Re: [asterisk-users] PRI Splitter
Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then splitting it up so that 6 channels went to a video system (h.320) and 17 channels to our PBX. You can also convert the signaling or send out on different type of connections like v.35. Pretty cool device and rock solid. We never had any problems with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, August 27, 2008 8:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI Splitter Asterisk? PaulH Jeremy Mann wrote: Does anyone know of a pri splitter device? Something that would take an incoming PRI, and based on DID send that out one of other multiple PRI ports? I'm needing to take a single PRI from the telco, and send it to two separate phone systems(one asterisk) based on DID. I know I could probably achieve the same thing with a 3 port PRI card in a server, but I'd like something braindead easy to configure from both a hardware and software perspective. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 ___ -- 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] Polycom LDAP Corporate Directory
Any more information on this? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of faraz Sent: Friday, April 18, 2008 6:30 PM To: Watkins, Bradley Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory please do!. how much did the 50 cost you? On Fri, 2008-04-18 at 18:22 -0400, Watkins, Bradley wrote: I actually just ordered 50 licenses to give this and the other applications a try. I'll post my results to the list once I get them and have had a chance to play around. Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of faraz Sent: Friday, April 18, 2008 6:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom LDAP Corporate Directory I havent tried it. I have quite a few polycoms and didnt even know polycom had this feature! :) This is obviously a separate peice of software that must be purchased and installed on the phones. Looks amazing though- any idea on pricing?. On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote: Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip /applications/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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
[asterisk-users] Polycom LDAP Corporate Directory
Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- 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] Hardphone SIP phone costs
I understand the maximizing pricing and branding aspect of phones but when you look at feature set it just doesn't make sense. And as far as purchasing the phone you can get it without a contract at the same price. When I starting thinking about it, can anyone else see a time when desk phones are replaced by smart phones? Why would a company pay for work cell phone and desk phone when one device could potentially do it all? I know there are issues that need to be considered like safety (911) for one. But can anyone else see where I'm coming from on this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 19, 2008 6:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardphone SIP phone costs On Wed, Mar 19, 2008 at 1:32 AM, John Faubion [EMAIL PROTECTED] wrote: when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Remember that the service providers such as ATT, Cingular, Sprint, Verizon and so forth, subsidize the cost of the phones because they make it up over the course of the contract. Hence the reason that some phones that have an initial cost when sold with a 1 year contract may be free initially with a 2 year contract. Even some VoIP phones and ATA's are done this way but only through service providers. Take the subsidies away and that iPhone is pretty pricey. John Cisco charges the premium because they can. The name alone commands the premium price. If people were not buying, they would obviously lower prices. I am sure their bean counters have computed the price to sales ratio to an exact science. It is called maximizing profits. I also have a habit of noting computers and phones when when watching TV. They have great product placement in many shows as does the Mac. The Stargate series is one of the few shows where they use Dell laptops. Product placement is a good way subtly put Cisco into your head, the other is their massive advertising campaigns on CNN and other business oriented channels. I stick with Polycom and have no complaints. Thanks, Steve Totaro ___ -- 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
[asterisk-users] Hardphone SIP phone costs
I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping RD costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- 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] Druid Open Source Edition
Is there a limit on how many phones you can use? I couldn't find anything on the website about this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Wilson Sent: Wednesday, March 12, 2008 1:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Druid Open Source Edition I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last year and I will now be making the switch to druid. It came with a user portal that was easy to use and had alot of great features. Has anyone downloaded this system today?? Can you please let me know what you think as well. -Josh ___ -- 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
[asterisk-users] Linksys SPA-942 Phones
Hello List, After seeing a few positive responses for the Linksys SPA-942 phones I was hoping to get some answers on the following questions: * How do the phones handling system wide paging? Is it similar to the Polycom phones? * Can a corporate directory be configured with the phones using Asterisk? * How is the speaker phone quality? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- 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
[asterisk-users] Script for seeding polycom phones with an extension directory
Hello List,
Not sure if this will be helpful but I made changes to the original
Cisco directory.php.txt script and applied them for use on the Polycom
phones. This will create an extension directory and alphabetize it
based on the sip registrations you have setup in sip.conf. Note that
this only seeds the phones and does not synchronize them. Anyway
thought it might save people some time. To run do: php scriptname
/home/polycom/-directory.xml.
?
header(Content-type: text/xml);
header(Connection: close);
header(Expires: -1);
// location of asterisk config files
$location = /etc/asterisk/;
// parse sip.conf
$sip_array = parse_ini_file($location.sip.conf, true);
while ($v = current($sip_array))
{ if (isset($v['name']))
{ $directory[] = fn. $v['name']./fn\n.
ct.key($sip_array)./ct\n;
}
next($sip_array);
}
sort($directory);
echo directory\n;
echo item_list\n;
foreach ($directory as $v) {
echo item\n;
echo $v;
echo /item\n;
}
echo /item_list\n;
echo /directory\n;
?
Roy Anciso
Director of Technology
Manistee Intermediate School District
1710 Merkey Road
Manistee, MI 49660
Ph: 231-723-4264
Fx: 231-723-1690
[EMAIL PROTECTED]
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[asterisk-users] Followme
I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which initiates followme) or to continue onto voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- 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] Followme
I've thought about that but is there a way to do this on whether or not they are configured in follow.conf. I didn't want to introduce unnecessary prompts for callers trying to reach people without followme enabled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Tuesday, January 22, 2008 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Followme In the dialplan you would just add a prompt and ask the caller to press 1 to locate or hold for voice mail. If they press 1 launch the followme app. On Jan 22, 2008 10:25 AM, Anciso, Roy [EMAIL PROTECTED] wrote: I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call comes in (external or internal) and rings extension with followme configured. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which initiates followme) or to continue onto voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- 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 -- Bruce Reeves Nortex Networks ___ -- 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] Voicemail - is it possible to automatically usethe extension being dialed from?
Heres what I do for this:
exten = *85,1,VoicemailMain(${CALLERID(NUM)})
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of arkda
Sent: Tuesday, January 22, 2008 5:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail - is it possible to automatically
usethe extension being dialed from?
Hi,
Is it possible to dial voicemail from a particular phone line and
automatically enter the extension that is being dialed from, thereby
only prompting for the password?
I've been searching around to find if this is possible, but I haven't
been able to find an example of this. I have a feeling it's more of a
endpoint function, but I thought I'd ask if anyone has accomplished this
with Asterisk.
Thanks!
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Re: [asterisk-users] cisco ip phne 7911G with asterisk
I'm running Asterisk 1.4.17 and as far as I know it only happens on the 7911g. And it's only issue when a user from a 7911g phone is leaving a message. Calls between sip users and PSTN sound good. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Friday, January 18, 2008 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote: Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the phone. Thanks I would try if a I have more time. Nowadays I have problems with the sound quality in the conversatio. 7941g with SIP-8-0-3 firwmware sounds well but 7911G with the same firmware version sounds really bad in an Asterisk 1.2.15. I listen echo and noise taps when I talk with other sip users of the same Asterisk. I have test the phone with the same firmware againts the lastest version of Asterisk (subversion stable tree) and there is no problem with the sound quality. Do you know what is the problem in this case? Or if there is a bug in Asterisk 1.2 that could affect in that way the audio quality? Thanks, However againts the 1.4.17 version there is no problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Wednesday, January 16, 2008 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Tue, Jan 15, 2008 at 01:14:42PM +, Christian Pinedo wrote: hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I thought that the phone should ask the SEPmac.xml.cnf file but it asks CTLSEPmac.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEPmac.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version. I would thank some clue. Thanks, It was a TFTP server issue. The classical TFTP server used in the unix world responds to queries with bad error codes. I finally used aTFTPD that does this well so the phone understands that there's no CTLSEP file and then asks for SEP file. -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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 -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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 ip phne 7911G with asterisk
Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the phone. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christian Pinedo Zamalloa Sent: Wednesday, January 16, 2008 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Tue, Jan 15, 2008 at 01:14:42PM +, Christian Pinedo wrote: hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I thought that the phone should ask the SEPmac.xml.cnf file but it asks CTLSEPmac.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEPmac.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version. I would thank some clue. Thanks, It was a TFTP server issue. The classical TFTP server used in the unix world responds to queries with bad error codes. I finally used aTFTPD that does this well so the phone understands that there's no CTLSEP file and then asks for SEP file. -- Christian Pinedo Zamalloa (zako) PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x828D0C80 Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80 ___ -- 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
[asterisk-users] Record calls then send them to users voicemail
Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ -- 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 79xx XML services
Although it's not LDAP I used a script that I found on the voip wiki and
changed it so it looked at only sip configuration files. It also
alphabetizes the output so it can be displayed that way on the phone.
Below are my notes on the subject. If someone is willing to post this
to the wiki and send me a link, that would be awesome.
