Re: [asterisk-users] Delay in IVR
Hi, sorry for my insistence but I would your aid for my problem. Thanks. -- Salvatore. - Original Message - From: Sasa s...@shoponweb.it To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 03, 2010 9:51 AM Subject: Re: [asterisk-users] Delay in IVR Hi, in trixbox I don't know what create an extension with letter but only with number. Thanks. -- Salvatore. - Original Message - From: Kingsley Tart kings...@skymarket.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 02, 2010 5:34 PM Subject: Re: [asterisk-users] Delay in IVR On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote: On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote: HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. I know nothing of Trixbox but I had a problem with my own dialplan where there was a delay with the user selecting 0 from my IVR menu. It turned out that because my extensions all started with 0 (they were real phone numbers), asterisk thought that the caller might be starting to type one of the valid extensions and so waited for the timeout (digit timeout I think) before it went further. To see if that's your problem, try seeing whether a menu option that won't match the start of any of your defined extensions happens more quickly. I got around it by having the IVR in a different context where the extensions started with a letter, so no entered digits would match. Salvatore, Did this help? -- Cheers, Kingsley. -- _ -- 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] Delay in IVR
Hi, here information request: extension number is 100/101 ring group number is 600 cpu : Intel(R) Pentium(R) D CPU 3.00GHz 3 GHz On another voip machine (always with Trixbox) I haven't this problem, I have tried with another phones and with XLite I have always this problem. About DTMF in SIP trunk I have (in USER Details) this parameter: dtmfmode=rfc2833 Another sip trunk configuration is: PEER Details: secret=yqyqyq nat=no context=from-pstn host=x.x.x.x insecure=very type=friend username=yyy In USER Details: canreinvite=no context=from-trunk dtmfmode=rfc2833 insecure=very nat=no port=5060 type=user I hope that informtions are enough for resolve my problem Thanks. -- Salvatore. - Original Message - From: mike mosier trixbo...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 09, 2010 3:36 PM Subject: Re: [asterisk-users] Delay in IVR I use to use trixbox its basically asterisk with free pbx. What are your extension numbers? Ring group number? What processor are you using? The more info the better. When I used trixbox I never had this problem. It could be DTMF, what is your dtmf in the trunk. What kind of trunk? Sip? What kind of phones. What is the drtmf setting on your phones? What kind of phone are you testing this with? I always have one test sip trunk that I know works great for testing. Its not likely in the code of the ivr. The problem is in how you setup everything else that leads to trunk. Respectfully Michael D Mosier Ftoc Certified On Jun 9, 2010 2:53 AM, Sasa s...@shoponweb.it wrote: Hi, sorry for my insistence but I would your aid for my problem. Thanks. -- Salvatore. - Original Message - From: Sasa s...@shoponweb.it To: Asterisk Users Mailing List - ... Sent: Thursday, June 03, 2010 9:51 AM Subject: Re: [asterisk-users] Delay in IVR Hi, in trixbox ... -- _ -- 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] Delay in IVR
Hi, I have tried with a some change in IVR configuration but the result
isn't changed, I have tried with Enable Directory and Enable Direct Dial
disabled, also I have tried with timeout=1 but nothing is changed !
My IVR configuration is:
trixbox1*CLI dialplan show ivr-2
[ Context 'ivr-2' created by 'pbx_config' ]
'1' =1. dbDel(${BLKVM_OVERRIDE})
[pbx_config]
2. Set(__NODEST=)
[pbx_config]
3. Goto(ext-group,600,1)
[pbx_config]
'2' =1. dbDel(${BLKVM_OVERRIDE})
[pbx_config]
2. Set(__NODEST=)
[pbx_config]
3. Goto(ext-group,600,1)
[pbx_config]
'fax' = 1. Goto(ext-fax,in_fax,1)
[pbx_config]
'h' =1. Hangup()
[pbx_config]
'hang' = 1. Playback(vm-goodbye)
[pbx_config]
2. Hangup()
[pbx_config]
'i' =1. Playback(invalid)
[pbx_config]
2. Goto(loop,1)
[pbx_config]
'loop' = 1. Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
[pbx_config]
2. GotoIf($[${LOOPCOUNT} 0]?hang,1)
[pbx_config]
3. Goto(ivr-2,s,begin)
[pbx_config]
'return' = 1. Set(MSG=custom/giornonew)
[pbx_config]
2. Set(_IVR_CONTEXT=${CONTEXT})
[pbx_config]
3.
Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT_${CONTEXT}}) [pbx_config]
4. Goto(ivr-2,s,begin)
[pbx_config]
's' =1. Set(MSG=custom/giornonew)
[pbx_config]
2. Set(LOOPCOUNT=0)
[pbx_config]
3. Set(__DIR-CONTEXT=default)
[pbx_config]
4. Set(_IVR_CONTEXT_${CONTEXT}=${IVR_CONTEXT})
[pbx_config]
5. Set(_IVR_CONTEXT=${CONTEXT})
[pbx_config]
6. GotoIf($[${CDR(disposition)} = ANSWERED]?begin)
[pbx_config]
7. Answer()
[pbx_config]
8. Wait(1)
[pbx_config]
[begin]9. Set(TIMEOUT(digit)=3)
[pbx_config]
10. Set(TIMEOUT(response)=1)
[pbx_config]
11. Set(__IVR_RETVM=)
[pbx_config]
12. ExecIf($[${MSG} != ]?Background(${MSG}))
[pbx_config]
13. WaitExten(,)
[pbx_config]
't' =1. Goto(loop,1)
[pbx_config]
Include ='ivr-2-custom'
[pbx_config]
-= 10 extensions (33 priorities) in 1 context. =-
Thanks.
--
Salvatore.
- Original Message -
From: David Backeberg dbackeb...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, June 02, 2010 8:46 PM
Subject: Re: [asterisk-users] Delay in IVR
On Mon, May 24, 2010 at 9:41 AM, Kingsley Tart kings...@skymarket.co.uk
wrote:
I know nothing of Trixbox but I had a problem with my own dialplan where
there was a delay with the user selecting 0 from my IVR menu. It turned
out that because my extensions all started with 0 (they were real phone
numbers), asterisk thought that the caller might be starting to type one
of the valid extensions and so waited for the timeout (digit timeout I
think) before it went further.
A similar thing can happen quite easily with FreePBX, where at least
in the past, the default was that every IVR had an implicit valid
selection of any extension in the system, unless you unchecked the box
that made that happen. It really explained why so many calls were
going to a particular phone before we found that default!
