similar to: Cisco 7960 Register Problem

Hi all I have a problem to register a cisco 7960 to an asterisk 1.2.2 I defined in sip.conf the next : ["phonenumber"] type=friend username="username" secret="password" host=dynamic context=work I am trying to catch the register requests with sip debug with no success (empty screen). I can only catch the register messages with ngrep on

Hi guys still have the problem to setup the IVR correctly. I am forwarding call from ser : if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("xxx.xxx.xxx.xxx:5061"); t_relay(); break; } } inside sip.conf

Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you can see below: 1. U Asterisk_IP:5060 -> OPENSIPS_IP:5060 2. OPTIONS sip:OPENSIPS_IP

Hi, I have a dataframe, hilodata, which looks like this: > hilodata sym date maxprice minprice ntick 1 ABK 19910711 11.1867461 0.0000000 108 2 ABK 19910712 11.5298979 11.1867461 111 3 ABK 19910715 11.7357889 11.4612675 52 4 ABK 19910716 11.5298979 11.3240068 51 5

Just responding in case this may be of help to somebody with firewalling issues. Not sure if this is off on a tangent to the original question... Here are three different forms of common firewall scripts and ways of getting SIP to work behind them. The third one has some additional stuff beyond just SIP although I can't remember why I wrote it that way. I've been having no fun using

This is a call coming in through the ISDN to 7940's. Answering with non-codec capability 1 - Is that the problem? SIP Debugging Enabled We're at 10.1.0.11 port 18406 Answering/Requesting with root capability 8 Answering/Requesting with preferred capability 4 Answering/Requesting with preferred capability 8 Answering with non-codec capability 1 <<<<<<------------- 12

Anything to do about portscans? Is there any way (should I) to see if the connection is a legit (only SIP currently) connection BEFORE my * answers? [2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0 -- Executing [s at default:1] Answer("SIP/sip.jmg.se-081dd730", "") in new stack [2007-10-17

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

I've attached two SIP debugs in reference to bug #116. They are from today's CVS build. 1. pickup.txt is a call from SIP(1) to SIP(2) with SIP(3) picking up the call. After which, SIP(2) rings for about 30 seconds then stops. 2. hangup.txt is a call from SIP(1) to SIP(2) with SIP(1) hanging up before the call is answered. SIP(1&3) are Cisco 7960's and SIP(2) is a Polycom

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test. - Cisco 7960

Hello all, taking examples from various pointers, I am attempting to put together an outbound dialing example using SIP (Cisco 7960) with 2 X100P. Everything seems to be working without generating errors, but the problem is the phone hangs up (102/Bye). Any pointers/advice are much appreciated Here is the section in extensions.conf: extensions.conf ; From CISCO at work ; exten =>

*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please what TTL means specifically and what is the best value og TTL and what is the lengh vale mean

I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple "sip debug" and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend

R 2.3.1 I wrote a little script to do some cross correlations. The symbols are in a text file like so: symbols.txt ibm dd csco """ require(tseries) symbols <- scan("symbols.txt", what = 'character') for(line in 1:(length(symbols)-1)) { assign(symbols[line], get.hist.quote(instrument = symbols[line], start = "2005-09-01", quote =

Greetings all, After a long day of fighting with GTalk and having it finally working, I wanted to setup DPMA on my Digium phone. So first of all, I had to reinstall it all and reconfigure it all, since it works only on certified versions, and my installation was not from the certified branch. It took a long time of recompiling, testing, adding missing stuff, but I got it straight. Now, I

Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite: Re-invite to non-existing call leg on other UA. SIP dialog

I've got three 7960s running v6 SIP firmware. My Asterisk setup has worked fine with grandstream devices, and basically, we're just upgrading to use nicer phones. Whilst I can make/receive calls from the 7960 to/from gossiptel). When I try to place a call, I get the following Jan 21 11:09:23 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to authenticate user "30"

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard

Dear all, I have installed asterisk 1.0.5 on redhat 9 I have installed also, asterisk-oh323 0.6.5 module (successfully compiled and installed) Now When I am trying to get asterisk communicate with a Radius (in my case: it's the VoiceMaster Radius) I was able to do the following: After installing all recommended to download and install radius client for asterisk