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Solving port blocking issue @ softphone
by feby.francis＠crosstel.com June 26, 2014

June 26, 2014

Experts, I have a customer who travels a lot and uses our softphone offering, I need a solution to overcome the situations when the port 5060 is blocked. Is there any work around to solve the port blocking problems? We use standard 5060 on the SBC and my switch vendor is Broadsoft. Thanks for all the help and advices, Feby Francis

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Call established before audio path
by briant＠socket.net June 26, 2014

June 26, 2014

We have customers who are reporting that when people call into them that the beginning of the greeting is not being heard. It appears that the phones think the call is ready before there is a complete audio path. We see this the most on calls that come in from the PSTN. Long distance or toll free calls are the worst. We have a Broadworks system that uses Lucent 5010 (Telica) switches for the connections to the PSTN. The symptom seems to be very similar for all of our PSTN connections, to multiple carriers. I'm just curious if anyone has any experience with this problem, or if anyone has any suggestions to get to the bottom of the issue. Any assistance or inspiration would be greatly appreciated. Thank you, Brian Taylor Socket Telecom, LLC brian at socket.net

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Monitoring tools
by jra＠baylink.com June 25, 2014

June 25, 2014

[ inadvertantly sent to NANOG. I hate RFC 2919 non-compliant mailers ] ----- Original Message ----- > From: "Joseph Jackson" <jjackson at aninetworks.net> > Check out Homer @ http://sipcapture.org we love it. There is also a > commercial version that has more features / support. Well, I don't know if Homer hsa the realtime viz tool I'm looking for, but if it lives up to the attitude of it's website and prezi, then it's probably a good thing that I know about it anyway, thanks. Wonder if anyone's working on Sonus support. Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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HIPAA compliance
by ujjval＠simplesignal.com June 25, 2014

June 25, 2014

Hi Team Voiceops: What is needed for an ITSP to be HIPPA complaint? Sounds like a really dark grey area! Interesting Article here: http://www.linkedin.com/groups/Recommended-HIPAA-Compliant-VoIP-Services-22… Ujjval Karihaloo

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Odd network problem with SIP
by jra＠baylink.com June 24, 2014

June 24, 2014

----- Original Message ----- > From: "Frank Bulk" <frnkblk at iname.com> > Except, by definition, jitter is ping-response (latency) deviation. In > calculus terms, jitter is the derivative (change in rate) of latency. I was assuming that jitter had to do with the sliding of packets around WRT strict isochronous positioning -- specifically, packets arriving out of sequence, because they are so late. That's not what jitter means in VoIP? :-) Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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Odd network problem with SIP
by jra＠baylink.com June 21, 2014

June 21, 2014

I have a client using the Spitfire dialler on Win7 to dial via SIP to a carrier called VoxTelecom; I infer they're a virtual carrier because the SIP goes to their Sonus SBC at a single IP, but the media sessions go all over creation. The client has been having completion trouble, and the dialler folks said "crappy circuit; too much jitter and packet loss", tested from the dialler using PingPlotter. I don't think it was jitter, I think it was ping response deviation; PingPlotter doesn't appear to actually test jitter. But to tick the boxes, I had road runner out; they replaced a ubee D2 modem with their New Hawtness Arris D3 modem; no change. All his analog measurements were pristine, he told me. So next step, check the router. Log in to the Fortigate 40D; control panel won't paint properly. This *may* have been an IE10 compatibility mode red-herring, but I temporarily swapped it for a 20C, which I could talk to. Set up the inbound DNATs for udp/5060 SIP and upd/49152-49252 RTP. Ran some calls. They're not seeing my ACK after they send me an SDP. They get the invite, but not the ACK. I send RTP, but they don't bother cause they think the call's not set up yet. Wireshark on the dialler... that ACK packet *has a bad IP header checksum*. Not the earlier packets; just the ACK. Huh? So, assume it's the OS, somehow; reboot. 65 Windows updates and an hour later... Set everything back up, and run more test calls. This time, the Invite has a bad IP header checksum. Look at clock, 6pm EDT. Give up, go home. I'm going to go back to the original router Monday morning, and check the compatibility mode theory, but has anyone ever seen "just certain sent packets show up with a bad IP header checksum, as monitored on-machine"? (I know that the original 40D router might well *have* a problem, and that the checksum thing is orthogonal to the original problem -- since older pcaps show clean setups -- but at least I can confirm I have the NATs configured right.) Cheers, -- jra -- Jay R. Ashworth Baylink jra at baylink.com Designer The Things I Think RFC 2100 Ashworth & Associates http://www.bcp38.info 2000 Land Rover DII St Petersburg FL USA BCP38: Ask For It By Name! +1 727 647 1274

