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Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: ntblade on November 30, 2006, 02:27:59 PM
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Hi,
Thanks for the fantastic contrib. Here is my setup:
Fresh install of SME 7.0 - no updates
Dell 420 sc, 2 x 160G drives in HW RAID (Dell SATA controller)
1G RAM
2 x X100Ps (pretty sure they're clones)
selintra-sail-2.1.14-347
smeserver-asterisk-1.2.10-1
smeserver-asterisk-zappri-MPP-1.2.6-1
Firstly, the cards aren't detected automatically. I have to do:
#modprobe zaptel
#modprobe wcfxo
#lspci
04:01.0 Communication controller: Motorola Wildcard X100P
04:02.0 Communication controller: Motorola Wildcard X100P
#dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.7 Echo Canceller: KB1
ACPI: PCI interrupt 0000:04:01.0[A] -> GSI 17 (level, low) -> IRQ 177
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X100P
ACPI: PCI interrupt 0000:04:02.0[A] -> GSI 18 (level, low) -> IRQ 201
wcfxo: DAA mode is 'FCC'
Found a Wildcard FXO: Wildcard X100P
Then when I probe for the PCI cards they're found:Loading zaptel...
Zaptel view of card in /proc/zaptel/1
Span 1: ZTDUMMY/1 "ZTDUMMY/1 1"
Zaptel view of card in /proc/zaptel/2
Span 2: WCFXO/0 "Wildcard X100P Board 1"
1 WCFXO/0/0 FXSKS (In use)
Zaptel view of card in /proc/zaptel/3
Span 3: WCFXO/1 "Wildcard X100P Board 2"
2 WCFXO/1/0 FXSKSand the trunks are added automatically.
Firstly, should ZTDUMMY still be there?
I've now taken delivery of a couple very cheap SIP phones which I've managed to make internal calls with and they also work with sjphone.
Secondly, I can make incoming calls but only one of the modems accepts calls. Ive swapped the modems, swapped the cables and lines but this makes no difference.
Can anyone help me with the above please?
Kind regards
NTBlade
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Firstly, the cards aren't detected automatically. I have to do:
Hi NTBlade
There is a section on our docs pages which shows how to identify your clones to the selintra database.
http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter258
Once done you can use the PCI panel to identify your cards and stop/start the pbx.
Firstly, should ZTDUMMY still be there?
Ideally, no. However, it is likely that dummy was left over from a previous PCI detect. If no boards are detected the database is udated to load dummy at startup. YOu can remove it by doing an initialise in PCI panel. You can manually remove it for a particular cycle by doing
rmmod ztdummy
It's difficult to comment upon why one of the X100's isn't taking calls because we aren't sure what state SAIL is in since you have somewhat circumvented the "normal" card detect stuff. Can we suggest you define the cards properly to the database and then load them through the PCI panel? If you do it this way and still have problems we will be able to simulate your problem here. As it stands we aren't 100% sure what state your system is in.
btw - what cheap SIP phones are you using? If they aren't ones we know we can update the database for the next out. Also, do they suppport remote provisioning?
Kind Regards
Selintra
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> Hi NTBlade
Hi Selintra, thanks for your reply.
>
> There is a section on our docs pages which shows how to identify your clones to the selintra database.
That worked a treat. In case this helps others, here's what I did...# lspci -vv
04:02.0 Communication controller: Motorola Wildcard X100P
Subsystem: Motorola: Unknown device 0000
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B-
Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR-
Latency: 64 (250ns min, 32000ns max)
Interrupt: pin A routed to IRQ 201
Region 0: I/O ports at dc00 [size=256]
Region 1: Memory at dfbff000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+)
Status: D0 PME-Enable- DSel=0 DScale=0 PME-
I have two identical cards and from your documentation the one above is in PCI slot 2 so...# lspci -vn
04:02.0 Class 0780: 1057:5608
Subsystem: 1057:0000
Flags: bus master, medium devsel, latency 64, IRQ 201
I/O ports at dc00 [size=256]
Memory at dfbff000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
I checked the Vendor and Card id on the PCI ID Repository which confirmed the card as a Motorola Wildcard X100P then I added the new database tuple like this...# /sbin/e-smith/db selintra-work set pci:1057:5608:1057:0000 sysdev
# /sbin/e-smith/db selintra-work setprop pci:1057:5608:1057:0000 probe wcfxo zzeor EOR
I then fired up my browser and opened the server-manager, chose globals in the sail section and issued a commit. Probing for the cards was successful but the dummy module was still there so I followed your instructions and unloaded it...# rmmod ztdummy
and re-probed for the cards and hey presto! Two cards recognised and no dummy! (I think I did an activate and commit the re-probe. - it was last night I did this) I can now make and take calls on both lines. I haven't tried both at the same time yet. Watch this space...
btw - what cheap SIP phones are you using? If they aren't ones we know we can update the database for the next out. Also, do they suppport remote provisioning?
