Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
On Jun 25, 2004, at 8:51 PM, [EMAIL PROTECTED] wrote: I highly recommend the UIP200. Although I haven't used all the phones out there, I know it performs and sounds a lot better than the Snom 200 and Grandstream...comparable to Cisco 7960 in sound quality. We use it for our Asterisk set up in the office and it is fully compatible. Couple of down sides are: no web server and no STUN support. Outbound proxy is what you have to use if behind a home router NAT. Uniden tech support said they focus on business deployment, so mass configuration using files and TFTP server is their choice for configuring. No web server is a little bit annoying when you just want to make changes to one phone and don't feel like editing a text file, but when you need to make a common change or firmware upgrade for 20 phones, you only need to change one file and reboot all phones for them to take effect...then this is great. It'd be nice if they support both methods! STUN is said to be available in next firmware release in a couple months along with addit ional features. I'm in the process of trying to get a UIP200. I say in process because I ordered one online, but was told they are on backorder for at least two weeks. I ordered mine from here: http://www.thevoipconnection.com/store/customer/home.php?cat=253 It looks like a great phone, I just wish I could get my hands on one soon and try it out. -- Seth veritas vos liberabit Mattinen [EMAIL PROTECTED] ___ 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
[Asterisk-Users] How to transfer call in case that I am the originator
Hi, I would like to make a call and then when I am connected to the destination to transfer the call to my coleague in the office. When we receive the call it is easy using #. But when I am the originator the # doesn't work. Can you give me some suggestions? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ 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
[Asterisk-Users] Nightly / Daily, Weekend / Weekday and Holiday regime of the Asterisk
Hi, I would like to have different type of behaviour of our IP PBX depending on the time and the day: Weekday Nightly - 18:30 to 08:30 Daily - 08:30 to 18:30 Weekend, Holiday, etc. For example Daily the IP PBX will rings to some phones, nightly will work IVR system. How can I do that? Also is it possible the manner of dialing plan to be different depending of the caller using Caller ID? Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, 11 August str., No. 43, 1202 Sofia, Bulgaria ___ 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
[Asterisk-Users] Setting up your own menu like voice mail
Hi all, Anyone know where/how I can setup my own menu to work like the voicemailmain menu. e.g. extension.conf exten = 888,1,mymenusystem exten = 888,2,Goto(s,6) then somewhere mymenusystem plays message and give options to goto exten 1, 2, 3 etc Thanks in advance. Dee ___ 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] Setting up your own menu like voice mail
On Sat, 2004-06-26 at 10:09 +0100, Dee Lowndes wrote: Hi all, Anyone know where/how I can setup my own menu to work like the voicemailmain menu. e.g. extension.conf exten = 888,1,mymenusystem exten = 888,2,Goto(s,6) then somewhere mymenusystem plays message and give options to goto exten 1, 2, 3 etc http://www.voip-info.org/wiki-Asterisk+config+extensions.conf The wiki has a *lot* of good info. Use it. Regards, Gonzalo ___ 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] X101P on a UK BT line ---- txgain issue
Is opermode set via asterisk or do you need to do
modprobe wcfxs opermode=UK
chris
- Original Message -
From: Nicolas Gudino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, June 26, 2004 12:43 AM
Subject: Re: [Asterisk-Users] X101P on a UK BT line txgain issue
Hi Richard,
These complex impedances are all supported in the Silabs chips used in
both the new TDM FXO module and the FXS module, but the driver currently
sets them to 600 Ohms.
I guess at some stage a patch will appear to perhaps set these depending
on the default tonezone set in the config files.
This was submited today to CVS (answer to your prays?):
Update of /usr/cvsroot/zaptel
In directory mongoose.digium.com:/tmp/cvs-serv10293
Modified Files:
wcfxs.c
Log Message:
Add support for international impedence matching (improves echo abroad!)