Cisco Phone Extension Directory Using Services Button
I used a PHP script that I found on the Internet and rewrote it to fit
our needs. Original code is found at:
http://users.marshall.edu/~twohig5/directory.php.txt
The new code only looks at the sip.conf file since we are only using sip
phones. My version also alphabetizes the directory. Below is the source
code:
Directory.php.txt
?
header(Content-type: text/xml);
header(Connection: close);
header(Expires: -1);
// location of asterisk config files
$location = /etc/asterisk/;
$dirname = MISD Directory;
// parse sip.conf
$sip_array = parse_ini_file($location.sip.conf, true);
while ($v = current($sip_array))
{ if (isset($v['name']))
{ $directory[] = Name. $v['name']./Name\n.
Telephone.key($sip_array)./Telephone\n;
}
next($sip_array);
}
sort ($directory);
echo CiscoIPPhoneDirectory\n;
echo Title.$dirname./Title\n;
foreach ($directory as $v) {
echo \nDirectoryEntry\n;
echo $v;
echo /DirectoryEntry\n;
}
echo \nPromptChoose Name and Press Dial/Prompt\n;
echo /CiscoIPPhoneDirectory\n;
?
From here you can schedule this to run every so often. Once the file is
created you must place it in your web directory on the server.
I chained the command and also wrote the output to an xml file in the
web directory. The command looks like this:
'php /etc/asterisk/directory.php.txt /var/www/html/directory.xml'
System Speeddials using Services Button
For speed dials I modified the php code to look to a specific file in
the asterisk directory called speeddials.conf. This file only contains
attributes that the php script will look for. This is great because you
only need to specify the number and name fields. Below is my example of
speeddial.php.txt (php code) and speeddials.conf (speed dials):
Speeddial.php.txt
?
header(Content-type: text/xml);
header(Connection: close);
header(Expires: -1);
// location of asterisk config files
$location = /etc/asterisk/;
$dirname = System Speed Dial;
// parse speeddials.conf
$ssd_array = parse_ini_file($location.speeddials.conf, true);
while ($v = current($ssd_array))
{ if (isset($v['name']))
{ $directory[] = Name. $v['name']./Name\n.
Telephone.key($ssd_array)./Telephone\n;
}
next($ssd_array);
}
sort ($directory);
echo CiscoIPPhoneDirectory\n;
echo Title.$dirname./Title\n;
foreach ($directory as $v) {
echo \nDirectoryEntry\n;
echo $v;
echo /DirectoryEntry\n;
}
echo \nPromptChoose Name and Press Dial/Prompt\n;
echo /CiscoIPPhoneDirectory\n;
?
Speeddials.conf
;System Speed Dial File
;This is used in conjuction with speeddial.php.txt
;
[9,7234264]
name=MISD Admin Office
[9,7236205]
name=MISD Special Ed Office
[9,7233521]
name=MAPS Supintendent Office
[9,7232547]
name=MAPS MHS
Once these files are create just run the php command:
'php speeddials.php.txt /var/www/html/speeddial.xml'
This will generate the speed dial file and place it in your web
directory.
You can also schedule this to run just like the extension directory
script.
Creating the Main Services Menu
To display these two items when the user presses the Services button you
first we need to create a file that contains the menus. I created a
file called services.xml and placed in the web directory /var/www/html/.
Then I wrote the menu structure using XML. I used the information found
on the Cisco website as guide to do this. Below is my services.xml file:
CiscoIPPhoneMenu
TitleInformation Services/Title
PromptPress to Enter/Prompt
MenuItem
NameExtension Directory/Name
URLhttp://192.168.1.94/directory.xml/URL
/MenuItem
SoftKeyItem
NameDir/Name
URL/URL
Position/Position
/SoftKeyItem
PromptPress to Enter/Prompt
MenuItem
NameSystem Speed Dial/Name
URLhttp://192.168.1.94/speeddial.xml/URL
/MenuItem
SoftKeyItem
NameDir/Name
URL/URL
Position/Position
/SoftKeyItem
/CiscoIPPhoneMenu
As you can see the example above uses the CiscoIPPhoneMenu tag. I
created a couple of menu items called Extension Directory System
Speed Dial which points to the directory.xml and speeddial.xml files we
created earlier.
For photos of how this looks on the phone visit:
http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones/photo#5135
353621777248450
http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones/photo
One major caveat is for some reason Cisco has a limit on how many
numbers you can display using the CiscoIPPhoneDirectory directive. I
believe it is 32. So to keep things sane I created a directory
/etc/asterisk/sip and departmentalized my sip registrations there.
You can tell asterisk to load the sip config files by inserting the
following in your main
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust list, which if populated, contains information about the servers to which the phone is attempting to connect and whether the server connection will be secure or nonsecure. Based on the information above an empty file will work just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, January 04, 2008 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, I don't remember if I had this same problem with a 7961G but I did figure out that you can not do an upgrade from factory default SCCP to the latest SIP 8.x.x firmware. In my case the phone just did not work properly. To make it work I downgraded the phone back to SIP 7.x firmware (iirc I used 7.5) and then upgraded to the latest SIP 8.x.x firmware. Hope this helps. Regards, Patrick ___ --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 7941G-GE with Asterisk and CTPSEP odyssee
NOT 09:28:58.221096 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:58.221849 tftpClient: retval = 10 NOT 09:28:58.222629 tftpClient: Non secure file requested NOT 09:28:58.235315 TFTP: [16]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:58.238209 TFTP: [16]:Finished -- rcvd 1 bytes NOT 09:28:58.241145 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:58.242856 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:58.244754 SECD: updateCTL: finished CTL update ERR 09:28:58.245704 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:02.648053 SECD: updateCTL: starting CTL update NOT 09:29:02.651331 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:29:02.652499 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:29:02.658547 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:29:02.661503 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:29:02.662335 tftpClient: look up server - 0 WRN 09:29:02.665405 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:29:02.668874 tftpClient: secVal = 0xa NOT 09:29:02.669746 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:29:02.671475 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:29:02.672277 tftpClient: retval = 10 NOT 09:29:02.673060 tftpClient: Non secure file requested NOT 09:29:02.684870 TFTP: [25]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:29:02.687805 TFTP: [25]:Finished -- rcvd 1 bytes NOT 09:29:02.691794 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:29:02.693428 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:29:02.695315 SECD: updateCTL: finished CTL update ERR 09:29:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon /english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust list, which if populated, contains information about the servers to which the phone is attempting to connect and whether the server connection will be secure or nonsecure. Based on the information above an empty file will work just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, January 04, 2008 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, I don't remember if I had this same problem with a 7961G but I did figure out that you can not do an upgrade from factory default SCCP to the latest SIP 8.x.x firmware. In my case the phone just did not work properly. To make it work I downgraded the phone back to SIP 7.x firmware (iirc I used 7.5) and then upgraded to the latest SIP 8.x.x firmware. Hope this helps. Regards, Patrick ___ --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 7941G-GE with Asterisk and CTPSEP odyssee
You're right. That was my mistake. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Friday, January 04, 2008 11:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee That is not the name the phone requests When uping my 7960, the empty file did the trick I so far am unable to go beyond 7.1 however, as Asterisk rejects anything I dial with 7.3 Anyone have SIMPLE sample config files? John Novack Anciso, Roy wrote: Try naming the empty file: SEP0019E7D16CD6.tlv Not CTLSEP0019E7D16CD6.tlv -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus Laube Sent: Friday, January 04, 2008 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Thanks for the hint. I just tried that although I only see my worries coming true: the CTLSEPmac.tlv file is the first one the phone requests when booting, no possibility to set something different as the SEPmac.cnf.xml should be loaded after the successful load of the CTL file. And thus the phone is looping with Configuring IP and CTLFile failure. Can I set this option by ssh? Thanks a lot and in advance, Christophorus In your SEPmac.cnf.xml file look for the setting below and set it to 0: deviceSecurityMode0/deviceSecurityMode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Glenn Cobb Sent: Friday, January 04, 2008 9:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee Here is a little more info... I hooked up the 7971G-GE to my pc and grabbed this with tera-term. Its the console output during the CTL update process. I am using SIP70.8-3-3. NOT 09:28:45.969295 DHCP: Restart - delay = 1 NOT 09:28:45.981198 DHCP: Sending Release... NOT 09:28:49.000449 DHCP: dhcpSendReq: status 0x12301000 NOT 09:28:49.001281 DHCP: Sending Request... NOT 09:28:49.015673 DHCP: ACK received NOT 09:28:49.016517 DHCP: Succeeded NOT 09:28:49.058273 DHCP: IP Address -- 10.10.10.247 NOT 09:28:49.059129 DHCP: Subnet Mask - 255.255.255.0 NOT 09:28:49.059960 DHCP: Default Gwy - NOT 09:28:49.073169 PAE: SIGIPCFG received... NOT 09:28:49.075897 ESP: send ADMIN, logging = 1, shell = 0, ipconfig = 1 WRN 09:28:49.120127 SECD: WARN:getCTLInfo: ** phone has no CTL WRN 09:28:49.127292 SECD: WARN:getCTLInfo: ** phone has no CTL NOT 09:28:49.140946 CDP-D: catchipcfg getdhcpinfo IP: a0a0af7 Chng:1 NOT 09:28:49.152532 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.178685 tftpClient: request server 1 --- NOT 09:28:49.201261 tftpClient: request server 0 --- 10.10.10.10 NOT 09:28:49.204518 ESP: server 0 = 10.10.10.10 NOT 09:28:49.228784 tftpClient: request server 1 --- NOT 09:28:49.233253 ESP: server 1 = NOT 09:28:49.319960 SECD: updateCTL: starting CTL update NOT 09:28:49.323284 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:49.324525 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:49.327942 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:49.331598 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:49.332439 tftpClient: look up server - 0 WRN 09:28:49.335498 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:49.339140 tftpClient: secVal = 0xa NOT 09:28:49.340260 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:49.341141 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:49.341897 tftpClient: retval = 10 NOT 09:28:49.342678 tftpClient: Non secure file requested NOT 09:28:49.356155 TFTP: [26]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:49.359594 TFTP: [26]:Finished -- rcvd 1 bytes NOT 09:28:49.363943 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:49.365631 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:49.367522 SECD: updateCTL: finished CTL update ERR 09:28:49.368469 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:28:53.768028 SECD: updateCTL: starting CTL update NOT 09:28:53.772517 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:53.773673 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:53.776093 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:53.778770 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:53.779616 tftpClient: look up server - 0 WRN 09:28:53.782887 SECD
Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee
/CTLFile.tlv NOT 09:28:53.818805 SECD: updateCTL: finished CTL update ERR 09:28:53.819756 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:28:58.199464 SECD: updateCTL: starting CTL update NOT 09:28:58.202780 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:28:58.203933 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:28:58.205791 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:28:58.208403 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:28:58.209244 tftpClient: look up server - 0 WRN 09:28:58.215701 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:28:58.219254 tftpClient: secVal = 0xa NOT 09:28:58.220320 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:28:58.221096 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:28:58.221849 tftpClient: retval = 10 NOT 09:28:58.222629 tftpClient: Non secure file requested NOT 09:28:58.235315 TFTP: [16]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:28:58.238209 TFTP: [16]:Finished -- rcvd 1 bytes NOT 09:28:58.241145 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:28:58.242856 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:28:58.244754 SECD: updateCTL: finished CTL update ERR 09:28:58.245704 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:02.648053 SECD: updateCTL: starting CTL update NOT 09:29:02.651331 SECD: ctlRequestFile: Socket 7 connected to /usr/tmp/tftpClientSock NOT 09:29:02.652499 SECD: ctlRequestFile: Request CTLSEP0019E7D16CD6.tlv NOT 09:29:02.658547 tftpClient: tftp request rcv'd from /usr/tmp/ctlSock, srcFile = CTLSEP0019E7D16CD6.tlv, dstFile = /usr/tmp/CTLFile.tlv NOT 09:29:02.661503 tftpClient: auth server - tftpList[0] = 10.10.10.10 NOT 09:29:02.662335 tftpClient: look up server - 0 WRN 09:29:02.665405 SECD: WARN:lookupCTL: CTL update in progress, no old CTL, assume TFTP 10.10.10.10 NONSECURE NOT 09:29:02.668874 tftpClient: secVal = 0xa NOT 09:29:02.669746 tftpClient: 10.10.10.10 is a NONsecure server NOT 09:29:02.671475 tftpClient: temp retval = SRVR_NONSECURE, keep looking NOT 09:29:02.672277 tftpClient: retval = 10 NOT 09:29:02.673060 tftpClient: Non secure file requested NOT 09:29:02.684870 TFTP: [25]:Requesting CTLSEP0019E7D16CD6.tlv from 10.10.10.10 NOT 09:29:02.687805 TFTP: [25]:Finished -- rcvd 1 bytes NOT 09:29:02.691794 SECD: ctlRequestFile: tftp Status 0 rcv'd ERR 09:29:02.693428 SECD: ctlVerifyFile: CTL file too small: /usr/tmp/CTLFile.tlv NOT 09:29:02.695315 SECD: updateCTL: finished CTL update ERR 09:29:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon /english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust list, which if populated, contains information about the servers to which the phone is attempting to connect and whether the server connection will be secure or nonsecure. Based on the information above an empty file will work just fine. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Sent: Friday, January 04, 2008 5:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee On Fri, 2008-01-04 at 09:11 +0100, Christophorus Laube wrote: Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEPmac.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, I don't remember if I had this same problem with a 7961G but I did figure out that you can not do an upgrade from factory default SCCP to the latest SIP 8.x.x firmware. In my case
Re: [asterisk-users] 7970 CTLFile.tlv?
I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv? I've got a Cisco 7970 that's not completing its network registration to Asterisk. The Registering message stays on the screen (with the moving time wheel). After a few minutes, the onscreen message flashes Updating CTL then Loading..., then the status messages update with: No valid CAPF server File Not Found: CTLFile.tlv No CTL installed SEPMAC.cnf.xml (where MAC is the phone's MAC addr minus :s) before repeating the cycle (forever). Where can I get a CTLFile.tlv , or remove the requirement for it? Or is there another way to fix this problem? TIA. Asterisk v1.2.9.1-BRIstuffed-0.3.0-PRE-1q SCCP firmware Load File: TERM70.7-0-1-0s App Load ID: Jar70.2-9-0-117.sbn JVM Load ID: CVM70.2-0-0-112.sbn OS Load ID: cnu70.2-7-4-134.sbn Boot Load ID: 7970_64060118.bin -- (C) Matthew Rubenstein ___ --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 7961 new firmware stops readingconfiguration files
Chad, You might want to upgrade to the latest firmware. I have 7961g on 8-3-3SR2S and works very well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Osmond Sent: Thursday, December 20, 2007 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files Hello, I have been running SIP41.8-2-2SR4S firmware on the Cisco 7961 and have recently upgraded to SIP41.8-3-2SR1S to add the DND button on the front of the phone and also to hopefully resolve some issues with the phones not registering after a long period. Once we upgraded the phones now display Error Verifying Config Info in the Status messages and will not process the configuration file. To make a change on the phone I have to downgrade to 8.2.2R4 and change the configuration, and then upgrade to 8.3.2R1, which is a bit of a pain. The tftp logs indicate that the phones is getting the correct SEPMAC.xml.cnf file but will not parse it, so that doesn't seem to be the issue. The Wiki pages for 79x1 indicate that it's a known issue, has anyone managed to get past the issue? I tried logging a call with Cisco TAC, but they're giving the We don't support SIP on anything other then CME... Thanks, Chad __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ --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
[asterisk-users] Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH
For those using Cisco 7911g phones, I am running into an issue with one the Cisco demo phones we have. The 7961 works great with asterisk no problems However, the 7911g gets audio clipping when recording voicemails or the unavailable message. Also when a call is transferred using the 7911g the music stops and then starts to play the beginning of the MOH file and continues until the voicemail recording takes over. The best way to describe it is like a dj scratching records. Does anyone else have issues like these? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Play Beep instead of MOH
Is there a way to tell asterisk to beep every few seconds rather than play MOH. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Digium TE120P versus Sangoma A101D-X
Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called Digium about this and their tech support does not recommend using their HLEC software canceller on T1 cards since it consumes so much CPU. I was ready to get the license keys for HLEC but when I was transferred to sales person they would not give me the keys stating that I have to have an analog card to obtain the license. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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] Music on Hold Problem w/ Transfers
I'm having this problem. Here is my output with verbosity on 10: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2524-099012b0, SIP/2523|15) in new stack -- Called 2523 -- SIP/2523-09905220 is ringing -- SIP/2523-09905220 answered SIP/2524-099012b0 -- Packet2Packet bridging SIP/2524-099012b0 and SIP/2523-09905220 -- Started music on hold, class 'default', on SIP/2524-099012b0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2523-0990f110, SIP/2500|15) in new stack -- Called 2500 -- SIP/2500-09913080 is ringing -- Stopped music on hold on SIP/2524-099012b0 == Spawn extension (default, 2523, 1) exited non-zero on 'SIP/2523-0990f110ZOMBIE' -- Nobody picked up in 15000 ms [Nov 21 13:25:05] NOTICE[14600]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/2500-09913080' not posted -- Executing [EMAIL PROTECTED]:2] VoiceMail(SIP/2524-099012b0, u2500) in new stack -- SIP/2524-099012b0 Playing '/var/spool/asterisk/voicemail/default/2500/unavail' (language 'en') == Spawn extension (default, 2500, 2) exited non-zero on 'SIP/2524-099012b0' -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 21, 2007 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold Problem w/ Transfers Lacy Brian, Could you please set verbosity to 10, then place your calls/holds/transfers and post the output? Both where it works and where it doesn't. Otherwise, helping you troubleshoot this will be difficult. Tony Plack On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works. After posting, I dialed my cellphone, and music on hold works in all situations. It's something having to do with internal calls. I don't really care if that isn't working. I didn't think to try that first. ___ --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] Music on Hold Problem w/ Transfers
Asterisk version 1.4.13 Also when I listened in on a transfer it sounds like the moh is trying to start but then immediately stop and tries to start again. Below is my musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh random=no -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 21, 2007 6:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold Problem w/ Transfers Started music on hold, class 'default', on SIP/2524-099012b0 -- Please post your [default] section of musiconhold.conf Also need to know what version of Asterisk, version of kernel. Do you have ztdummy loaded (lsmod)? ___ --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
[asterisk-users] Cisco phones and 32 directory object limit
Hello List, For those of you with Cisco phones and XML directories and large user bases, how do you handle the 32 directory object limit? I know you can create multiple xml files with 32 objects in each but this just seems really sloppy. I would like to have one large directory. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Page Command
Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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 7911/7941/7970/7971 Softkey XML Files
Sorry forgot the images: http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Thursday, November 15, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files The softkeys translate fine. Things like redial, new call, call forward, transfer, conference, hold, end call, do not disturb (for DND you have to go through a few more menus) line selection, services. I'll try attaching a screenshot of the softphone I have setup. I've setup the services button so you can browse the local extension directory (based on the sip.conf file) and I also setup a script to generate system speeds dials for all the phones. It also alphabetizes them automatically. I'm hoping to use a nonstandard template to make things like DND a bit more accessible. I just received the 7941 7911g phones from our Cisco rep I'm working on loading the SIP image on those. Oh the other thing I created is a script for auto generation of your SEPmac.cnf.xml file for each phone. You just enter in the mac address the sip extension, password, display name and phone label and the xml file is automatically generated. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, November 15, 2007 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... Without softkey configuration files, I've heard you cannot translate menus when connecting a Cisco SIP phone to any non-Cisco SIP server. -Greg ___ --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 7911/7941/7970/7971 Softkey XML Files
The softkeys translate fine. Things like redial, new call, call forward, transfer, conference, hold, end call, do not disturb (for DND you have to go through a few more menus) line selection, services. I'll try attaching a screenshot of the softphone I have setup. I've setup the services button so you can browse the local extension directory (based on the sip.conf file) and I also setup a script to generate system speeds dials for all the phones. It also alphabetizes them automatically. I'm hoping to use a nonstandard template to make things like DND a bit more accessible. I just received the 7941 7911g phones from our Cisco rep I'm working on loading the SIP image on those. Oh the other thing I created is a script for auto generation of your SEPmac.cnf.xml file for each phone. You just enter in the mac address the sip extension, password, display name and phone label and the xml file is automatically generated. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, November 15, 2007 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... Without softkey configuration files, I've heard you cannot translate menus when connecting a Cisco SIP phone to any non-Cisco SIP server. -Greg ___ --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 7911/7941/7970/7971 Softkey XML Files
The Cisco Documentation states that you can modify standard and nonstandard softkey templates. They may not be xml files. I just assumed they were xml since that is what is used to configure the phone. Here is snip from the 7911G documentation that states you can configure the private key (which is really all I need) and modifying the softkeys: Configuring Softkey Templates Using Cisco Unified CallManager Administration, you can manage softkeys associated with applications that are supported by the Cisco Unified IP Phone 7906G and 7911G. Cisco Unified CallManager supports two types of softkey templates: standard and nonstandard. Standard softkey templates include Standard User, Standard Feature, Standard IPMA Assistant, Standard IPMA Manager, and Standard IPMA Shared Mode Manager An application that supports softkeys can have one or more standard softkey templates associated with it. You can modify a standard softkey template by making a copy of it, giving it a new name, and making updates to that copied softkey template. You can also modify a nonstandard softkey template. To configure softkey templates, select Device Device Settings Softkey Template from Cisco Unified CallManager Administration. To assign a softkey template to a phone, use the Softkey Template field in the Cisco Unified CallManager Administration Phone Configuration page. Refer to Cisco Unified CallManager Administration Guide, Cisco Unified CallManager System Guide for more information. Here is the link to configuring a 7911G phone with SIP and instructions for modifying the softkeys. http://cisco.com/en/US/customer/products/hw/phones/ps379/products_admini stration_guide_chapter09186a0080798562.html#wp1058838 So again, if anyone can post the softkey templates I would greatly appreciate it. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Oliver Sent: Wednesday, November 14, 2007 3:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. I have done packet captures to prove it out from CCM 5.x and 6.0. Sorry, no xml files to accomplish it. Maybe one day they will be less of basterds?!?!?!?!? -Greg Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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 7911/7941/7970/7971 Softkey XML Files
There is an option to specify a softkey file in SEPmac.cnf.xml. I have an email into our Cisco rep. I'm hoping he can shed some light on this. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, November 13, 2007 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. As far as I know (and I might be very wrong), you can't change the soft key configuration of Cisco phones with the SIP Firmware. Maybe you can with Cisco's CallManager - I don't know. Someone PLEASE correct me if I'm wrong because I've been wanting to do this for a year ___ --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 IP Communicator with Asterisk
