search for: 519961

...l work 3/ With a special cable/dongle (a la wikki) I am looking at getting several 20 x 7960 (not Gs) to work with * and a NetGear FSM7326P switch. Do I also need to get a PowerDSine converter dongle for each phone? TIA - -- Ron Wellsted http://www.wellsted.org.uk ron@wellsted.org.uk FWD: 519961 N 52.567623, W 2.137621 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.6 (GNU/Linux) iQEVAwUBQjAYlEtP/KMNOfRbAQK/PQgAseiWaL9WD7EEbsXVitARMHch4ewVyJva WyOW7HqV1wh/kQj8RPAANyIwDuLcBUiJ1SzMLZeNM7qq4YEAuvqua8mFUlh/VjnR yA0RkIM83im54RZQzYELwUOGtWH0znbdlGJc6qFoGgNAkA9BBA/pBmYrZb2syGgX IrbMhQTSlPs8hE8...

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

...ng 1.0.7 without this problem, I upgraded to fix a problem with Monitor (the call stopped monitoring when transfered, 1.2.5 has fixed this). Does any one have any suggestions? - -- Ron Wellsted ron@wellsted.org.uk http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.2.2 (GNU/Linux) iQEVAwUBRCFQrktP/KMNOfRbAQJD6wf+KLQHBoI48MSrfRsetvgeB3xdsednTarf aeKx1SnZssv9TELa4a3Kg8EcC/nbIXBCV+NMH28xqmS8/+PXfx+q8oIg8noHAYrO VU/zFw6+O+CtDadlDnd0MjInvG4wQNHo7ec29fLpTBn44yMw1UiYkd02WjSj/hCv PIVEZXqnGLerOxoD3SvtILI1OXZdz6K7j7juQ...

I have a connect to * via H.323/g711 from device A and want to connect to B which want for H.323/g729 h323.conf contains disallow=all allow=alaw allow=g729 but outgoing faststart/TCS contains only g711 (from h323_request(format) i think) and so no codec negotiation and no voice. Howto run up g711/H323 -> * -> g729/H323 PS intel's g729 was used. ast 1.0.3-6 PPS stupid -

This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering "stop now" kills both the console attachment as well as the background process. I need to attach to the running asterisk in order to do "init keys" but once I do that, it seems I cannot just

My VoIP provider allows me to have more than one call outbound on the same line simultaneously, for some reason. I am pretty sure that they do not want this to happen, so I'd like instead to limit each line to one call. I do not want the users to have to dial another prefix to go out on another line. Is there any way to add multiple accounts for my _9. extension and have Asterisk

Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i "resume" the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had

Anyone have screen size + resolution for the cisco 7940g, 7960g? I'm having a hard time finding it anywhere. No vendors have the information and cisco doesn't list it. All I've been able to find is the rez of the 7940g : 100x145. -Dan

is there a way to support login and logout functionality in a phone? We are using Cisco 7940 and 7960 phones and have 2 shift. We want to be able to use the same phone using like 2 different extensions. The phone will then "remember" your settings if possible, if anyone has left you a voice mail etc. Is this possible? Regards, Paul -------------- next part -------------- An HTML

Hi All, Anyone have a working CentOS 6 kickstart file that they are using to bind a host to Active Directory 2008 R2? I'm working on a full AD/Linux environment and would like to stand on the shoulders of others if they are already doing such a thing. I'm thinking I need to enable LDAP and Kerberos, although Winbind might also be the key here. The config will ideally get the UID and

...hine? > > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on the exact procedure for your distribution. > > HTH > > - -- > Ron Wellsted > http://www.wellsted.org.uk > ron@wellsted.org.uk > FWD:519961 Gossiptel:9309811 > N 52.567623, W 2.137621 > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.2.6 (GNU/Linux) > Comment: Using GnuPG with Thunderbird - > http://enigmail.mozdev.org > > iQEVAwUBQi4T10tP/KMNOfRbAQJajAgAso8fLd3qYmBhgfzUBrMDQ8jDE/kWH/4r > jcTiVHcsMxbm1kBxAL5...

I have been trying for sometime to upgrade this phone to either sip, or the latest sccp (at this point I don't care) with no success. It has an sccp image (don't know the version off the top of my head) that is incompatible with chan_sccp. When I boot up the phone, it asks the tfpt server for the .tlv file, then the SEP<mac> file which it receives successfully. It then asks for

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

I am trying to enable dial-by-email by using LDAPget to query an Active Directory server. I've got it retrieving the phone number fine. Unforunately, the numbers stored in active directory are either in the format: (xxx) xxx-xxxx or xxx-xxx-xxxx. Is there any way to parse characters out of the dialed phone number so that I only end up with digits (remove spaces, parenthesis and dashes)?

what happens when you drop your gains? use /etc/asterisk/zaptel.conf and fiddle with tx and rx values. Works, most of the time. -----Original Message----- From: John McEntee [mailto:john@mcentee87.freeserve.co.uk] Sent: Wednesday, October 11, 2006 12:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Echo problems on ISDN. (mainly incoming calls) OK I

I installed Asterisk on Gentoo using emerge. At first, emerge tried installing version 0.9 but reading the wiki showed how to get the latest stable. I'm running Gentoo kernel 2.6.11-gentoo-r9. Asterisk seems to be working just fine, but I'm concerned that since I don't have any Digium hardware, I may need a timer source. When I executed emerge zaptel, it installed zaptel 1.0.7

Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have

Maybe not the place for this but thought I'd post the info for others. I purchased a cisco 7960 off ebay and needed to convert to SIP for *. I know * supports SCCP but I wont go into that here. I'd read on voip-info.org that a contract could be purchased for approx $8 to allow me to download the firmware. I though, being in the UK, i'd get one through a reseller in the UK.

I have some problems starting asterisk with a hfc card using zaphfc: [chan_zap.so]Mar 8 17:53:06 WARNING[2447]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death Mar 8 17:53:06 WARNING[2447]: loader.c:391 load_modules: Loading module chan_zap.so failed! I'm using asterisk from debian/sid: Asterisk

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm