Re: [asterisk-users] Uniden UIP200 phones
Lyle Giese wrote: Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Do you mean you're using SetVar(Alert-Info: ...) instead of SIPAddHeader(Alert-Info: ...) ? Thanks, Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uniden UIP200 phones
Mojo with Horan Company, LLC wrote: Lyle Giese wrote: Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Do you mean you're using SetVar(Alert-Info: ...) instead of SIPAddHeader(Alert-Info: ...) ? Thanks, Moj I WAS using SetVar with * v1.0.x. For version 1.4.x, I had to ask what the new syntax was for the same functionality. Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uniden UIP200 phones
Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Uniden UIP200 phones
Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Thanks, Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering
On 2/7/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: Removing this line will likely fix the problem. Since you don't have a NAT,the qualify= setting doesn't help keep the port(s) open. At the same time,most SIP devices have a NAT Keep Alive option, if that is an issue. HelloIt did fix my problem, thank you for this.Wonder why this use to work with Asterisk earlier than 1.2.x ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Uniden UIP200 and Asterisk v1.2.4: problem notregistering
It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent Uniden SIP Phone p2 Ver BS4.70 for peer uip200 but after about 5s I will get something like: UIP200 is now unreachable. It appears that, for whatever reason, the packet being sent to the phones from Asterisk to check if they're still around is not being replied to. qualify=3000; send udp every 2 seconds, to keep nat Removing this line will likely fix the problem. Since you don't have a NAT, the qualify= setting doesn't help keep the port(s) open. At the same time, most SIP devices have a NAT Keep Alive option, if that is an issue. Obviously, this doesn't explain what changed to cause the issue, but this should at least have you up and running until you do figure it out. Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 Issues
Jeff Herring wrote: Phone won't register on LAN port registers but doesn't work on PC port. SIP to SIP works. Anyone have a Configuration that works out there? Phone has 4.63 Firmware Make sure you have nat=never (or nat=route). Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ --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] Uniden UIP200 Opinions
On Fri, 2005-08-05 at 13:30 -0400, Jim Feniello wrote: I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. We've deployed about 50 here. They work, mostly. Hold works (* does MOH when on hold), transfers kinda work (using the XFER button, the phone does seem to occasionally get confused afterwards tho, but * does MOH), DND and Forwarding both work. But, I would fall short of recommending them. Would really like to see the transfer problems resolved. That and the documentation is sub-par. -- David Zanetti [EMAIL PROTECTED] Team Leader, Systems Administration Catalyst IT Limited +64-4-8032233 +64-21-402260 signature.asc Description: This is a digitally signed message part ___ 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] Uniden UIP200 Opinions
I can only advise against them they are like a wallmart special. Barely work and sound is really bad. For the cost of those phones you can almost get polycoms which are a lot better in sound quality. S. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Feniello Sent: August 5, 2005 10:30 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Uniden UIP200 Opinions Hi, I've read through the archives, and wanted to get an updated opinion on the Uniden UIP200 phone. Seems like there were a lot of opinions that it was a good phone, but there were a few items that people were waiting for firmware updates for, but that was in 2004. I'm going to be using them in an office, 12 phones, on a LAN connected to an asterisk box. Thanks for any advice or opinions. -jim ___ 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] Uniden UIP200, please help
