RE: [Asterisk-Users] WiFi Phones
Hi I have a Zyxel 2000W wifi phone, setup is easy and quick to perform. However I have found the range less that satisfactory. I have a Cisco 1200 AP and our wireless laptop devices can acccess the network fine, however the Zyxel is pretty rubbish. For example I can be 5 metres away with only a single brick wall in the way and hardly have signal. It could be this particular handset has a problem. I would be interested to see if anyone else has a similar experience or could it be my phone? Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA Sent: 07 October 2005 09:41 To: 'Andy Hamilton'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] WiFi Phones Can you try zyxel. I has graphical interface to do the configuration. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton Sent: Friday, October 07, 2005 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] WiFi Phones
Agreed, but my laptops and PDA's work fine at 25m+. It could be the particual phone I have, I think I'll box it back up this weekend and get a replacement set out as it sonly a week old! Thanks. Matt -Original Message- From: Angus Comber [mailto:[EMAIL PROTECTED] Sent: 07 October 2005 10:21 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones wireless generally struggles with brick walls. - Original Message - From: Matt Love [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, October 07, 2005 9:55 AM Subject: RE: [Asterisk-Users] WiFi Phones Hi I have a Zyxel 2000W wifi phone, setup is easy and quick to perform. However I have found the range less that satisfactory. I have a Cisco 1200 AP and our wireless laptop devices can acccess the network fine, however the Zyxel is pretty rubbish. For example I can be 5 metres away with only a single brick wall in the way and hardly have signal. It could be this particular handset has a problem. I would be interested to see if anyone else has a similar experience or could it be my phone? Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA Sent: 07 October 2005 09:41 To: 'Andy Hamilton'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] WiFi Phones Can you try zyxel. I has graphical interface to do the configuration. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Hamilton Sent: Friday, October 07, 2005 4:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WiFi Phones Anyone have good words to say about any of the WiFi handsets currently available? The UTStarCom F1000 (an 802.11b device) works pretty well. It's about half the $$$ of a Cisco 7920 (which are also pretty nice), but it seems like most of the config is done from the keypad. There is a TFTP option, but it seems that isn't quite perfect. You could check the manual (I programmed the unit without that, except to find that the default password is 88). The unit, I'm guessing, was designed somewhere in Asia, and the language translation shows it a little bit. Sound quality seems pretty good for the few calls I've passed through it. I only have one AP in my house, so I can't comment on roaming. The headset for my cell phone is stereo, and I think the phone would be most happy with a standard 3 conductor plug, but I imagine a headset on a phone is a headset on a phone. The keypad is a touch small, and sometimes I hit the wrong key (and my fingers aren't terribly fat). I also seemed to have a problem transferring calls (using the built in transfer function -- # should still work). Despite many vendors' pages saying that it does 802.1x authentication, it sure looks like WEP is the only available security option. Overall: I would recommend purchasing one, for testing at the very least. They are well priced and of good quality. Battery life seems to be pretty good, too. -A ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Reduce ring time to answer on Asterisk @Home 1.5
Hi, Is there a way to reduce the ring time to answer on the Asterisk @ Home platform. Currently it sometimes takes 3 UK Rings before asterisk picks up the call. In AMP I have Setup-General Settings- Extension of Fax machine DISABLED Below is my ZAPATA.conf file if this is any help. Thanks in advance! Matt ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes ;usecallerid=uk usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf ;Include AMP configs #include zapata_additional.conf ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] FW: channel offhook state