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Asterisk Inconsistently sending RTP for deterministic set of orig DIDs
by beckman＠angryox.com June 19, 2014

June 19, 2014

I'm dealing with a strange situation and I'm hoping there might be someone who can see an easy answer. 1. When a call comes into Asterisk, we answer the call and send a SIP 200 OK. Then we play an audio clip, then bridge the call to a Dial() with the 'r' option to play ringing to the origination side of the call. This works with all of Carrier B's DIDs and most of Carrier A's DIDs. The RTP streams start sending as soon as we answer() the inbound call. 2. For other DIDs, we answer() and send a 200 OK and do not play any audio but bridge the call directly to a Dial(). The audio is still passed to the caller and they hear ringing generated from Asterisk, not locally. 3. For the problem DIDs we are working on right now, they all look like #2 but no audio is passed to the caller, and the RTP stream is not sent (based on tcpdumps). However when the Dial()ed call leg answers, the RTP stream begins (inconsistently, but that's another issue). Because this issue is happening with just a small subset of DIDs on one specific carrier of ours, and that we have made NO changes to our Asterisk configuration or our AGI that handles calls, and that the same AGI handles all inbound calls the same way, I'm looking for any troubleshooting advice I can find. This started very suddenly after several years of no issues three days ago. My Asterisk server has enough inodes, very few open files (no where close to ulimit levels), and no indication that there are any problems. There are no limitations on inbound or outbound ports for RTP (no firewall rules restricting that traffic). Reinvites are not enabled. Beckman --------------------------------------------------------------------------- Peter Beckman Internet Guy beckman at angryox.com http://www.angryox.com/ ---------------------------------------------------------------------------

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Adtran Help
by david＠ringfree.biz June 18, 2014

June 18, 2014

I have an adtran 912 that I'm seeing one way audio on origination calls (very intermittently). I'm exporting pcap's from the machine to a tftp server over the internet. But the 912 runs out of memory and pauses the pcap process and thus far I haven't been able to get a single decent pcap to show the carrier. I've been on with Adtran support and their answer is to put the tftp server closer to the adtran so that exports finish faster. I'm doing that and will have it done in the am. However I thought I'd ask if there's a magic process that anyone has found for getting good pcaps from these devices. They seem to work well, but troubleshooting is a bitch. You can even log the debug. You have to keep a damn terminal open and hit a key every 4 minutes to not disconnect. Thanks David -- -- www.ringfree.biz 828-575-0030

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Canadian SIP Trunk Providers
by ciscoplumber＠gmail.com June 13, 2014

June 13, 2014

> > Hi Jeff > > Are you looking for a partner to originate Canadian TF and DID traffic or > LD term or both? > > Thanks > > Ivan > Looking to both originate and terminate. Thanks. On Fri, Jun 13, 2014 at 4:55 PM, Ivan Kovacevic < ivan.kovacevic at startelecom.ca> wrote: > Hi Jeff > > Are you looking for a partner to originate Canadian TF and DID traffic or > LD term or both? > > Thanks > > Ivan > On Jun 13, 2014 5:50 PM, "Jeff Anderson" <ciscoplumber at gmail.com> wrote: > >> I work for a service provider that sells SIP trunking services. We are >> looking at adding a Canadian SIP peering partner. >> >> Does anyone have a list of the major players in this space that would be >> a good place to start our research? >> >> Thanks, >> >> Jeff >> >> _______________________________________________ >> VoiceOps mailing list >> VoiceOps at voiceops.org >> https://puck.nether.net/mailman/listinfo/voiceops >> >>

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Canadian SIP Trunk Providers
by ciscoplumber＠gmail.com June 13, 2014

June 13, 2014

I work for a service provider that sells SIP trunking services. We are looking at adding a Canadian SIP peering partner. Does anyone have a list of the major players in this space that would be a good place to start our research? Thanks, Jeff

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