Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error
On 02/14/2011 12:04 PM, James Miller wrote: I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm snip Do you actually have those files in your TFTP directory? You need both the RINGLIST.DAT file that specifies what files are available and what they are called, PLUS the actual ring files themselves. All of my Cisco ringer files are .pcm files, like ATT,pcm, ATT2.pcm, etc. Tom -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 10:31 AM, James Miller paramedi...@gmail.com wrote: I did that and this is what I got when I tried to play the 24 ringtone: 13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency ring_emergency.pcm octet That line should read something like: blah.. RRQ ring_emergency.pcm octet According to the line you send, the phone is requesting the file: Emergency ring_emergency.pcm In the ringlist.dat file in the first column I typed the display name then hit the tab key. Now on some it only moved a couple of spaces over, on others, it tabbed way over. Not sure whats going on there with that. Not sure what editor you are using, but are you certain that it is inserting Tabs, and not spaces when you hit the tab key? If you want, you can send me the file off-list and I'll take a look at it. -Jonathan -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
Yes, nothing changed EXCEPT for the software image the phone pulled down. All of the files are still in the exact same locations with the exact same names as they had in 8.9. I'm at a loss as to what's causing this issue and so apparently is Cisco given they have yet to respond to my follow up information. Regards. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. snip Do you actually have those files in your TFTP directory? You need both the RINGLIST.DAT file that specifies what files are available and what they are called, PLUS the actual ring files themselves. All of my Cisco ringer files are .pcm files, like ATT,pcm, ATT2.pcm, etc. Tom -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
Problem has been resolved with the assistance of Jonathan. Appears to be an issue with my text editors not properly tabbing the file correctly. Regards. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
Better to report a BUG to cisco. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller Sent: Monday, February 14, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. I have asterisk 1.8.2.2 installed with the FreePBX module with a 7960 just recently flashed to 8.12. Not sure what else you all may need but any help would be greatly appreciated. Respectfully, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. image001.gif-- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
That's the problem, I am not sure if the problem lies with Cisco, or if it lies with Asterisk. I figured I'd try here first before running in circles with a TAC Case. Regards. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faisal Hanif Sent: Monday, February 14, 2011 8:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Better to report a BUG to cisco. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller Sent: Monday, February 14, 2011 6:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error Sensitivity: Confidential Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. I have asterisk 1.8.2.2 installed with the FreePBX module with a 7960 just recently flashed to 8.12. Not sure what else you all may need but any help would be greatly appreciated. Respectfully, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. image001.gif-- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote: Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I’ve tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. Have you run a tcpdump on the tftp server to make sure it is requesting the correct file? It might be asking for RingList.dat, ringlist.dat, RINGLIST.DAT, etc. as capitalization seems to not be one of Cisco's concerns. (FYI, mine was RINGLIST.DAT, but I have no more 79x0's around to test with) Try running this as root on the tftp server and look for a request for the file: # tcpdump -nn port 69 -Jonathan -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm . . . However, when you attempt to play one it says Loading Ringer File but it doesn't do anything. So now it's at least seeing the file, now it just won't play them. Thanks for the help thus far! James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, February 14, 2011 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 asterisk 1.8.22 ringlist.dat error On Mon, Feb 14, 2011 at 5:40 AM, James Miller paramedi...@gmail.com wrote: Error! Filename not specified. Good Day everyone, Yesterday I upgraded the firmware on my 7960 to Sip 8.12 as provided by Cisco, however now the phone does not and will not read the RINGLIST.dat file. I've tried rebooting the phone, tried resetting the phone back to factory, have deleted the RINGLIST.dat file and reloaded the phone then reinstalled the RINGLIST.dat, and still the bloody phone will not read the file. I have not been able to locate anything in google about this kind of issue and am at a loss as to what in the world is the issue. Have you run a tcpdump on the tftp server to make sure it is requesting the correct file? It might be asking for RingList.dat, ringlist.dat, RINGLIST.DAT, etc. as capitalization seems to not be one of Cisco's concerns. (FYI, mine was RINGLIST.DAT, but I have no more 79x0's around to test with) Try running this as root on the tftp server and look for a request for the file: # tcpdump -nn port 69 -Jonathan -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
On Mon, Feb 14, 2011 at 9:04 AM, James Miller paramedi...@gmail.com wrote: I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm . You should only see the description of the file on the display. Make sure that the description and filename are tab-separated, since spaces are allowed in the description. However, when you attempt to play one it says Loading Ringer File but it doesn’t do anything. So now it’s at least seeing the file, now it just won’t play them. You can run the same command ( tcpdump -nn port 69 ) to view what file the phone is attempting to download from the tftp server. My guess is that it isn't pulling anything down or something like 24 24-ring-tone-1.raw if the file is not tab separated. -Jonathan -- _ -- 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 7960 asterisk 1.8.22 ringlist.dat error
I did that and this is what I got when I tried to play the 24 ringtone: 13:29:16.573318 IP 192.168.1.103.50849 192.168.1.60.69: 39 RRQ Emergency ring_emergency.pcm octet In the ringlist.dat file in the first column I typed the display name then hit the tab key. Now on some it only moved a couple of spaces over, on others, it tabbed way over. Not sure whats going on there with that. Thank you for your help. I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 Let us never forget our fallen men and women of the armed forces who's future's were lost protecting the future's of the free world. -Original Message- I did the command listed, and its actually requesting RINGLIST.DAT, so I changed the filename to match its request but now its showing in the ring type setting: Chirp 1 Chirp 2 24 24-ring-tone-1.raw Att1 ring_att1.pcm . You should only see the description of the file on the display. Make sure that the description and filename are tab-separated, since spaces are allowed in the description. However, when you attempt to play one it says Loading Ringer File but it doesnt do anything. So now its at least seeing the file, now it just wont play them. You can run the same command ( tcpdump -nn port 69 ) to view what file the phone is attempting to download from the tftp server. My guess is that it isn't pulling anything down or something like 24 24-ring-tone-1.raw if the file is not tab separated. -Jonathan -- _ -- 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 7960 phone can't leave a queue
Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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 7960 phone can't leave a queue
Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find the dialplan.xml? As far as I can tell, there is no TFTP server in this network. I found the IP address that the phone tries to use for TFTP (192.168.1.7 in this case) but there is nothing at that device. Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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 7960 phone can't leave a queue
It may be pulling a tftp server from dhcp, or it may just have an old config. Do all the phones (even the ones that work properly) use the same tftp address? Thanks, --Warren Selby On Jul 25, 2010, at 4:47 PM, Kevin Keane subscript...@kkeane.com wrote: Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find the dialplan.xml? As far as I can tell, there is no TFTP server in this network. I found the IP address that the phone tries to use for TFTP (192.168.1.7 in this case) but there is nothing at that device. Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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 -- _ -- 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 7960 phone can't leave a queue
I'm not sure if ALL use the same TFTP address, but I believe so. My guess is that it is actually the TFTP server that the previous phone vendor used for the phone's initial configuration before shipping it to us. So in that sense it would be an old config. Is there a way to extract the current configuration somehow to regenerate the XML? To make matters worse, I don't have the phone's admin password (it's not the default cisco). Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 1:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue It may be pulling a tftp server from dhcp, or it may just have an old config. Do all the phones (even the ones that work properly) use the same tftp address? Thanks, --Warren Selby On Jul 25, 2010, at 4:47 PM, Kevin Keane subscript...@kkeane.com wrote: Stupid question (sorry, I'm pretty much an Asterisk beginner) - where do I find the dialplan.xml? As far as I can tell, there is no TFTP server in this network. I found the IP address that the phone tries to use for TFTP (192.168.1.7 in this case) but there is nothing at that device. Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Warren Selby Sent: Sunday, July 25, 2010 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue Check your dialplan.xml file that the affected phones are loading. Thanks, --Warren Selby On Jul 25, 2010, at 10:52 AM, Kevin Keane subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause thi s! 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 -- _ -- 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 -- _ -- 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 7960 phone can't leave a queue
On Sun, Jul 25, 2010 at 9:52 AM, Kevin Keane subscript...@kkeane.comwrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I’d appreciate help with tracking down which setting might cause this! Thanks! I would also check to see if they are static members. That would explain why some can leave and some can't. Mike -- _ -- 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 7960 phone can't leave a queue
