[asterisk-users] Poor VoIP voice quality in one direction from three providers
We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make outbound calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we don't want to maintain the DSL line or deal with the hassles of analog/digital conversion any more. So we want to switch to a reliable VoIP provider and move the asterisk server to one of our colocation data centers. We've tried getting test accounts with three VoIP providers: FlowRoute, CallCentric, and Vitelity. In our tests, outbound calls now go from softphones - asterisk - VoIP provider - outside world. We use ulaw all the way through. But with all three providers, we see a curious thing: The audio quality in the direction from our softphones to the outside world still sounds as good as POTS, but the audio quality in the inbound direction (outside world - VoIP Provider - asterisk - softphone) is noticeably worse. It sounds overcompressed or slightly robotic somehow, with a decrease in dynamic range. It's not lagged or echoey; it just sounds like it's maybe using a crappier codec than ulaw, in that direction only. I'm baffled by this. Both legs of the calls show as Format: 0x4 (ulaw) in sip show channel. Testing the first provider, I just assumed that their analog-digital conversion was inferior to what the Zaptel cards offer (i.e., that they were injecting inferior sound quality into their ulaw connection)... but we're getting exactly the same results with all three providers, which makes me think it's us. Why might this happen? Is there any possible reason other than all three of the VoIP providers are decreasing the audio quality before injecting it into the ulaw stream? -- Robert L Mathews ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip phones how to dial a # sign?
C F [EMAIL PROTECTED] wrote: Use the latest stable or CVS HEAD and modify features.conf. You can change it there. FYI, only CVS HEAD (not stable) supports the new features.conf options. -- Robert L Mathews, Tiger Technologieshttp://www.tigertech.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
C F [EMAIL PROTECTED] wrote:
I your case the problem is with the grandstream, the GS will not
display callerID correctly, take out the name from the callerid string
like this:
exten = ${EXTEN},PRI,SetCallerID(${CALLERIDNUM})
Actually, Tony Mountifield pointed out that the problem is a bug in CVS
stable:
http://bugs.digium.com/bug_view_page.php?bug_id=0003557
(Thanks, Tony!)
With this bug fixed, the Grandstream phones work fine even with the
caller ID name present. They don't display the name, but they do display
the number properly. The problem I was mentioning made it not even
display the number correctly, on any SIP phone.
--
Robert L Mathews, Tiger Technologieshttp://www.tigertech.net/
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Re: [Asterisk-Users] Is there a Caller ID issue in the latest CVSStable
Nicol?s Gudi?o [EMAIL PROTECTED] wrote: Paul, 1.0.5 stable suffers from caller id issues as well, at least for SIP channels. What fixed things for me was swapping in app_dial.c from 1.0.2 stable (didn't try others). You could also just diff app_dial.c between versions to find the problem but I took the lazy way out the first time around. Drumkilla reverted the callerid changes on the latest stable (thanks Russell!). You will be fine if you checkout stable from CVS now. Hmmm; I think I'm still having problems with it, using a completely fresh checkout and compile: Connected to Asterisk CVS-v1-0-02/11/05-17:34:08 I have two Zap FXS lines and two SIP phones, and: - Zap channel to Zap channel, caller ID works (displays correctly on the analog phone display). - SIP phone to Zap channel, caller ID works. - SIP phone to ZIP phone, caller ID does NOT work (Grandstream phone displays Err). - Zap channel to SIP phone, caller ID does NOT work. - Incoming Free World Dialup calls to Zap channel extension, caller ID works. - Incoming Free World Dialup calls to SIP phone extension, caller ID does NOT work. So it seems that asterisk stable, as of today, does not send correct caller ID on calls that end up on SIP phones, unless I'm doing something boneheaded (although I used almost-identical config files on 1.0.2 with no trouble). A tcpdump shows that asterisk is sending this in the SIP INVITE header to the phone: From: asterisk sip:[EMAIL PROTECTED]; (IP address obscured; it's correct in the original.) But somehow asterisk appears instead of the correct caller ID. Wasn't that the bug other people were seeing that the stable update was supposed to fix? Have I missed something obvious? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P FXS works only if two lines are off hook?
