[asterisk-users] cisco sip firmware update for cisco 7970
I'm trying to buy the cisco firmware update but it seems that i cannot order online because I bought my 7970 on ebay. Is there any other chance to get this update? ... anyone can make me a favour and send it to me by email? thank you ___ --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: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
there is any way to configure a 7970 without using the display, I have my LCD broken so I cannot see what I'm doing :) but the phone works fine. 2006/12/13, Paul A Brown [EMAIL PROTECTED]: Hi Is NAT set to NO? It needs to be set to NO in 8.0.3 or it just sits there at registering as you say Thanks - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 9:08 AM Subject: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2) Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --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 ___ --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 ___ --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 ___ --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
[asterisk-users] Answer a call that is not ringing on your extension
Answer a call that is not ringing on your extension. I want to pick up an external call that is ringing on another extension that is not mine. Now in my old standard pbx I press 5 and I get the call. How to do this with asterisk? thank you. ___ --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
[asterisk-users] Fwd: Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --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
[asterisk-users] SIP - IAX Attended transfer
need help on this.. I have configured all my internal extensions as SIP phones, i have one ATA-186 and one softphone. When I try to transfer calls between internalt, I use *1 as it configured on feature.conf. But my external line configured a IAX. When there is an incoming call from outside, the line 1 from my ATA186 rings. If I want to transfer my call using *1 is not possible. So I need to use flash button only when call are from IAX terminal. This is normal? I'm using Asterisk 1.2.16 on a mac osx . ___ --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
[asterisk-users] SIP - IAX Attended transfer
Need help on this.. I have configured all my internal extensions as SIP phones, i have one ATA-186 and one softphone. When I try to transfer calls between internalt, I use *1 as it configured on feature.conf. But my external line configured a IAX. When there is an incoming call from outside, the line 1 from my ATA186 rings. If I want to transfer my call using *1 is not possible. So I need to use flash button only when call are from IAX terminal. This is normal? I'm using Asterisk 1.2.16 on a mac osx . ___ --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
[asterisk-users] Asterisk 1.2.16 AIX2 - SIP Attended transfer
Need help on this issue, I have a problem, when I receive a call from IAX extension (my external DID, all incoming calls from outside), I cannot transfer my calls using atxfer = *1 That is really weird because all my SIP phones can transfer calls between them. any help? dp ___ --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] Re: Voicemail and OSX 10.4 Intel
Well, yes I'm sorry I'm using 1.2.13, all compile is ok, also I've installed mpg123. all modules loaded fine, and all codecs too. One thing keep my attention and is that when I change format for recording, it ever uses wav|wav49, I tried to change on voicemail.conf format to only gsm, but is not recording in this format. format=gsm 2006/10/27, Martin Joseph [EMAIL PROTECTED]: On 2006-10-27 09:59:10 -0700, David Parcerisa [EMAIL PROTECTED] said: Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav. I think tha could be something related to this : x=0, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav49, 0x518fe0 -- x=1, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav, 0x180a200 but I don't know what this means ... something I need to compile extra? thanyou in advance Why are you using 1.2.1? try updating to something a bit fresher like 1.2.12.1;~) I have never seen any issue with this on my mac asterisk systems so I don't think it's something extra to build. You should see these in your /usr/lib/asterisk/modules by default. Did you mess around with your module loading or your modules? You might have screwed things up that way... Dunno really, just reaching, Marty ___ --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 ___ --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
[asterisk-users] Voicemail and OSX 10.4 Intel
Hello; I have a problem with voicemail and my asterisk 1.2.1 on a OS X Mac Pro intel box. When I try to record a message from an incoming call or a greeting message from internal phone using voicemail, It's like something is not doing well. I can heard anything, only a distorsion sound that is equal to lenght of the message left. First I thoug that could be something with format=gsm|wav. I think tha could be something related to this : x=0, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav49, 0x518fe0 -- x=1, open writing: /var/spool/asterisk/voicemail/default/11/unavail format: wav, 0x180a200 but I don't know what this means ... something I need to compile extra? thanyou in advance Dp. ___ --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