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mstults tds.net Sent: Sunday, July 25, 2010 3:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 phone can't leave a queue On Sun, Jul 25, 2010 at 9:52 AM, Kevin Keane subscript...@kkeane.commailto:subscript...@kkeane.com wrote: I recently inherited an Asterisk system (PBX in a Flash, based on Asterisk 1.4 and FreePBX). The phones are mostly Cisco 7960 phones with the SIP firmware. The Asterisk setup relies heavily on queues with dynamic agents. The problem I am having is that on SOME (but not all) the Cisco phones, the phone will not allow dialing a second *. As a result, the agent can log in to queue 600 by dialing 600* but cannot log out again with 600**. Is this due to a setting on the phone, or within Asterisk? I suspect it is on the phone, since not all devices are affected. I'd appreciate help with tracking down which setting might cause this! Thanks! I would also check to see if they are static members. That would explain why some can leave and some can't. Mike Most people indeed are static members, but this problem only affects dynamic members. It is really a dialing issue, not a queue issue (the phone won't let them dial the second *). Thanks for the thought! -- _ -- 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 7960 become UNREACHABLE behind pix firewall
Alyed wrote: From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify If you turn on *qualify* in the configuration of a SIP device in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, asterisk will send a SIP OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER functionhttp://www.voip-info.org/wiki/view/Asterisk+func+sippeer, and inversely this function will only provide status information for peers which have *qualify=yes*. My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it. Sorry, should have mentioned that all these phones have qualify=yes and nat=yes in sip.conf. Thanks. -- James On Sat, Mar 27, 2010 at 8:17 AM, James Lamanna jlama...@gmail.com wrote: Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] -- _ -- 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 7960 become UNREACHABLE behind pix firewall
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX Are you able to TFTP new phone configs? Assuming so, and it's for only 10 phones, try decreasing the registration time. I've got a 7960 on my desk and documented it with a TFTP-ready config: http://help.cloudvox.com/faqs/sip-phones/cisco-7900-ip-phone It's at the end, commented out. I don't think that config's been used much - most Cloudvox folks are just using SIP to test their AGI apps, not as primary phones. If you want another data point that still crosses your NAT boundary, feel free to sign up for and register with Cloudvox and see whether your registration lasts, using that same config. We switched to pay-as-you-go pricing, so even the free accounts include SIP. If your registrations to Cloudvox also time out, it's probably the PIX. Troy -- Cloudvox -- http://cloudvox.com/ Asterisk in the cloud -- AGI, HTTP/JSON, SIP, REST, live in minutes -- _ -- 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 7960 become UNREACHABLE behind pix firewall
On Mon, Mar 29, 2010 at 12:25 AM, Troy Davis t...@yort.com wrote: sip fixup is enabled on the PIX Try disabling the sip fixup on the PIX and see if that helps. You may have to adjust the configs on the phones themselves when you do this. -- Thanks, --Warren Selby 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 7960 become UNREACHABLE behind pix firewall
From: http://www.voip-info.org/wiki/view/Asterisk+sip+qualify If you turn on *qualify* in the configuration of a SIP device in sip.confhttp://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, Asterisk will send a SIP OPTIONShttp://www.voip-info.org/wiki/view/SIP+method+optionscommand regularly to check that the device is still online. If the device does not answer within the configured (or default) period (in ms) Asterisk considers the device off-line for future calls. This status can be checked by the SIPPEER functionhttp://www.voip-info.org/wiki/view/Asterisk+func+sippeer, and inversely this function will only provide status information for peers which have *qualify=yes*. My guess is that your Nat/firewall is closing the connection after some time the phone is idle, so this way Asterisk will make sure to always have communication going trhough that connection so your NAT/firewall won't just close it. try playing with qualifyfreq as well. Let us know if it helped. Alyed 2010/3/27 James Lamanna jlama...@gmail.com Hi, I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall. After some period of time, asterisk says that some of them are unreachable, and the phones lose their registration. The only way to make the phones recover is to clear the NAT translation tables for the phones on the PIX (clear xlate...) Does anyone know how to fix this? As you can imagine, it is quite annoying. And it does not happen to all the phones either. sip fixup is enabled on the PIX phone config parts: nat_enable : 1 nat_received_processing : 0 nat_address: [public ip of PIX] Thank you. -- James (Please CC me on all replies) -- _ -- 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 7960 Multiline phone
Hi You could also do it with one extension but set the call limit for the extension in the sip.conf to something like call-limit=3 Which would allow 3 concurrent calls to the one extension Ish Jimmy Ezell wrote: Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomming4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Tuesday, August 11, 2009 10:05 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can’t register to one ‘extension’ afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jimmy Ezell *Sent:* Tuesday, August 11, 2009 12:18 PM *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. *From:* Jimmy Ezell *Sent:* Tuesday, August 11, 2009 9:15 AM *To:* 'asterisk-users@lists.digium.com' *Subject:* Cisco 1760 Multiline phone I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file = [incomming1] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming2] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming3] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 === *Jimmy Ezell** *Assistant IT Manager *(408) 487-2200** * http://www.hmhca.com/ * * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
D you are a genius! Thank you very much, this does exactly what I want. Worked like a charm. Just a little extra information for the archive. I changed my PhoneMacAddress.cnf file .cnf to have the phone configuration lines listed in D's post. I also changed my extensions.conf file as he suggested. I changed my sip.conf file to have a single section for all of the extensions: [incoming] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 Thanks again, Jimmy From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of D Tucny Sent: Tuesday, August 11, 2009 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Multiline phone With that phone what you really probably want to do is just configure them all with the same details... i.e. # Line 1 appearance line1_name: incoming line1_shortname: Incoming (Line1) line1_authname: incoming line1_password: password # Line 2 appearance line2_name: incoming line2_shortname: Incoming (Line2) line2_authname: incoming line2_password: password # Line 3 appearance line3_name: incoming line3_shortname: incoming (Line3) line3_authname: incoming line3_password: password # Line 4 appearance line4_name: incoming line4_shortname: Incoming (Line4) line4_authname: incoming line4_password: password # Line 5 appearance line5_name: incoming line5_shortname: incoming (Line5) line5_authname: incoming line5_password: password # Line 6 appearance line5_name: 102 line5_shortname: Ext. 102 (Line1) line5_authname: 102 line5_password: password in the phone config file... Then, in extensions.conf exten = workhours,1,Dial(SIP/incoming) exten = workhours,n,Voicemail(100,u) ... The phone will only actually register multiple times for 'incoming' though asterisk just handles that and calls to 'incoming' will come through on the lowest available line and show as call waiting with an 'Answer' soft key allowing the next call to be answered placing the current call on hold... Seems to be exactly what you want... d 2009/8/12 Jimmy Ezell jez...@hmhca.com Sorry for not being real clear. What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? Jimmy Ezell From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 Multiline phone Jimmy, To clarify, you want to configure the phones like this where p means phone and l means logical line: Phone 1: P1l1 P1l2 P1l3 Phone 2: P2l1 P2l2 P2l3 Phone 3: P3l1 P3l2 P3l3 It sounds like (and looks like) you're dialing
Re: [asterisk-users] Cisco 7960 Multiline phone
Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomm ing4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 10:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can't register to one 'extension' afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. From: Jimmy Ezell Sent: Tuesday, August 11, 2009 9:15 AM To: 'asterisk-users@lists.digium.com' Subject: Cisco 1760 Multiline phone I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file = [incomming1] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming2] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming3] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 === Jimmy Ezell Assistant IT Manager (408) 487-2200 http://www.hmhca.com/ image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
Jimmy, To clarify, you want to configure the phones like this where p means phone and l means logical line: Phone 1: P1l1 P1l2 P1l3 Phone 2: P2l1 P2l2 P2l3 Phone 3: P3l1 P3l2 P3l3 It sounds like (and looks like) you're dialing all of the extensions on one phone at the same time, which is why they're ringing and ringing. What you want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1) in the dial command to ring them simultaneously. asterisk will then fail through if none of the phones answer in time. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Multiline phone Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomming4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 10:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can't register to one 'extension' afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. From: Jimmy Ezell Sent: Tuesday, August 11, 2009 9:15 AM To: 'asterisk-users@lists.digium.com' Subject: Cisco 1760 Multiline phone I have a cisco 1760 phone running sip and I need to configure for our receptionist so that she can answer calls on more then one extension. What is the easiest way to configure this so that incomming calls go to the next availble extension? Does each extension on the phone need to be set seperately in the sip.conf file (see below for my example)? sip.conf file = [incomming1] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming2] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 [incomming3] type=friend context=internal host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100 === Jimmy Ezell Assistant IT Manager (408) 487-2200 [cid:image001.jpg@01CA1A99.E2624550]http://www.hmhca.com/ inline: image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