Index: wcfxs.c
===
RCS file: /usr/cvsroot/zaptel/wcfxs.c,v
retrieving revision 1.73
retrieving revision 1.74
diff -u -d -r1.73 -r1.74
--- wcfxs.c 23 Jun 2004 18:24:21 - 1.73
+++ wcfxs.c 25 Jun 2004 14:34:07 - 1.74
@@ -28,7 +28,6 @@
#include linux/errno.h
#include linux/module.h
#include linux/init.h
-#include linux/usb.h
#include linux/errno.h
#include linux/pci.h
@@ -90,6 +89,95 @@
{43,LOOP_CLOSE_TRES_LOW,0x1000},
};
+static struct fxo_mode {
+ char *name;
+ int ohs;
+ int ohs2;
+ int rz;
+ int rt;
+ int ilim;
+ int dcv;
+ int mini;
+ int acim;
+} fxo_modes[] =
+{
+ { FCC, 0, 0, 0, 0, 0, 0x3, 0, 0 },/* US, Canada */
+ { TBR21, 0, 0, 0, 0, 1, 0x3, 0, 0x2 },/* Austria, Belgium,
Denmark, Finland, France, Germany,
+
Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
+
Norway, Portugal, Spain, Sweden, Switzerland, and UK */
+ { ARGENTINA, 0, 0, 0, 0, 0, 0x3, 0, 0 },
+ { AUSTRALIA, 1, 0, 0, 0, 0, 0, 0x3, 0x3 },
+ { AUSTRIA, 0, 1, 0, 0, 1, 0x3, 0, 0x3 },
+ { BAHRAIN, 0, 0, 0, 0, 1, 0x3, 0, 0x2 },
+ { BELGIUM, 0, 1, 0, 0, 1, 0x3, 0, 0x2 },
+ { BRAZIL, 0, 0, 0, 0, 0, 0, 0x3, 0 },
+ { BULGARIA, 0, 0, 0, 0, 1, 0x3, 0x0, 0x3 },
+ { CANADA, 0, 0, 0, 0, 0, 0x3, 0, 0 },
super big snip
--
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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[Asterisk-Users] problem with zaphfc
hi everybody, I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA (NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I user both phones at the same time, the sound is very, very crappy, as if it is played at a slower speed (like playing a 7'' single at 33 speed - in those old venyl days). I have not modified my NTBA with a second ohm resistors - can this be the problem? thank you! ___ 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
[Asterisk-Users] RE: NO AUDIO IN BOTH DIRECTIONS
lets see if we help Jeremy (and ourselves) to narrow down the timeframe when this problem startet. I have the following release running with the recommended pw/openh323 libs. Audio is working fine and I use faststart (must). Asterisk CVS-04/13/04-22:41:25 Does anyone have a newer release running that works with audio?. I had one myself but lost due to HD crash I think it was from May-20 but I dont have the exact date. Freddi ___ 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] Problem with music on hold...
At 16:47 25/06/2004 -0400, you wrote: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59s-mh4 (2000/Oct/27). Looks a bit old to me... I'll try to install a newer release. You need version r this is the only one that works well with asterisk ___ 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
[Asterisk-Users] IAX FWD, No authority found?
Hi Folks,
Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk
to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with
port 5036 forwarded:
$IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT
--to-destination 172.16.20.200:5036
I can make outgoing calls just fine, but when I receive an inbound call (FWD call me
service) I get the following errors from iax2 debug...
Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00088ms SCall: 2 DCall: 00109 [65.39.205.121:4569]
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 9ms SCall: 00040 DCall: 0 [65.39.205.121:4569]
VERSION : 2
CALLED NUMBER : 398xxx
CALLING NUMBER : 511
CALLING NAME: [EMAIL PROTECTED]
LANGUAGE: en
USERNAME: iaxfwd
FORMAT : 4
CAPABILITY : 4
ADSICPE : 2
DATE TIME : 148517838
Asterisk*CLI
Jun 26 12:36:10 NOTICE[98311]: chan_iax2.c:4439 socket_read: Rejected connect attempt
from 65.39.205.121
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 1ms SCall: 3 DCall: 00040 [65.39.205.121:4569]
CAUSE : No authority found
Asterisk*CLI
Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT
Timestamp: 1ms SCall: 3 DCall: 00040 [65.39.205.121:4569]
CAUSE : No authority found
Asterisk*CLI
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL
Timestamp: 0ms SCall: 00040 DCall: 3 [65.39.205.121:4569]
I've googled for this and have found pointers to problems with using call groups (I'm
not using them) and context issues. I've checked out the contexts, excerpts below...
iax.conf:
[general]
port=5036
bandwidth=high
disallow=lpc10
jitterbuffer=no
tos=lowdelay
context=from-iax
;Register with IAX Peers
;
register = 398xxx:[EMAIL PROTECTED]
[free_world_dialup]
type=user
auth=rsa
inkeys=freeworlddialup
extensions.conf
[globals]
PHONE1=SIP/1001
PHONE2=SIP/1002
FWDNUMBER=398xxx
[from-iax]
;Handle iax calls from FWD to my FWD#
exten = {FWDNUMBER},1,Dial(${PHONE1}${PHONE2},20,tr)
exten = {FWDNUMBER},2,Hangup
If anyone has any other suggestions I'd be very grateful :)
-Cheers Max.
