Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On 12/14/2013 01:29 AM, Martin wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. I thought that was fixed in the latest 9.x? Where do u download the SIP firmware usually for your Cisco IP Phones? I have a 7961 and just registered at cisco.com then logged in, did a search and was offered the firmware files for free. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1 (cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded somewhere. Version 9.x is broken, SIP only works over TCP. Where do u download the SIP firmware usually for your Cisco IP Phones? Search for cmterm-7941_7961-sip.8-3-1.zip I also have some other files here but I don't remember what was the reason for them :-( Martin Your kindly help is highly appreciated. Regards Bilal I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. --- Tato zpráva neobsahuje viry ani jiný škodlivý kód - avast! Antivirus je aktivní. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
Solve it. :) Found this link: http://www.voip-info.org/wiki/view/SCCP-HOWTO2 Cheers, Alex 2012/3/22 Alexandre Rodrigues alex...@gmail.com: Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 tooooooooooooooooo
i can make bouns mints to asterisk and elastix just give me the ips and i will add mints send the ip or host to civic_t...@yahoo.com From: Alexandre Rodrigues alex...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, March 22, 2012 6:28 PM Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk 1.8.4.2 to Hello, Facing the same problem with the following debug skinny log: -- Asked to indicate 'Stop tone' condition on channel Skinny/500@duba-23 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Received Alarm Message: 32: Name=SEP001X Load= SCCP11.8-3-4SR1S : Invalid SCCP message! : ID :83 Did you solved the problem? Thanks in advance, Alex, 2011/6/25 bilal ghayyad bilmar...@yahoo.com: Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooo in the handset and we do not hear voice (even when we dial the digits, we only hear t .. but it dials and destination answer). Also if we call to these phones, and we pickup handset of the 7942G, I am hearing too and no voice (no one hear voice .. source and destination are not hear). What about be? Is it related to skinny channel that does not work? In that case, skinny channel is not working fine and that means, I have to use SIP ! Did any one face like this problem? Another problem, if I did changes in the extensions.conf and I need to reload, then I can not reload only the extensions.conf, I have to do reload and that will cause a reset for the Phones. Any advise? Did anyone tried skinny and faced those problems? Regards Bilal wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk
Thanks a lot. OK, from where you got these files? I am trying to know the source so I can get from it any missing file that the phone is needed. Regards Bilal - bilal ghayyad wrote: Dears; The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it register but no voice can be heared). My phones are Cisco 7940s, so it may be a different layout then expected: cat dialplan.xml DIALTEMPLATE TEMPLATE MATCH=0 Timeout=2 User=Phone/ !-- Local operator-- TEMPLATE MATCH=8. Timeout=0 User=Phone/ !-- Unpark call-- TEMPLATE MATCH=9,.11 Timeout=0 User=Phone/ !-- Service numbers -- TEMPLATE MATCH=9,1.. Timeout=0 User=Phone/ !-- Long Distance -- TEMPLATE MATCH=9,... Timeout=0 User=Phone/ !-- Local numbers -- TEMPLATE MATCH=5... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=44 Timeout=0 User=Phone/ !-- Paging-- TEMPLATE MATCH=700 Timeout=0 User=Phone/ !-- Parking a call-- TEMPLATE MATCH=45 Timeout=0 User=Phone/ !-- Previous Page-- TEMPLATE MATCH=4... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=3... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=\*... Timeout=0 User=Phone/ !-- 3 digit Speed Dials -- TEMPLATE MATCH=\*.. Timeout=2 User=Phone/ !-- 2 digit Speed Dials -- TEMPLATE MATCH=\*. Timeout=1 User=Phone/ !-- 1 digit Speed Dials -- TEMPLATE MATCH=..\* Timeout=0 User=Phone/ !-- 2 digit Access Codes -- TEMPLATE MATCH=* Timeout=15/ !-- Anything else -- /DIALTEMPLATE Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk
bilal ghayyad wrote: from where you got these files? I found the link on http://www.voip-info.org I searched for Cisco 7940 http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk
Dears; The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it register but no voice can be heared). But now when we need to dial any number from the Cisco IP Phone 7942, it gives busy (the phone send the call for the asterisk just by dialing the first digit). So, do I have to place the dialing.xml file in the TFTP to be given for the Phone? Or what I have to do? I do not have this dialing.xml file, who has it? Also if possible to get any other required files other than XMLDefault.cnf.xml and SEP.cnf.xml, what do I need? Thanks for the help in advance. Regards Bilal -- You do not need sccp.conf if you are not using chan_sccp. It has different features(bugs) than chan_skinny, but yes it would also reset the phones (if it supports reload, and I have no idea if it does). Also if the phone is in a call it will not reset until after the user hangs up. Reloading the channel triggers a soft reset that causes the phone to request its configuration, which may have changed. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942 and Skinny/SIP in asterisk