Sorry for not being real clear. What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? Jimmy Ezell From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 Multiline phone Jimmy, To clarify, you want to configure the phones like this where p means phone and l means logical line: Phone 1: P1l1 P1l2 P1l3 Phone 2: P2l1 P2l2 P2l3 Phone 3: P3l1 P3l2 P3l3 It sounds like (and looks like) you're dialing all of the extensions on one phone at the same time, which is why they're ringing and ringing. What you want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1) in the dial command to ring them simultaneously. asterisk will then fail through if none of the phones answer in time. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 3:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 Multiline phone Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomm ing4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: Tuesday, August 11, 2009 10:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can't register to one 'extension' afaik) and ring them in order: 1,1,Dial(SIP/xx) 1,n,Dial(SIP/xx1) 1,n,Dial(SIP/xx2) Or ring them at the same time: 1,1,Dial(SIP/xxSIP/xx1SIP/xx2) Someone else may have better solution though. -Dave From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jimmy Ezell Sent: Tuesday, August 11, 2009 12:18 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Cisco 1760 Multiline phone Sorry I mean to say cisco 7960 phone. From: Jimmy Ezell
Re: [asterisk-users] Cisco 7960 Multiline phone
What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? You could probably use DEVICE_STATE to check the status first : http://www.voip-info.org/wiki/view/Asterisk+func+device_State hth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Multiline phone
With that phone what you really probably want to do is just configure them all with the same details... i.e. # Line 1 appearance line1_name: incoming line1_shortname: Incoming (Line1) line1_authname: incoming line1_password: password # Line 2 appearance line2_name: incoming line2_shortname: Incoming (Line2) line2_authname: incoming line2_password: password # Line 3 appearance line3_name: incoming line3_shortname: incoming (Line3) line3_authname: incoming line3_password: password # Line 4 appearance line4_name: incoming line4_shortname: Incoming (Line4) line4_authname: incoming line4_password: password # Line 5 appearance line5_name: incoming line5_shortname: incoming (Line5) line5_authname: incoming line5_password: password # Line 6 appearance line5_name: 102 line5_shortname: Ext. 102 (Line1) line5_authname: 102 line5_password: password in the phone config file... Then, in extensions.conf exten = workhours,1,Dial(SIP/incoming) exten = workhours,n,Voicemail(100,u) ... The phone will only actually register multiple times for 'incoming' though asterisk just handles that and calls to 'incoming' will come through on the lowest available line and show as call waiting with an 'Answer' soft key allowing the next call to be answered placing the current call on hold... Seems to be exactly what you want... d 2009/8/12 Jimmy Ezell jez...@hmhca.com Sorry for not being real clear. What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? *Jimmy Ezell * -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Tuesday, August 11, 2009 12:39 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone Jimmy, To clarify, you want to configure the phones like this where p means phone and l means logical line: Phone 1: P1l1 P1l2 P1l3 Phone 2: P2l1 P2l2 P2l3 Phone 3: P3l1 P3l2 P3l3 It sounds like (and looks like) you’re dialing all of the extensions on one phone at the same time, which is why they’re ringing and ringing. What you want to do is place the extensions for line 1 of each phone (p1l1,p2l1,p3l1) in the dial command to ring them simultaneously. asterisk will then fail through if none of the phones answer in time. -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jimmy Ezell *Sent:* Tuesday, August 11, 2009 3:05 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Cisco 7960 Multiline phone Thanks for the help, I really appreciate the feedback. I tried ringing them all at the same time as you suggested: exten = workhours,1,Dial(SIP/incomming1SIP/incomming2SIP/incomming3SIP/incomming4SIP/incomming5) but it does very strange stuff: - I have to push the extension button twice to answer. - More then one extension shows off hook at the same time (Maybe 2 or 3 of the 5 will show off hook on the phone) - When I hang up the phone starts to ring again even though there is no caller I tried ringing them in order: exten = workhours,1,Dial(SIP/incomming1,5,r) exten = workhours,n,Dial(SIP/incomming2,5,r) exten = workhours,n,Dial(SIP/incomming3,5,r) exten = workhours,n,Dial(SIP/incomming4,5,r) exten = workhours,n,Dial(SIP/incomming5,5,r) exten = workhours,n,Macro(voicemail,100) Now I see the call march along each of the extensions until it gets to the end goes to voice mail. What I really want is for the call to go to only one of the unused lines and then fall straight through to voicemail after the timeout. Anyone have some thoughts on getting it to work that way? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Tuesday, August 11, 2009 10:05 AM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Cisco 1760 Multiline phone Yes each extension needs to be configured separately in the cisco CNF file. I use a distinct extension on each phone (2 phones can’t register to one ‘extension’ afaik) and ring them in order: 1,1,Dial(SIP/xx
Re: [asterisk-users] Cisco 7960 Multiline phone
On Tue, Aug 11, 2009 at 5:12 PM, Jimmy Ezell jez...@hmhca.com wrote: Sorry for not being real clear. What I have is 1 front desk phone only with 6 lines Front Desk Phone line 1 - incoming extension 1 Front Desk Phone line 2 - incoming extension 2 Front Desk Phone line 3 - incoming extension 3 Front Desk Phone line 4 - incoming extension 4 Front Desk Phone line 5 - incoming extension 5 Front Desk Phone line 6 - inside office extension If incoming line 1 is busy I want the next incoming call to come in on line 2. If incoming line 2 and 3 are busy but 1 is free the next call should got to line 1. So lines 1 and 2 might get a lot of calls but only on really busy days will calls make it up to lines 4 and 5. Does that make sense? Anyone have the solution? *Jimmy Ezell* What is the purpose of having the incoming lines show as different line appearances if you are just going to use them as a hunt group? I as because the easiest solution is to route all the incoming calls to the same line appearance. Each line can have multiple calls coming in at the same time, and you can handle easily using the soft buttons. Then you could reuse the extra buttons as speed dials, other specific extensions (i.e. the boss' DID) etc. -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CIsco 7960 + asterisk: hepl needed
Thanks Guys, I managed to get it working the problem was NAT; in the sip.conf [general] nat=yes however in the SIPMAC.cnf there was nothing about NAT. It took me a while to spot it since both asterisk and phone were in same network and I did not think about NAT. Solutions: 1) add in sip.conf in [55] nat=no OR 2) add in SIPDefault : nat_enable: 1 nat_address: nat_received_processing: 1 Thanks again Tuesday, July 28, 2009, 11:14:52 PM, you wrote: Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing 55 phone icon with x so it looks like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI sip show peers ... 55/55 (Unspecified)D N 5060 UNKNOWN ... What am I missing? Best regards, Lukasz sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid=Lukasz Cisco 7960 55 canreinvite=no host=dynamic dtmfmode=rfc2833 qualify=200 mailbox=55 SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;unicast sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIPMAC.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: Lukasz Cisco7960; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Castle ; add a space at the end, looks neater phone_password: cisco ; Limited to 31 characters (Default - cisco) user_info: none telnet_level: 2 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, pepesz76mailto:pepes...@o2.pl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CIsco 7960 + asterisk: hepl needed
Hello Mark, I managed to make it work - see my previous post Since you have those phones - does: voip_control_port: 5060 start_media_port: 1 end_media_port: 10050 works in your case? I tried to put those in SIPDefault but looks like the phone ignores those and always says: start media port 16384 end media port 32766 Any ideas? Thanks :) Thursday, July 30, 2009, 3:21:54 AM, you wrote: Pepesz, Did you get your 7960 working? We have about 40 of them running. your config looks ok. I can compare it to my setup tomorrow. look at your sip config on the phone and see if it matches what you expect. also check the status on the phone and see if there are any errors. try quotes around the password. Mark On Jul 28, 2009, at 5:14 PM, pepesz76 wrote: Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing 55 phone icon with x so it looks like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI sip show peers ... 55/55 (Unspecified)D N 5060 UNKNOWN ... What am I missing? Best regards, Lukasz sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid=Lukasz Cisco 7960 55 canreinvite=no host=dynamic dtmfmode=rfc2833 qualify=200 mailbox=55 SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;unicast sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIPMAC.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: Lukasz Cisco7960; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Castle ; add a space at the end, looks neater phone_password: cisco ; Limited to 31 characters (Default - cisco) user_info: none telnet_level: 2 -- Best regards, pepesz76mailto:pepes...@o2.pl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CIsco 7960 + asterisk: hepl needed