--
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Re: [asterisk-users] Delay in IVR
Hi, in trixbox I don't know what create an extension with letter but only with number. Thanks. -- Salvatore. - Original Message - From: Kingsley Tart kings...@skymarket.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 02, 2010 5:34 PM Subject: Re: [asterisk-users] Delay in IVR On Mon, 2010-05-24 at 14:41 +0100, Kingsley Tart wrote: On Mon, 2010-05-24 at 15:09 +0200, Sasa wrote: HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. I know nothing of Trixbox but I had a problem with my own dialplan where there was a delay with the user selecting 0 from my IVR menu. It turned out that because my extensions all started with 0 (they were real phone numbers), asterisk thought that the caller might be starting to type one of the valid extensions and so waited for the timeout (digit timeout I think) before it went further. To see if that's your problem, try seeing whether a menu option that won't match the start of any of your defined extensions happens more quickly. I got around it by having the IVR in a different context where the extensions started with a letter, so no entered digits would match. Salvatore, Did this help? -- Cheers, Kingsley. -- _ -- 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
[asterisk-users] Delay in IVR
HI, I have in 'inbound route' a IVR, with press 1 or 2 the destination call is always a ring group called '600', my problem is that after press 1 (but this problem is present also with press 2) before that the inbound call is transfer to extension pass 10/11 seconds ! In attach log file about incoming call. I use Trixbox with Asterisk-1.6.0.10. Thanks. -- Salvatore. -- _ -- 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
[asterisk-users] PCI analog cards on * vs. Quintum
What is the verdict? There was one positive response, but would like to hear a few more. In addition, what I am looking at is FXO ports to be used with GSM gateways, so any recommendations for specific cards are welcomed. From my experience with PRI cards, I am a little biased toward Sangoma. 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
[asterisk-users] chan_mobile handle 92 log flood
Dear all, Picked up some more BT usb adapters and got a flood of error messages as follows: hci_scodata_packet: *hci0 SCO packet for unknown connection handle 92* Anyone has any idea how to deal with this? Sasa Bobek ___ -- 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] chan_mobile one device per dongle?
yes, only one device per USB dongle. On Sat, Jul 18, 2009 at 4:22 PM, Steve Totaro stot...@totarotechnologies.com wrote: Hello, I read on the wiki that chan_mobile supports one device per dongle. Is this still the case? From the official website, there is very little info but this line Channel Groups for implementing ‘GSM Gateways’ which leads me to believe (or hope at least) that more than one phone can be paired to a dongle. Dongles are so cheap I guess it doesn't really matter other than more complexity. Anyone know for sure? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] Latest chan_mobile
In general, I found it hard to get chan_mobile working straight out of the box, and although there is a great effort to make it so, phone manufacturers are not helping by making command sets and BT implementations different from device to device, SW version to SW version. Elastix seems to have the most trouble free implementation out there and has certainly saved me a lot of time and money and I recommend you give it a go, before banging your head over code. You can check the buglist on Digium for further info or the list of compatible phones on voip-info.org, but it may be a USB dongle issue as well (CSR seems to be the safest bet after they fixed the error log flood). On Sat, Jul 18, 2009 at 3:27 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Hello, I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm -qa | grep blue pulseaudio-module-bluetooth-0.9.12-10.1 bluez-utils-3.36-7.1 kdebluetooth4-0.3-4.1.1 libbluetooth-devel-3.36-3.1 gnome-bluetooth-0.11.0-26.2 bluez-test-4.22-6.1.1 libbluetooth3-4.22-6.1.1 libbluetooth2-3.36-3.1 Thanks in advance! Carlos. ___ -- 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] Latest chan_mobile
Yes, chan_mobile works great on Elastix. If the migration is complicated, you may consider installing/testing it on an old computer. On Sun, Jul 19, 2009 at 2:21 AM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Thank for your time. Do you used chan_mobile with Elastix distribution successfully? If so, I will consider the switch. I can't jump to another distribution easily because I have a working environment that will make really hard the migration. On Sat, Jul 18, 2009 at 10:57 AM, Sasa Bobek sasa.bobek...@gmail.comwrote: In general, I found it hard to get chan_mobile working straight out of the box, and although there is a great effort to make it so, phone manufacturers are not helping by making command sets and BT implementations different from device to device, SW version to SW version. Elastix seems to have the most trouble free implementation out there and has certainly saved me a lot of time and money and I recommend you give it a go, before banging your head over code. You can check the buglist on Digium for further info or the list of compatible phones on voip-info.org, but it may be a USB dongle issue as well (CSR seems to be the safest bet after they fixed the error log flood). On Sat, Jul 18, 2009 at 3:27 PM, Carlos Ruiz Diaz carlos.ruizd...@gmail.com wrote: Hello I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm -qa | grep blue pulseaudio-module-bluetooth-0.9.12-10.1 bluez-utils-3.36-7.1 kdebluetooth4-0.3-4.1.1 libbluetooth-devel-3.36-3.1 gnome-bluetooth-0.11.0-26.2 bluez-test-4.22-6.1.1 libbluetooth3-4.22-6.1.1 libbluetooth2-3.36-3.1 Thanks in advance! Carlos. ___ -- 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 ___ -- 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] open source call center application for Asterisk
Truth is you don't need anything more then Asterisk to configure a call center On Mon, Jul 13, 2009 at 2:19 PM, ashish chauhan ashishchauhan07...@gmail.com wrote: Dear all, I am new to asterisk.i like to configure call center using asterisk.please can anyone tell me open source application to fulfill my requirement. thanks Ashish Kumar Chauhan M T S ,C D A C Chennai ___ -- 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] Using a mobile phone via USB as an extension
Just google/bing it. http://voip-info.org/wiki/view/chan_mobile On Thu, Jul 9, 2009 at 12:56 PM, Olivier oza-4...@myamail.com wrote: 2009/7/2 Carlos Ruiz Diaz carlos.ruizd...@gmail.com Check chan_mobile. Now is mature enough to be used in a server with low CPS. The USB connectivity will be introduced in the close future (I think) but by now it can be connected via bluetooth device. Where did you get this info (USB connectivity for chan_mobile) ? Is there a way to learn a bit more ? ___ -- 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] chan_mobile help.
Could not agree more. I had chan_mobile up and running with an older version of Trix, but never managed to recreate it with the latest versions. Other people I talked to even suggested that it was made on purpose. With elastix the only problem I had was the missing mobile.conf.example, but you can create one from the Trix instructions from scratch or download it from the SVN. On Tue, Jul 7, 2009 at 7:56 PM, Razza razz...@gmail.com wrote: Seems the only option is to give Elastix a go. ___ -- 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] chan_mobile help.
I had loads of issues when trying i on trix, but the same procedure worked
like a charm with elastix.
On Sun, Jul 5, 2009 at 7:09 PM, Razza razz...@gmail.com wrote:
I've been failing to get chan_mobile working, so am looking to the experts
to help :o)
I have followed this guide -
http://www.voipphreak.ca/2008/10/30/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/
and this guide -
http://www.geek-pages.com/articles/asterisk/howto_build_and_configure_chan_mobile_on_trixbox.html
and tried hybrids of the two which is dangerous! ;o)
When I pair my phone to my asterisk server, I see no adverstised
bluetooth services (such as headset etc.), is that correct?