Robert Burcham wrote: I have seen no responses to my earlier post: http://lists.digium.com/pipermail/asterisk-users/2005-February/089944.html and my problem persists. Would someone please share their configs and firmware versions? I sent you an email (off-list) the other day with configs attached. If you didn't receive it ping me off-list and I'll resend. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Uniden UIP200
Lyle, Can you point me to the location on the Uniden site where I can find UIP200 firmware upgrades. Thanks, Ed McKinnon http://www.crmi.com *** REPLY SEPARATOR *** On 12/23/2004 at 9:43 AM Lyle Giese wrote: Firmware v 4.63 has been released on the Uniden website. No docs yet to explain the extras. Does anyone know how to turn off the call logs on the phone? It's very annoying in my SOHO environment as all incoming calls ring this phone(business line and personal line) and then if you pick up the call elsewhere, the phone tags it as a missed call. I have not had time to try putting the phone behind a NAT yet either. Thanks, Lyle ___ 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 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] Uniden UIP200
On Mon, 2004-27-12 at 13:34 -0800, Edward J. McKinnon wrote: Lyle, Can you point me to the location on the Uniden site where I can find UIP200 firmware upgrades. Thanks, Ed McKinnon http://bcs.uniden.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] Uniden UIP200 firmware v4.63
I had th same problem, had to finally connect the phone directly to a tftp server to get it to work. -Original Message- From: Lyle Giese [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Dec 2004 15:36:44 -0600 Subject: [Asterisk-Users] Uniden UIP200 firmware v4.63 I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Does the UIP 200 work across a nat yet? If it does, care to share your config for it? Thanks, Lyle ___ 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 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] Uniden UIP200 firmware v4.63
I have a tftp server on the local subnet and it's picking up the config files just fine. It can call out and I have two-way audio. Asterisk cann't seem to talk to the phone and therefore you cann't call the phone. That's as far as I could get with it. I played with the proxy and registrar settings on the phone and the nat= settings in sip.conf as well as the port parameter with no success. Lyle - Original Message - From: Charles S. Antrim [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Lyle Giese [EMAIL PROTECTED] Sent: Friday, December 24, 2004 5:25 PM Subject: Re: [Asterisk-Users] Uniden UIP200 firmware v4.63 I had th same problem, had to finally connect the phone directly to a tftp server to get it to work. -Original Message- From: Lyle Giese [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Dec 2004 15:36:44 -0600 Subject: [Asterisk-Users] Uniden UIP200 firmware v4.63 I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Does the UIP 200 work across a nat yet? If it does, care to share your config for it? Thanks, Lyle ___ 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 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] Uniden UIP200
Thank you! I will try again tomarow On Tue, 2004-12-14 at 14:05 -0500, Leif Madsen wrote: On Tue, 14 Dec 2004 03:22:47 -0600, Justin Carlson [EMAIL PROTECTED] wrote: If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could you please post it. Hi Justin, I am using the UIP200 here at home. Find pasted my configuration files. Note that I haven't tested everything in the file, but basic functionality (inbound and outbound calling) definately works. As does the MWI. This is based on the 4.59a firmware. Different firmwares may have different configurations? unidenMACOFPHONE.txt -- # UIP200 Mass Configuration System Mac-based File # Notes: Lines start with '#' are comments # To leave a field value unchanged (as saved on local phone), leave value to blank. # To disable a field, use '-' as value # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: BS.459a # Firmware. The items listed in this Firmware section must be in this order. # FirmwareVersion and FirmwareFileName only used if AutoFirmwareUpdate is YES # FimrwareFileName only used if FirmwareVersion differ from firmware ver in Flash AutoFirmwareUpdateYES #choices are YES and NO FirmwareFileName uip200_459aenc.pac FirmwareVersion BS4.59a # Sip