Are there any other phones registered, or is it just this phone that is having issues? The first thing that I see is the qualify=200 line, and I have not had good experience with Cisco devices and any qualify setting. I would try leaving that out. I also have double quotes around the line1_* parameters. See my comments inline. On Tue, Jul 28, 2009 at 2:14 PM, pepesz76 pepes...@o2.pl wrote: Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing 55 phone icon with x so it looks like the phone is not registered. The phone and the asterisk are in the same local network. On asterisk side: Cawdor*CLI sip show peers ... 55/55 (Unspecified)D N 5060 UNKNOWN ... sip.conf: [55] type=friend defaultuser=55 secret=12345655 context=home_castle callerid=Lukasz Cisco 7960 55 canreinvite=no host=dynamic dtmfmode=rfc2833 Remove: qualify=200 Add: disallow=all allow=ulaw (Or whatever codecs you are using) buggymwi=yes SIPDefault.cnf: image_version: P0S3-8-12-00 proxy1_address: 192.168.1.109 ; Can be dotted IP or FQDN proxy_register: 1 messages_uri: 80 phone_password: cisco ; Limited to 31 characters (Default - cisco) sntp_mode: directedbroadcast ;unicast sntp_server: 192.168.1.77 time_zone: GMT+01/00 ; assuming you're in GMT time_format_24hr: 1 ; to show the time in 24hour format date_format: D/M/Y ; format you would like the date in dial_template: dialplan SIPMAC.cnf: image_version: P0S3-8-12-00 line1_name: 55 line1_name: 55 line1_authname: 55 line1_password: 12345655 line1_shortname: 55 line1_displayname: Lukasz Cisco7960 line1_authname: 55 line1_shortname: 55 ; displayed on the phones softkey line1_password: 12345655 line1_displayname: Lukasz Cisco7960; the caller id proxy1_port: 5060 proxy1_address: 192.168.1.109 # Phone Label (Text desired to be displayed in upper right corner) phone_label: Castle ; add a space at the end, looks neater phone_password: cisco ; Limited to 31 characters (Default - cisco) user_info: none telnet_level: 2 If that still doesn't work, then telnet into the phone and see what is going on. Commands like show config show register etch are very useful for this kind of troubleshooting. If the phone was attached to a CallManager using SIP before, then there could be some bad configuration still stuck in the phone. If you don't specify a new value, these phones cache the old config. Try factory defaulting the phone if all else fails. I have quite a few of these phones working without issue. Good luck! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
As a last resort (if qualify doesn't help), you could enter this (global) to increase the timeout on UDP translations: ip nat translation udp-timeout 300 (or greater if you prefer) It is likely a NAT timeout issue. When you call outbound, you 'reactivate' the SIP session in your NAT device, allowing calls to come in until it expires (default on many devices is 60 seconds). You may also receive inbound calls when the phone reregisters regularly. Try 'qualify=yes' in your phones section in sip.conf to send keepalives (option packets in this case) every two seconds to the phone to keep it from going idle. You can see the state of the phone from the console with a 'sip show peers', if unreachable, your NAT device has killed the NAT forward. Should look like one of these: xxx/xxx x.x.x.x D N 5060 OK (46 ms) xxx/xxx x.x.x.x D N 5060 UNREACHABLE As another troubleshooting step, you can telnet to the phone and have it reregister with Asterisk manually (register line 1 1) to see if that brings it back to life. If qualify doesn't do it, see if you can increase UDP timeouts in your firewall/NAT device. I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0 crypto isakmp profile IKE-PROFILE match identity group VPN-Users client authentication list default isakmp authorization list default client configuration address initiate client configuration address respond virtual-template 1 ! ! crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Very cool, I believe that did the trick. Thank you for your time. On Sat, Oct 18, 2008 at 7:42 PM, Darryl Dunkin [EMAIL PROTECTED] wrote: Oh, you are using ip inspect as well. I have this setup on a few routers when using the FW feature set: ip inspect udp idle-time 900 -Original Message- From: Stephen Reese [mailto:[EMAIL PROTECTED] Sent: Saturday, October 18, 2008 14:41 To: Asterisk Users Mailing List - Non-Commercial Discussion; Darryl Dunkin Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming calls I tried increasing the value and even set it to never and added the qualify line but that did not help. Do I need to poke any holes in the firewall on the nat device for the udp traffic to stay persistent? I have included my routers configuration in case someone notices something I may need to make the connection work correctly. Also when I call the phone within the OK reachable time after the call disconnects the status immediately become UNREACHABLE. ns1*CLIsip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 UNREACHABLE 3 sip peers [Monitored: 0 online, 1 offline Unmonitored: 2 online, 0 offline] [Oct 18 16:55:09] NOTICE[21216]: chan_sip.c:15231 handle_response_peerpoke: Peer '101' is now Reachable. (217ms / 2000ms) ns1*CLI sip show peers Name/username HostDyn Nat ACL Port Status vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored vitel-inbound/rsreese 64.2.142.1165060 Unmonitored 101/10168.156.63.118D N 1038 OK (217 ms) 3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 2 online, 0 offline] [Oct 18 17:24:16] NOTICE[21216]: chan_sip.c:19339 sip_p oke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 134 CISCO CONF FOLLOWS: ! version 12.4 service timestamps debug datetime msec service timestamps log datetime service password-encryption ! hostname 3725router ! boot-start-marker boot system flash:/c3725-adventerprisek9-mz.124-21.bin boot-end-marker ! logging buffered 8192 debugging logging console informational enable secret 5 ! aaa new-model ! ! aaa authentication login default local aaa authentication ppp default local aaa authorization exec default local aaa authorization network default local ! aaa session-id common clock timezone EST -5 clock summer-time PCTime date Apr 6 2003 2:00 Oct 26 2003 2:00 network-clock-participate slot 1 network-clock-participate slot 2 no ip source-route ! ip traffic-export profile IDS-SNORT interface FastEthernet0/0 bidirectional mac-address 000c.2989.f93a ip cef ! ! no ip dhcp use vrf connected ip dhcp excluded-address 172.16.2.1 ip dhcp excluded-address 172.16.3.1 ! ip dhcp pool VLAN2clients network 172.16.2.0 255.255.255.0 default-router 172.16.2.1 dns-server 205.152.144.23 205.152.132.23 option 66 ip 172.16.2.10 option 150 ip 172.16.2.10 ! ip dhcp pool VLAN3clients network 172.16.3.0 255.255.255.0 default-router 172.16.3.1 dns-server 205.152.144.23 205.152.132.23 ! ! ip domain name neocipher.net ip name-server 205.152.144.23 ip name-server 205.152.132.23 ip inspect name SDM_LOW cuseeme ip inspect name SDM_LOW dns ip inspect name SDM_LOW ftp ip inspect name SDM_LOW h323 ip inspect name SDM_LOW https ip inspect name SDM_LOW icmp ip inspect name SDM_LOW netshow ip inspect name SDM_LOW rcmd ip inspect name SDM_LOW realaudio ip inspect name SDM_LOW rtsp ip inspect name SDM_LOW sqlnet ip inspect name SDM_LOW streamworks ip inspect name SDM_LOW tftp ip inspect name SDM_LOW tcp ip inspect name SDM_LOW udp ip inspect name SDM_LOW vdolive ip inspect name SDM_LOW imap ip inspect name SDM_LOW pop3 ip inspect name SDM_LOW esmtp ip auth-proxy max-nodata-conns 3 ip admission max-nodata-conns 3 ip ips sdf location flash://256MB.sdf ip ips notify SDEE ip ips name sdm_ips_rule vpdn enable ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! ! crypto pki trustpoint TP-self-signed-995375956 enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-995375956 revocation-check none rsakeypair TP-self-signed-995375956 ! ! crypto pki certificate chain TP-self-signed-995375956 certificate self-signed 01 quit username user privilege 15 secret 5 ! ! ip ssh authentication-retries 2 ! ! crypto isakmp policy 3 encr 3des authentication pre-share group 2 ! crypto isakmp policy 10 hash md5 authentication pre-share crypto isakmp key cisco address 10.0.0.2 no-xauth ! crypto isakmp client configuration group VPN-Users key dns 2 domain neocipher.net pool VPN_POOL acl 115 include-local-lan netmask 255.255.255.0
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED] wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060 Unmonitored
vitel-inbound/rsreese 64.2.142.1165060 Unmonitored
101/10168.156.63.118D N 1038 Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0 offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco)
sntp_server:10.10.10.1
time_zone: EST
dial_template: DIALPLAN
nat_enable: 1
nat_address: 172.16.2.1
nat_received_processing: 1
outbound_proxy_port: 5060
outbond_proxy: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name: scott
#line2_authname: scott
#line2_shortname: 201
#line2_password: tiger
#line2_displayname: Larry Ellison; # Line 2 Display Name (Display
name to use for SIP messaging)
# Phone Label (Text desired to be displayed in upper right corner)
phone_label: Stephen Reese ; Has no effect on SIP messaging
# Phone Password (Password to be used for console or telnet login)
phone_password: goaway ; Limited to 31 characters (Default - cisco)
# User classifcation used when Registering [ none(default), phone, ip ]
user_info: none
telnet_level: 2
Any ideas or help would be great, thanks.
I'm still unable to wrap my head around this problem. I can recieve a
call after I first call out from the line/phone. I didn't think it's a
NAT issue since it kind of works.
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Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.
Should look like one of these:
xxx/xxx x.x.x.x D N 5060 OK (46 ms)
xxx/xxx x.x.x.x D N 5060 UNREACHABLE
As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
101/10168.156.63.118D N 1038
Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco)
sntp_server:10.10.10.1
time_zone: EST
dial_template: DIALPLAN
nat_enable: 1
nat_address: 172.16.2.1
nat_received_processing: 1
outbound_proxy_port: 5060
outbond_proxy: ns1.neocipher.net
SIP0112B9EAFF72.cnf
image_version: P0S3-08-9-00
# Line 1 Setup
line1_name: 101
line1_authname: 101
line1_shortname: Line 101
line1_password: test
line1_displayname: Stephen Reese; # Line 1 Display Name (Display
name to use for SIP messaging)
# Line 2 Setup
#line2_name
Re: [asterisk-users] Cisco 7960 not always receiving incoming calls
Sorry, I missed the Cisco router bit.
As a last resort (if qualify doesn't help), you could enter this
(global) to increase the timeout on UDP translations:
ip nat translation udp-timeout 300 (or greater if you prefer)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl
Dunkin
Sent: Friday, October 17, 2008 17:28
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
It is likely a NAT timeout issue. When you call outbound, you
'reactivate' the SIP session in your NAT device, allowing calls to come
in until it expires (default on many devices is 60 seconds). You may
also receive inbound calls when the phone reregisters regularly. Try
'qualify=yes' in your phones section in sip.conf to send keepalives
(option packets in this case) every two seconds to the phone to keep it
from going idle. You can see the state of the phone from the console
with a 'sip show peers', if unreachable, your NAT device has killed the
NAT forward.
Should look like one of these:
xxx/xxx x.x.x.x D N 5060 OK (46 ms)
xxx/xxx x.x.x.x D N 5060 UNREACHABLE
As another troubleshooting step, you can telnet to the phone and have it
reregister with Asterisk manually (register line 1 1) to see if that
brings it back to life.
If qualify doesn't do it, see if you can increase UDP timeouts in your
firewall/NAT device.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Reese
Sent: Friday, October 17, 2008 17:04
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Cisco 7960 not always receiving incoming
calls
On Wed, Oct 15, 2008 at 7:57 PM, Stephen Reese [EMAIL PROTECTED]
wrote:
I've searched around and found a few similar situations where the
phone will call out when using a Asterisk server but not receive
inbound calls. My issue is a little stranger. If I call out from the
phone then the phone will receive the next inbound call. The phone
will not receive another inbound call until a call out again from it
first. Any ideas?
I am using SIP and am using the latest phone image from Cisco to date.
