similar to: call drops after a few seconds

I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via:

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does

Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to

Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. _________________________________________________________________ Hotmail: Trusted email with Microsoft?s powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/

Hello, We have 2 SIP trunks from Bandwidth.com and if both are in use and someone tries to dial out, they cause another call to get one-way audio (the caller hears us, we cannot hear them). This happens 100% of the time and Bandwidth.com doesn't offer any support. I don't see any setting that tells Asterisk that there are 2 channels available from Bandwidth.com's IP. I'm

Hi guys,   I have two data sets of prices: endprice0, endprice1   I use the Wilcox test:   wilcox.test(endprice0, endprice1, paired = TRUE, alternative = "two.sided",  conf.int = T, conf.level = 0.9)   The result is with V = 1819, p-value = 0.8812.   Then I calculated the z-value of the test: z-value = -2.661263. The corresponding p-value is: p-value = 0.003892, which is different from

It might be worth mentioning the voip call is coming from a number we have thru bandwidth.com in case anyone uses them. James Shigley From: James A. Shigley Sent: Wednesday, December 02, 2009 3:06 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Variable Name needed That wasn't it either. I tried a few other likely fields from

We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An

your Asterisk server at all. Try doing a packet trace on the network segment where the calling SIP agent is and see where it's trying to send the ACK to. My guess would be your firewall is incorrectly handling the SIP messages. Generally it's very bad news to use an ALG or firewall to mangle SIP packets as they almost always get it wrong. In your case there is a Record-Route header in the

I have an issue with receiving inbound calls. I've got bandwidth.com trunks incoming to my asterisk box, bandwidth sends all incoming traffic to one of two IP addresses, and requires outbound traffic go to either of the same two IP addresses. I've got to use fromuser=<DID> on outgoing calls so they apply the right caller ID. My issue is that I want incoming calls to match on a

Safe Harbor Games (http://www.safeharborgames.net/) is a popular site in the backgammon community. I am one of several people trying to get the game server to work on Linux with Wine. I am using a Asus EEE 1000HE with the latest version of Easy Peasy (Ubuntu) and Wine. The game software requires browser to launch and I was successful in getting this to work by loading windows version of

i own a CyberPower OR2200LCDRM2U and was wondering if any nut drivers support this ups. none i have tried have worked although usbhid-ups did tell me that the productid=0601 was not supported yet. i see in the compatability list on the nut website that the PR2200 is supported with the powerpanel driver but the OR2200 is not listed. i am interested in what i am doing wrong or if this is a support