bilal ghayyad wrote: Dears; The Cisco 7942 worked in SIP and did not work in skinny firmware (in skinny, it register but no voice can be heared). My phones are Cisco 7940s, so it may be a different layout then expected: cat dialplan.xml DIALTEMPLATE TEMPLATE MATCH=0 Timeout=2 User=Phone/ !-- Local operator-- TEMPLATE MATCH=8. Timeout=0 User=Phone/ !-- Unpark call-- TEMPLATE MATCH=9,.11 Timeout=0 User=Phone/ !-- Service numbers -- TEMPLATE MATCH=9,1.. Timeout=0 User=Phone/ !-- Long Distance -- TEMPLATE MATCH=9,... Timeout=0 User=Phone/ !-- Local numbers -- TEMPLATE MATCH=5... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=44 Timeout=0 User=Phone/ !-- Paging-- TEMPLATE MATCH=700Timeout=0 User=Phone/ !-- Parking a call-- TEMPLATE MATCH=45 Timeout=0 User=Phone/ !-- Previous Page-- TEMPLATE MATCH=4... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=3... Timeout=0 User=Phone/ !-- Corporate Dial plan-- TEMPLATE MATCH=\*... Timeout=0 User=Phone/ !-- 3 digit Speed Dials -- TEMPLATE MATCH=\*.. Timeout=2 User=Phone/ !-- 2 digit Speed Dials -- TEMPLATE MATCH=\*.Timeout=1 User=Phone/ !-- 1 digit Speed Dials -- TEMPLATE MATCH=..\* Timeout=0 User=Phone/ !-- 2 digit Access Codes -- TEMPLATE MATCH=* Timeout=15/ !-- Anything else -- /DIALTEMPLATE Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dear Dan; I have to do something in the compilation to have chan_sccp? Because, I do not have this channel and I have only chan_skinny. Even in the /usr/lib/asterisk/module/, I did not find chan_sccp. Maybe that is the reason why I do not have the sccp.conf file? So, using the sccp channel, will also face the same problem that the phones will restarted if I did reload? Regards Bilal --- On Mon, 6/20/11, Dan Austin dan_aus...@phoenix.com wrote: From: Dan Austin dan_aus...@phoenix.com Subject: RE: Cisco IP Phones and Skinny in asterisk To: bilal ghayyad bilmar...@yahoo.com Date: Monday, June 20, 2011, 7:09 PM It would be best to keep this on the list, I just had not had a chance to reply yet. Your first issue is just how the SCCP protocol works. Every keypress is relayed to the server, so the phones must maintain an active connection to the PBX. You can avoid this by just reloading the modules you update and not the whole PBX- ie- sip reload or module reload chan_sip The second issue is likely a firmware issue on the phone, and Likely one where the phone software is too new. You might also Not have the correct definition in skinny.conf I did use chan_sccp years ago, but have not kept up with it. The configuration should be with the source package for that channel. The configuration is similar, but you cannot rename the files as there are key differences. Dan -Original Message- From: bilal ghayyad [mailto:bilmar...@yahoo.com] Sent: Monday, June 20, 2011 3:40 PM To: Dan Austin Subject: Re: Cisco IP Phones and Skinny in asterisk Dear Dan; Because you are using skinny with your Cisco IP Phones in the office, so I beleive you might help me really to resolve my problem (please). First of all, are u using skinny channel or sccp channel? Actually, I tried skinny and I faced two major problems (so if I am going to face same problems in sccp then no need to use sccp, so please advise). The two problems that I faced them are: 1) When I do reload then the skinny channel is reloaded and that will cause a restart for the Cisco IP Phones (that are registered to skinny channel). Is the same thing happening with u when u r using sccp channel? 2) When I called the Phone, it is ringing, when we pickup the handset to answer the call, we hear t and we do not hear what source is talking and source does not hear us even .. but if we select music on hold, then caller will hear the music. Also, when we tried to use the Ciscp IP Phone to place a call, while we are dialing, the too tone is always existed and it is ringing at destination but no voice (always t). So if I used sccp then I will not face these problems? From the other side, if I need to use sccp (if we assumed the above problems are not existed) then can u please help for below: 1) If i used sccp and I gave the IP Phone the IP address TFTP server, and no configuration files were existed on TFTP, then it will register on the asterisk sccp channel? 2) The sccp.conf file, where I can find it? Is it the same as the skinny.conf file? 3) To use sccp instead of the skinny channel, all what I need is to unload the skinny from the modules.conf file and load the sccp channel in the modules.conf, and I can use the skinny.conf file for the configuration? About the firmware on the Phone, it will stay the same? I appreciate the kindly help please. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
If memory serves isn't that support contract include broken phones / parts too? I thought I read that if my phone Is broken - it is covered From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, June 20, 2011 9:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.com wrote: You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) The support contract from Cisco is only US $8.99 on CDW I really hate to link to my own blog, but I do have a post on there that details how to setup a 79x1 phone using SIP firmware with asterisk. Click the link in my signature and go to the Blog and you should be able to easily find the relevant post. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
You do not need sccp.conf if you are not using chan_sccp. It has different features(bugs) than chan_skinny, but yes it would also reset the phones (if it supports reload, and I have no idea if it does). Also if the phone is in a call it will not reset until after the user hangs up. Reloading the channel triggers a soft reset that causes the phone to request its configuration, which may have changed. Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Tuesday, June 21, 2011 1:15 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk Dear Dan; I have to do something in the compilation to have chan_sccp? Because, I do not have this channel and I have only chan_skinny. Even in the /usr/lib/asterisk/module/, I did not find chan_sccp. Maybe that is the reason why I do not have the sccp.conf file? So, using the sccp channel, will also face the same problem that the phones will restarted if I did reload? Regards Bilal --- On Mon, 6/20/11, Dan Austin dan_aus...