--
Max Lock
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[Asterisk-Users] Asterisk Eating Digits
When I call a PBX system and enter digits, Asterisk is
eating away some digits. For example when I call ATT
and when the system prompts me to enter my phone
number, Asterisk eats away some digits, so ATT does
not get the number that I entered. I am using the
extensions.conf as it came from the install with some
additions. I added longdistance to the default
context. Please help!
[default]
include = mainmenu
include = longdistance
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
Thank you,
Naren
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Re: [Asterisk-Users] Problem with music on hold...
Thanks all, It's working now with version r. Francois At 16:47 25/06/2004 -0400, you wrote: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59s-mh4 (2000/Oct/27). Looks a bit old to me... I'll try to install a newer release. You need version r this is the only one that works well with asterisk ___ 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 ___ 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
[Asterisk-Users] I need DIDs in Canada and USA with roll over option
I need a provider of DIDs with multiple inbounds. regards joe baptista ___ 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] IAX FWD, No authority found?
Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call (FWD call me service) I get the following errors from iax2 debug... Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00088ms SCall: 2 DCall: 00109 [65.39.205.121:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 9ms SCall: 00040 DCall: 0 [65.39.205.121:4569] FWD uses iax2, and the proper port number to allow in your firewall is udp 4569 (as shown in the Timestamp line above), not 5036. ___ 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] FAX x Echo Cancellation
I installed a TDM04b and a TDM40b with aggressive echo suppression and it's working almost perfectly. The problem is that all extensions are fax machines and people uses it for both purposes, voice and fax. AFAIK, I cannot use aggressive suppression for fax extensions, but when I turn it off terrible echos happen. Is there any workaround for this case? The patch to correct echo went into Head cvs, and internationalization went in a few hours later. Upgrade to latest head cvs and look for echotraining=800 in config samles. I don't have a clue how it might effect fax calls. Rich ___ 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
[Asterisk-Users] Problems Compiling and Loading asterisk-oh323 0.6.2
Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make I also applied the patch that is said that is needed for openh323 1.13.5. And I get the following errors: make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.2/wrapper' make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.2/asterisk-driver' gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE - I/usr/src/asterisk/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c:660: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) chan_oh323.c:660: initializer element is not constant chan_oh323.c:660: (near initialization for `oh323_ep_list.lock') I have been sucessful before in compiling all packages before. I still have the libraries installed from the wrapper package. I decided to try and download a newer version of openh323 and pwlib, but they did not compile correctly either, so I went back to the versions that I listed above, because I knew they would compile correctly. I still have the successfully compiled and installed modules, and before attempting to upgrade to the newer versions of pwlib and openh323, I ran asterisk -. This is the error I got : [chan_oh323.so]Jun 25 13:45:13 WARNING[1024]: loader.c:242 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: __tf6PMutex Jun 25 13:45:13 WARNING[1024]: loader.c:423 load_modules: Loading module chan_oh323.so failed! So, I am wondering what is wrong and whether the packages I have built are compatible. Any help on this is greatly appreciated. -- Brian Wilkins [EMAIL PROTECTED] Heritage Communications Corporation Melbourne, FL USA 32935 ___ 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] IAX2 authentication confusion (bug 1928)
Andres wrote: I just tried this myself and it behaves as you have described it. No need to use a username. When the call comes in on the remote Asterisk, the iax.conf simply tries to match the password to any entry. The first entry with a matching password gets used. I suggest you open a bug to at least get this documented. Done, as bug 1928, although the notes for 1458 imply that Mark is aware of this issue and the code is not faulty... he wants it work this way. Personally I cannot see the value in allowing completely anonymous IAX connections, especially since they can connect as _any_ user you may have specified in your iax.conf file by just guessing the password. Granted, if your IAX users are on fixed IP addresses you can use IP-based access control, and if you can use keys then that also solves the problem even for users with dynamic IPs. However, I'd like to see some explanation of why anonymous connections are allowed to iax.conf user entries with secrets specified; at best, I would think that anonymous connections should only be allowed to user entries with _no_ secret specified. Reading way between the lines and taking an educated guess, I'd suggest the reasoning behind Mark's architectual thoughts are likely to relate to providing peer-to-peer call completion capabilities, as opposed to forcing all * systems to pass through some service-provider's-voip- switch. If implemented correctly, you control how anonymous calls are handled/allowed via contexts, and not through simple password schemes. In all liklihood, the code is probably not totally implemented as yet to achieve the objective. ___ 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] Using Soxmix on extensions.conf
Erm, wont the TIMESTAMP value have changed during the monitor ?