Asterisk (every 30 seconds ish) reports -
-- Bluetooth Device PersX1 has connected.
-- Bluetooth Device PersX1 has disconnected, reason (104).
Also, when I run mobile search from the CLI, I get mismatched MAC's
agains phones (e.g. the MACs and phones swap?!) -
*CLI mobile search
Address Name Usable TypePort
-- Bluetooth Device PersX1 has connected.
-- Bluetooth Device PersX1 has disconnected, reason (104).
00:17:83:16:DD:85 Ray(Work) No Headset 0
00:1C:CC:63:15:DD Ray pers No Headset 0
00:23:45:32:78:57 BlackBerry PMWork 8310 YesHeadset 1
00:16:41:63:3C:5C Ray pers No Headset 0
00:1D:F6:C6:C1:77 BTG209604 No Headset 0
*CLI mobile search
Address Name Usable TypePort
-- Bluetooth Device PersX1 has connected.
-- Bluetooth Device PersX1 has disconnected, reason (104).
00:16:41:63:3C:5C BTG209604 No Headset 0
00:23:45:32:78:57 Ray pers YesHeadset 1
00:17:83:16:DD:85 Ray pers No Headset 0
00:1C:CC:63:15:DD Ray(Work) No Headset 0
00:1D:F6:C6:C1:77 BlackBerry PMWork 8310 No Headset 0
*CLI
Apparently my dongle is fully supported, lsusb -v yields -
Bus 002 Device 002: ID 0a12:0001 Cambridge Silicon Radio, Ltd Bluetooth
Dongle (HCI mode)
Device Descriptor:
bLength18
bDescriptorType 1
bcdUSB 2.00
bDeviceClass 224 Wireless
bDeviceSubClass 1 Radio Frequency
bDeviceProtocol 1 Bluetooth
bMaxPacketSize064
idVendor 0x0a12 Cambridge Silicon Radio, Ltd
idProduct 0x0001 Bluetooth Dongle (HCI mode)
bcdDevice 31.64
iManufacturer 0
iProduct0
iSerial 0
bNumConfigurations 1
My /etc/bluetooth/hcid.conf is as follows -
# HCId options
options {
autoinit yes;
security auto;
pairing multi;
passkey 0202;
}
device {
name AstTest;
class 0x3e0100;
iscan enable; pscan enable;
discovto 0;
lm accept;
lp rswitch,hold,sniff,park;
}
options {
autoinit yes;
security auto;
pairing multi;
pin_helper /etc/bluetooth/pin;
}
My /etc/asterisk/mobile.conf is as follows -
[general]
interval=30 ; Number of seconds between trying to connect to
devices.
; The following is a list of adapters we use.
; id must be unique and address is the bdaddr of the adapter from
hciconfig.
; Each adapter may only have one device (headset or phone) connected at a
time.
; Add an [adapter] entry for each adapter you have.
[adapter]
id=blue
address=00:10:60:D0:EC:66
;forcemaster=yes; attempt to force adapter into master mode.
default is no.
;alignmentdetection=yes ; enable this if you sometimes get 'white noise' on
asterisk side of the call
; its a bug in the bluetooth adapter firmware,
enabling this will compensate for it.
; default is no.
[PersX1]
address=00:23:45:32:78:57 ; the address of the phone
port=2 ; the rfcomm port number (from mobile
search)
context=incoming-mobile ; dialplan context for incoming calls
adapter=blue; adapter to use
group=1 ; this phone is in channel group 1
;nocallsetup=yes; set this only if your phone reports that
it supports call progress notification, but does not do it. Motorola L6 for
example.
Help!
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Re: [asterisk-users] *Sort of Commercial* TracFone's $45 unlimited offer to 'stun' rivals
Chan_mobile supports SMS with a limited number of phones On Fri, Jul 3, 2009 at 4:14 PM, Steve Totaro stot...@totarotechnologies.com wrote: Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data. http://www.fiercewireless.com/story/leap-metropcs-come-under-pressure-tracfone/2009-07-02 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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] NOT chan_mobile
Same here. On Fri, Jun 26, 2009 at 2:40 PM, Razza razz...@gmail.com wrote: Hi all, does anyone know of an application that will run in Windows (in my case users PC's) and behave in a similar fasion to chan_mobile? I'd like the app to register with asterisk, then talk to a (or a number of) mobiles over bluetooth thus creating an FXO port? I'm not interested in SMS etc. just voice. Thanks in advance Ray. ___ -- 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] Working chan_mobile/bluez anyone?
Hi all, Before I start with analog GSM gateways I wanted to check if maybe someone actually got a working combination of chan_mobile and bluez. If you do please share specifics like versions, phone, BT chipset, any other relevant info. Thanks, Sasa Bobek ___ -- 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 Auth
Hi, also with your template I have always the same problem ! Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, June 22, 2009 2:41 PM Subject: Re: [asterisk-users] Cisco 7941G Auth Hey Sasa, I have templates of all the files you need here (SEP file, extension file): http://dave.vc/wordpress/wp-content/uploads/2008/11/phoneadd.zip If you need further assistance, let me know. Thanks Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sasa Sent: Monday, June 22, 2009 4:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Jonathan Thurman wrote: What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device Hi, I have already writed also in SEPMacAddress.cnf.xml file (other at XMLDefault.cnf.xml file) the parameter: loadInformationSIP41.8-0-2SR1S/loadInformation ..but the problem isn't resolved !. Can I try to change some parameters ?..are desperate ! I think I have tried everything ! Thanks. -- Salvatore. - Original Message - From: Jonathan Thurman jthurma...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 6:04 PM Subject: Re: [asterisk-users] Cisco 7941G Auth What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file
[asterisk-users] GSM mobile trunks
Hi all, We have been planing for a long time to set up GSM mobile trunks for termination, and were planing on going with analog GSM adapters connected to a VoIP gateway. Should we be concerned with such a set-up as far as voice quality and other issues are concerned? Any experiences with GSM terminal chipsets? 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
Re: [asterisk-users] GSM mobile trunks
The price difference is HUGE. Analog i about 66% cheaper. On Tue, Jun 23, 2009 at 12:44 PM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Tue, 23 Jun 2009, Sasa Bobek wrote: Hi all, We have been planing for a long time to set up GSM mobile trunks for termination, and were planing on going with analog GSM adapters connected to a VoIP gateway. Should we be concerned with such a set-up as far as voice quality and other issues are concerned? Any experiences with GSM terminal chipsets? Why not SIP based GSM devices? e.g. Portech? Gordon ___ -- 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] GSM mobile trunks
Yup, even costlier than Portech. And we also tried chan_mobile as a low cost alternative, but that seems to be very buggy. On Tue, Jun 23, 2009 at 1:05 PM, Duncan Turnbull dun...@e-simple.co.nzwrote: Yip the VoiceBlue SIP units are very good but a bit pricey Gordon Henderson wrote: On Tue, 23 Jun 2009, Sasa Bobek wrote: Hi all, We have been planing for a long time to set up GSM mobile trunks for termination, and were planing on going with analog GSM adapters connected to a VoIP gateway. Should we be concerned with such a set-up as far as voice quality and other issues are concerned? Any experiences with GSM terminal chipsets? Why not SIP based GSM devices? e.g. Portech? Gordon ___ -- 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] GSM mobile trunks