@phoenix.com wrote: From: Dan Austin dan_aus...@phoenix.com Subject: RE: Cisco IP Phones and Skinny in asterisk To: bilal ghayyad bilmar...@yahoo.com Date: Monday, June 20, 2011, 7:09 PM It would be best to keep this on the list, I just had not had a chance to reply yet. Your first issue is just how the SCCP protocol works. Every keypress is relayed to the server, so the phones must maintain an active connection to the PBX. You can avoid this by just reloading the modules you update and not the whole PBX- ie- sip reload or module reload chan_sip The second issue is likely a firmware issue on the phone, and Likely one where the phone software is too new. You might also Not have the correct definition in skinny.conf I did use chan_sccp years ago, but have not kept up with it. The configuration should be with the source package for that channel. The configuration is similar, but you cannot rename the files as there are key differences. Dan -Original Message- From: bilal ghayyad [mailto:bilmar...@yahoo.com] Sent: Monday, June 20, 2011 3:40 PM To: Dan Austin Subject: Re: Cisco IP Phones and Skinny in asterisk Dear Dan; Because you are using skinny with your Cisco IP Phones in the office, so I beleive you might help me really to resolve my problem (please). First of all, are u using skinny channel or sccp channel? Actually, I tried skinny and I faced two major problems (so if I am going to face same problems in sccp then no need to use sccp, so please advise). The two problems that I faced them are: 1) When I do reload then the skinny channel is reloaded and that will cause a restart for the Cisco IP Phones (that are registered to skinny channel). Is the same thing happening with u when u r using sccp channel? 2) When I called the Phone, it is ringing, when we pickup the handset to answer the call, we hear t and we do not hear what source is talking and source does not hear us even .. but if we select music on hold, then caller will hear the music. Also, when we tried to use the Ciscp IP Phone to place a call, while we are dialing, the too tone is always existed and it is ringing at destination but no voice (always t). So if I used sccp then I will not face these problems? From the other side, if I need to use sccp (if we assumed the above problems are not existed) then can u please help for below: 1) If i used sccp and I gave the IP Phone the IP address TFTP server, and no configuration files were existed on TFTP, then it will register on the asterisk sccp channel? 2) The sccp.conf file, where I can find it? Is it the same as the skinny.conf file? 3) To use sccp instead of the skinny channel, all what I need is to unload the skinny from the modules.conf file and load the sccp channel in the modules.conf, and I can use the skinny.conf file for the configuration? About the firmware on the Phone, it will stay the same? I appreciate the kindly help please. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dear Warren; It look like u have a good experience in 791x series and in selecting SIP formware, so I am sure you might be able to help in the following: As u know, there are SIP firmware for Cisco phones to be used with Call Manager and other firmware to be used with Generic SIP Server (other than Cisco Unified Call Manager). Actually the firmware that start by P0S is that used for Generic SIP Server and that start by cmterm is used for Cisco Unified Call Manager. Can I understand from ur blog, that u can use the files that its name start by cmterm to make the Cisco IP Phone to be SIP image that can be used by Asterisk? In my case, as the Phones are 7942G, then there are two files are available, really I do not know the difference between them (if u can advise): cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? Appreciate your kindly help. Regards Bilal -- You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) The support contract from Cisco is only US $8.99 on CDW I really hate to link to my own blog, but I do have a post on there that details how to setup a 79x1 phone using SIP firmware with asterisk. Click the link in my signature and go to the Blog and you should be able to easily find the relevant post. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
wow I think someone needs to just spend some time reading and playing. Getting these phones working is not rocket science and there are similarities with how to do firmware / config pushes. Not to sound mean but RTFM Sent from my iPhone On Jun 21, 2011, at 7:45 PM, Warren Selby wcse...@selbytech.com wrote: On Tue, Jun 21, 2011 at 5:35 PM, bilal ghayyad bilmar...@yahoo.com wrote: Dear Warren; Please, keep all discussions to the list. There's no need to email me personally about this. snip cmterm-7942_7962-sip.9-2-1.cop.sgn (which is written that it is SIP IP Phone load) and cmterm-7942_7962-sip.9-2-1.zip which is written that it is SIP IP Phone firmware files only. So what is the difference between them the load and the firmware? The .sgn file is basically just a zip container that the Cisco Call Manager uses. You'll want to grab the zip file, extract the contents of the file into your tftp root directory. The latest firmware that I've used was 8.5.2, in which most everything I needed worked for me. I don't know specifics about the later versions of Cisco's SIP releases. Now, when I need to do the upgrade for the Phone, then I have to determine in the xml files the needed firmware? You should have, at least with firmware 8.5.2, the following files in your tftproot directory after unzipping the zip file: apps41.8-5-2TH1-9.sbn cnu41.8-5-2TH1-9.sbn cvm41sip.8-5-2TH1-9.sbn dsp41.8-5-2TH1-9.sbn jar41sip.8-5-2TH1-9.sbn SIP41.8-5-2S.loads term41.default.loads term61.default.loads XMLDefault.cnf.xml SEP[_MAC-ADDR_].cnf.xml I provide samples of the last two files on the blog post mentioned earlier. The last file, that starts with SEP, should contain the actual mac address of the phone you are trying to provision. So, for example, it would be SEP0003C9DD5624.cnf.xml, if the mac address of your phone was 0003.C9DD.5624. The example files are pretty much all you need, just go through them and change any location specific variables (such as _USER_, _IPADDR_, or _PASSWD_) to the proper values for your environment. Once you've got your tftp server setup properly with all of the appropriate config files, plug your phone in and follow the instructions at the bottom part of my blog post that explain how to get the phone reflashed to the SIP image and registered to your asterisk server. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dears; OK, I have two things now: 1) When I do reload from the asterisk CLI, then all the skinny phones are reset. This is very bad thing, how to avoid this from happening in each reload? Even if the reload will be done to take sip configuration !! 2) The line tone that is heared (the normal too tone which is heared when picking up the handset to place a call), now: while dialing the digits, I stay hear the tooo !!! It start ringing at the destination and I am still hearing the too, the destination answer the call and I am still hearing the t. How to resolve this? Please note that currently I am not giving the Phone any files from the TFTP, I just give the Phone the TFTP IP address (which takes it from the DHCP option) and it come to asterisk and register. I am able to call the extension of this Phone and it rings, but when pickup the handset to answer the call, I just hear t even the caller with me in the line and he is saying Hellooo but at Cisco Skinny Phone, I do not hear his voice, I just hear the to. Appreciate the kindly help. Regards Bilal --- The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Cisco has changed the file name format a few times, so you may want to copy xmlDefault.CNF.XML to XMLDefault.cnf.xml The more important steps is how have you configured the phone to locate the TFTP server? Are you using option 150 in DHCP, or manually setting the TFTP server address on the phone. Technically you do not need a TFTP server, since the Skinny phones will try to use the TFTP server address for registration, so you can just set that address to point to your asterisk server. A TFTP server is needed if you want custom ringtones or to manage software updates. For small setups or my home, I skipped setting up the TFTP server until I wanted to update the phone firmware. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Monday, June 20, 2011 2:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dear Stefan; First of all, I tried skinny and I faced two major problems (so if I am going to face same problems in sccp then no need to use sccp, so please advise). The two problems that I faced them are: 1) When I do reload then the skinny channel is reloaded and that will cause a restart for the Cisco IP Phones (that are registered to skinny channel). Is the same thing happening with u when u r using sccp channel? 2) When I called the Phone, it is ringing, when we pickup the handset to answer the call, we hear t and we do not hear what source is talking and source does not hear us even .. but if we select music on hold, then caller will hear the music. Also, when we tried to use the Ciscp IP Phone to place a call, while we are dialing, the too tone is always existed and it is ringing at destination but no voice (always t). So, with sccp no problem? From the other side, if I need to use sccp (if we assumed the above problems are not existed) then can u please help for below: 1) If i used sccp and I gave the IP Phone the IP address TFTP server, and no configuration files were existed on TFTP, then it will register on the asterisk sccp channel? 2) The sccp.conf file, where I can find it? Is it the same as the skinny.conf file? 3) To use sccp instead of the skinny channel, all what I need is to unload the skinny from the modules.conf file and load the sccp channel in the modules.conf, and I can use the skinny.conf file for the configuration? About the firmware on the Phone, it will stay the same? I appreciate the kindly help please. Regards Bilal --- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Where do u download the SIP firmware usually for your Cisco IP Phones? Your kindly help is highly appreciated. Regards Bilal --- I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) Sent from my iPhone On Jun 20, 2011, at 6:44 PM, bilal ghayyad bilmar...@yahoo.com wrote: If I need to use SIP, from where to get the suitable firmware for these Cisco IP Phones 7942G? Where do u download the SIP firmware usually for your Cisco IP Phones? Your kindly help is highly appreciated. Regards Bilal --- I'm using the sip firmware.. It's alright.. I feel like I'm not receiving all the features I should.. But MWI works and multiple call appearance.. On Mon, Jun 20, 2011 at 5:38 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; snip Have you thought about perhaps just flashing the phones to use the SIP firmware? -- Thanks, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On Mon, Jun 20, 2011 at 6:10 PM, Robert-iPhone rhuddles...@gmail.comwrote: You are supposed to go via cisco and support contract BUT Google is your friend (JFGI) The support contract from Cisco is only US $8.99 on CDW I really hate to link to my own blog, but I do have a post on there that details how to setup a 79x1 phone using SIP firmware with asterisk. Click the link in my signature and go to the Blog and you should be able to easily find the relevant post. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
I've no experience with that phone model or protocol. But if you run a tftp trace you'll see what files the phone is looking for. Check my old thread on pbxinaflash forums for details. i -- Original Message -- Received: 04:59 AM COT, 06/16/2011 From: bilal ghayyad bilmar...@yahoo.com To: ianworthing...@usa.net, rswago...@gmail.com, s...@open-t.co.uk, cass...@cassius.org, wcse...@selbytech.com, asterisk-users@lists.digium.com Subject: Cisco IP Phones 7942G (skinny): TFTP and required files Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith -- On 6/16/11 4:59 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Cisco has changed the file name format a few times, so you may want to copy xmlDefault.CNF.XML to XMLDefault.cnf.xml The more important steps is how have you configured the phone to locate the TFTP server? Are you using option 150 in DHCP, or manually setting the TFTP server address on the phone. Technically you do not need a TFTP server, since the Skinny phones will try to use the TFTP server address for registration, so you can just set that address to point to your asterisk server. A TFTP server is needed if you want custom ringtones or to manage software updates. For small setups or my home, I skipped setting up the TFTP server until I wanted to update the phone firmware. Dan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones 7942G (skinny): TFTP and required files
On 16/06/11 19:12, Cassius Smith wrote: Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith Agreed -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dears; OK, I start beleive that the problem in the TFTP and the files that I placed there. Now, I am using the Phone as skinny, and the files that are placed in the directory /var/lib/tftpboot/ as following: CTLSEPB8BEBF22AB62.tlv SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml Well, actually the CTLSEPB8BEBF22AB62.tlv is totally empty, so should I place any thing in it? Anyone has a format for the file CTLSEPB8BEBF22AB62.tlv? Also, what do I miss other files that the Phone needs it? From the other side, what should do about the chown and the chmod for the directory tftpboot? Appreciate the kindly help and advise. Regards Bilal - Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
2011/6/13 bilal ghayyad bilmar...@yahoo.com Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ - -- (o_ Stefan Gofferje| SCLT, MCP, CCSA //\ Reg'd Linux User #247167 | VCP #2263 V_/_ Heckler Koch - the original point and click interface -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.16 (GNU/Linux) iEYEARECAAYFAk32H6YACgkQbQKZlCdPOMPq4QCgknv5BoRc2q18JjsO/2a9Sz8O gAsAoKER5vgiSu0ro+46OhBqbXsX6Qwx =HwKD -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
On Mon, 13 Jun 2011, bilal ghayyad wrote: I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? tcpdump will show you if the tftp configuration is working. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Skinny in asterisk
I also had trouble w/ these phones at first. There was a DHCP option (?81?) you'll have to google it. The phones would not talk to tftp until I set dhcp option. The console aux cable is easy to build and VERY useful Sent from my iPhone On Jun 13, 2011, at 8:31 PM, Mark Engelhardt ma...@intuitiveengineering.com wrote: Bilal, I suggest you turn on logging on your tftp server to see what files are actually being requested, and if the the tftp server is dishing them out... Try adding a few v's to your tftp setup: File: /etc/xinetd.d/tftp Line to change: server_args = -s /tftpboot -v -v -v Look in /var/log/messages for the output. Also, I believe your 7942G has a console/aux port which is a serial port, you can learn what is happening as the phone boots up with that too. Good Luck! Mark On Jun 13, 2011, at 3:02 PM, bilal ghayyad wrote: Dears; The Asterisk version is 1.8.3.2 The Cisco IP Phone is 7942G and it is running now skinny. The used TFTP is tftp-server which is installed in fedora. I placed the following two files (but look like it was not taken from the TFTP, as nothing appeared in the messages), but I am able to to ping from the asterisk box to the vlan that the Phone is connected, so no problem in the reachability: SEPB8BEBF22AB62.cnf.xml xmlDefault.CNF.XML Are the files name correct? Or the Cisco IP Phone 7942G are not working fine with Asterisk or the tftp-server? Regards Bilal Hi All; Can anyone advise if using Cisco IP Phones Which model(s) are you planning to use ? in skinny protocol is fine or not? Or it is better to use it in SIP protocol? -- Hi, On 06/13/2011 01:04 PM, bilal ghayyad wrote: Can anyone advise if using Cisco IP Phones in skinny protocol is fine or not? Or it is better to use it in SIP protocol? SCCP works better than SIP in my opinion as there are more features. Check out http://chan-sccp-b.sourceforge.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
On 03-31-2011 01:10, bilal ghayyad wrote: I can not do the configuration from the web based of the Phone? it depends on the model. 7912 has a web ui, 79[46]0 does not. * Can I understand that working with Asterisk does not give a chance to have IP Phones with featues assigned on the buttons, so the only way to use the features is to be by access code and can not be by button? No way to assign the access code to the button? basically yes, but again, it depends on model and firmware. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. 3) As you know that it is required to have a correct username and password to login, so where to give the username and password in the Cisco IP Phone to be able to login for the SIP account? same as 1) rgds a. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
Kindly find below my notes preceded by ( * ). Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. * So I will need to have a TFTP server to place the configuration file on it and to be downloaded when the Phone is booting? I can not do the configuration from the web based of the Phone? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. * Can I understand that working with Asterisk does not give a chance to have IP Phones with featues assigned on the buttons, so the only way to use the features is to be by access code and can not be by button? No way to assign the access code to the button? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