Settings MyLcdDisplay 1001 MyDialNumber 1001 DisplayNameApartment 1406 UserNameForProxy 1001 PasswordForProxy 1001 UserNameForRegistrar 1001 PasswordForRegistrar 1001 # Programmable Keys. Key functionality must go before key values. ProgrammableKey1 OneTouchDial ProgrammableKey2 OneTouchDial ProgrammableKey3 OneTouchDial ProgrammableKey4 OneTouchDial ProgrammableKey5 TwoTouchDial ProgrammableKey6 DoNotDisturb ProgrammableKey7 VMA ProgrammableKey8 Mute # One and Two-touch keys. Must go after Programmable keys functionality definitions. # Refer to Programmable and Fixed Function Keys for usage guide # OneTouchKeyX value is used ONLY when ProgrammableKeyX is OneTouchDial OneTouchKey1 1000 OneTouchKey2 OneTouchKey3 OneTouchKey4 1601 OneTouchKey5 2001 OneTouchKey6 OneTouchKey7 8500 OneTouchKey8 TwoTouchDigit0 TwoTouchDigit1 TwoTouchDigit2 TwoTouchDigit3 TwoTouchDigit4 TwoTouchDigit5 TwoTouchDigit6 TwoTouchDigit7 TwoTouchDigit8 TwoTouchDigit9 # Hotline and vmwi numbers --Must be placed after OneTouchDial's HotLineNumber- VmaDirectCallNo 8500#value associating with VMA Programmable key. VmwiLampIndicatorEnable TimeDisplay Enable #end of file unidencom.txt - # UIP200 Mass Configuration System Generic File # Notes: # 1. Lines start with '#' are comments # 2. To leave a field value unchanged (as saved on local phone), leave value to blank. # 3. To set a field's value to empty, use '-' as value. # 4. To NOT overwrite user local settings of: programmable key, one/two touch keys, VMA #number, VMWILampIndicator, set OverwriteLocalSetting = NO. Default is YES. This #key will ALSO affect whether or not THESE settings in unidenMAC.txt be used. # 5. Any duplicate parameters exist in both unidencom.txt and unidenMAC.txt, MAC settings #will be used. # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: 4.59a #Overwrite user local settings of programmable keys, one/two touch keys, vma settings #If set to no, these current settings on the phone will not be overwritten. OverwriteLocalSettingsYES # must be placed on top of config file # Sip Settings --If only ProxyServer needed, set OutboundProxy1/Port same as ProxyServer/Port ProxyServer 192.168.1.1# can be an IP address or FDQN ProxyServerPort 0 # 0 to use default port OutboundProxy1192.168.1.1# can be an IP address or FQDN OutboundProxy1Port0 # enter a port number or 0 for default (5060) Registrar1192.168.1.1# can be an IP address or FQDN Registrar1Port0 # enter a port number or 0 for default (5060) RegisterExpireSec 3600 RegisterRetrySec 90 Q_Param 50 RegisterExpireLimitPercent10 FailoverRetrySec 8 SipPort 5060 SRVRecordName - #_sip._udp.unisip.com # options are ON or OFF SessionTimerSupport ON # options are ON or OFF SessionTimerRefresher ON SessionTimerMin 60 TimerInterval0300 TimerInterval1150 # Audio Settings G711MuTxPacketLength
Re: [Asterisk-Users] Uniden UIP200
On Tue, 2004-14-12 at 03:22 -0600, Justin Carlson wrote: Hello all, I have a uip200 for testing and I can't seem to get the phone to register to my * server. I have configured the unidencomm.txt and the unidenMACOFPHONE.txt files and the phone tries to register but * comes back with a 403 Forbidden message in sip debug, the phone simply displays #3 Register error. I have snom 200/190's and grandstreams, these phones we hoped could replace the (less than quality) grandstreams. If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could you please post it. The config file options depend largely on your firmware version. Here is a snippet of the important parts: unidencom.txt: -- # Sip Settings --If only ProxyServer needed, set OutboundProxy1/Port same as ProxyServer/Port ProxyServer 192.168.1.102# can be an IP address or FDQN ProxyServerPort 0 # 0 to use default port OutboundProxy1192.168.1.102# can be an IP address or FQDN OutboundProxy1Port0 # enter a port number or 0 for default (5060) Registrar1192.168.1.102# can be an IP address or FQDN Registrar1Port0 # enter a port number or 0 for default (5060) Registrar2192.168.1.102# can be an IP address or FQDN Registrar2Port0 # enter a port number or 0 for default (5060) unidenmac.txt -- # Sip Settings MyLcdDisplay 204 MyDialNumber 204 DisplayName204 UserNameForProxy 204 PasswordForProxy 1234 UserNameForRegistrar 204 PasswordForRegistrar 1234 ___ 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] Uniden UIP200
On Tue, 14 Dec 2004 03:22:47 -0600, Justin Carlson [EMAIL PROTECTED] wrote: If anyone has a working unidencomm.txt and unidenMACOFPHONE.txt file Could you please post it. Hi Justin, I am using the UIP200 here at home. Find pasted my configuration files. Note that I haven't tested everything in the file, but basic functionality (inbound and outbound calling) definately works. As does the MWI. This is based on the 4.59a firmware. Different firmwares may have different configurations? unidenMACOFPHONE.txt -- # UIP200 Mass Configuration System Mac-based File # Notes: Lines start with '#' are comments # To leave a field value unchanged (as saved on local phone), leave value to blank. # To disable a field, use '-' as value # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: BS.459a # Firmware. The items listed in this Firmware section must be in this order. # FirmwareVersion and FirmwareFileName only used if AutoFirmwareUpdate is YES # FimrwareFileName only used if FirmwareVersion differ from firmware ver in Flash AutoFirmwareUpdateYES #choices are YES and NO FirmwareFileName uip200_459aenc.pac FirmwareVersion BS4.59a # Sip Settings MyLcdDisplay 1001 MyDialNumber 1001 DisplayNameApartment 1406 UserNameForProxy 1001 PasswordForProxy 1001 UserNameForRegistrar 1001 PasswordForRegistrar 1001 # Programmable Keys. Key functionality must go before key values. ProgrammableKey1 OneTouchDial ProgrammableKey2 OneTouchDial ProgrammableKey3 OneTouchDial ProgrammableKey4 OneTouchDial ProgrammableKey5 TwoTouchDial ProgrammableKey6 DoNotDisturb ProgrammableKey7 VMA ProgrammableKey8 Mute # One and Two-touch keys. Must go after Programmable keys functionality definitions. # Refer to Programmable and Fixed Function Keys for usage guide # OneTouchKeyX value is used ONLY when ProgrammableKeyX is OneTouchDial OneTouchKey1 1000 OneTouchKey2 OneTouchKey3 OneTouchKey4 1601 OneTouchKey5 2001 OneTouchKey6 OneTouchKey7 8500 OneTouchKey8 TwoTouchDigit0 TwoTouchDigit1 TwoTouchDigit2 TwoTouchDigit3 TwoTouchDigit4 TwoTouchDigit5 TwoTouchDigit6 TwoTouchDigit7 TwoTouchDigit8 TwoTouchDigit9 # Hotline and vmwi numbers --Must be placed after OneTouchDial's HotLineNumber- VmaDirectCallNo 8500#value associating with VMA Programmable key. VmwiLampIndicatorEnable TimeDisplay Enable #end of file unidencom.txt - # UIP200 Mass Configuration System Generic File # Notes: # 1. Lines start with '#' are comments # 2. To leave a field value unchanged (as saved on local phone), leave value to blank. # 3. To set a field's value to empty, use '-' as value. # 4. To NOT overwrite user local settings of: programmable key, one/two touch keys, VMA #number, VMWILampIndicator, set OverwriteLocalSetting = NO. Default is YES. This #key will ALSO affect whether or not THESE settings in unidenMAC.txt be used. # 5. Any duplicate parameters exist in both unidencom.txt and unidenMAC.txt, MAC settings #will be used. # MAXIMUM FILE SIZE IS 10KB # Current Limitation: No spaces allowed for a setting's value # Version: 4.59a #Overwrite user local settings of programmable keys, one/two touch keys, vma settings #If set to no, these current settings on the phone will not be overwritten. OverwriteLocalSettingsYES # must be placed on top of config file # Sip Settings --If only ProxyServer needed, set OutboundProxy1/Port same as ProxyServer/Port ProxyServer 192.168.1.1# can be an IP address or FDQN ProxyServerPort 0 # 0 to use default port OutboundProxy1192.168.1.1# can be an IP address or FQDN OutboundProxy1Port0 # enter a port number or 0 for default (5060) Registrar1192.168.1.1# can be an IP address or FQDN Registrar1Port0 # enter a port number or 0 for default (5060) RegisterExpireSec 3600 RegisterRetrySec 90 Q_Param 50 RegisterExpireLimitPercent10 FailoverRetrySec 8 SipPort 5060 SRVRecordName - #_sip._udp.unisip.com # options are ON or OFF SessionTimerSupport ON # options are ON or OFF SessionTimerRefresher ON SessionTimerMin 60 TimerInterval0300 TimerInterval1150 # Audio Settings G711MuTxPacketLength 20 G711MuJitterBufferLength 10 G711MuJitterBufferMax 200 G711ATxPacketLength 20 G711AJitterBufferLength 10 G711AJitterBufferMax 200 G729TxPacketLength20 G729JitterBufferLength
Re: [Asterisk-Users] Uniden UIP200 -- configured, but not working?