The version I have is 2.1.2.0. It makes for a really nice software sip phone:) The other thing I should note is that you only need the SEPXXX.cnf.xml file and dialplan.xml file in your tftp directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen Sent: Friday, November 09, 2007 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco IP Communicator with Asterisk Great info. Could you specify which version of IPCommunicator you got to work like this? Thanks! ___ --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
[asterisk-users] Cisco IP Communicator with Asterisk
I'm not sure if anyone has done this before or not but, I was able get the Cisco IP Communicator soft phone to work with Asterisk using SIP. Thought I would share my experiences. The key is on the installation. To have the software use the SIP protocol type the following command: msiexec /i CiscoIPCommunicatorSetup.msi /qb SIP=1. After installation configuration is just like configuring a Cisco 7970 hard phone. I used the configuration instructions outlined by Kerry Garrison at Asterisk Tutorials http://www.asterisktutorials.com/showproduct.php?ProductID=10. Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Selecting OSLEC for zaptel-1.4.6
Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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] Selecting OSLEC for zaptel-1.4.6
Thanks I was trying to patch 1.4.6 using the 1.4.1.patch. The 1.4.4 patch did the trick:) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Tuesday, November 06, 2007 4:53 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6 Dave Fullerton wrote: Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. snip / Roy Anciso You shouldn't need to. As long as you have applied the oslec-zaptel patch it should be selected automatically. You can double check it by looking in zconfig.h I can confirm this - I just upgraded from 1.4.5.1 to 1.4.6 today and the zaptel-1.4.4 oslec patch applied cleanly to the zaptel source tree (with a bit of fuzz) and the build was good too. When the zaptel module loads (the from the init script) OSLEC is installed automatically. Alan ___ --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] Outgoing PRI CID?
I do this to tie extensions to a particular number:
exten = _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD2317231516)
exten = _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology2317234264)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Turbo
Fredriksson
Sent: Thursday, November 01, 2007 2:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Outgoing PRI CID?
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...
- s n i p -
[default]
include = outgoing
include = priin
[outgoing]
exten = _NX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local
number (w/o areacode) - Stockholm
exten = _0NX.,1,Macro(dial,${EXTEN},30,r)
[priin]
exten = _X.,1,Dial(IAX2/graham/${EXTEN},30,r)
[macro-dial]
exten = s,1,NoOp(Trying extension/number: ${ARG1} from
${CALLERID(num)})
;exten = s,n,Set(CALLERID(num)=${CALLERID(num)})
exten = s,n,Dial(Zap/g1/${ARG1},${RINGTIME},r)
exten = s,n,Playback(connection-failed)
exten = s,n,Congestion()
- s n i p -
This * is only for PRI connection. The actual routing is done
in an * installation running under a XEN domain...
Incoming works exactly as planed. So is the 'macro-dial'
with the exception that the numer shown in the receiving
end is '500' (the switchboard). This even if/when I call
from '528'...
Any ideas?
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Re: [asterisk-users] Cisco Phones
That is correct. Our Cisco rep is sending us a 7911G and 7941G so we can test with asterisk. We plan on converting them over to SIP for testing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Thursday, October 25, 2007 3:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco Phones Can you comment on the use of these phones with asterisk with the Skinny images? I think you're talking about Cisco phones converted to using the SIP image. Moj Alex Balashov wrote: Roy, While there is a difference in the feature set provided by the SIP and Skinny images for the Cisco phones, the loss is not appreciable in my view. There are some differences in interface aesthetics as well. ___ --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
[asterisk-users] Cisco Phones
For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] Cisco phones with Asterisk
Hello List, For those of you using Cisco phones, did you have to purchase a 'SIP license' for each phone? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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
[asterisk-users] What web GUI are people happy with?
Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --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] What web GUI are people happy with?
Thanks for your suggestion, I saw mention of the asterisk-gui in a previous post but didn't see much response on it. As I mentioned in my original message I have installed Asterisk from source and I also have a good understanding of how and why asterisk works. However I would like to make it simple for some of our sysadmins who are not comfortable with CLI to make simple changes hence the web gui. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alan Lord Sent: Monday, October 15, 2007 2:16 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What web GUI are people happy with? Anciso, Roy wrote: Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Why don't you just install the Asterisk GUI? Get it from SVN: svn co http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui Instructions to build are in the latest TFOT book. But FWIW, I had a quick look but decided against it. And other GUIs too. My installation is quite small, admittedly, but I really want to understand how and why my system is configured the way it is. The only way to truly get that is to do it by hand... You only need Vi :-) Alan -- The way out is open! http://www.theopensourcerer.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 ___ --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