Don't you need to set a CALLFILENAME var, just once and re-use it.
T.
Carlos Medina wrote:
Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix
on my linux shell it works so i can mixed two calls into one
consolidated call. I want to do the process automatically since
extensions.conf but it doesnt work. My extensions.conf looks like this:
exten = 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor)
exten = 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)
exten = 407,3,Dial(SIP/407|20|t)
exten = 407,4,System(soxmix
${MONITORDIR}/${TIMESTAMP}.${CALLERIDNUM}-in.wav
${MONITORDIR}/${TIMESTAMP}.${CALLERIDNUM}-out.wav
${MONITORDIR}/${CALLERIDNUM}.wav)
exten = 407,5,Hangup
It creates the 2 files but dont do the mix between them. I dont know
what the problem is.
Thanks for your help.
Carlos Andres Medina
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Re: [Asterisk-Users] IAX2 authentication confusion (bug 1928)
Rich Adamson wrote: Reading way between the lines and taking an educated guess, I'd suggest the reasoning behind Mark's architectual thoughts are likely to relate to providing peer-to-peer call completion capabilities, as opposed to forcing all * systems to pass through some service-provider's-voip- switch. If implemented correctly, you control how anonymous calls are handled/allowed via contexts, and not through simple password schemes. In all liklihood, the code is probably not totally implemented as yet to achieve the objective. Mark's response to the bug entered explained the situation fairly well, and I have updated the IAX2 wiki page with a note about this issue. Basically, the simple solutions are: - use only RSA keys for authentication (can't be guessed) - use IP-based access control for any type=user entries in iax.conf that would provide access to services that you don't want anonymous users to be able to steal - as a last resort, provide a guest user entry in iax.conf (no secret specified), which goes to a limited context (possibly just Congestion)... Asterisk will always choose this no-secret-specified user entry first for any anonymous incoming IAX2 connections, without proposing any kind of secret match/challenge with the caller I don't see a problem with having all these options. One, or a combination, should provide everything everyone needs. I'm reviewing the current chan_iax2 code right now, and I'm going to write a new wiki page for IAX2 Authentication to document all this stuff more clearly so others don't have to figure it out the way I did :-) ___ 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] Transfer - to your own number
Hi! - method a) SetGroup() and GetGroupcount() in extensions.conf - method b) incominglimit= and outgoinglimit= in sip.conf But could you actually *prevent* the transfer? Or would you have to wait() and dial() again? I have the feeling you are looking at this the wrong way - probably the best idea is to put the secretary into her own context where he(!)/she is not allowed to call herself, by it through direct dialing or through transfer or what-have-you. Cheers, Philipp ___ 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] Nightly / Daily, Weekend / Weekday and Holiday regime of the Asterisk
Hi! I would like to have different type of behaviour of our IP PBX depending on the time and the day: Do some reading about context and how they can be included based upon the time of day: ; This includes the context daytime - timing list for includes is ; ; time range|days of week|days of month|months ; ;include = daytime|9:00-17:00|mon-fri|*|* Cheers, Philipp ___ 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
[Asterisk-Users] ZyXEL Prestige 200w - should I return it ?
Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.2' sip debug gives me this output: Sip read: REGISTER sip:192.168.1.3:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bk621e84e2b7df7d From:sip:[EMAIL PROTECTED];user=phone;tag=157CC2EBBCE4671B357 To:sip:[EMAIL PROTECTED];user=phone Call-ID:[EMAIL PROTECTED] CSeq:297 REGISTER DIGEST username=zyxel,realm=,nonce=,uri=sip:192.168.1.3,algorithm=MD5,response=da0b30f7f1093eb9d5df85ff8b1d888c User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.2 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bk621e84e2b7df7d From: sip:[EMAIL PROTECTED];user=phone;tag=157CC2EBBCE4671B357 To: sip:[EMAIL PROTECTED];user=phone;tag=as30a92f43 Call-ID: [EMAIL PROTECTED] CSeq: 297 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Problem 2: choppy sound I can dial in to asterisk demo even if the phone is not registered correctly. The sound is breaking up making it hard to hear the asterisk demo. Problem 3: What drivers are the latest It seems that there are newer drivers to the phone (http://lists.digium.com/pipermail/asterisk-users/2004-June/049265.html) But where can I get them? Problem 4: the phone keeps booting/initializing The phone boots all the time. About every 3 minutes. Problem 5: Product is not mentioned in ZyXEL support forums It seems that the product is non existing on ZyXEL web pages (http://www.zyxel.com/support/supportnote.php). A search in ZyXEL knowledgebase (http://www.zyxel.com/support/knowledgebase2.php) gives zero results on the keyword 2000w If I did not have it here in my hand I wold think that it did not exist. My setup: ZyXEL Prestige 2000w Software Version: WJ.00.07 Bootrom Version: B.00.13 Release Date: Mar 01 2004 sip.conf [zyxel] context=default type=friend host=192.168.1.3 ;tried dynamic as well dtmfmode=rfc2833 username=zyxel secret=ADG Regards Terje ___ 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
[Asterisk-Users] Echo worse after new echo patch
Hi all, I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo. There was still some occassionally that I couldn't figure out how to get rid of. I used MARK2 without the AGRESSIVE cancel option with the following zaptel.conf settings before the latest path: echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=8.0 after applying the patch my settings are the same except for: echotraining=800 Problem: I'm seeing is that echo is actually worse on the XLite SIP phones! Also, my wife is telling me that the Vtech 5.8 Ghz cordless attached to the TDM400P now actually has some echo, when it never did before. (I received the infamous, What did you do to the phone...it sounds terrible!) Any ideas or suggestions? Will the path affect the rxgain setting? Will that need to be re-adjusted? What would cause echo to show up on the TDM400P FXS phones? Is the success people are seeing with the echo patch using the AGRESSIVE suppressor? Thanks in advance, Ed Rubright ___ 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] ZyXEL Prestige 200w - should I return it ?
If this is their wireless model similar/identical to the wisip. I would suggest using g729 to get rid of the choppy sounds. Not sure what the issue is with registration off the top of my head. I had some problems with a test phone last week with wep turned on. If possible you might try testing with wep off just to rule that out as an issue. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Terje Christensen [EMAIL PROTECTED]: Hi all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.1.2' sip debug gives me this output: Sip read: REGISTER sip:192.168.1.3:5060 SIP/2.0 Via:SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bk621e84e2b7df7d From:sip:[EMAIL PROTECTED];user=phone;tag=157CC2EBBCE4671B357 To:sip:[EMAIL PROTECTED];user=phone Call-ID:[EMAIL PROTECTED] CSeq:297 REGISTER DIGEST username=zyxel,realm=,nonce=,uri=sip:192.168.1.3,algorithm=MD5,response=da0b30f7f1093eb9d5df85ff8b1d888c User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone Contact: sip:[EMAIL PROTECTED]:5060;transport=udp Expires: 3600 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.1.2 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bk621e84e2b7df7d From: sip:[EMAIL PROTECTED];user=phone;tag=157CC2EBBCE4671B357 To: sip:[EMAIL PROTECTED];user=phone;tag=as30a92f43 Call-ID: [EMAIL PROTECTED] CSeq: 297 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Problem 2: choppy sound I can dial in to asterisk demo even if the phone is not registered correctly. The sound is breaking up making it hard to hear the asterisk demo. Problem 3: What drivers are the latest It seems that there are newer drivers to the phone (http://lists.digium.com/pipermail/asterisk-users/2004-June/049265.html) But where can I get them? Problem 4: the phone keeps booting/initializing The phone boots all the time. About every 3 minutes. Problem 5: Product is not mentioned in ZyXEL support forums It seems that the product is non existing on ZyXEL web pages (http://www.zyxel.com/support/supportnote.php). A search in ZyXEL knowledgebase (http://www.zyxel.com/support/knowledgebase2.php) gives zero results on the keyword 2000w If I did not have it here in my hand I wold think that it did not exist. My setup: ZyXEL Prestige 2000w Software Version: WJ.00.07 Bootrom Version: B.00.13 Release Date: Mar 01 2004 sip.conf [zyxel] context=default type=friend host=192.168.1.3 ;tried dynamic as well dtmfmode=rfc2833 username=zyxel secret=ADG Regards Terje ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] Echo worse after new echo patch