On Mon, 2004-29-11 at 00:21 -0500, Ken D'Ambrosio wrote: no calls actually take place, either in-bound or out-bound. With sip debug going, I get this: The phone's firmware rev. is BS4.59a The unidenmac.txt file is as follows: MyLcdDisplay22 MyDialNumber22 UserNameForRegistrar22 PasswordForRegistrarfoo TimeDisplay Enable Try adding : UserNameForProxy 22 PasswordForProxy foo Lastly, if, in the unidencom.txt file, I put a proxy bit in, I get an honest-to-goodness busy signal, which certainly seems better than nothing. But I'm not using a proxy -- I'm going straight to the Asterisk box. I don't use a proxy either, but always define ProxyServer and OutboundProxy1 in unidencom.txt (in addition to Registrar1/2) Ryan PS: Try using Ethereal to debug problems like this. ___ 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] Uniden UIP200 -- configured, but not working?
Ryan Courtnage wrote: If it registers fine with Asterisk, then what exactly is the problem? Does the Uniden phone display an error? Asterisk? Can you make/receive calls? The firmware version and unidenmac.txt might also be relevant to the problem. Jeepers. You want a description of the actual problem? (*thwaps* self) Ummm. Okay. Sorry 'bout that. The problem is that everything seems to register fine, but no calls actually take place, either in-bound or out-bound. With sip debug going, I get this: Nov 29 00:15:41 DEBUG[337904]: chan_sip.c:757 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' The phone's firmware rev. is BS4.59a The unidenmac.txt file is as follows: MyLcdDisplay22 MyDialNumber22 UserNameForRegistrar22 PasswordForRegistrarfoo TimeDisplay Enable Lastly, if, in the unidencom.txt file, I put a proxy bit in, I get an honest-to-goodness busy signal, which certainly seems better than nothing. But I'm not using a proxy -- I'm going straight to the Asterisk box. Any ideas? Thanks again, -Ken P.S. Yes, I've looked at the uip200 WIKI. ;-) ___ 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] Uniden UIP200 -- configured, but not working?
On Fri, 2004-26-11 at 21:25 -0500, Ken D'Ambrosio wrote: Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? If it registers fine with Asterisk, then what exactly is the problem? Does the Uniden phone display an error? Asterisk? Can you make/receive calls? The firmware version and unidenmac.txt might also be relevant to the problem. Cheers -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.coalescentsystems.ca ___ 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] Uniden UIP200 -- configured, but not working?
Ken D'Ambrosio wrote: Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? [22] type=friend host=dynamic context=local-access canreinvite=no qualify=300 callerid=Uniden SIP Phone 22 mailbox=22 secret=bar nat=no Try: dtmfmode=rfc2833 (The UIP200 only support rfc2833) and: nat=never (The UIP200 does not like rfc3581 (rport)) Check out: http://www.voip-info.org/wiki-UIP200 My colleague wrote the Issues with the UIP200 and Asterisk section. Regards, -- Jason Becker Director CEO Coalescent Systems Inc. 403.244.8089 www.coalescentsystems.ca ___ 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] Uniden UIP200 configuration -- manual MIA?