I am also using a Cisco router at the gateway. Is there anything
special I should to to make this work? Note my soft phone does not
have any issues using the same dialing rules and extension
information. Here is some of my config stuff:
ns1*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
vitel-outbound/rsreese 64.2.142.22 5060
Unmonitored
vitel-inbound/rsreese 64.2.142.1165060
Unmonitored
101/10168.156.63.118D N 1038
Unmonitored
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 3 online, 0
offline]
Inbound call in progress when the SIP Cisco phone doesn't ring
Verbosity is at least 5
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Inbound call in progress when the SIP Cisco does ring after I first
make an outbound call
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Goto(SIP/rsreese-082a8358,
default,101,1) in new stack
-- Goto (default,101,1)
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/rsreese-082a8358,
SIP/101SIP/[EMAIL PROTECTED],30) in new stack
== Using SIP RTP CoS mark 5
-- Called 101
== Using SIP RTP CoS mark 5
-- Called [EMAIL PROTECTED]
-- SIP/101-0825cab8 is ringing
-- SIP/vitel-outbound-08270130 is making progress passing it to
SIP/rsreese-082a8358
-- SIP/vitel-outbound-08270130 is ringing
== Spawn extension (default, 101, 1) exited non-zero on
'SIP/rsreese-082a8358'
Extensions.conf, which I don't think is relevent, I've changed it to
just a simple dial the sip phone and it still fails.
exten = 101,1,Dial(SIP/101SIP/[EMAIL PROTECTED],30)
exten = 101,n,GotoIf($[${DIALSTATUS} =
CHANUNAVAIL]?lbl_default_1:)
exten = 101,n,GotoIf($[${DIALSTATUS} = NOANSWER]?lbl_default_1:)
exten = 101,n(lbl_default_0),Hangup()
exten = 101,n(lbl_default_1),Dial(SIP/[EMAIL PROTECTED],30)
exten = 101,n,Goto(lbl_default_0)
Cisco phone stuff from a Cisco 7960:
SIPDefault.cnf
image_version: P0S3-08-9-00
proxy1_address: neocipher.net; Can be dotted IP or FQDN
proxy_register: 1
messages_uri: 100
phone_password: cisco ; Limited to 31 characters (Default - cisco
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Thank you, Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 08:26, Fri 10 Oct 08, David Gibbons wrote: You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Or, if you dont want to use outside modules use Asterisk 1.6 (which has been released as well) with the chan_skinny driver. A lot of development went into it and it's much more useable then the 1.2 version. Myself uses chan_skinny in production without too much trouble. Specially when you use the 7960 phones it's a nice setup. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. What driver are you referring to ? It must be something outside of the core asterisk, because a lot of commits went into chan_skinny the last year or so. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Yes, chan_skinny in 1.4 is better then the 1.2 version, but the real stuff happened in the 1.6 version. 1.6.0 is released, so why not use that one instead of 1.4? Thank you, Wayne. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the day. I shall take another look at both options. Thank you Wayne. Michiel van Baak wrote: On 08:26, Fri 10 Oct 08, David Gibbons wrote: You need to check out the chan_sccp-b mainling lists on sourceforge. There is active development in SVN but not in tarball releases. http://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion It is very stable. Or, if you dont want to use outside modules use Asterisk 1.6 (which has been released as well) with the chan_skinny driver. A lot of development went into it and it's much more useable then the 1.2 version. Myself uses chan_skinny in production without too much trouble. Specially when you use the 7960 phones it's a nice setup. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wayne Sent: Thursday, October 09, 2008 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4 Hi All, I'm thinking of creating a new asterisk server using the latest 1.4 stable release to replace my ageing Asterisk SVN-branch-1.2-r7231 (its been a while!). My only concern - my phones are Cisco 7960's (with sccp firmware 7.2 loaded) and to support them better, I remember compiling in a skinny(?) driver to replace the (from what I could tell) basic in built sccp support. After digging around a little it would appear that the original creator of the skinny driver has not done any development for ages. What driver are you referring to ? It must be something outside of the core asterisk, because a lot of commits went into chan_skinny the last year or so. Simple question, has 1.4 got better native support for sccp now without having to add in anything extra to make everything work ok?, if not, is there a version that someone may have carried forward of the skinny driver that will work with 1.4? Yes, chan_skinny in 1.4 is better then the 1.2 version, but the real stuff happened in the 1.6 version. 1.6.0 is released, so why not use that one instead of 1.4? Thank you, Wayne. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp, Skinny and 1.4
On 21:28, Fri 10 Oct 08, Wayne wrote: Thanks both, The only thing I have a little concern over is that 1.6 is that its still a development release (if I understand things correctly). No, 1.6.0 has been released. This is indeed the first public 'final' release of the 1.6 series. But it's not in beta or release-candidate anymore. Basically, it's the latest and greatest version that should be stable. Stability is the main thing for me (its only a very small set up) but there are no technical people around if something were to go wrong through the day. You do know it's just another daemon an a linux box right ? If you cant afford downtime you should not bet on one server, but make every part of your network redundant. That means at least: connectivity power hardware locations backups all the other stuff I forgot -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 audible hold reminder?
On Fri, Aug 15, 2008 at 12:27:16PM -0500, [EMAIL PROTECTED] wrote: Hello, I have recently setup my first PBX and am wondering if there might be a way to send audible notification to the cisco 7960 phone when a call is put on hold. We lost a call due to a customer being on hold and forgotten about (yikes). Is there a way to get the phone to beep or ring down the same or other SIP channels after a certain amount of time on hold? Yes and no. (I am on the SIP version 8.9) In the config file for the phone: call_hold_ringback: 1 This option means that if there is a call on hold, and the handset is replaced (say, after ending another call) then the held call will ring again at the handset. I don't think there is a way (on the handset) to set a held call timeout to re-ring on the phone. If you park the call with asterisk instead of holding it, then the call park option allows calls to come back to the person who parked them after a set timeout. You may be able to do something else in asterisk, though not sure what. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 134-138 Borough High Street, London SE1 1LB Registered in England 3137929 at 3 Park Road, Peterborough, PE1 2UX ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960
An educated guess is: reverse the SIP trunk buttons, so the preferred provider is the top button, and voila, your speed dial going to the first trunk is now what you want. On Wed, Aug 13, 2008 at 7:44 PM, Shawn L [EMAIL PROTECTED] wrote: This one is a little off-topic, it's more about the phone than asterisk itself. I have a cisco 7960 configured with 2 lines to 2 different sip providers (cant get asterisk to register with the 2nd provider, but that's another story). Is there a way yo determine which direction speed-dial buttons will go out? I'd like to have speed-dial buttons that will go out on line2 instead of line 1. Anyone know if this is possible? Thanks Shawn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 odd behaviour ...
remove callprogress=yes and busydetect=yes lotusscript wrote: Been using the Snom 360 and 190 for a while and decided to try the Cisco 7960. The problem I'm seeing is the call terminates between 2:34 and 3:00 minutes. This only happens when using Zap channels. Internal calls work fine. No probs with the Snoms. No errors show on the * box when the line drops. Anyone seen this? Asterisk 1.2.14 Cisco Firmware: P0S3-08-8-00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
As expected, Jim took care of me WRT the Cisco upgrade. It is now far more usable than when it was SCCP... I gave up on trying to get SCCP working in Asterisk after upgrading to 1.4 from 1.0. Due to his generosity, I feel I owe him to recommend his termination\origination services. The one or two times I've had any issue, he has been quick to respond and took care of me. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP Upgrade
That I am. I'll contact you off list. -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: Sigma Networks [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 04, 2008 12:34 PM Subject: Re: [asterisk-users] Cisco 7960 SIP Upgrade Mike Hammett wrote: I couldn't figure it out on my own. I tried to purchase a Smartnet for the phone, but the original 7960 is not supported. Is it technically possible and if so, what would it cost me to have someone remote into my network and upgrade my SCCP 7960 to the latest SIP firmware? -- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Mike, I know you are a very happy customer of Sigma Networks ( :-) )... I'd be happy to upgrade the phone to 8.3.3SR2 for you. Jim ph: 408-701-9929 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 to 2 SIP servers?
Yes it's work for me... (with olds 7940 phones...) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Shawn Laemmrich Envoyé : mercredi 5 décembre 2007 23:43 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Cisco 7960 to 2 SIP servers? Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2 different SIP servers @ the same time? I currently have an asterisk box @ home with several sip extensions and a Nortel C15k phoneswitch at work (not the pbx, the full phone switch). I can connect from the SIP phone to the Nortel phone switch, but cannot make asterisk talk to it at all (if anyone has any ideas on this one, I'd be hugely grateful). So I thought if I could have the cisco ip phone on my desk talk to both servers (like a line1 is my home asterisk server, line 2 is the nortel switch) I'd be all set. Does anyone know if this is possible, and if so how to do it? Thanks in advance Shawn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 to 2 SIP servers?