I have a TDM400P with one FXO module and two FXS modules in it. I also have a Wildcard X101P. After trying hard to get things working on various Intel computers, but having echo problems that made it not really usable, I decided to try it on some older PowerPC (Macintosh) hardware running Yellow Dog Linux. Things started off smoothly. Both zaptel and asterisk seemed to compile okay, and both cards are detected: kernel: Found a Wildcard FXO: Wildcard X101P kernel: PCI: Enabling device 00:0f.0 (0004 - 0007) kernel: Freshmaker version: 63 kernel: Freshmaker passed register test kernel: Module 0: Installed -- AUTO FXO (FCC mode) kernel: Module 1: Installed -- AUTO FXS/DPO kernel: Module 2: Installed -- AUTO FXS/DPO kernel: Module 3: Not installed I can dial out from a SIP phone through the FXO ports on the X101P or the TDM400P (with almost no echo), so some things are basically working. However, the FXS lines didn't work properly: there is no audio in either direction. If I call these channels from a SIP phone, they do ring properly, and if I pick up a phone connected to them, the console correctly shows, for example: -- Starting simple switch on 'Zap/3-1' But there is no dialtone and no voice audible in either direction when they are called. Outgoing calls from these FXS channels don't work; pressing numbers on the keypad beeps but has no other effect. Then by accident I picked up both FXS lines at the same time, and both of them work perfectly! I get dialtones, I can dial and make calls with them, audio works in both directions -- nothing wrong at all. So as long as they're both off the hook at the same time, everything is fine. But as soon as I hang up either one of the lines, the sound on the other line will *also* go dead again within a second. A little more experimentation: having just one FXS module on the TDM400P (removing the other) doesn't work at all. The only way the FXS lines work is with both FXS modules installed and off hook simultaneously. This problem occurs with the released versions (zaptel 1.0.4 with the wcfxs driver and asterisk 1.0.5), and with cvs head (using the wctdm driver). Does anyone have any idea why this would happen, and how I could fix it? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for Asterisk
At 1/16/04 7:25 AM, Andrew Kohlsmith [EMAIL PROTECTED] wrote: That's pure bullshit -- I use software RAID *specifically* because I value my data. I don't want to buy two hardaware RAID controllers to have one sit on the shelf just in case the first dies... and if the second dies you're SOL because they've lasted long enough that they're no longer available. Linux software RAID is available on any Linux system and if the system blows up I can put the drives in another system and *not* worry about it not being detected. Yeah, I couldn't agree more. We originally thought hardware RAID was the way to go, and we bought a couple of fully loaded Dell PowerEdge 2550s with SCSI hardware RAID 5 arrays at about $4500 a pop. We also bought a PowerEdge 600SC for around $900 with lots of disk space to use as a network backup machine (backing up the 2550s) with Linux software RAID 5. I've also had a crappy old desktop machine running Linux software RAID 1 for a couple of years. It turns out that the software RAID is just as reliable (more so, in fact -- we have had a number of lockups on the 2550s that appear to be due to the hardware RAID subsystem locking up, and the software RAID machines have never done that, even though the backup server does more disk I/O than the others). The software RAID on the 600SC is faster than the hardware RAID in bonnie tests. In addition, the Dell PowerEdge mailing lists are full of people with horror stories about their hardware RAID systems -- if that dies on mine, I'm screwed until I can convince Dell to come out and fix it (which they often won't do until they've spent hours on the phone with you trying various things). We should have simply bought 4 600SCs (instead of 2 2550s and a 600SC), using one as a hot standby, and saved ourselves around $6000. In fact, we're planning on moving to that and selling the 2550s on eBay to improve our overall reliability. If the power supply, motherboard or RAM of a 600SC dies, we can easily move the disks to the spare machine and be back up within a few minutes without relying on anyone else. In the worst case (RAID corruption/machine catches on fire), I'm still going to be okay, because I can restore from backups in a couple of hours. The key thing to me is that at no point do we have to rely on any other company to get things up and running again, which is far more important than any putative risk of data corruption from software RAID (which I have not seen even under very heavy disk loads, and which I think is pretty much a myth these days; look at the Dell PowerEdge mailing lists if you think hardware RAID is more reliable -- those stories of hardware RAID problems from real users have scared me to the point that I'll never consider buying any sort of proprietary disk subsystem again). -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ I am not able rightly to apprehend the kind of confusion of ideas that could provoke such a question. -- Charles Babbage ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
At 11/3/03 6:57 PM, Anthony Wood [EMAIL PROTECTED] wrote: Internals can use the IP address of the NAT box as the Asterisk Server IP and then it should work. This doesn't work on my NAT box, unfortunately. Devices behind the NAT can't connect to the public IP address and talk to other devices behind the NAT. Don't know why (cheapo NAT box, most likely; it's part of my DSL modem), but I believe this situation is fairly common. -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