You haven't said which model Cisco phone your working with. There are several different models and they all have different configuration options. Also, asterisk in itself doesn't have anything to do with button assignments on phones. Cisco phones tend to be harder to manipulate soft-keys than say, Polycom or Aastea phones. But no matter, whichever phone you chose, you'll likely have to do any custom button assignments in the phone's config, whether that be a file or a webapp. Thanks, --Warren Selby, dCAP On Mar 30, 2011, at 6:10 PM, bilal ghayyad bilmar...@yahoo.com wrote: Kindly find below my notes preceded by ( * ). Good morning, from the last question i assume you're looking for a SIP-based configureation. On 03-30-2011 00:16, bilal ghayyad wrote: 1) How I can assign for each button an extension? you can configure them as lines (at least in my 7940). look for linex_name, linex_authname and linex_password settings in the config file. * So I will need to have a TFTP server to place the configuration file on it and to be downloaded when the Phone is booting? I can not do the configuration from the web based of the Phone? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? AFAIK you can't reprogram the softkeys. There are two buttons which you can use for programming (well, sort of). You can define the mailbox extension which can be any extension. You can write a dial plan for a specific function and then use it as voicemail. The other button is the service button which can be programmed to access any HTTP url. I'm using mine to switch my desk lamp on or off. * Can I understand that working with Asterisk does not give a chance to have IP Phones with featues assigned on the buttons, so the only way to use the features is to be by access code and can not be by button? No way to assign the access code to the button? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and Asterisk
The answer to all of your questions are the same - the config file that you create for your phone. Thanks, --Warren Selby, dCAP On Mar 29, 2011, at 5:16 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; I need to use Cisco IP Phones with Asterisk and I have some questions to know how to use it if someone can advise: 1) How I can assign for each button an extension? 2) How I can assign for specific button a feature to be used (like call forward or call pickup .. etc)? 3) As you know that it is required to have a correct username and password to login, so where to give the username and password in the Cisco IP Phone to be able to login for the SIP account? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk
CallFwd should be one of the soft keys on your Cisco phones. Are you re-flashing the Cisco phones with SIP? -Cassius On 1/2/11 3:50 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to configure the buttons in the Cisco IP Phones to be used for different functionalities like Call Forward, Call Pickup, ... etc? For example, if I need to assign one of the buttons existed at Cisco IP Phone to be used for CallFrw, how to do this in Asterisk? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Ip phones
Hi, Michiel van Baak wrote: What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive I got mine from www.centralpoint.nl As far as I know they only deliver the phones with SCCP image. But as you can read in my previous mail this is no problem, simply install chan_sccp. If you want the phones to run SIP, you have to buy a license for the SIP image. Centralpoint has them too. My company is a cisco supplier too, maybe we can arrange some pricing strategies together. However, Cisco remains an expensive phone. Be aware, you cannot really compare delivery from any dutch supplier to what you find on Ebay. We only deal in new stock, nothing refurb, and yes, they are expensive. Florian ___ --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] Cisco Ip phones
Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap You probably should do a little reading from the wiki and past postings as there is no lack of information on this topic. Cisco and Polycom phones rank the highest in terms of overall quality by those that have been exposed to lots of sip phones. Lots of sip phones in the middle, while the most inexpensive phones tend to be rated lower quality for many different reasons. When working with non-technical people and sip phones, they tend not to like Snom's and Grandstreams (and others) due to what technical people think are silly things. Those silly things are things like: - light weight phones that slide around the desk - displays that aren't readable unless you stand up - poor display images (including letters) - function keys that are not intuitive (or don't work as expected) - buttons that are hard to press - speaker phone functions that should never have been included since they don't work in a reasonable office environment - menues that are difficult to use by non-technical users, or are layered so deep it takes time to find commonly used functions - etc, etc, etc. ___ --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] Cisco Ip phones
Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. Florian ___ --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] Cisco Ip phones
On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco Ip phones
Have you tested Aastra. Works great with * and reasoable pricing Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: den 20 september 2005 20:57 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco Ip phones