look again, what you are looking for is there. -Original Message- From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:17 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] Uniden UIP200 configuration -- manual MIA? Hi, all. Got my Uniden UIP200 today (ordered from thetwistergroup.com), and was very excited to set it up... until I came to the realization that there were no docs with it whatsoever. There was, however, a sheet of paper with the stock warnings (don't use the phone in the tub, etc.), AND a URL -- for documentation. Score! Well, no. The URL it gave me, bcs.uniden.com, does indeed have docs: for the user. Nothing about the administrator, how TFTP needs to be configured, how to unlock config (an LCD panel option), etc. As far as I can tell, all of this is simply missing. Now, maybe I'm just dumb, but I'm thinking that there's an admin manual I didn't get. Anyone have any pointers? Either web-wise, or simple stuff like the password for the unlock config option? Thanks much... Ken D'Ambrosio ___ 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] Uniden UIP200 Call Waiting Hold
oi geli wrote: I am using Uniden UIP200 SIP Phone. While I was talking in one line, another call came in. I tried the to put the first call on hold. It would not put the call on hold. But I could switch between the lines with Flash. When there is one call, the hold works fine. Has anybody else found this problem? Yes. It was described in the mailing list a few weeks ago. Basically call waiting on these phones does not work. ___ 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] Uniden uip200
I moved the phone to the same subnet as the * server and I got a bit further as you indicate is the way it needs to be for now. It's giving me a #3 registration error. Could still use a couple of pointers on the uniden*.txt files as to what they really need in there. I still have something wrong in there. I have a GS 101 working, so I am not completely lost, but the lack of error messages... I turned on Sip debug and it looks like I get a lot of empty sip messages when I have the UIP200 turned on and don't really see any traffic from it in sip debug. It can pickup an ip address from a dhcp server of course and does pull down the .txt files from the tftp service I have running so it can communicate with the network. Lyle - Original Message - From: Ryan Courtnage [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Tuesday, September 21, 2004 11:44 PM Subject: Re: [Asterisk-Users] Uniden uip200 Lyle, If you are behind NAT, and * isn't, I'm afraid I have some bad news for you. According to Uniden, STUN support is a Feature Under Development. To furthur complicate things for you, the UIP200 currently does not respond (at all) to an INVITE that has 'rport' in the SIP Via field. In other words, unless you want to tweak * source code, you have to use nat=never in your sip.conf. More info here: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 BS4.59a is the latest firmware. Your best bet is to call Uniden support and open a ticket with them. I think i heard that the next firmware version is coming out mid-Oct ... if your lucky, that firmware will better support your environment. Ryan Lyle Giese wrote: I got a Uniden UIP200 and started to configure it and I am lost I have a tftp server setup on my * server and have the files unidencom.txt and unidenmac.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I have the configuration files on the tftp server setup correctly or not. Where does one put in a STUN server? What do they mean by proxy server? I tried to dial 124 and it just dropped into voicemail... Any ideas? Thanks, Lyle sip conf ;uip200 1 [124] type=friend context=local callerid=Lyle 124 username=124 secret= host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;outgoinglimit=1 ;incominglimit=1 mailbox=101 disallow=all ;allow=gsm allow=ulaw allow=alaw ;allow=g723.1 Extensions.conf exten = 124,1,Dial(SIP/124,24,Ttr) exten = 124,2,VoiceMail(u101) exten = 124,3,Hangup ___ 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 -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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] Uniden uip200
Lyle Giese wrote: Could still use a couple of pointers on the uniden*.txt files as to what they really need in there. I still have something wrong in there. I'll send you my config files in a separate email. ___ 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] Uniden uip200