It can be attached to 6 if I remember correctly. However, each is a separate line. Cisco will not perform a seamless connection to multiple servers for a single line as some sort of fail over system. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Thursday, December 06, 2007 11:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7960 to 2 SIP servers? Yes it's work for me... (with olds 7940 phones...) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Shawn Laemmrich Envoyé : mercredi 5 décembre 2007 23:43 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Cisco 7960 to 2 SIP servers? Is it possible for a Cisco 7960 phone with SIP firmware to connect to 2 different SIP servers @ the same time? I currently have an asterisk box @ home with several sip extensions and a Nortel C15k phoneswitch at work (not the pbx, the full phone switch). I can connect from the SIP phone to the Nortel phone switch, but cannot make asterisk talk to it at all (if anyone has any ideas on this one, I'd be hugely grateful). So I thought if I could have the cisco ip phone on my desk talk to both servers (like a line1 is my home asterisk server, line 2 is the nortel switch) I'd be all set. Does anyone know if this is possible, and if so how to do it? Thanks in advance Shawn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 or 7960G
On Sun, Sep 02, 2007 at 03:47:45PM +0100, Chris Bagnall wrote: There's both a 7960 and a 7960G (and a 7961 to confuse matters further). The 7960 is the earlier version. The easiest way to identify it from a picture is to look at the messages/services/etc. buttons. On the 7960 the words messages and services are written on them. On the G, there's an envelope and a globe on the buttons themselves, and the words messages and services are provided on a surround sticker (one assumes to make internationalization easier). ...although I don't think Cisco ever produced any other languages for the 7960G anyway, but 7960 and 7960G are pretty much identical. 7961 is a completely different phone with totally different software, although it has a better screen and much better audio quality than the 7960. 7960 was end-of-life a while ago by Cisco. Not sure about the 7960G though. If you run them in SIP Only mode, they are quite limited when it comes to actual functionality when compared to what other phones are offering. 7961, although a better bit of hardware, does not offer much noticable improvement for SIP. The functionality is about exactly the same, but with more possibilities for integration via XML than the 7960. 7961 does support standard 802.3af PoE and not Cisco's legacy proprietary PoE system which they introduced before 802.3af. You need a Cisco switch or a switch that supports legacy PoE (Foundry FES for example) to make the 7960s power on, but 7961 works with standard 802.3af PoE kit. Contact me off-list if you want my list of specific limitations of the 7960/SIP, as there are many. Rob -- Robert Lister - London Internet Exchange - http://www.linx.net/ sip:[EMAIL PROTECTED] - inoc-dba:5459*710- tel: +44 (0)20 7645 3510 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 or 7960G
On 9/2/2007 at 9:32 AM, Joe Acquisto [EMAIL PROTECTED] wrote: Is there more than one version of the Cisco 7960? I see some items advertised as 7960 or 7960G, but searching on 7960 only brings up 7960G info, or ambiguous stuff. joe a. A partial never mind, it appears they are two different models. Yet the differences are not readily apparent. joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 or 7960G
There's both a 7960 and a 7960G (and a 7961 to confuse matters further). The 7960 is the earlier version. The easiest way to identify it from a picture is to look at the messages/services/etc. buttons. On the 7960 the words messages and services are written on them. On the G, there's an envelope and a globe on the buttons themselves, and the words messages and services are provided on a surround sticker (one assumes to make internationalization easier). Apart from the minor cosmetic differences, I don't know if there are any actual feature differences between them. I've one of each here and I've yet to find any worthwhile feature differences. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp
On 18:59, Fri 31 Aug 07, Joe Acquisto wrote: What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? You'll need the firmware and an TFTP server to get the firmware on the phone. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp
On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote: On 18:59, Fri 31 Aug 07, Joe Acquisto wrote: What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? You'll need the firmware and an TFTP server to get the firmware on the phone. I guess my question is more along the line of how difficult Cisco is about this? I know router firmware is not always just for the asking. Hmm, I guess I *could* ask Cisco . . . joe a. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 sccp
On 09:17, Sat 01 Sep 07, Joe Acquisto wrote: On 9/1/2007 at 7:46 AM, Michiel van Baak [EMAIL PROTECTED] wrote: On 18:59, Fri 31 Aug 07, Joe Acquisto wrote: What is involved in getting SIP firmware into a Cisco 7960 with sccp installed? Expensive image from Cisco? Plated in unobtanium? You'll need the firmware and an TFTP server to get the firmware on the phone. I guess my question is more along the line of how difficult Cisco is about this? I know router firmware is not always just for the asking. You'll need a Cisco smartnet account to get the firmware. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
Dan Austin wrote: Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan Well, he's also using chan_sccp, so no amount of upgrading is going to help with that. In my opinion (and I think Dan and several others would agree), chan_skinny is far more stable (and active...) than chan_sccp. -- Jason Parker Digium ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
On 13:52, Fri 31 Aug 07, Jason Parker wrote: Dan Austin wrote: Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan Well, he's also using chan_sccp, so no amount of upgrading is going to help with that. In my opinion (and I think Dan and several others would agree), chan_skinny is far more stable (and active...) than chan_sccp. as on of the 'several outhers' I totally agree. We used to run chan_sccp for our kirk setup and some cisco phones. The switch to chan_skinny made everything usable again :) The random crashes and lockups you get with chan_sccp are too annoying :) -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Won'
Jason wrote: Dan Austin wrote: Shawn wrote: I'm having a wierd problem with a Cisco 7960 (sccp2) and asterisk (1.4.2) If the call that I'm trying to make goes through, everything works fine. But if there's any sort of error (like me messing around in my extensions.conf, etc). I can't get the connection to drop. ie: If I get the conjestion tone and hang up the phone, I can do a sccp show channels I can see that the channel is still in use (even after several minutes). If I pick up the phone to attempt to make another call, I get an error that it can't put the current call on hold to start the new call. What am I missing? An upgrade. The sccp channel in early 1.4 had quite a number of problems, and it was completely broken in 1.4.3 to 1.4.6 Any version after 1.4.7 should work better, with the latest being the best choice. Dan Well, he's also using chan_sccp, so no amount of upgrading is going to help with that. In my opinion (and I think Dan and several others would agree), chan_skinny is far more stable (and active...) than chan_sccp. Bugger! I should have noted the 'sccp show channels' command. I tend to swap skinny/SCCP automatically, since Cisco uses both in the documentation, and had it in my head that he meant skinny Yes, chan_skinny in 1.4.7+ has had major love applied. I only have a couple test phones hooked up for development, so my impression of stability is not worth much, but I think we have managed to fix up the most hideous bugs. If we can keep up the pace, chan_skinny in 1.6 is going to rock. Sorry for the confusion. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960
can you give a bit more info? I know that you need nat=never for example - Original Message - From: Khaled To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 -- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960
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RE: [asterisk-users] Cisco 7960
I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From: Khaled mailto:[EMAIL PROTECTED] To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Cc: [EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. _ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. *___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960
Dear Khaled, What is the softphone u r using? Thx MAG Khaled wrote: I am using firmware version pos3-07-500 Kindly can you provide me with the basic configuration for cisco ip phone and asterisk config file *I have nat=never at my asterisk config file and nat enabled N0 at cisco phone *I have an out bound proxy ip and port 5060 at cisco phone *Voip control port is 5061 My problem is my soft phone can call the cisco phone with normal RTP and Bye message,but my cisco phone cant dial my soft phone. Asterisk sends bye message for my soft phone. Thanks --- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wireless Sent: Tuesday, February 27, 2007 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 can you give a bit more info? I know that you need nat=never for example - Original Message - From:Khaled To:'Asterisk Users Mailing List - Non-Commercial Discussion' Cc:[EMAIL PROTECTED] Sent: Tuesday, February 27, 2007 10:03 AM Subject: [asterisk-users] Cisco 7960 Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 - * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 TFTP Timeout Error on RINGLIST.DAT and dialplan.xml
On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote: I've looked around and couldn't find much on this, but using two different TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP Timeout Errors. The SIP configuration files load fine. Any ideas? Have you made sure that the file has the proper rights? Iirc it needs to be 644. You can also use Wireshark (former Ethereal) to sniff the traffic and see what the Cisco requests. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 TFTP Timeout Error on RINGLIST.DAT and dialplan.xml
On Thu, 2007-02-08 at 13:27 -0500, Brian M. Arlinghaus wrote: I've looked around and couldn't find much on this, but using two different TFTP servers (linux / windows), my Cisco 7960s won't load the RINGLIST.DAT and dialplan.xml files. On both the TFTP servers and the phone, I get TFTP Timeout Errors. The SIP configuration files load fine. Any ideas? Take the phone out of the equation. Make sure iptables isn't getting in the way -- not likely since you can get the SIP files. sudo /etc/init.d/iptables stop Set lots'o -v's in the server_args in /etc/xinet.d/tftp and try it from the tftp server's command line: tftp tftp-host-name -c get dialplan.xml and hope that tftp says something useful or tftpd logs something useful in the sytem error log. Move on to another host and repeat the tftp command. Try it again from the phone. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 / SIP tftp configs
On Wed, 2005-08-24 at 12:44 -0400, Asterisk User Group wrote: I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it forget what it has programmed and listen only to the download? Change it to UNPROVISIONED 2nd - Has anyone figured out how to get the Message button to launch a dial to VoicemailMain? messages_uri: 3rd - How do I display on the LCD an alias to the registered line? line1_name: 2000 line1_authname: 2000 line1_password: ** line1_shortname: Home The doc seems to suggest that line1_name is what it registers with and line1_authname is what it uses if challenged during the authentication. This doesn't make any sense to me. I am looking for the line to be 2000 but the display to say Home or Business, etc. Thanks, dbc. ___ 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 provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
Which asterisk release are you running chan_skinny under? - Original Message - From: Will Roy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phone Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7] What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regards Wil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
I am running 1.4.0-beta2 Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST)From: Anthony LaMantia [EMAIL PROTECTED]Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=utf-8Which asterisk release are you running chan_skinny under?- Original Message -From: Will Roy [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phoneBefore I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny.The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag:Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :) regardsWil ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
Well, I've never actually been able to make chan_skinny work with 79xx
phones.