Hi, Yes, I had the same. Incoming calls were fine it was just when I made outgoing calls the line would sometimes hang and I would get all circuits are busy. Putting a butt (test) phone on the line in parallel indicated the line had dropped back to an on hook state, although asterisk wouldn't use it for some time. 20 mins. In the log it showed an error indicating it could not create a ZAP channel when I tried to create an outbound line. In the end I had to remove the card from the PC, run * without the card and run genzaptelconf to remove the zap-auto entries. I also removed all the outbound routing and removed by 4 ZAP trunks from the configs. I then shutdown the machine and re-installed the card and let * find the hardware and then re-ran genzaptelconf again. Im sure there is another more appropriate solution, but im an * newbie and I was clutching at straws!!! Regards Matt _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacqueline Lee Sent: 26 September 2005 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] FW: channel offhook state Has anyone else experienced the same problem, where a Zap channel gets stuck in off-hook state? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 1:45 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FW: channel offhook state -Original Message- From: Jacqueline Lee [mailto:[EMAIL PROTECTED] Sent: Friday, September 23, 2005 11:46 AM To: asterisk-users@lists.digium.com Subject: channel offhook state We are using a digium card (TDM400) with asterisk for our access to the PSTN. Initially when the server starts, all the zap channels on the card are in the onhook state. As soon as a channel is used (for inbound or outbound PSTN calls) the corresponding channel goes into offhook state, and stays in offhook state, even after the call ends; Asterisk log shows that the channel was hungup. Most of the time, the channel is still usable to make more PSTN calls, even though it shows in offhook state. Occasionally the channel becomes unusable for making PSTN calls (usually channel 1). The symptom is Asterisk and the client show the PSTN call was established, but the destination PSTN number never really receives the call. Shouldn't the channel go back to onhook state once the call hangs up? Is the persistent offhook state causing the channel to eventually become unusable? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.5/110 - Release Date: 9/22/2005 File: ATT00068.txt -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 9/23/2005 File: ATT00080.txt attachment: winmail.dat___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Unable to create ZAP channel - All circuits are busy
Hello,
I have [EMAIL PROTECTED] 1.5 installed and all is working fine for
incoming calls and sometimes outgoing calls. Installed in the box is a digium
TDM04B (4xFXO Ports)
setup as ZAP1 to
ZAP4. I have incoming calls coming in on lines 1-4 in that order and
outgoing calls prefering ZAP4 then ZAP3 then ZAP2.
When i try to dial
out to the PSTN from a SIP phone it sometimes works (normally after a reboot)
and a few calls later i get the voiceprompt "All circuits are
busy."
I know the PSTN line
is on hook and ok to use.
Looking in the log
file i can see a line saying unable to create a ZAP channel. (See extract
below)
Sep 16 07:47:46 VERBOSE[2012]: -- Executing
SetVar("SIP/200-361a", "DIAL_NUMBER=850220") in new stackSep 16 07:47:46
VERBOSE[2012]: -- Executing SetVar("SIP/200-361a", "DIAL_TRUNK=3") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Executing AGI("SIP/200-361a",
"fixlocalprefix") in new stackSep 16 07:47:46 VERBOSE[2012]: -- Launched AGI
Script /var/lib/asterisk/agi-bin/fixlocalprefixSep 16 07:47:46
VERBOSE[2012]: fixlocalprefix: Could not parse
/etc/asterisk/localprefixes.confSep 16 07:47:46 VERBOSE[2012]: -- AGI Script
fixlocalprefix completed, returning 0Sep 16 07:47:46 VERBOSE[2012]: --
Executing SetVar("SIP/200-361a", "OUTNUM=850220") in new stackSep 16
07:47:46 VERBOSE[2012]: -- Executing Cut("SIP/200-361a", "custom=OUT_3|:|1") in
new stackSep 16 07:47:46 DEBUG[2012]: _expression_ is '0'Sep 16 07:47:46
VERBOSE[2012]: -- Executing GotoIf("SIP/200-361a", "0?19") in new stackSep
16 07:47:46 DEBUG[2012]: Not taking any branchSep 16 07:47:46 VERBOSE[2012]:
-- Executing Dial("SIP/200-361a", "ZAP/2/850220") in new stackSep 16
07:47:46 NOTICE[2012]: Unable to create channel of type 'ZAP'Sep 16
07:47:46 VERBOSE[2012]: == Everyone is busy/congested at this timeSep 16
07:47:46 DEBUG[2012]: Exiting with DIALSTATUS=CHANUNAVAIL.Sep 16 07:47:46
VERBOSE[2012]: -- Executing Goto("SIP/200-361a", "s-CHANUNAVAIL|1") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Goto
(macro-dialout-trunk,s-CHANUNAVAIL,1)Sep 16 07:47:46 VERBOSE[2012]: --
Executing NoOp("SIP/200-361a", "Dial failed due to CHANUNAVAIL") in new
stackSep 16 07:47:46 VERBOSE[2012]: -- Executing Macro("SIP/200-361a",
"outisbusy") in new stackSep 16 07:47:46 VERBOSE[2012]: -- Executing
Playback("SIP/200-361a", "allison7/all-circuits-busy-now") in new stackSep
16 07:47:46 DEBUG[2012]: Ooh, format changed from unknown to ulawSep 16
07:47:46 DEBUG[2012]: Scheduling timer at 160 sample intervalsSep 16
07:47:46 VERBOSE[2012]: -- Playing 'allison7/all-circuits-busy-now' (language
'en')Sep 16 07:47:46 DEBUG[2012]: Stopping retransmission on
'[EMAIL PROTECTED]' of Response 102: FoundSep
16 07:47:47 DEBUG[2012]: Setting NAT on RTP to 0Sep 16 07:47:48 DEBUG[2012]:
Scheduling timer at 0 sample intervalsSep 16 07:47:48 VERBOSE[2012]: ==
Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/200-361a' in
macro 'outisbusy'Sep 16 07:47:48 VERBOSE[2012]: == Spawn extension
(from-internal, 9850220, 4) exited non-zero on 'SIP/200-361a'Sep 16 07:47:48
VERBOSE[2012]: -- Executing Macro("SIP/200-361a", "hangupcall") in new
stackSep 16 07:47:48 VERBOSE[2012]: -- Executing ResetCDR("SIP/200-361a",
"w") in new stackSep 16 07:47:48 DEBUG[2012]: cdr_mysql: inserting a CDR
record.
Ive tried rebooting, genzaptel,
rebuilt zaptel drivers etcDoes anyone have any suggestions as
to what i can do to debug.
Any help would be
appreciated.
Thanks
___
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RE: [Asterisk-Users] Seperate Incoming calls on TDM02?
Hi Hatton, Could you provide some examples of the config files for this. Im trying to do the same. Im confused with some of the other posts (its not hard to confuse me!) Some say its just the zapata and some say theres way more to it. I have 4 FXO ports, 2 on one number and 2 on another and want to have different incoming rules\IVR depending upon channel called. Is it as simple as changing the contexts in the zapata.conf or is there more to it. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Hatton Humphrey Sent: 16 September 2005 15:17 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Seperate Incoming calls on TDM02? Yeah, in your zapata.conf just give each channel a different context setting. It's slightly more complicated with [EMAIL PROTECTED] and/or AMP, you need to use the zapata_custom.conf file, instead. You also need to use the extensions_custom.conf file, too, though there might be a better way I don't know about. I'm fighting with this right now and I'm hitting a serious frustration point - right now all incoming calls are getting handled by the from_pstn context which is how it honestly should be according to the current conf files. However when I change the context from from_pstn to aa_1 and aa_2 respectively it doesn't change anything in the way the system is answering the lines. Hatton zapata.conf: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=no threewaycalling=no transfer=no cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf zapata_additional.conf is empty zapata-auto.conf: ; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ; Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to be a complete zapata.conf. Rather, it is intended ; to be #include-d by /etc/zapata.conf that will include the global settings ; callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 ; channel 1, WCTDM, inactive. ; channel 2, WCTDM, inactive. signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from_pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from_pstn group=0 channel = 4 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Getting email of voicemail to work
Hi, Can someone point me in the direction of getting the voicemail - Email to work on [EMAIL PROTECTED] 1.5 Ive put in the email addresses of voicemail users eg [EMAIL PROTECTED] But i cant find where to set the email server up. we have a company email server an idealy i would like to relay the mails to it. However i cant find within the AMP where to setup either the Sendmail server or put in a relay address for an external server. Can anyone help. (you may have guessed im an @ newbie) Thanks Matt ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Setting up multiple trunk groups with different internal ring groups
Hi, I have 4 analogue PSTN lines on my legacy PBX, 2 lines on one number in a rollover group ZAP1 ZAP2and 2 lines on another number ZAP3 ZAP4. Is it possible to have a group of phones ring when lines ZAP1 2 are called and a DIFFERENT set of extensions ring when ZAP3 or ZAP4 receive a caller? Any suggestions would be appreciated, or perhaps point me to the correct documentation. Im a newbie to Asterisk and have found finding documentation tricky. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