Thanks for the info Gordon. Just what I was looking for. I think I have seen one of the telecom FM units, it actually has a whole phone inside :) In my end of the world things are quite different :) Portech costs an average of 160E per port, and the cost of the GSM adapter including the cost of the FXS/FXO port is about 50E. Sasa On Tue, Jun 23, 2009 at 1:17 PM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: On Tue, 23 Jun 2009, Sasa Bobek wrote: The price difference is HUGE. Analog i about 66% cheaper. But you then need some sort of analogue adapter/interface to feed the analogue GSM module... Although if you've already got this, it's a obviously a cheaper option. However, here in the UK, the price difference isn't that bad, but it might be that the number of ports you'rea fter makes a difference - e.g. a 2-port SIP Portech units is £321, a single (analogue) port Telecom FM unit is £119. (so actually slightly cheaper for 2 ports with SIP Ethernet Interfaces here), but if you're looking at a dozen channels it might well be different.. However - to your original question - I've used both the Portech and Telecom FM units (on a TDM 400 card) and not really been able to tell the difference. The Portech dials quicker, the Telecom FM obviously needs to get the number passed via DTMF, but since GSM is ... GSM which is pretty poor speech quality to start with, it didn't make any difference I could tell. Gordon On Tue, Jun 23, 2009 at 12:44 PM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net gordon%2baster...@drogon.net gordon%252baster...@drogon.net wrote: On Tue, 23 Jun 2009, Sasa Bobek wrote: Hi all, We have been planing for a long time to set up GSM mobile trunks for termination, and were planing on going with analog GSM adapters connected to a VoIP gateway. Should we be concerned with such a set-up as far as voice quality and other issues are concerned? Any experiences with GSM terminal chipsets? Why not SIP based GSM devices? e.g. Portech? Gordon ___ -- 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 Auth
Jonathan Thurman wrote: What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device Hi, I have already writed also in SEPMacAddress.cnf.xml file (other at XMLDefault.cnf.xml file) the parameter: loadInformationSIP41.8-0-2SR1S/loadInformation ..but the problem isn't resolved !. Can I try to change some parameters ?..are desperate ! I think I have tried everything ! Thanks. -- Salvatore. - Original Message - From: Jonathan Thurman jthurma...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 6:04 PM Subject: Re: [asterisk-users] Cisco 7941G Auth What does your SEPMacAddress.cnf.xml file look like? In my experience, the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had to specify the firmware version in each SEP file. I am using 8-4-4S, but for you this would be something like this: device loadInformationSIP41.8-0-2SR1S/loadInformation /device And you shouldn't need the tlv file. -Jonathan On Fri, Jun 19, 2009 at 8:25 AM, Sasa s...@shoponweb.it wrote: David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had
[asterisk-users] Cisco 7941G Auth
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. ___ -- 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 Auth
John Novack wrote: I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary ...with CTLSEPmac_address.tlv in tftp dir in log file I have: Using local port 3131 [19/06 17:14:02.816] CTLSEPmac_address.tlv: sent 1 blk, 0 bytes in 0 s. 0 blk resent [19/06 17:14:02.863] Connection received from 192.168.1.61 on port 49188 [19/06 17:14:06.988] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 17:14:06.988] ..and the problem isn't resolved. Thanks. -- Salvatore. - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:38 PM Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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 Auth
David Gibbons wrote: I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. I have tried with tftp on linux machine but the result isn't changed. Thanks. -- Salvatore. - Original Message - From: David Gibbons d...@videon-central.com To: novacks...@gmail.com; 'Asterisk Users MailingList - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, June 19, 2009 4:50 PM Subject: Re: [asterisk-users] Cisco 7941G Auth I've found that different types of TFTP servers return differing errors when a file doesn't exist. You don't need the TLV file, but you do need a distro that tells the phone it's not there correctly. I have not had ANY luck with windows tftp servers, only linux. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack Sent: Friday, June 19, 2009 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7941G Auth Sasa wrote: Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco 7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem is that Cisco phone isn't authenticated on Asterisk. In tftp directory I have: apps41.1-1-1-15.sbn cnu41.3-1-1-15.sbn copstart.sh cvm41sip.8-0-1-18.sbn dialplan.xml dsp41.1-1-1-15.sbn jar41sip.8-0-1-18.sbn load115 load308 load309 load30018 SIP41.8-0-2SR1S.loads term41.default.loads term61.default.loads XMLDefault.cnf SEPmac_address.cnf.xml ..and in tftp log I have: Connection received from 192.168.1.61 on port 49153 [19/06 10:16:35.968] Read request for file CTLSEPmac_address.tlv. Mode octet [19/06 10:16:35.968] File CTLSEPmac_address.tlv : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:35.968] Connection received from 192.168.1.61 on port 49154 [19/06 10:16:36.109] Read request for file SEPmac_address.cnf.xml. Mode octet [19/06 10:16:36.109] Using local port 3995 [19/06 10:16:36.109] SEPmac_address.cnf.xml: sent 15 blks, 7239 bytes in 0 s. 0 blk resent [19/06 10:16:36.171] Connection received from 192.168.1.61 on port 49155 [19/06 10:16:40.046] Read request for file /mk-sip.jar. Mode octet [19/06 10:16:40.046] File \mk-sip.jar : error 2 in system call CreateFile Impossibile trovare il file specificato. [19/06 10:16:40.046] Connection received from 192.168.1.61 on port 49156 [19/06 10:16:40.984] Read request for file Italy/g3-tones.xml. Mode octet [19/06 10:16:40.999] File Italy\g3-tones.xml : error 3 in system call CreateFile Impossibile trovare il percorso specificato. [19/06 10:16:40.999] Connection received from 192.168.1.61 on port 49164 [19/06 10:16:42.843] Read request for file dialplan.xml. Mode octet [19/06 10:16:42.859] Using local port 3998 [19/06 10:16:42.859] dialplan.xml: sent 1 blk, 104 bytes in 0 s. 0 blk resent [19/06 10:16:42.906] In XMLDefault.cnf I have: loadInformation309 SIP41.8-0-2SR1S/loadInformation309 ..and on 7941G I have: App Load IDjar41sip.8-0-1-18.sbn Boot Load ID7941G_64-02070631Amd64megRel.bin VersionSIP41.8-0-2SR1S Thanks. -- Salvatore. I have had sucess with creating a zero length file named CTLSEPmac_address.tlv Or whatever the damn thing wants, and it then seems to be happy. With Cisco 7960's Your results may vary John Novack ___ -- 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 -- Dog is my co-pilot ___ -- 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] Portech MV3770 Caller-ID
Hi, I have modified in Mobile/Setting the parameter SIP From from tel/user to tel/tel and now I view the correct incoming number. Thanks. -- Salvatore. - Original Message - From: Christian Victor christ...@victormedia.de To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, March 09, 2009 6:51 PM Subject: Re: [asterisk-users] Portech MV3770 Caller-ID 2009/3/9 Sasa s...@shoponweb.it Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is ... ..now what parameter can I modify for to view the external phone number ? Thank in advance. Hi Salvatore! Can you verify if the number is submitted to Asterisk? If not maybe you need to change the way the number is transmitted from the gateway to the Asterisk box. I can't remember the exact parameter but it is on the Mobile-Settings page. There should be four choices in a drop-down list - various combination if SIP-ID/number/incoming number. Chris ___ -- 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] Portech MV3770 Caller-ID
Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is configured on MV-370. The MV-370 configuration is: Mobile to Lan Table : 0 * 192.168.1.1 Lan to Mobile Table: 0 * # SIP Setting: Display Name: Portech User Name: 1005 Register Name: 1005 Register Password: passwd Domain Server: 192.168.1.1 Proxy Server: 192.168.1.1 Outbound Proxy: Satus: Registered In Asterisk I have created a SIP Trunk with this configuration: TRUNK Name: gsm in PEERS Details: context=from-pstn host=192.168.1.2 type=friend USER Context: 1005 in USERS Details: canreinvite=no context=from-trunk host=192.168.1.2 nat=yes qualify=yes secret=passwd type=friend username=1005 ..and I have created the extension '1005' with this configuration: [1005] deny=0.0.0.0/0.0.0.0 type=friend secret=passwd qualify=yes port=5060 pickupgroup= permit=0.0.0.0/0.0.0.0 nat=yes mailbox=1...@device host=dynamic dtmfmode=rfc2833 dial=SIP/1005 context=from-pstn canreinvite=no callgroup= callerid=device 1005 accountcode= call-limit=50 ..now what parameter can I modify for to view the external phone number ? Thank in advance. - Salvatore. ___ -- 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] Problem with Portech
Hi, I use Asterisk-1.2.26 (with Trixbox-2.1.12) and Portech MV-370 and my problem is that when I try to call an external mobile phone via Portech I have alway busy and in log file: Called Portech/348xxx -- Got SIP response 486 Busy Here back from 192.168.1.2-- SIP/Portech-086e5ee0 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing Goto(SIP/200-08701488, s-BUSY|1) in new stack. My configuration is: [1005] type=friend secret=1005 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=no [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 dial=SIP/1005 context=from-internal canreinvite=no callerid=device 1005 ..and sip trunk with in PEER details: type=peer host=192.168.1.2 context=from-trunk ..and in USER details: canreinvite=no context=from-trunk host=dynamic nat=yes qualify=yes secret=1005 type=friend username=1005 On Portech in LAN To Mobile Table I have: Item -- 0 URL -- * Call Num: # and in Service Domain Settings extension 1005 is registred and in Mobile Status the SIM instllation is ok. Thanks in advance. - Salvatore. ___ -- 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 7906g SIP
Hi Duncan, when my Cisco phone is started I don't view nothing in my tftp logs, in other words when cisco phone startup it don't call my tftp server for to try search configuration files. Regards. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 17, 2008 11:49 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore Have you checked the tftp logs in any event? Its important to check the tftp logs and see if anything is being requested. I have had this before but usually its still trying to grab its first couple of files, and from that you can get an idea of where its getting stuck. If it says upgrading it means its trying to change from one version to another and failing, so you need to go backwards to a version it can cope with. If its not asking for any files then usually what I have done is to go to the lowest SIP version 2 or 3 for changing from the call manager to SIP and reset the phone to factory defaults and try and get it to start the change again Cheers Duncan Sasa wrote: Hi Duncan, yes I have a tftp server (I use also Cisco 7941G that use tftp server for upload configuration) and I know this function, but now my problem is that the phone is stopped on the initial screen that show 'upgrading' and MAC address and the process not continued. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 8:52 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore Do you have a TFTP server that serves the phone configuration files? This is very separate to the phone, i.e. on a server/pc somewhere, and will log all the file requests it receives. You can check this irrespective of the phone Have you checked whether tftp requests are being made, usually they come before the system goes into the upgrading state. I have had that before and it was caused by having different load files from that specified in the OS79XX.TXT file which for my phones usually have P003-08-6-00 but for upgrading I start from P0S30202 For SIPDefault.cnf you also need the image version to match #Image Version image_version:P0S3-08-6-00 ; But for conversion I first go to this image image_version:P0S30202 ; And I go from that to this image_version:P0S3-06-2-00 ; then to the current version And I have these files on my tftpserver which are the respective firmwares -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin I can't recall if I need all the 08-6 versions Cheers Duncan Sasa wrote: Hi Duncan, I have tried more times to make the reset phone but is displays always and only 'upgrading' and MAC address and I cann't access the phone configuration. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 11:41 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED
Re: [asterisk-users] Cisco 7906g SIP
Hi Duncan, yes I have a tftp server (I use also Cisco 7941G that use tftp server for upload configuration) and I know this function, but now my problem is that the phone is stopped on the initial screen that show 'upgrading' and MAC address and the process not continued. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 8:52 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore Do you have a TFTP server that serves the phone configuration files? This is very separate to the phone, i.e. on a server/pc somewhere, and will log all the file requests it receives. You can check this irrespective of the phone Have you checked whether tftp requests are being made, usually they come before the system goes into the upgrading state. I have had that before and it was caused by having different load files from that specified in the OS79XX.TXT file which for my phones usually have P003-08-6-00 but for upgrading I start from P0S30202 For SIPDefault.cnf you also need the image version to match #Image Version image_version:P0S3-08-6-00 ; But for conversion I first go to this image image_version:P0S30202 ; And I go from that to this image_version:P0S3-06-2-00 ; then to the current version And I have these files on my tftpserver which are the respective firmwares -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin I can't recall if I need all the 08-6 versions Cheers Duncan Sasa wrote: Hi Duncan, I have tried more times to make the reset phone but is displays always and only 'upgrading' and MAC address and I cann't access the phone configuration. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 11:41 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] Cisco 7906g SIP
Hi Duncan, I have tried more times to make the reset phone but is displays always and only 'upgrading' and MAC address and I cann't access the phone configuration. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 11:41 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- 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 7906g SIP
Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL
Re: [asterisk-users] Cisco 7906g SIP
Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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 ___ -- 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
Re: [asterisk-users] Cisco 7906g SIP