I found the chan_sccp to work quite well:
http://chan-sccp.berlios.de/
plus this patch for a problem on MeetMe (I don't remeber where I found
it, but it works!):
diff -uNr chan_sccp-20060408.org/sccp_pbx.c chan_sccp-20060408/sccp_pbx.c
--- chan_sccp-20060408.org/sccp_pbx.c 2006-04-08 14:20:17.0 +0200
+++ chan_sccp-20060408/sccp_pbx.c 2006-05-17 17:14:15.0 +0200
@@ -290,6 +290,12 @@
static int sccp_pbx_answer(struct ast_channel *ast) {
sccp_channel_t * c = CS_AST_CHANNEL_PVT(ast);
+ // if channel type is undefined, set to SCCP
+ if (!ast-type) {
+ sccp_log(1)(VERBOSE_PREFIX_3 SCCP: Channel type
undefined, sett
ing to type 'SCCP'\n);
+ ast-type = SCCP;
+ }
+
if (!c || !c-device || !c-line) {
ast_log(LOG_ERROR, SCCP: Answered %s but no SCCP
channel\n, as
t-name);
return -1;
I recommend using SIP firmware anyway... the conversion process is a bit
annoying but
as far as now 7940/7960 are really stable IP phones.
I am currently using chan_sccp only for 7902 phones (I've just got 2 of
them)
which do not support SIP firmware.
Will Roy ha scritto:
Before I got down the path of converting a Cisco 7960 I have over to
SIP I wanted to try and set it up using Skinny.
The phone registers ok with Asterisk. When I call a SIP softphone
extension on my network the call is made and I can answering it.
However no voice is heard over the call.
When I debug Skinny on the console after the call has connected I see
the following messag:
Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]
What additional information would be required to troubleshoot this? or
should I stop wasting time and just convert the phone to SIP? :)
regards
Wil
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P.IVA 01697420030
P.le Lombardia, 4 - 28100 Novara - Italy
Tel. 0321-499508
Fax 0321-492974
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Re: [asterisk-users] cisco 7960 not registering after * restart
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. And that's *exactly* the sort of information that makes me wonder, Aaron: do you guys write a (publically accessible) blog on the goings on in your telecoms dept/asterisk project? I know you're a little closer to In The Real World than some folks, which might militate against... but you're a college, too. :-) And it seems that you're gonna know a whole lot of stuff. Just wondering... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
Heh, well, I actually just started a blog to keep track of various goings on, but I just started it so it's kinda scarce. I intend to update it in and out with various information I email to people so everyone can benefit from the questions and answers people use. I'd like to see other people register and start posting stuff there as well as it's got free registration and basically unlimited storage. Voip-info.org is great for learning how to do the basics, but I'd like to see more people join together and disseminate information about how they do things. Check it out: http://asterisk.mdaniel.net On Thu, 2006-10-12 at 10:16 -0400, Jay R. Ashworth wrote: On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. And that's *exactly* the sort of information that makes me wonder, Aaron: do you guys write a (publically accessible) blog on the goings on in your telecoms dept/asterisk project? I know you're a little closer to In The Real World than some folks, which might militate against... but you're a college, too. :-) And it seems that you're gonna know a whole lot of stuff. Just wondering... Cheers, -- jra -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Aaron On Wed, 2006-10-11 at 15:35 +0200, Louis-David Mitterrand wrote: Hello, When I restart asterisk the cisco 7960/7940 phones (sip fw 7.5) fail to re-register themselves with asterisk, even though I put timer_register_expires: 60 in SIPDefault.cnf Is there a way to have these phones register themselves every 60 seconds? Alternatively, can asterisk be made to remember its dynamic sip hosts' registration after a restart? Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7960 not registering after * restart
On Wed, Oct 11, 2006 at 09:11:43AM -0500, Aaron Daniel wrote: That's a bug with the 7.5 firmware. I would suggest upgrading to the 8.4 version, we've been running it for a few weeks in a test environment and everyone's been pretty satisfied with the new firmware (read: nobody's complained). If the server goes out, they re-register after the timeout without problems. Thanks for your helpful answer, What is the cisco part number for the appropriate smartnet contract required to obtain 79XX firmware? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 Double Natted
On the 7960 with a SIP image, Press the Settings button and go to option 4 SIP Configuration. Scroll down to line 24 NAT Enabled and set it to yes. Then set 25 NAT Address to the external IP address. This will need to be manually changed every time the phone's router pulls a new DHCP lease. In your sip.conf, make sure that you have nat=yes and qualify=yes. I have had double-NATed 7960s work with this setup, but you are at the mercy of the routers involved in performing the NAT. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Sunday, September 24, 2006 5:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco 7960 Double Natted Hi All Yes I know double Nat is a problem But I have a Cisco 7960 which is remote from the * PBX ad connected via the Internet. Each side has NAT (1) Sometimes it will work often it won't. And when it decides to work is random Always (2) The Register side works fine. SIP SHOW PEERS has the phone listed with the correct IP address and an average Qualify time (121 ms) Always (3) You can make calls outbound with the Cisco phone through the * PBX Problem (4) You can not receive any calls (when not working correctly) (a) The Phone rings but not voice goes through (b) Sometimes get a 481 Call Leg Does Not Exist (c) Sometimes get a -- is circuit-busy (5) On a reload of the * box you will 95 % sure loose the connection if it was working ? (6) SIP 5060 - 5063 and RTP 1 - 25000 is open and port forwarded on both sides (7) All calls are VoIP and terminate or originate via a VoIP Provider Anybody got any ideas, I have tried everything Thanks All Barry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Double Natted
Thanks for the input Yes I have nat=yes and qualify=yes I know in the SIPMacAddress.cnf file I have # NAT/Firewall Traversal nat_enable: 1 nat_received_processing: 1 nat_address: phone's public IP Address Do I still need to set it again in SIP Configuration ? Thanks all Barry Hughes, Sam wrote: On the 7960 with a SIP image, Press the Settings button and go to option 4 SIP Configuration. Scroll down to line 24 NAT Enabled and set it to yes. Then set 25 NAT Address to the external IP address. This will need to be manually changed every time the phone's router pulls a new DHCP lease. In your sip.conf, make sure that you have nat=yes and qualify=yes. I have had double-NATed 7960s work with this setup, but you are at the mercy of the routers involved in performing the NAT. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Double Natted
Barry Fawthrop wrote: Hi All Yes I know double Nat is a problem But I have a Cisco 7960 which is remote from the * PBX ad connected via the Internet. Each side has NAT (1) Sometimes it will work often it won't. And when it decides to work is random Always (2) The Register side works fine. SIP SHOW PEERS has the phone listed with the correct IP address and an average Qualify time (121 ms) Always (3) You can make calls outbound with the Cisco phone through the * PBX Problem (4) You can not receive any calls (when not working correctly) (a) The Phone rings but not voice goes through (b) Sometimes get a 481 Call Leg Does Not Exist (c) Sometimes get a -- is circuit-busy (5) On a reload of the * box you will 95 % sure loose the connection if it was working ? (6) SIP 5060 - 5063 and RTP 1 - 25000 is open and port forwarded on both sides (7) All calls are VoIP and terminate or originate via a VoIP Provider Anybody got any ideas, I have tried everything Thanks All Barry You could try giving up and not wasting anymore time. At least that was my experience after spending MANY hours working on a solution. Well I came up with a solution, and it was to remove the double NAT, at least to layer 3 of the stack. OpenVPN saved the day. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 part numbers ...
Neither is technically the product you need. The CP-7960G-CH1 is a Cisco 7960G phone, with a CallManager client license, preloaded with SCCP firmware. The CP-7960G-CCME is a Cisco 7960G phone, with a CallManager Express client license, preloaded with SCCP firmware. To my knowledge, Cisco does not ship phones pre-loaded with SIP. They ship with SCCP. What you would need would be the following. (1) Cisco CP-7960G= (Global Spare) (1) Cisco SW-SM-UL-7960= (SIP MGCP License for Single 7960 IP Phone) (1) CON-SNT-7960 (Smarnet 8X5 NBD IP Phone 7960 MGR Set) You will also need a Cisco authorized telephony partner to register your smartnet contract for you, based on the serial number of the phone you purchase. When you receive the phone, and your contract is registered, you can go to Cisco's website, apply for a CCO login, and your login permissions will allow you to download the SIP firmware. You can then migrate your phone firmware from SCCP to SIP, or your reseller may do so for you. If you are not in the US, you will need the (ASIAPAC) version of the Smartnet contract. Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 800.398.VoIP X3402email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Cesc To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 19, 2006 7:29 PM Subject: [asterisk-users] Cisco 7960 part numbers ... Hi,I requested a quote from a cisco reseller (or something like this) for 2 cisco 7960 phones, ideally preloaded with SIP firmware ... and i got the quote back with: 1x CP-7960-CH1 and 1x CP-7960-CCME. My question is, what is the difference between the two? If these are not the part number for the pre-loaded SIP phones, what part number is the correct? and what about the service contract ... when is it needed? Thank you very much ...Cesc ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 part numbers ...