Lyle, If you are behind NAT, and * isn't, I'm afraid I have some bad news for you. According to Uniden, STUN support is a Feature Under Development. To furthur complicate things for you, the UIP200 currently does not respond (at all) to an INVITE that has 'rport' in the SIP Via field. In other words, unless you want to tweak * source code, you have to use nat=never in your sip.conf. More info here: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 BS4.59a is the latest firmware. Your best bet is to call Uniden support and open a ticket with them. I think i heard that the next firmware version is coming out mid-Oct ... if your lucky, that firmware will better support your environment. Ryan Lyle Giese wrote: I got a Uniden UIP200 and started to configure it and I am lost I have a tftp server setup on my * server and have the files unidencom.txt and unidenmac.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I have the configuration files on the tftp server setup correctly or not. Where does one put in a STUN server? What do they mean by proxy server? I tried to dial 124 and it just dropped into voicemail... Any ideas? Thanks, Lyle sip conf ;uip200 1 [124] type=friend context=local callerid=Lyle 124 username=124 secret= host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;outgoinglimit=1 ;incominglimit=1 mailbox=101 disallow=all ;allow=gsm allow=ulaw allow=alaw ;allow=g723.1 Extensions.conf exten = 124,1,Dial(SIP/124,24,Ttr) exten = 124,2,VoiceMail(u101) exten = 124,3,Hangup ___ 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 -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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] Uniden uip200
Try nat=route to correct the rport issue mentioned earlier. I'm told by sources within Uniden that the firmware supporting STUN will be released soon. -Curt On Tue, 21 Sep 2004 22:44:34 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Lyle, If you are behind NAT, and * isn't, I'm afraid I have some bad news for you. According to Uniden, STUN support is a Feature Under Development. To furthur complicate things for you, the UIP200 currently does not respond (at all) to an INVITE that has 'rport' in the SIP Via field. In other words, unless you want to tweak * source code, you have to use nat=never in your sip.conf. More info here: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 BS4.59a is the latest firmware. Your best bet is to call Uniden support and open a ticket with them. I think i heard that the next firmware version is coming out mid-Oct ... if your lucky, that firmware will better support your environment. Ryan Lyle Giese wrote: I got a Uniden UIP200 and started to configure it and I am lost I have a tftp server setup on my * server and have the files unidencom.txt and unidenmac.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I have the configuration files on the tftp server setup correctly or not. Where does one put in a STUN server? What do they mean by proxy server? I tried to dial 124 and it just dropped into voicemail... Any ideas? Thanks, Lyle sip conf ;uip200 1 [124] type=friend context=local callerid=Lyle 124 username=124 secret= host=dynamic nat=yes canreinvite=no dtmfmode=rfc2833 ;outgoinglimit=1 ;incominglimit=1 mailbox=101 disallow=all ;allow=gsm allow=ulaw allow=alaw ;allow=g723.1 Extensions.conf exten = 124,1,Dial(SIP/124,24,Ttr) exten = 124,2,VoiceMail(u101) exten = 124,3,Hangup ___ 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 -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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] Uniden UIP200 Review
pbx*CLI show version Asterisk CVS-06/29/04-12:20:13 built by [EMAIL PROTECTED] on a i686 running Linux I am probably using a version of * before they fixed the first bug. Now I wonder if the phone would work behind a NAT with this update and nat=never. On Mon, 23 Aug 2004 20:35:01 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: spectro wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. Do you have nat=never with your RC2? I use CVS-D2004.06.29.15.30 Here's the bug that introduced rfc5581 support: http://bugs.digium.com/bug_view_page.php?bug_id=0001862 And here's the bug that made it optional: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 ___ 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] Uniden UIP200 Review
Do you have the UIP200 working behind nat when you have nat=never set? I've only been able to connect from the same IP range when I have nat=never set. I can connect with nat=yes from behind nat, but it won't stay connected. I'm using CVS-HEAD-08/24/04-10:03:52 Ryan Courtnage wrote: spectro wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. Do you have nat=never with your RC2? I use CVS-D2004.06.29.15.30 Here's the bug that introduced rfc5581 support: http://bugs.digium.com/bug_view_page.php?bug_id=0001862 And here's the bug that made it optional: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 ___ 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] Uniden UIP200 Review
Devon Stephens wrote: Do you have the UIP200 working behind nat when you have nat=never set? I've only been able to connect from the same IP range when I have nat=never set. I can connect with nat=yes from behind nat, but it won't stay connected. I'm using CVS-HEAD-08/24/04-10:03:52 Sorry, I can't help you there. We always use our uip200s in the same net as *. Maybe running with sip debug will reveal why you can't stay connected? ___ 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] Uniden UIP200 Review