I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 3:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP When the 'upgrading' process fails, it means that one or more of the required files is missing from the TFTP root folder. Check the logs to see which file it fails on, get that file and you should be good to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, I have try again with your method but after that the phone reboot I have on the screen phone displayed 'upgrading' with MAC address but the reset process is stopped ! Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 4:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Please send the TFTP log after using the regular factory reset method I described. Thanks Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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
Re: [asterisk-users] Cisco 7906g SIP
Hi, if possible use 7906G without callmanager software but only with SIP protocol support ? Thanks. -- Salvatore. - Original Message - From: Stefan Gofferje [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 1:27 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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 7906g SIP
Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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 7906g SIP
Hi Dave, I have tried restore to factory default value (as you have recommended to me) but without success, however also with only files: SEPMAC.conf file contents of the cop file ..but the result isn't changed ! Thanks in advance. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:59 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, Sometimes I have to do a hard reset of the phone in order to get it to load the SIP firmware, even when the load file is specified in the SEPMAC.conf file. Make sure that only the contents of the cop file and the SEPmac.cnf file are present in your tftp root. Then unplug the phone and press and hole the # key. Plug the phone back in, still holding the # key. When the line buttons begin turn on and off in sequence, press 123456789*0#. This will factory reset the phone and should cause it to check the termxx.default.loads file for the proper image. It will then read the SIP image name from that file and flash itself with the SIP image. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Thursday, October 09, 2008 8:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Dave, the cmterm-7911_7906-sip.8-0-4sr1.cop is a compressed file and on the inside has: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads I use Cisco7941 without callmanager software but only with SIP support. Thanks. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 09, 2008 2:30 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa, You can actually just rename the .cop file to a .tar.gz file. Cisco doesn't have (to my knowledge) any non-callmanager SIP software. The SIP load is just a SIP load, not a SIP load unique to generic SIP or callmanager. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Gofferje Sent: Thursday, October 09, 2008 7:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP Sasa schrieb: I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? cmterm is the callmanager software. You need to get the non-callmanager SIP-software. Contact your local Cisco representative to buy a license for that. Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- 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 ___ -- 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 7906g SIP
Hi, sorry for my insistence but for me is a big problem ! :-( ...someone have the same problem ? Thanks in advance. -- Salvatore. - Original Message - From: Sasa [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 07, 2008 2:53 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi, in tftp server I have the followings files: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads ..and on 7906g in status menu I have: load file: sccp11.8-3-2s app load id: jar11sccp.8-3-1-22.sbn jvm load id: cvm11sccp.8-3-1-22.sbn os load id: cnu11.8-3-1-22.sbn boot load id: tnp06.3-0-1-31.bin dsp load id: dsp11.8-3-1-22.sbn I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? Thanks in advance. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 07, 2008 1:04 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- 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 ___ -- 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 7906g SIP
Hi, in tftp server I have the followings files: apps11.1-1-3-15.sbn cnu11.3-1-3-15.sbn copstart.sh cvm11sip.8-0-3-16.sbn dsp11.1-1-3-15.sbn jar11sip.8-0-3-16.sbn load307 load369 SIP11.8-0-4SR1S.loads term06.default.loads term11.default.loads ..and on 7906g in status menu I have: load file: sccp11.8-3-2s app load id: jar11sccp.8-3-1-22.sbn jvm load id: cvm11sccp.8-3-1-22.sbn os load id: cnu11.8-3-1-22.sbn boot load id: tnp06.3-0-1-31.bin dsp load id: dsp11.8-3-1-22.sbn I need other files other than those obtained with cmterm-7911_7906-sip.8-0-4sr1.cop ?? Thanks in advance. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 07, 2008 1:04 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- 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
[asterisk-users] Cisco 7906g SIP
Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- 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
[asterisk-users] ISDN card freeze
Hi, I use Asterisk 1.2.17 with BRIstuffed-0.3.0-PRE-1y-e (with Trixbox 2.2.12) and I have three ISDN card with chipset HFC on PCI slot, my problem is that after a inactivity period one o two isdn card are disconnected: asterisk1*CLI zap show status HFC-S PCI A ISDN card 0 [TE] layer 1 AC HFC-S PCI A ISDN card 1 [TE] layer 1 AC HFC-S PCI A ISDN card 2 [TE] layer 1 D ..and when I have this condition with ISDN card 2 isn't possible to do a call from internal extension to external number phone but is possible to receive external call direct to phone number that's referred isdn card 2 ! In this condition if arrive to external direct to ISDN card 2 this card is again available up and I have: asterisk1*CLI zap show status HFC-S PCI A ISDN card 0 [TE] layer 1 AC HFC-S PCI A ISDN card 1 [TE] layer 1 AC HFC-S PCI A ISDN card 2 [TE] layer 1 AC Why the isdn card is freeze after an inactived period ? My zapata.conf is: [trunkgroups] [channels] language=it signalling=bri_cpe_ptmp rxwink=300 pridialplan=local prilocaldialplan=local switchtype=euroisdn pmp_l1_check=no nodialtone=no te_choose_channel=no usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 context=from-pstn channel=1-2 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming #include zapata-auto.conf group=1 context=from-pstn channel=4-5 #include zapata_additional.conf #include zapata-BRI-HFC.conf My zaptel.conf: # Span 1: ZTHFC1 HFC-S PCI A ISDN card 1 [TE] span=1,1,3,ccs,ami bchan=1-2 dchan=3 # Span 2: ZTHFC2 HFC-S PCI A ISDN card 2 [TE] span=2,1,3,ccs,ami bchan=4-5 dchan=6 # Span 3: ZTHFC3 HFC-S PCI A ISDN card 3 [TE] span=3,1,3,ccs,ami bchan=7-8 dchan=9 My zapata-BRI-HFC.conf: resetinterval=never immediate=no switchtype=euroisdn signalling=bri_cpe_ptmp pridialplan=dynamic prilocaldialplan=local nationalprefix=0 internationalprefix=00 usecallingpres=yes echocancel=yes echocancelwhenbridged=yes echotraining=100 context=from-zaptel group=0 channel = 1-2 group=1 channel=4-5 group=2 channel=7-8 My asterisk.conf is: [directories] astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astagidir = /var/lib/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk [options] transmit_silence_during_record = yes Thanks. -- Salvatore. ___ -- 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] busy/congestion random
Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn pmp_l1_check=no nodialtone=no usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 context=from-pstn channel=1-2 channel=4-5 callgroup=1 pickupgroup=1 immediate=no faxdetect=incoming #include zapata-auto.conf group=1 context=from-pstn channel=1-2 channel=4-5 #include zapata_additional.conf #include zapata-BRI-HFC.conf ..the log is: Executing Macro(SIP/206-090a7dd8, dialout-trunk|1|348241||) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK=1) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_NUMBER=348241) in new stack -- Executing Set(SIP/206-090a7dd8, ROUTE_PASSWD=) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?noauth) in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing GotoIf(SIP/206-090a7dd8, 0?disabletrunk|1) in new stack -- Executing Set(SIP/206-090a7dd8, _NODEST=) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK_OPTIONS=tT) in new stack -- Executing Set(SIP/206-090a7dd8, GROUP()=OUT_1) in new stack -- Executing Macro(SIP/206-090a7dd8, user-callerid|SKIPTTL) in new stack -- Executing NoOp(SIP/206-090a7dd8, user-callerid: device 206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSER=206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?report) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?start) in new stack -- Executing Set(SIP/206-090a7dd8, REALCALLERIDNUM=206) in new stack -- Executing NoOp(SIP/206-090a7dd8, REALCALLERIDNUM is 206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSER=206) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSERCIDNAME=Centralino) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?report) in new stack -- Executing Set(SIP/206-090a7dd8, AMPUSERCID=206) in new stack -- Executing Set(SIP/206-090a7dd8, CALLERID(all)=Centralino 206) in new stack -- Executing Set(SIP/206-090a7dd8, REALCALLERIDNUM=206) in new stack -- Executing NoOp(SIP/206-090a7dd8, TTL: ARG1: SKIPTTL) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?continue) in new stack -- Goto (macro-user-callerid,s,23) -- Executing NoOp(SIP/206-090a7dd8, Using CallerID Centralino 206) in new stack -- Executing Macro(SIP/206-090a7dd8, record-enable|206|OUT) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/206-090a7dd8, recordingcheck|20080115-131850|asterisk-12308-1200399530.1395) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20080115-131850|asterisk-12308-1200399530.1395: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/206-090a7dd8, No recording needed) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 0?skipoutcid) in new stack -- Executing Set(SIP/206-090a7dd8, DIAL_TRUNK_OPTIONS=tT) in new stack -- Executing Macro(SIP/206-090a7dd8, outbound-callerid|1) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?start) in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp(SIP/206-090a7dd8, REALCALLERIDNUM is 206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set(SIP/206-090a7dd8, USEROUTCID=) in new stack -- Executing Set(SIP/206-090a7dd8, EMERGENCYCID=) in new stack -- Executing Set(SIP/206-090a7dd8, TRUNKOUTCID=) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf(SIP/206-090a7dd8, 1?usercid) in new stack -- Goto (macro-outbound-callerid,s,18) -- Executing GotoIf(SIP/206-090a7dd8, 1?report) in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp(SIP/206-090a7dd8, CallerID set to Centralino 206) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing AGI(SIP/206-090a7dd8, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set(SIP/206-090a7dd8, OUTNUM=348241) in new stack -- Executing Set(SIP/206-090a7dd8, custom=ZAP/g0) in new stack -- Executing GotoIf(SIP/206-090a7dd8, 1?gocall) in new stack -- Goto
Re: [asterisk-users] Fw: Remove a TDM Card
Tzafrir Cohen wrote: New: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1,2 dchan=3 span=2,1,3,ccs,ami bchan=4-6 dchan=6 ..in zapata.conf I have: ; new part: switchtype=euroisdn signalling = bri_net priindication=outofband group = 1 channel = 1-2 group = 2 channel = 4-5 ..therefore I must only modify zaptel.conf and zapata.conf ?..and I don't must unload modules ? But when PC started without TDM card isn't a problem that is loaded wctdm24xxp module (that is present in rc.modules and rc.modules-2.4.33.3) on boot ? Thanks. -- Salvatore. ___ --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] Fw: Remove a TDM Card
Tzafrir Cohen wrote: You have been quite short on details. For instance: what distribution of Linux? What version of Zaptel? Do you have another Zaptel card? It seems you either have two zaphfc cards or one dual-BRI card. If so, the procedure is slightly more complicated, as you basically have to reconfigure the system afterwards. As I mentioned, genzaptelconf can be handy for that. I don't know what Linux distribution is installed but the kernel version is 2.6.19.2, the zaptel version is zaptel-1.2.12 and is present one TDM Card and two zaphfc cards..with this architecture is correct my procedure for remove TDM card ? Thanks. -- Salvatore. ___ --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] Fw: Remove a TDM Card
Hi, my problem isn't on new voip box with latest asterisk version...my problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this steps for remove rightly TDM Card: - remove line configuration about tdm card in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv - rmmod wctdm24xxp - halt - remove physically card tdm from pc (box voip 1) - restart box voip 1 ..this procedure is ok ? Thanks ! -- Salvatore. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 29, 2007 1:50 AM Subject: Re: [asterisk-users] Fw: Remove a TDM Card On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote: Hi, sorry but perhaps I don't have explained clearly my problem...now I have a box voip that must be replace with another box voip but I want to do test before remove the old voip from production. With later versions of Zaptel you have zapconf and genzaptelconf . Use either of them to generate /etc/zaptel.conf and to generate a sample zapata.conf snippet in /etc/asterisk/zapata-channels.conf . -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Fw: Remove a TDM Card
Hi, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. -- Salvatore. - Original Message - From: Sasa [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card Hi, I would like remove a Digium TDM2400P from Asterisk (version 1.2.13) box but when I remove card from the PC after reboot Asterisk not started correctly. On box now with TDM Card I have: [EMAIL PROTECTED]:~# lsmod Module Size Used by zaphfc 167966 wctdm24xxp635525 zaptel 192132 26 zaphfc,wctdm24xxp The kernel version is: 2.6.19.2 ..then I have in /etc/rc.d/rc.modules: rc.modules rc.modules-2.4.33.3 ..both files are identical: #!/bin/sh /sbin/modprobe zaptel /sbin/modprobe zaphfc modes=3 /sbin/ztcfg -vv /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv ..in zapata.conf I have: signalling=fxs_ls group = 3 channel = 1-5 ..and in zaptel.conf I have: fxsls=1-24 ..now my dobious is about correct steps for remove TDM Card from PC, I think that: - remove line configuration in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv - rmmod wctdm24xxp - reboot ..this procedure is ok ? Thanks. -- Salvatore. ___ --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] Fw: Remove a TDM Card