On Tue, 2006-09-19 at 19:58 -0400, Cory Andrews wrote: [snip] What you would need would be the following. (1) Cisco CP-7960G= (Global Spare) (1) Cisco SW-SM-UL-7960= (SIP MGCP License for Single 7960 IP Phone) (1) CON-SNT-7960 (Smarnet 8X5 NBD IP Phone 7960 MGR Set) You will also need a Cisco authorized telephony partner to register your smartnet contract for you, based on the serial number of the phone you purchase. When you receive the phone, and your contract is registered, you can go to Cisco's website, apply for a CCO login, and your login permissions will allow you to download the SIP firmware. You can then migrate your phone firmware from SCCP to SIP, or your reseller may do so for you. If you are not in the US, you will need the (ASIAPAC) version of the Smartnet contract. Thanks for the info, good to know. One question: do Europeans also need to order the ASIAPAC version of the smartnet contract or is there also a European version? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 part numbers ...
I can't answer definitively in regard to the Smartnet, but I only see the US and ASIAPAC versions SKU'd up with US Cisco distributors (Ingram, Comstor, TechData) and I do not see an EU version on the GPL. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 800.398.VoIP X3402 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 19, 2006 10:32 PM Subject: Re: [asterisk-users] Cisco 7960 part numbers ... On Tue, 2006-09-19 at 19:58 -0400, Cory Andrews wrote: [snip] What you would need would be the following. (1) Cisco CP-7960G= (Global Spare) (1) Cisco SW-SM-UL-7960= (SIP MGCP License for Single 7960 IP Phone) (1) CON-SNT-7960 (Smarnet 8X5 NBD IP Phone 7960 MGR Set) You will also need a Cisco authorized telephony partner to register your smartnet contract for you, based on the serial number of the phone you purchase. When you receive the phone, and your contract is registered, you can go to Cisco's website, apply for a CCO login, and your login permissions will allow you to download the SIP firmware. You can then migrate your phone firmware from SCCP to SIP, or your reseller may do so for you. If you are not in the US, you will need the (ASIAPAC) version of the Smartnet contract. Thanks for the info, good to know. One question: do Europeans also need to order the ASIAPAC version of the smartnet contract or is there also a European version? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 won't download dialplan.xml
Put this line in your SIPDefault.cnf file (or the individual phone's): dial_template: dialplan Just cut the .xml off the filename and the phone will pull that particular dialplan :) On Fri, 2006-09-01 at 16:39 -0400, Peter Pauly wrote: I'm monitoring my tftp servers' logs and my Cisco 7960 test phone won't download dialplan.xml to the phone. I know this from the logs and from the behavior of the phone. I see it downloading other files like the ring tone file, etc. Is there something that needs to be set in the cnf files to download the dialplan? I thought it is included automatically. I've also tried reseting the phone to factory presets. I'm running POS3-08-2-00. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
Hi on my 7940 Phones here, this is the first Part of the Factory Reset Procedure after Step 3 and the Status Message you have to hit all Keys on the Number Pad (1 - 2 - 3 - 4 - #) and then answer the Question by hitting Number 2 Cu David Maxx Lobo schrieb: Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
If the phone already had the SIP image running. Check the SIPDefault.cnf file there may be a phone_password= string this is the phone's current password use it remember to change to number or uppercase if need be Ferguson, Michael wrote: Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks - - -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 password reset
David and Barry, Thanks for the help. 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop Sent: Wednesday, August 16, 2006 6:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset If the phone already had the SIP image running. Check the SIPDefault.cnf file there may be a phone_password= string this is the phone's current password use it remember to change to number or uppercase if need be Ferguson, Michael wrote: Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks - - -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 password reset
Thanks. Will this action blow away the SIP images I already have on the phone? 'preciate it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 password reset
Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 password reset
What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7960 password reset
Maxx, Thanks much for the feedback. I will check into it and follow up with your instructions. 'preciate it. Best wishes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 5:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset What Cisco image is the phone running? If it is really old (lower than P0S030203) then yeah, this won't work. If you upgrade the phone to P0S30203 and from there on to P0S3-06-3-00, and then these instructions will work fine. This should be pretty straightforward using ATFTP and the Cisco images. In response to your other question, a factory reset TMK does not wipe out the SIP image. Just the settings. --Maxx Ferguson, Michael wrote: Maxx, That did not work. Any other ideas? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxx Lobo Sent: Tuesday, August 15, 2006 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7960 password reset Fastest way (wipes everything out): 1. Power off the phone completely. 2. Hold down the # key, then power the phone on. 3. Continue holding the # key until the LCD gives you a status message. 4. Follow the prompts to do a full factory reset, which resets the password as well. --Maxx Ferguson, Michael wrote: G'Day List, I am trying, once again, to configure my 7960 to work with asterisk. Where abouts do I go to reset the password on the phone? Thanks - - -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Call Waiting Beep
The only thing I can find is here on page 157: http://www.cisco.com/application/pdf/en/us/guest/products/ps2156/c2001/ccmigration_09186a00801d1972.pdf Hope this helps. On 7/26/06, Cory Andrews [EMAIL PROTECTED] wrote: Anyone aware of a way to turn off the call waiting beep via tftp for cisco 7960's? Disabling this through the call menu doesn't appear to work. This would be for sip firmware Thanks Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Call Waiting Beep
I'll take a look at this, thanks! Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 26, 2006 10:12 PM Subject: Re: [asterisk-users] Cisco 7960 Call Waiting Beep The only thing I can find is here on page 157: http://www.cisco.com/application/pdf/en/us/guest/products/ps2156/c2001/ccmigration_09186a00801d1972.pdf Hope this helps. On 7/26/06, Cory Andrews [EMAIL PROTECTED] wrote: Anyone aware of a way to turn off the call waiting beep via tftp for cisco 7960's? Disabling this through the call menu doesn't appear to work. This would be for sip firmware Thanks Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 - automated send DTMF digits after dialing?
On Jul 20, 2006, at 8:47 PM, [EMAIL PROTECTED] wrote: Is it possible to make a 7960 speed dial automatically send DTMF digits some specific number of seconds after dialing? I'd like to automate dialing into a PBX. We do this with 'internal extensions' So in extensions.conf we have defined some extensions that use dial's D param. have a look at the cli command: show application dial Michiel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 - automated send DTMF digits after dialing?
On Fri, 21 Jul 2006, Michiel van Baak wrote: On Jul 20, 2006, at 8:47 PM, [EMAIL PROTECTED] wrote: Is it possible to make a 7960 speed dial automatically send DTMF digits some specific number of seconds after dialing? I'd like to automate dialing into a PBX. We do this with 'internal extensions' So in extensions.conf we have defined some extensions that use dial's D param. have a look at the cli command: show application dial I was hoping to avoid this... -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP 8-3-0
Are you using the Non-CallManager version? _ Mobilcom http://www.mobilcom.net - Original Message - From: Tong [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 8:56 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 if you don't report it to cisco they won't know that bug exisit. - Original Message - From: Daryl Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 4:05 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP 8-3-0
Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 SIP 8-3-0
if you don't report it to cisco they won't know that bug exisit. - Original Message - From: Daryl Johnson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 4:05 PM Subject: Re: [asterisk-users] Cisco 7960 SIP 8-3-0 Tim, I have seen the same 400 errors and the broken MWI... I backed up to 7.3... We'll see if Cisco corrects these in the next release... Daryl - Original Message - From: Tim Connolly [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, July 17, 2006 12:06 PM Subject: [asterisk-users] Cisco 7960 SIP 8-3-0 Looks like the MWI broke on 8-3 also... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.1/390 - Release Date: 7/17/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7960 Softkey templates
I thought I saw somewhere that the 7960 would not, but the 7961 would. There was a chart somewhere on Cisco's website. The 7961 can read an xml file for this info. On 7/5/06, Scott Higginbotham [EMAIL PROTECTED] wrote: Does anyone know how to (if even at all possible) remap the softkeys on theCisco 7960's for various conditions (without having to purchase Call Manager)Example:Cisco 7960 (running SIP Version 8.2) has a call in progress and displays thefollowing softkeys:'Hold''EndCall' 'Confrn''More'Pressing 'More' gets you: 'Trnsfer' and 'BlndXfer'I would much rather re-order the softkeys to be:'Hold''EndCall' 'Trnsfer' 'More'Instead of the default.I know with Call Manager you can modify the SIP soft key templates, but if one doesn't have call manager, is there still away to do this?Thanks.Scott HigginbothamSystems / Network Operations Manager215.259.2185 or 1.800.835.5710 ext 2185 [EMAIL PROTECTED]___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 BLA
Works great using SCCP. On 6/13/06, Steve Glaus [EMAIL PROTECTED] wrote: While I'm frantically scouring this list, does anyone have anyinformation about getting BLA (busy line appearance) working on Cisco 7960? The last I heard was that this wasunsupported in Cisco's SIP firmware___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Lacy MooreAspendora, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 International
Shaun wrote: I'm having a problem with my Cisco 7960 phones with the SIP image. When i try to dial a international number i keep getting a busy signal but i dont see anything on the asterisk console (-vc) like i do when i dial local or long distance numbers. sip debug peer your-phone-extension-number-here and check your debug messages for what your phone is sending to asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)
[EMAIL PROTECTED] ha scritto: On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote: [EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess intenal is not the righe context :-) Sergio The from-sccp-internal is almost an exact copy of my from-sip-internal context, which works fine there's a typo in your sccp.conf intenal instead internal, so of course the context does not exists Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)
[EMAIL PROTECTED] ha scritto: context = from-sccp-intenal I guess intenal is not the righe context :-) Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