On 21:30, Tue 20 Sep 05, Anders Svensson wrote: Have you tested Aastra. Works great with * and reasoable pricing Nope, haven't seen any phone of them in real life yet. Right now we deploy snom's for the price/quality rate they deliver. I find them very stable and nice phones. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco Ip phones
We have tested this phone with a Asterisk system and deliver the phone with pre installed SIP-firmware without License What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 20:57 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 20:38, Tue 20 Sep 05, Florian Overkamp wrote: Hi Sander, Sander wrote: Hi there does any of you use ip phones from cisco on asterisk and how is the quality of this configuration ? i have to make a price of an asterisk server with 100 ip phones but i need stable phones snom is nice but still i have trouble with echo on them and budgetone is cheap and feels cheap Cisco phones work fine using SIP, good reports have also been seen with SCCP/Skinny, although my own experience on that is limited. We use SwissVoice a lot and others have reported great success with Polycom. I been using some Cisco phones for a while now. I started with converting them to SIP so they could connect to * Now with chan_sccp I reverted them all back to SCCP and they work awesome. Too bad they are so darn expensive, otherwise I wouldn't use anything else. Just my experience :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco Ip phones
I have a snom 360 installed but the woman that is operating it complains about it all the time i looked at it and sometimes when sh transfers a phonecall it will just hang and stays in the phone the snom does not have connection to the line you can only see the line is still there in the display it tells you connected i think it's something like she don't push the buttons in good enough. But they complain about many things mostly they have to look inside there company phones are ringing but nobody answers them :) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Michiel van Baak Verzonden: dinsdag 20 september 2005 22:01 Aan: asterisk-users@lists.digium.com Onderwerp: Re: [Asterisk-Users] Cisco Ip phones On 21:30, Tue 20 Sep 05, Anders Svensson wrote: Have you tested Aastra. Works great with * and reasoable pricing Nope, haven't seen any phone of them in real life yet. Right now we deploy snom's for the price/quality rate they deliver. I find them very stable and nice phones. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco Ip phones
On 22:28, Tue 20 Sep 05, Sander wrote: We have tested this phone with a Asterisk system and deliver the phone with pre installed SIP-firmware without License What about the license?? And do you have to buy a license and changing the phone to sip protocol looks scary :( and time consuming with 100 phones not all suppliers will do it for you, and does any of you know a supplier in the netherlands with good pricing neonova is way too expensive I got mine from www.centralpoint.nl As far as I know they only deliver the phones with SCCP image. But as you can read in my previous mail this is no problem, simply install chan_sccp. If you want the phones to run SIP, you have to buy a license for the SIP image. Centralpoint has them too. Changing the phones to SIP is really easy. Simply edit the lddefault.cfg so it will list the SIP image file. Put the SIP image and the lddefault.cfg file on your tftp server and every cisco rebooting will be converted to SIP. Reverting this process is the same (I just did it 3 weeks ago). Put the lddefault.cfg that states the SCCP image and the SCCP image on the tftp server and reboot the phones. I haven't tested the bigger cisco phones, but the 7905 has totally no trouble when converting from SCCP to SIP and viceversa. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called 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] Cisco IP Phones on Asterisk: chan_sip or chan_sccp
you need to pay for both sip or sccp image licences for cisco phones :-( PJ Joseph wrote: Johann Steinwendtner wrote: Hello ! I 'd like to connect Cisco IP phones to *. (7940 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? SIP will cost you an extra $100 per phone to license the SIP software. But the SIP has been working for a long time with * and is gernerally quite stable. On the other hand, SCCP comes with the phone, and the phone has many more features. However chan_sccp has not been tested heavily and is likely to have a few bugs in it. I would recommend that you set it up both ways and see for yourself. The phone definitely feels nicer in sccp. ___ 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] Cisco IP Phones on Asterisk: chan_sip or chan_sccp
Johann Steinwendtner wrote: Hello ! I 'd like to connect Cisco IP phones to *. (7940 7960) Shall I use SIP or SCCP. Which approach provides better support of features of the Cisco IP phones ? SIP will cost you an extra $100 per phone to license the SIP software. But the SIP has been working for a long time with * and is gernerally quite stable. On the other hand, SCCP comes with the phone, and the phone has many more features. However chan_sccp has not been tested heavily and is likely to have a few bugs in it. I would recommend that you set it up both ways and see for yourself. The phone definitely feels nicer in sccp. -- respectfully, Joseph ___ 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] Cisco IP Phones
On Sat, 22 Jan 2005, Mike Dent wrote: On Fri, 21 Jan 2005 19:25:06 -0500, Glenn Powers [EMAIL PROTECTED] wrote: Mike Dent wrote: What do you mean by provisioning? loading the config files, with proxy servers, usernames, passwords, etc. So basically its just a silly word for configuring? Maybe its Cisco speak? maybe its just management mumbo-jumbo :) [mixed top and bootom posting sorted out] Provisioning is a common term in communications and refers to the act of setting something up. See http://en.wikipedia.org/wiki/Provisioning. Peter ___ 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] Cisco IP Phones