At 11/3/03 10:00 AM, Martin Pycko [EMAIL PROTECTED] wrote: Is externip and new parameter?? It's new. It prevents asterisk from putting the private IP in the messages that asterisk sends with SIP. Does it take an IP address, like externip=1.2.3.4? And does it then force the SIP messages for that channel to use the externip value instead of the server's local IP address? If so, that's useful; it will help people who know in advance that a certain phone is on one side of a NAT or the other. However, it would be nicer still if it could fix the SIP messages only when necessary, using a subnet mask or STUN, as has been proposed. The reason is that hard-coding an IP address to use when communicating with a certain client means you can't have a phone in an office (on the same side of the NAT as Asterisk) during the day, then take the phone home at night (on the other side of the NAT) and have it work without changing sip.conf. -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind LinkSys NAT Routing
At 11/3/03 2:41 PM, Martin Pycko [EMAIL PROTECTED] wrote: It's not for phones, it's for asterisk behind a NAT. My apologies; I'm not making my question clear. I realize this option is for Asterisk behind a NAT, but of course Asterisk uses this parameter to talk to SIP clients (which I referred to, perhaps too specifically, as phones), and that's what I meant. In other words, Asterisk might be talking to SIP phones on either side of the NAT. A given SIP phone acting as an extension may be on the same private network as Asterisk, or it may be on the other side of the NAT (out on the public Internet, possibly even behind its own NAT on the other end). Imagine I have both Asterisk and a SIP phone on my local office network using private IP addresses, and I also have a second SIP phone that is in another location, at someone's home office on the public Internet. The externip=a.b.c.d doesn't help in this situation, because it forces Asterisk to use the external IP address in all cases, which breaks the functionality for local phones. That is, the new option presumably makes it possible to have *all* your SIP phones on the other side of the NAT from Asterisk, but you can't some phones on both sides. (Indeed, I just tried it, and using externip=something prevents SIP phones on the same private network as Asterisk from working.) In Bug ID 104, a patch was suggested that takes the netmask into effect and makes the right decision for phones on either side of the NAT. However, the code that was added for externip in the current CVS isn't that patch; it's just a way of giving me a choice of having SIP phones on the outside of the NAT working, or having SIP phones on the inside of the NAT working, but not both at the same time. I guess I'm curious why the hard-coded global option was used, because it doesn't really solve the problem in the general case. The whole trouble with NAT is that Asterisk may need to use a different IP address depending on the IP address of the SIP client it's communicating with, and that address needs to be determined on the fly. In a perfect word, this would all be handled by magic so it required no configuration (e.g., STUN), but the patch in 104 would at least allow phones on both sides of the NAT to work with a small amount of configuration, which isn't possible now with the CVS code. Thanks again for the hard work you're putting in to Asterisk! -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX gain TX gain
At 10/30/03 11:36 PM, Dan [EMAIL PROTECTED] wrote: Have you tried to use values like 0.5 or 0.8? Hmmm, good suggestion, but it didn't help, unfortunately. However -- I did some more testing, and found that using extremely large negative values such as -20.0 does make it noticeably quieter (I hadn't tried anything below -10.0 before). So I can confirm for others having such trouble that negative values do work, but you might need to make them bigger than you think. -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RX gain TX gain
At 10/30/03 12:21 PM, Jared Smith [EMAIL PROTECTED] wrote: It's my understand that they are db levels. (And, if I remember my electrical engineering classes from college, a 3db increase effectively doubles the volume.) As a slight aside on the subject of gain It seems that most people asking about RX/TX gain want to increase their volume. I have the opposite problem: I have a Digium TDM10B FXS card that generates sound far too loud (in the earpiece) with the RX gain set at 0.0, or commented out. That is, routing an analog line = X101P = Asterisk = TDM10B = analog phone is MUCH louder than if I just plug the same phone into the same analog line directly. Some people have suggested that using a negative gain will make it quieter, but I haven't had any luck with this. I *can* make it even louder by increasing the gain -- if I use rxgain = 10 on the TDM10B, for example, it's so loud it sounds like the phone is going to explode -- but using things like rxgain = -3.0 or rxgain = -10.0 doesn't make it any quieter. I can't get it below the rxgain = 0 value. I've been meaning to dig around the source and see what's up, but since it's being discussed... anyone know how to use rxgain to lower the earpiece volume? -- Robert L Mathews, Tiger Technologies http://www.tigertech.net/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
