is also connected to other trunks we have outside our network, there is no problem here, registration is fine. Problem seems to be...

...the customer to open up UDP ports 5060 and 10000 - 20000. Calling the number gets a SIP request from Simwood. The customer's machine then sends a SIP 401 response. Simwood send an ACK ..... and then nothing. Nothing appears in the Asterisk CLI; to get the SIP trace I used the command # ngrep -t -q -n -q -Wbyline -deth0 1283 port 5060 (note that 1283 = the STD code from which the call is originating, so it should show up in any related packets.) ########## sip.conf ########## [simwood_in_slough] type=friend host=178.22.140.34 fromdomain=178.22.140.34 permit=178.22.140.34/255.255.2...