They are identical to theses http://www.tigernetcom.com/Products_All%20VoIP%20Phone_IPPH%20202.htm but are unbranded and have the single RJ45. They appear to be the same as the NetComm V85 but without the buttons down the side. There is a web interface for setup but I'm not sure about how to check if they support provisioning. ??
The manual that comes with them on CD isn't really clear on how to use the functions but I'll just have to keep trying. :-)
Many thanks for your help :-)
NTB
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Hi,
Is there info on minimising the echo I'm getting when making calls via PSTN?
I've tried the quite line test on BT and also *52* and the echo is very apparent. Any pointers please?
Oh, looks like the X100P clones are meant for the FCC/Japan market as mine have the si3012 ks DAAs :-( Will that make a huge diference in the UK?
Thanks again
NTB
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s there info on minimising the echo I'm getting when making calls via PSTN?
Try here
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html
and here
http://www.voip-info.org/wiki/view/Asterisk+fxotune
Good start point for zapata in UK is
rxgain=8.0
txgain=-2.0
echotraining=800
You can get FXOtune and ZTMonitor from our download site here...
ftp://81.149.154.14/utilities
to fetch them in to your server use
wget ftp://81.149.154.14/utilities/fxotune
wget ftp://81.149.154.14/utilities/ztmonitor
We can also give you a fix to load fxotune parameters at asterisk start-up.
usually with a couple of hours headbanging and fiddlin' about you can get echo down to an acceptable level but the gain settings should be done first in isolation. Get them right (ish) and the echo canceler has less work to do.
Japanese telephony is, AFAIK, pretty much vanilla Ma Bell so you should be OK.
Kind Regards
selintra
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Thanks. Before trying what you've suggested I would first lke to try seting the DAA mode on the card. I found this while googling - http://www.sigsegv.cx/sip-2.html...
Zapata sources must be compiled with the linux26 target and run under Linux 2.6 to provide echo cancellation when using the X100P card. With the MARK2 algorithm selected in the Makefile the echo cancellation is quite reasonable. It may be necessary to adjust the txlevel a bit to ensure that the people on the other side hear you OK and there is no distortion.
It is also essential to select the corect DAA mode. The card defaults to DAA mode 0 which is FCC and which does not set the line impedance and gain correctly for UK lines. This is immediately visible in dmesg: wcfxo: DAA mode is 'FCC' (correct for EU is CTR21). Note that the information on X100p clones in the VOIP Wiki is not quite correct. The newer Wildcards can be programmed for correct line parameters and setting opermode definitely makes a difference. For UK modprobe needs to be passed a "opermode=1".
Would it be worth giving this a try and then attempting the echo cancelation?
How do I pass the parameter to modprobe if the modules are already being loaded automatically?
Cheers
NTB
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Would it be worth giving this a try and then attempting the echo cancelation?
In the spirit of adventurers and shit kickers the world over - anything is worth a try - :-)
How do I pass the parameter to modprobe if the modules are already being loaded automatically?
By being fiendishly wos'name, er... clever...
Open up /etc/modprobe.conf in your favourite editor and look for the line which reads
install wcfxo /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
Gently replace it with
install wcfxo opermode=UK /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
Save it back and run...
depmod
and.... Robert's your pater's masculine sibling!
:-)
Kind Regards
Selintra
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Thanks again,
I did...
install wcfxo opermode=UK /sbin/modprobe --ignore-install wcfxo && /sbin/ztcfg
I tried with opermode=1 and opermode=UK and I still get...DAA mode is 'FCC'
Also...# fxotune -i
Tuning module /dev/zap/1
Unable to set impedance on fd 4
Failure!
Tuning module /dev/zap/2
Unable to set impedance on fd 4
Failure!
/dev/zap/3 absent: No such file or directory
/dev/zap/4 absent: No such file or directory
So, it looks like my X100ps aren't suitable for UK use.
Is all lost?
NTB :cry: :cry:
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Hi Ntb
Is all lost?
I'm afraid it doesn't look too good from here. Short of sticking them together with superglue and advertising them to Her Majesty's Navy as an anchor, we're pretty much out of ideas on this one. You can still set the gains correctly using the ztmonitor and you should be able to set echo training on. It may be OK. Doesn't hurt to give it a try.
We've just received a shipment of AX100P's to try. They aren't expensive so we'll let you know how we get on with them.
Kind Regards and commiserations
Selintra
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Been looking at AX100Ps as well but no idea of what DAAs they have.
Do you know yet?
Also, do I need to reboot / restart asterisk after setting the echocancel in /etc/asterisk/zapata.conf ?
NTBlade
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Been looking at AX100Ps as well but no idea of what DAAs they have.
Do you know yet?
Not yet - we'll let you know
Also, do I need to reboot / restart asterisk after setting the echocancel in /etc/asterisk/zapata.conf ?
Sure do. Asterisk can load most settings by just doing a reload at the console; but not for zap.
Kind Regards
Selintra
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Thanks for the reply again,
I'll leave the X100 config for now. I've purchased a VOIP account from Voiptalk and an 0845 number which I've pointed to my (IAX) IP address and forwarded port 4569 to my server. However, when I dial out the asterisk CLI gives me:
WARNING[5049]: chan_iax2.c:7075 socket_read: Call rejected by 217.14.138.49: No such context/extension
I've a feeling that maybe I've configured the trunk incorrectly.
Incoming calls seem OK apart from some echo :?
Any ideas anyone?
Thanks
NTB
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Hi NTB
Well, the good news is you are getting to Telappliant OK. Their IAX IP address is 217.14.138.49. The rejection is probably occuring because you have miscoded your voiptalk password OR, as is sometimes the case with their somewhat confusing site, you have purchased an inbound ONLY 0845 number.
If you want us to have a proper look at it then send your iax.conf file to us at admin@selintra.com.
On the inbound,you should NOT be getting echo unless you are using a softphone or low-quality handset. Telappliant are, in our opinion, the best pre-pay supplier/carrier in the UK right now and they don't give echo(they're also mates of ours which makes us somewhat biased but still... they're pretty damn good :-))
Kind Regards
Selintra
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Thanks for the help with this.
I read in another post that you were soon to offering VOIP services yourself. Wish I'd seen it sooner.!
Anyway. The site is mighty confusing. Can't seem to find much.
I wasn't aware that there was an inbound only number! WTF?
Off to bed now. I've mailed my iax.conf.
NTB
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Hi NTB,
You should try sipgate at sipgate.co.uk, they work great with SAIL/Asterisk and you get a local STD code if you are located in the UK (they check your IP) and an 0845 number if you are outside the UK. I helped my mate in the UK to set his SAIL/Asterisk server and he has a 01905 number with them, works great and sipgate to sipgate calls are of course free. It costs nothing to give them a try. Only thing to beware of is the user name you choose is NOT the user name for setting up the trunk, this caught me out :oops: The user name is in the email they send you.
Regards,
Del :pint:
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Hi Del,
I'll give sipgate a go if I'm still having problems in a couple of days. Meanwhile, I've bought call credit and an 0845 number so I'm not giving up yet.
N
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Hi NTB,
You can have more than one trunk :D Just tell your routes to use your Voiptalk for outgoing calls :wink:
Regards,
Del
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So far, the support from Selintra and here has been great . The support from Voiptalk has also been helpful...
I ran the test here, and we can make an outgoing call from the acocunt
with no problem.
but I still have the same problem.
Here's the latest response...
There should be something in your extensions.conf such as:
exten => _0[1-9].,1,Dial(IAX2/USERID@voiptalk/44${EXTEN:1})
exten => _00.,1,Dial(IAX2/USERID@voiptalk/${EXTEN:2})
If you follow the guide here you should have no problems:
http://www.voiptalk.org/products/iaxconfig
Is this something that should have been added automatically or do I have to manually add it?
Thanks
N
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Is this something that should have been added automatically or do I have to manually add it?
No, SAIL does all of this stuff on-the-fly in its AGI code. However, something isn't quite right in your case.
So far we've only seen the error message from your log. Can you do us favour? Run the call again but before you do turn on the agi debugger with
agi debug
then run your call and send us the log up to the failure. When you've finished, turn the agi debugger off with
agi no debug
If you are still getting the same error message, it does seem to suggest a problem at Telappliant rather than at your end.
Kind Regards
Selintra
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Doh!!!!
Sometimes you just have a bad day or two. Telappliant will only accept E164. Go to the trunk you created for your voiptalk account and in the transformation mask put...
00: 0:44
Job's a good 'un.
Regards, apologies and .... :oops:
Selintra
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OH YES BABY!
Thanks very much indeed for the help. Quick test to my own number identifed as my 0845 number.
Sweet! However, I still get...Peer Dunpender not found.when I click on the status icon in the trunklines panel. Not a big problem though.
Now for some experimentation...
All the best
Norrie