I was excited to see the announcement on the list regarding the fix for the echo problems on Digium FXO cards! I have 2 X101P's, TDM400P with 4 FXS modules and couple of XLite softphones. A few months back,I had gone thru the recommendation on the list to remove echo from the SIP phones(I never did have any echo on the TDM400P FXS phones), and had removed about 90% of the echo. There was still some occassionally that I couldn't figure out how to get rid of. I used MARK2 without the AGRESSIVE cancel option with the following zaptel.conf settings before the latest path: echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=8.0 after applying the patch my settings are the same except for: echotraining=800 Problem: I'm seeing is that echo is actually worse on the XLite SIP phones! Also, my wife is telling me that the Vtech 5.8 Ghz cordless attached to the TDM400P now actually has some echo, when it never did before. (I received the infamous, What did you do to the phone...it sounds terrible!) Any ideas or suggestions? Will the path affect the rxgain setting? Will that need to be re-adjusted? What would cause echo to show up on the TDM400P FXS phones? Is the success people are seeing with the echo patch using the AGRESSIVE suppressor? I've not tested the x101p card with the newest patch. I'd have to guess your rxgain is set to high. For my tdm04b fxo card, I've commented out the gain adjustments from zapata.conf totally and let them default. No echo here at all. Every- thing else is stock compile (no aggressive, etc). Seems to me (but I could be wrong) that I seen some code somewhere that implied there might be some auto-gain things going on. Certainly won't hurt to comment them out and restart everything. If memory serves correctly, need to stop asterisk and reload the zaptel stuff as well. Rich ___ 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] problem with zaphfc
the problem not only occures when I use both phones - when I'm using phone 1 and annother calls knocks on for example - the sound is also not ok. any hints? I'm using a VIA C3 600 MHz with a dual-riser from VIA (make 2 PCI-slots out of one). maybe this is the problem ?! - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 26, 2004 12:52 PM Subject: [Asterisk-Users] problem with zaphfc hi everybody, I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA (NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I user both phones at the same time, the sound is very, very crappy, as if it is played at a slower speed (like playing a 7'' single at 33 speed - in those old venyl days). I have not modified my NTBA with a second ohm resistors - can this be the problem? thank you! ___ 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 ___ 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] X101P on a UK BT line ---- txgain issue
Chris Stenton wrote: Is opermode set via asterisk or do you need to do modprobe wcfxs opermode=UK You need to do modprobe wcfxs opermode=UK This will only work if you have the TDM400 FXO modules. The X101P is a 600 ohm US/JATE card only. Richard ___ 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] NO AUDIO IN BOTH DIRECTIONS
T. Chan wrote: Jeremy, any way to fix that? Thanks again. I've spent many many days trying to duplicate any of these problems and absolutely cannot. I have tried everything from my mini-itx to my celeron based laptop to my dual xeon dell 1750s and every single one of them work 100% successfully in both directions with the cvs -head and chan_h323. I've also very successfully tested interop with 5300s, Quintium A800, some multi-tech box someone in IRC let me push a few calls thru (sorry forgot your nick) and even my 7960 here now runs the H.323 load and then OpenPhone works perfectly... I simply cannot duplicate any such problems. I even manage a few different production systems with 5300s and they are running absolutely perfectly on asterisk cvs -head with chan_h323. Jeremy McNamara ___ 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] NO AUDIO IN BOTH DIRECTIONS
On Sat, 2004-06-26 at 15:27, Jeremy McNamara wrote: I even manage a few different production systems with 5300s and they are running absolutely perfectly on asterisk cvs -head with chan_h323. Can you post the config from your 5300s? -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] problem with zaphfc
What you describe as slow sound is most likely not a problem with the driver.. atleast.. very unlikely. Neither with hardware... You need proper 100ohm termination on your bus, please check that. Slow sound is a very very wierd problem... you can experience clicks, noise and/or frame (voice) drops under bad conditions (improper configuration of hardware or software). If possible try to issolate the source of the problem, using different hardware, changing asterisk version or whatever you see fit. Regards, Michael FastJack wrote: the problem not only occures when I use both phones - when I'm using phone 1 and annother calls knocks on for example - the sound is also not ok. any hints? I'm using a VIA C3 600 MHz with a dual-riser from VIA (make 2 PCI-slots out of one). maybe this is the problem ?! - Original Message - From: FastJack [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, June 26, 2004 12:52 PM Subject: [Asterisk-Users] problem with zaphfc hi everybody, I'm running my asterisk with a HFC-S card in NT-mode with a modded NTBA (NT1) (=simply crossed cable) and two ISDN-phone behind it. Now, when ever I user both phones at the same time, the sound is very, very crappy, as if it is played at a slower speed (like playing a 7'' single at 33 speed - in those old venyl days). I have not modified my NTBA with a second ohm resistors - can this be the problem? thank you! ___ 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 ___ 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 ___ 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] NO AUDIO IN BOTH DIRECTIONS