We are running CVS-06/29/04-12:20:13 and our UIP200 works fine behind nat with nat=yes. Our asterisk is one of the last V1-0_stable releases before rc1, chan_sip.c is version 1.292.2.27. The fix for Bug 1862 added rport to the VIA header, Bug 1935 added the option nat=never so it does not send rport to phones with broken firmware (UIP200). The problem is nat=never disables NAT so the UIP200 doesn't work behind one. Maybe adding a new option to sip.conf like rport=yes/no ? This way you can say: nat=yes rport = no On Wed, 25 Aug 2004 12:02:01 -0600, Devon Stephens [EMAIL PROTECTED] wrote: Do you have the UIP200 working behind nat when you have nat=never set? I've only been able to connect from the same IP range when I have nat=never set. I can connect with nat=yes from behind nat, but it won't stay connected. I'm using CVS-HEAD-08/24/04-10:03:52 ___ 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] Uniden UIP200 Review
Maybe adding a new option to sip.conf like rport=yes/no ? This way you can say: nat=yes rport = no This sounds like a good solution to me. I have tried every combination of settings that I can think of to get the Uniden to receive incoming calls from behind a NAT with no luck (without putting an SER box between them anyway). Uniden is supposedly coming out with a new firmware one of these days with STUN support that would help as well, but since we already have put in a workaround for their phones issue we should probably modify it to support the phone behind a NAT as well. ___ 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] Uniden UIP200 Review
spectro wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. Do you have nat=never with your RC2? I use CVS-D2004.06.29.15.30 Here's the bug that introduced rfc5581 support: http://bugs.digium.com/bug_view_page.php?bug_id=0001862 And here's the bug that made it optional: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 ___ 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] Uniden UIP200 Review
spectro wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. Do you have nat=never with your RC2? I use CVS-D2004.06.29.15.30 Here's the bug that introduced rfc5581 support: http://bugs.digium.com/bug_view_page.php?bug_id=0001862 And here's the bug that made it optional: http://bugs.digium.com/bug_view_page.php?bug_id=0001935 ___ 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] Uniden UIP200 Review
On Sun, 22 Aug 2004 18:37:41 -0600, Ryan Courtnage [EMAIL PROTECTED] wrote: Tim // NCS wrote: - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). Are you running RC1 or RC2?. We are running a pre-RC1 version of * and nat=yes works fine. After we upgraded to RC2 we were unable to call the UIP200 extension. ___ 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] Uniden UIP200 Review
Tim // NCS wrote: Im new to this list, and ran across this post about a Uniden UIP200. Since its been a few months now, I was wondering how it's turned out so far. Tim, We have deployed several UIP200 phones (22 to be exact). The phone hardware is of exceptional quality, and it contains some very nice features like programmable buttons (speed-dials), headset jack, tilt-up display, 10/100 switch, and PoE support. There are, however, some firmware issues that Uniden has yet to resolve: - Audio prompts get clipped in several situations (ie: while navigating voicemail menus in voicemailmain, or using the Directory application). We've notified Uniden of this issue. Uniden has _not_ yet acknowledged this problem, however it is a common one (for uip200+asterisk users anyways) - You MUST use nat=never in sip.conf. The UIP200 does not like rfc3581 (rport), and will not reply to requests that contain it. Using 'nat=never' in sip.conf disables *'s support for this rfc. Uniden has acknowledged the issue (DR#60). - If you wish to disable call-waiting, you will need to do it at the server-side. There is a bug in the UIP200 firmware that will cause the phone to drop calls if call-waiting is disabled in the phone's config file. Uniden has acknowledged the issue (DR#61). - Even more serious, is random phone rebooting using certain uip200 firmware versions. The latest version of the firmware, 4.59a, exhibits this problem . We're forced to stick with 4.55 (which has been stable). Uniden has been notified, but has _not_ yet acknowledged the issue. Aside from the audio clipping issue, the #1 complaint we hear is the inability to cancel a consultive transfer (ie: If the person you are transferring to does not want to take the call, there is no way to return yourself to the original caller). Uniden has this item on their development road-map, but it keeps getting pushed ahead. Asterisk users are the minority of uip200 customers. With that in mind, issues that occur only in an */uip200 environment will probably not be treated as top-priority by Uniden. Despite the issues, the UIP200 is still a good value for the price, and it stands out as a winner among the similarly priced SIP phones that are currently available. Hope this helps in you decision making, -- Ryan Courtnage Director CTO Coalescent Systems Inc 403.244.8089 www.voxbox.ca ___ 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
