Re: [Asterisk-Users] Phones that work well through NAT
Polycoms (the IP501s at any rate) work EXCEPTIONALLY well through NAT. It's as literally dead-simple as plug-and-go. No configuration on the phone, and all you want is a nat=yes in their sip.conf entry. That's it. Seriously. In addition to nat=yes, I recommend adding qualify=yes for all phones behind NAT. Otherwise the router doing NAT may flush out the port mappings relative to your phone. The qualify essentially sends a keep-alive. We have Polycom IP500s and 501s and this works very well for them (one sitting right here on my desk). -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/STUN Server
I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb If your asterisk server is on a WAN IP, then you may not even need STUN. Add qualify=yes nat=yes to the sip.conf entry for each device. This will have asterisk ignore the IP in the SIP packets and look at the TCP header. Also, qualify will send a 'keep-alive' to keep NAT from losing the association of ports. HTH, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I have also issues with jitter over wan (cdma), I'm trying to debug how dejitter buffer is working (using iax2 jb debug), but nothing happens/no debug output on asterisk console :-( is any way how to monitor iax jitter buffer? thx PJ I'm really hoping to see some working settings from some people here. The jitterbuffer is one of the main features I've been looking forward to in 1.2. Here are my current settings, if anyone notices a major problem please let me know. I'm using dropcount of 2 hoping that a shrink in the jitterbuffer will happen a little faster as a trade-off. Am I thinking correctly on this? I moved the resyncthreshold way up since people are having issues with it. My thoughts on minexcessbuffer=60 is to immediately get a decent buffer going, as this is much higher than the jitter I usually see (~20ms). jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=60 jittershrinkrate=1 maxjitterinterps=10 resyncthreshold=1500 Thanks, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP601 Question
[test] type=friend secret=blahpoly insecure=yes host=dynamic qualify=500 nat=no mailbox=testmailbox callerid=Yourname test conext=local disallow=all allow=ulaw progressinband=no here is the local section of the dial plan. exten = 850,1,Goto(Mercury-Network,850,1) exten = 888,1,VoiceMailMain(@Mercury-Network-Emp) Try adding dtmfmode=rfc2833 to your sip entry. Also, check the permissions for the file on your boot server. HTH, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call AGI when agent answers call in queue... ?
I would like to kick off an AGI script when an agent answers a call... thus passing the phone that answered the call, the CID, etc. Anyone know how I could do this? TIA -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] automatically detecting failed registration
Hello all, Has anyone figured out a way to send email notifications etc. due to failed IAX2 registration attempts? Thanks -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on nerdvittles: http://mundy.org/blog/index.php But I'm not sure that this actually does concurrent calls. I'm thinking my best bet is writing some fast agi to parse a mysql database, then create call files. Use asterisk manager interface to monitor calls and that way I can keep the preset concurrent limit. Any ideas? TIA! -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
Ron Senykoff [EMAIL PROTECTED] wrote: I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Can you tell me which party this is for, so I can ensure I never vote for them? It's a basic GOTV (Get Out The Vote) drive, with just a short message to encourage people to come out to the polls. It has nothing to do with asking for any money, etc. Just a short message to people who belong to the party from their candidate. -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
I just did a quick office poll and everyone agreed if a party candidate did this to them, they would vote for the candidate's opponent. The office is rarely unanimous in political matters so this was a pretty interesting result to me. I'm pretty sure the feeling is universal. Like I said there is a special place in hell for people who do this. Unsolicited is unsolicited, no matter the content. The last thing people want is an unsolicited call from a _politician_. If you want to rant about politics, take it to the right forum. alt.politics If you want to PM me, go for it. But don't turn this professional VoIP forum into your own soapbox. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls
Thanks for all your responses. The reason we would not go through a provider is that I run Asterisk phone systems, we have access to bandwidth, and I can do this myself for a fraction of the cost. Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What ATA should I buy?
I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Since you have no Digium hardware (and thus no connection to POTS or PRI)... are you routing your phone calls via VoIP? If so, it is not recommended to run FAX via VoIP. The two don't mix. FAX is not able to handle packet loss like VoIP. Also, any codec other than uLaw will not even come close to working, as the codecs are designed to compress voice. HTH, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Teliax - Codec Preference effective?
It does take between 1 and 12 hours for the new settings to take effect. Dan I've recently had a problem with codec changes taking affect, but they were nice enough to on-the-fly move an 800 number to route from one site to another. It seems there was some kind of cacheing issue, as I changed our server to route through a different gateway (voip-co2 instead of voip-co3) and immediately it went to uLaw. The real problem was that the uLaw site had no g.729 codecs installed... so there was no option to wait. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do we need a QOS switch ?
We have 10 people on our network and each person will have a SIP phone connected to our Asterisk server. All phones, Asterisk, other servers and users workstations will be using the same network. The question is: would I need a QOS device to give SIP traffic a chance? Our internal network is 100M. We will have a ISDN30 for outgoing calls. No calls will be made over the internet. We have dealt with this issue in small offices by using phones that contain a switch (Polycom IP500s) and do their own QoS. In other words, all the users' PCs are hooked into their phone, so any excessive traffic does not interfere with the phone. Since the phones then hook directly into the same switch that the PBX (Asterisk) hangs off, quality has been fine. Keep in mind this is for small offices like you describe. Provided the topology of your switches is OK, you should be fine. Just don't uplink to another switch where you can create a non-QoSd bottleneck link. -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom 501 horrible echo
One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). That has got to be the problem! I'll let you know how the results go. Upgrading to the correct sip.cfg fixed the problem. The Polycoms are back to their great speakerphone-ness. A gotcha is that the new sip.cfg now contains ntp settings. You'll need to modify these to fit your timeserver setup. -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Polycom 501 horrible echo
One thing I was pondering: you are not, by chance, using the same sip.cfg between version 1.4.1 and version 1.6.2 are you? The file has changed significantly between these versions, and certain acoustic settings that worked with 1.4.1 may not work with 1.6.2 (Not to mention that ipmid.cfg and sip.cfg were merged in the 1.5.x release). That has got to be the problem! I'll let you know how the results go. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 horrible echo
I've been running 1.6.4.0064 for the last few weeks.. I've had no problems with it, I haven't done a whole lot of speaker phone with it yet though.. Once my IP4000 reboots It'll be running it as well so that will be a good test. Which bootrom version are you using? -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 horrible echo
We also have noticed a poor server config can cause this in testing. Noticed when I had one person building * servers using Debian. Had them rebuilt with FC4 and have no issues - yet:) I recently upgraded all our phones to the latest Polycom firmware 1.6.2 and went from great speakerphone to tons of feedback. I would hate to have to go back to the old firmware. Although Polycom recommends keeping the older bootrom unless you need https provisioning, I'm going to try the new bootrom and see if it fixes the problem. This is being experienced across 3 corporate offices with 3 separate Asterisk servers. And I have to reiterate... all was good until the firmware upgrade. -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 horrible echo
I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? It was the old firmware from when we first got the phones actually. 1.4.x I think. Then I read that they fixed the CID issue and decided we needed an upgrade. I tried it out on my phone, but didn't really notice the problem until we had upgraded the rest. Oh well... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 501 horrible echo
I'm running 1.6.2.0041 according to my phone. Which firmware worked for you? It was the old firmware from when we first got the phones actually. 1.4.x I think. Then I read that they fixed the CID issue and decided we needed an upgrade. I tried it out on my phone, but didn't really notice the problem until we had upgraded the rest. Oh well... Also, these are IP 500 SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