Eric Wieling wrote: Can you post the config from your 5300s? They are customer owned gateways, but I can try. Jeremy McNamara ___ 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] NO AUDIO IN BOTH DIRECTIONS
Hi, Jeremy, thanks for your help and dedication in resolving the problem. There must be something that could have caused the problem. Why don't I provide detailed information on what hardware I use and how I installed the Asterisk and I would suggest that other colleagues who had or are having this problem might want to do the same in order for Jeremy to help us. I have tried two different hardware configuration with the same result. The first Asterisk server I use a Pentium Xeon 2.4G with 512M Ram without any digium card, I use Redhat 7.3 with Kernel upgraded to 2.4.20-28.7smp, ie. enabling Hyperthreading. The second Asterisk server, I use a Pentium4 3.0G with 512M Ram with same OS version and Kernel version. I read somewhere that the system should be more stable without hyperthreading, I have tried using 2.4.20-28.7 Kernel but do not find any difference in terms of stability nor voice quality at all. I have tried many many times the following steps on both servers. 1. Get pwlib 1.5.2 and openh323 1.12.2 (ones as suggested by Jeremy) and under pwlib, do ./configure, make clean, and then make both (I even tried doing just a make opt here), and then openh323, do ./configure, make clean, and then make opt. 1. Obtain asterisk, libpri, zaptel (although I don't need without digium card) from cvs development head by doing CVS checkout asterisk libpri zaptel. Everytime when I do this step, I will erase old directories to make sure I have everything cleaned. 2. Do, make clean and make install on all directories, except that for asterisk directory, I will go in ../asterisk/channels/h323 and do a make clean and then make (without the install) before doing a make install under the asterisk directory. 3. Asterisk ready. I tried calling from another Asterisk running a January cvs into one of these servers and out to cisco (or quintum or yet another Asterisk with digium), but I got no audio on both servers. I tried calling from SJPhone into one of these servers and out to cisco or quintum or another Asterisk with digium and same thing with no audio both ends on both servers. I tried calling from cisco, passing through one of these servers and out to another cisco or endpoints above, same thing. No matter what I did, there was just no audio on both ends at all. Now, I have kept everything the same except that I changed to the stable CVS and did the same thing as described above, and I could get audio now. Jeremy, I hope that would give you some idea if I did anything wrong, and probably most colleagues out there were doing similar to what I did and have unknowingly made some mistakes somewhere. By the way, I was using a h323.conf that is practically the same as your sample. Meantime, can you tell us if you will be incorporating options like fast start, h245 tunneling, early alerting...into the driver? Thanks, and I hope you can resolve this as soon as possible, thanks again for your support. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Saturday, June 26, 2004 4:28 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS T. Chan wrote: Jeremy, any way to fix that? Thanks again. I've spent many many days trying to duplicate any of these problems and absolutely cannot. I have tried everything from my mini-itx to my celeron based laptop to my dual xeon dell 1750s and every single one of them work 100% successfully in both directions with the cvs -head and chan_h323. I've also very successfully tested interop with 5300s, Quintium A800, some multi-tech box someone in IRC let me push a few calls thru (sorry forgot your nick) and even my 7960 here now runs the H.323 load and then OpenPhone works perfectly... I simply cannot duplicate any such problems. I even manage a few different production systems with 5300s and they are running absolutely perfectly on asterisk cvs -head with chan_h323. Jeremy McNamara ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ 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] NO AUDIO IN BOTH DIRECTIONS
On Sat, 2004-06-26 at 16:32, Jeremy McNamara wrote: Eric Wieling wrote: Can you post the config from your 5300s? They are customer owned gateways, but I can try. Heck, post the Cisco configs, the chan_h323 config, and sample Dial lines. The more info the better. 8-) -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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
