RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems
Before I forgot, yes indeed in Nat Transversal don't use outbound proxy, etc... :) Just disable NAT in your phone config settings, and everything below should be disabled. Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)Sent: Monday, 09 May 2005 20:37To: Asterisk Users Mailing List - Non-Commercial Discussion; ThoreSubject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Alex, Asterisk does not have a Outbound SIP Proxy. Remove any Proxy configuration from your Phone. I guess that part is called Registrar Server. Omit that information here and it should work. Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander ScheerschmidtSent: Monday, May 09, 2005 2:18 PMTo: 'Thore'; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Oh yeah, i forgot, do you hav installed the latest firmware ? If not, download it and install. My config (Zyxel phone): SIP PROXY SIP URI sip: @ 10.0.0.10 : 5060 SIP Server Address SIP Server Port Registrar Server Address Registrar Server Port Register Expiry Time(sec.) OPTIONS Interval Timer Session Expiry Time(sec.) Display Name Authentication Registrar Username Registrar Password Registration Status Registered PHONE SETTINGS Default Voice Codec G.729, 8kG.711u, 64kG.711a, 64k Speaking Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 Listening Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 RTP Port Jitter Buffer Small Medium Large Voice Frames per Packet Small Medium Large DTMF Relay disableinband(RFC2833)outband DTMF Payload(0~127) Regards, Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ThoreSent: Monday, 09 May 2005 10:20To: ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Zyxel 2000W (WI-FI) Problems
Seshu, Hmm, I think here in that specific case the name "proxy" was misplaced. This is my working config (incoming and outgoing calls). I seems to be the same mistake as nat=yes / no in your sip.config. Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)Sent: Monday, 09 May 2005 20:37To: Asterisk Users Mailing List - Non-Commercial Discussion; ThoreSubject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Alex, Asterisk does not have a Outbound SIP Proxy. Remove any Proxy configuration from your Phone. I guess that part is called Registrar Server. Omit that information here and it should work. Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander ScheerschmidtSent: Monday, May 09, 2005 2:18 PMTo: 'Thore'; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Oh yeah, i forgot, do you hav installed the latest firmware ? If not, download it and install. My config (Zyxel phone): SIP PROXY SIP URI sip: @ 10.0.0.10 : 5060 SIP Server Address SIP Server Port Registrar Server Address Registrar Server Port Register Expiry Time(sec.) OPTIONS Interval Timer Session Expiry Time(sec.) Display Name Authentication Registrar Username Registrar Password Registration Status Registered PHONE SETTINGS Default Voice Codec G.729, 8kG.711u, 64kG.711a, 64k Speaking Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 Listening Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 RTP Port Jitter Buffer Small Medium Large Voice Frames per Packet Small Medium Large DTMF Relay disableinband(RFC2833)outband DTMF Payload(0~127) Regards, Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ThoreSent: Monday, 09 May 2005 10:20To: ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Zyxel 2000W (WI-FI) Problems
Alex, Asterisk does not have a Outbound SIP Proxy. Remove any Proxy configuration from your Phone. I guess that part is called Registrar Server. Omit that information here and it should work. Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander ScheerschmidtSent: Monday, May 09, 2005 2:18 PMTo: 'Thore'; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Oh yeah, i forgot, do you hav installed the latest firmware ? If not, download it and install. My config (Zyxel phone): SIP PROXY SIP URI sip: @ 10.0.0.10 : 5060 SIP Server Address SIP Server Port Registrar Server Address Registrar Server Port Register Expiry Time(sec.) OPTIONS Interval Timer Session Expiry Time(sec.) Display Name Authentication Registrar Username Registrar Password Registration Status Registered PHONE SETTINGS Default Voice Codec G.729, 8kG.711u, 64kG.711a, 64k Speaking Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 Listening Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 RTP Port Jitter Buffer Small Medium Large Voice Frames per Packet Small Medium Large DTMF Relay disableinband(RFC2833)outband DTMF Payload(0~127) Regards, Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ThoreSent: Monday, 09 May 2005 10:20To: ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] Zyxel 2000W (WI-FI) Problems
Oh yeah, i forgot, do you hav installed the latest firmware ? If not, download it and install. My config (Zyxel phone): SIP PROXY SIP URI sip: @ 10.0.0.10 : 5060 SIP Server Address SIP Server Port Registrar Server Address Registrar Server Port Register Expiry Time(sec.) OPTIONS Interval Timer Session Expiry Time(sec.) Display Name Authentication Registrar Username Registrar Password Registration Status Registered PHONE SETTINGS Default Voice Codec G.729, 8kG.711u, 64kG.711a, 64k Speaking Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 Listening Volume(-14~14) -14-13-12-11-10-9-8-7-6-5-4-3-2-101234567891011121314 RTP Port Jitter Buffer Small Medium Large Voice Frames per Packet Small Medium Large DTMF Relay disableinband(RFC2833)outband DTMF Payload(0~127) Regards, Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ThoreSent: Monday, 09 May 2005 10:20To: ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 ___ 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] Zyxel 2000W (WI-FI) Problems
Hi, This is my Zyxel P2000W config : [2000] type=friend username=myusername secret=mypassword host=dynamic context=from-sip (whatever) mailbox=100 disallow=all allow=g729 dtmfmode=rfc2833 ... it works fine for incoming and outgoing calls Regards, Alexander From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ThoreSent: Monday, 09 May 2005 10:20To: ASTERIKSSubject: [Asterisk-Users] Zyxel 2000W (WI-FI) Problems Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 ___ 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] Zyxel 2000W (WI-FI) Problems
Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729 dtmfmode=rfc2833 Sip debug: headers, 0 lines Retransmitting #4 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 to 60.64.250.254:5060 Retransmitting #5 (NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.253.149:5060;branch=z9hG4bK31323424795a48;received=60.64.250.254;rport=5060 From: ;tag=C8355813679C716AFCA To: ;tag=as3bcc72b4 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Proxy-Authenticate: Digest realm="pbx.com", nonce="1bed12f1" Content-Length: 0 ___ 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] ZyXEL 2000W
Hi, Yeah, hope so with the new release comming up. But the current firmware is already an improvement to previous releases. Now it looks already a bit more professional. Regards, Alexander. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher KennaSent: Sunday, 08 May 2005 14:38To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] ZyXEL 2000W Anyone have any luck with the ZyXEL 2000W WiFi & Call Waiting? Does this phone even support it? Chris ___ 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] ZyXEL 2000W
There is a version 2 firmware coming out the first week of next month that you can upgrade the phone to. It is supposed to address a lot of issues - Original Message - From: Christopher Kenna To: asterisk-users@lists.digium.com Sent: Sunday, May 08, 2005 7:38 AM Subject: [Asterisk-Users] ZyXEL 2000W Anyone have any luck with the ZyXEL 2000W WiFi & Call Waiting? Does this phone even support it? Chris ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] ZyXEL 2000W
Anyone have any luck with the ZyXEL 2000W WiFi & Call Waiting? Does this phone even support it? Chris ___ 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] ZyXEL 2000w unregistering and no audio
Hi all, I'm trying to get a ZyXEL 2000w (with newest firmware) working with our in-office Asterisk 1.0 server. We have other SIP phones working. I've set up the Zyxel using the web interface and having it using g711ulaw compression. The first call after restarting the phone seems to work great. After that, however, things go down hill. The phone randomly starts saying "Unregistered", and will still dial but will not play any audio, or send any audio (the line picking up a call from the 2000w doesn't hear anything). Is anyone else having this problem, or has anyone fixed this problem? Thanks, Chris -- Chris TenHarmsel Software Journeyman Atomic Object, LLC ___ 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 2000w In Call Menu/Hold configs
Hi Everyone,
After a fair amount of faffing ive managed to get the 2000w working with
asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls
over our BT line). The final solution is to set up outgoing VoIP calls but
I now know that without a SIP aware router I can think again! (damn you
iptables!)
In the mean time I'm trying to figure out why I can't get the Zyxel to give
me the hold and transfer options on the screen, thus allowing me to pass
calls around once they are inside our network. I'm expecting a few desk
phones to turn up in a few weeks, but zyxel are adamant that this thing
supports it, so does anyone know how to get it working?
I was led to believe that you would need the 't' at the end of the dial
string to enable the called party to transfer the call about, is this
correct?
My dial plan for the relevant contexts looks a little like this:
[ Context 'local-extensions' created by 'pbx_config' ]
'0' =>1. Goto(2000|1) [pbx_config]
'2001' => 1. Dial(SIP/2001|20|tr) [pbx_config]
2. Voicemail(u2001) [pbx_config]
102. Voicemail(b2001) [pbx_config]
103. Hangup() [pbx_config]
'2002' => 1. Dial(SIP/2002|20|tr) [pbx_config]
2. Voicemail(u2001) [pbx_config]
102. Voicemail(b2001) [pbx_config]
103. Hangup() [pbx_config]
[ Context 'always-out-pots' created by 'pbx_config' ]
'_9XX.' => 1. Dial(Zap/1/WW${EXTEN:1}|tr)[pbx_config]
2. Goto(102) [pbx_config]
102. Congestion() [pbx_config]
103. Hangup() [pbx_config]
[ Context 'from-analog' created by 'pbx_config' ]
'h' =>1. Hangup() [pbx_config]
'i' =>1. Hangup() [pbx_config]
's' =>1. Dial(SIP/2001&SIP/2002|45|tr) [pbx_config]
2. VoiceMail(u2001) [pbx_config]
3. Hangup() [pbx_config]
102. VoiceMail(b2001) [pbx_config]
103. Hangup() [pbx_config]
If anyone has any advice it would be appreciated
Regards,
jd
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Adelix Ltd
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Re: [Asterisk-Users] ZyXEL 2000W
Thanks Jason. I have spoken to ZyXEL support and they have also confirmed that these advertised features are not found in the phone. They claim they will have a new firmware out in the next two or three weeks that will allow the phone to hold & transfer. Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ On 20/07/2004, at 1:26 AM, Jason Williams wrote: On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager <[EMAIL PROTECTED]> wrote: Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer Regards Jason ___ 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] ZyXEL 2000W
On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager <[EMAIL PROTECTED]> wrote: > Does anyone have the call hold feature working? If you do... how did > you make it work? The instructions say to press the left button to > place the call on hold, and the right button to take it off - except > when I am in a call, these keys have no effect. > > I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 > firmware - but none work, so I'm wondering if this feature just simply > isn't implemented, or if there is likely to be something wrong with my > asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer Regards Jason ___ 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 2000W
Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. Thanks in advance, Andrew _ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568 http://www.rwts.com.au/ _ ___ 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