Keith Burns wrote: I think you need to look at a few other factors. ... 2. Line power - Cisco uses one standard, other phones use another... but Cisco is the 900# gorilla in the powered switch market... your call... I'm curious about this point.. Most if not all vendors that support PoE are not already support 802.3af standard ? ... ___ 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] Cisco IP Phones
Cisco came up with PoE before the standard was set and so it differs. The polarity is switched, so using a dumb power injector and a crossed cable one could make it work anyway. Quoting Julio Arruda [EMAIL PROTECTED]: Keith Burns wrote: I think you need to look at a few other factors. ... 2. Line power - Cisco uses one standard, other phones use another... but Cisco is the 900# gorilla in the powered switch market... your call... I'm curious about this point.. Most if not all vendors that support PoE are not already support 802.3af standard ? ... ___ 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] Cisco IP Phones
On Sat, 22 Jan 2005 15:54:21 +0100, Kristian larsson [EMAIL PROTECTED] wrote: Cisco came up with PoE before the standard was set and so it differs. The polarity is switched, so using a dumb power injector and a crossed cable one could make it work anyway. Did you try it? and what are the pinouts for such a crossed cable? ___ 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] Cisco IP Phones
C F wrote: Did you try it? and what are the pinouts for such a crossed cable? I have been using this to power our 7940's with a 3COM injector. http://www.voip-info.org/tiki-index.php?page=Cisco%20POE Calvin ___ 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] Cisco IP Phones
Hi Glenn, What do you mean by provisioning? Thanks Mike On Fri, 21 Jan 2005 03:02:45 -0500, Glenn Powers [EMAIL PROTECTED] wrote: provision a hundred, so the process for provisioning one is going to seem a bit overwhelming. ___ 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] Cisco IP Phones
Since the provisioning can be done on the phone itself, I think what you are writing is not true. If one is using it in a SOHO environment, one can just provision it from the phone. On Fri, 21 Jan 2005 03:02:45 -0500, Glenn Powers [EMAIL PROTECTED] wrote: I'm considering put this on the voip-info.org Wiki, but I thought I'd throw it out a few observations here first: * Cisco IP Phones are designed for enterprise deployments. They are designed to be provisioned by the hundred or thousand. They are not designed to be deployed for a single user or even a small office. Sure, they work great in either of these settings, but they require more knowledge and infrastructure than most small offices have. If you're a consultant or reseller, buying one or two and spending an afternoon figuring out how to provision them makes sense. Once you know how to provision one, provisioning a hundred is not difficult. If you're an end user or a small office, you're not going to need to provision a hundred, so the process for provisioning one is going to seem a bit overwhelming. Other VoIP equipment is clearly designed for at-home installation, with web-based interfaces, etc. I think people should be aware of this when comparing IP Phone options. cheers, glenn ___ 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] Cisco IP Phones
Mike Dent wrote: Hi Glenn, What do you mean by provisioning? loading the config files, with proxy servers, usernames, passwords, etc. cheers, glenn ___ 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] Cisco IP Phones
I think you need to look at a few other factors. 1. Some IP phones are really flakey (had some serious issues with a couple of vendors MGCP Business line package). 2. Line power - Cisco uses one standard, other phones use another... but Cisco is the 900# gorilla in the powered switch market... your call... 3. Feature sets. Cisco puts a lot into their SCCP image... cos... well, its their (ok, Selsius') standard, but not a great deal into their SIP image (can anyone say 7914 ?) 4. Perception. Yep, it matters... want to put a Freedom Fries phone on a customer's desktop when they have all Cisco switches and routers... if they are so technically obtuse they need someone to put a telephony system in for them, they will probably believe the hype and want Cisco. Anyway, my 2c (and given the value of the Euro vs the USD, I guess my opinion ain't worth that much) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Glenn Powers Sent: Friday, January 21, 2005 5:25 PM To: Mike Dent; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco IP Phones Mike Dent wrote: Hi Glenn, What do you mean by provisioning? loading the config files, with proxy servers, usernames, passwords, etc. cheers, glenn ___ 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] Cisco IP phones, SIP, Call-Manager Contracts
Hell yes!!! The SIP firmware offers so much more and is better supported with * On Sat, 2004-11-13 at 06:58, Derek Conniffe wrote: Hi, There is a lot of talk about Cisco phones, SIP firmware and Contracts to download same. Does using a 7940/60 or other with SIP firmware offer better features/compatibility with Asterisk over using the [default?] Call-Manager firmware and chan_sccp? A lot of people here must have started with Call-Manager then moved on, with all the work that entails, and installed the SIP firmware - I'd love to hear someone's opinion of the difference in using the phones before after. Thanks, Derek --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.778 / Virus Database: 525 - Release Date: 15/10/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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