Hi, sorry but perhaps I don't have explained clearly my problem...now I have a box voip that must be replace with another box voip but I want to do test before remove the old voip from production. The box voip (named 1) that now is in production have three card, two isdn card and TDM2400P that I want remove for to install in the new box voip (named 2). On the box voip 1 I have: Asterisk version 1.2.13 The kernel version is: 2.6.19.2 ..but on the box voip 2 I have the new asterisk version and kernel. On box voip 1 I have: zaptel.conf: loadzone=it defaultzone=it span=2,1,3,ccs,ami bchan=25-26 dchan=27 span=3,1,3,ccs,ami bchan=28-29 dchan=30 fxsls=1-24 ..in zapata.conf I have: [channels] language=it ... ... ;Linee ISDN immediate=no switchtype=euroisdn signalling = bri_net priindication=outofband group = 1 channel = 25-26 group = 2 channel = 28-29 ;Linee tdm immediate=yes .. .. cidstart=ring signalling=fxs_ls group = 3 channel = 1-5 ..and always on box voip 1: [EMAIL PROTECTED]:~# lsmod Module Size Used by zaphfc 167966 wctdm24xxp635525 zaptel 192132 26 zaphfc,wctdm24xxp ..in /etc/rc.d/rc.modules: rc.modules rc.modules-2.4.33.3 ..my problem is how remove TDM Card from voip box 1 without stop this box voip, if I remove in correctly mode the TDM Card I can configure this card on new voip box 2 and to do test before of put new box voip in production in replacement the box voip 1. The stesp that I think correctly for remove TDM Card: - remove line configuration about tdm card in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv - rmmod wctdm24xxp - halt - remove physically card tdm from pc (box voip 1) - restart box voip 1 ..this procedure is ok ? Thanks ! -- Salvatore. - Original Message - From: Jon Pounder [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, November 28, 2007 3:53 PM Subject: Re: [asterisk-users] Fw: Remove a TDM Card Quoting Tony Plack [EMAIL PROTECTED]: Hi, sorry for my insistence but this is a big problem for me..my steps for remove card are ok ? Thanks. its not much help but I have found in general asterisk is not too graceful about zap numbering and even starting when the cards in place don't match the configuration. yeah the config is wrong but sometimes there are legitimate reasons for that like taking a card out for 5min to try something you should be able to still boot the remaining system without it, or add the configs before putting the card in. Bigger issue is getting around the renumbering of channels when you remove hardware at the bottom -- Salvatore. - Original Message - From: Sasa [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, November 26, 2007 4:25 PM Subject: [asterisk-users] Remove a TDM Card Hi, I would like remove a Digium TDM2400P from Asterisk (version 1.2.13) box but when I remove card from the PC after reboot Asterisk not started correctly. On box now with TDM Card I have: [EMAIL PROTECTED]:~# lsmod Module Size Used by zaphfc 167966 wctdm24xxp63552 5 zaptel 192132 26 zaphfc,wctdm24xxp The kernel version is: 2.6.19.2 ..then I have in /etc/rc.d/rc.modules: rc.modules rc.modules-2.4.33.3 ..both files are identical: #!/bin/sh /sbin/modprobe zaptel /sbin/modprobe zaphfc modes=3 /sbin/ztcfg -vv /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv ..in zapata.conf I have: signalling=fxs_ls group = 3 channel = 1-5 ..and in zaptel.conf I have: fxsls=1-24 ..now my dobious is about correct steps for remove TDM Card from PC, I think that: - remove line configuration in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv - rmmod wctdm24xxp - reboot ..this procedure is ok ? Thanks. -- Salvatore. 1.2.13 has some exploits and you should consider running something newer. 1.2 branch has no official support. However... You should make sure to load ztdummy in place of your tdm card. If I remember correctly 1.2 requires a timing source. modules-2.4.33.3 is for an old kernel and shouldn't apply to the current 2.6.19.2. Normally the TDM card provides this 1000Hz timing source when available, but ztdummy can mimic this timing from the kernel clock source. ztdummy may (or may not) have issues with your kernel version as a timing source. I believe kernel 2.6.21 or better solved a few problems with clock sources. Other than that, your steps are fine. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com
[asterisk-users] Remove a TDM Card
Hi, I would like remove a Digium TDM2400P from Asterisk (version 1.2.13) box but when I remove card from the PC after reboot Asterisk not started correctly. On box now with TDM Card I have: [EMAIL PROTECTED]:~# lsmod Module Size Used by zaphfc 167966 wctdm24xxp635525 zaptel 192132 26 zaphfc,wctdm24xxp The kernel version is: 2.6.19.2 ..then I have in /etc/rc.d/rc.modules: rc.modules rc.modules-2.4.33.3 ..both files are identical: #!/bin/sh /sbin/modprobe zaptel /sbin/modprobe zaphfc modes=3 /sbin/ztcfg -vv /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv ..in zapata.conf I have: signalling=fxs_ls group = 3 channel = 1-5 ..and in zaptel.conf I have: fxsls=1-24 ..now my dobious is about correct steps for remove TDM Card from PC, I think that: - remove line configuration in zapata.conf and zaptel.conf - remove in rc.modules and rc.modules-2.4.33.3 line: /sbin/modprobe wctdm24xxp /sbin/ztcfg -vv - rmmod wctdm24xxp - reboot ..this procedure is ok ? Thanks. -- Salvatore. ___ --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 7940G licensing with asterisk
Hi, also I have called Cisco suport to ask how to use SIP protocol on Cisco 7941G (and my Astersik), the their answer is the following: ..SIP Firmware for the 7941G phone only works with Call Manager 5.x. You must have CCM 5.x to use this firmware, is needeful to buy a CCM license for use SIP protocol Asterisk. -- Salvatore. - Original Message - From: Glenn Cobb [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, October 01, 2007 4:21 PM Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk In trying to verify licensing requirements I called Tech-Data and spoke to the Cisco licensing reps there (my company is set up as a reseller through Tech-Data) and was informed by them that a license for Cisco VoIP phones is only required if connecting it to a Call Manager or any other Cisco voice technology solution such as a Cisco router. If you are connecting a Cisco phone to any other pbx they consider it a third party solution and licensing requirements for that vendor are your responsibility. Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Thursday, September 27, 2007 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186 a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 7940G licensing with asterisk
Hi, on 7941G is needful the Call Manager license, the firmware for SIP use is available (with login) on 7912 and 7940. Thanks. -- Salvatore. - Original Message - From: Patrick [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 28, 2007 4:33 AM Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk On Thu, 2007-09-27 at 14:58 -0500, Erick Perez wrote: Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can handle the 7940G ? The 7941G does conform to the standard but it only support SCCP (shame on cisco). The 7941 7961 also support SIP if you load the appropriate firmware from the Cisco website (login required). Regards, Patrick ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 7940G licensing with asterisk
Hi, sorry for my intrusion... I have the same problem with Cisco 7941G, can I do buy the the Smartnet registration also for 7941G or this license is available only for 7940G ? Thanks. -- Salvatore. - Original Message - From: Cory Andrews [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 27, 2007 6:48 PM Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk You need to purchase a Smartnet license for your phone, and have it registered by a Cisco authorized reseller. The Smartnet registration will run you $10-$20 per phone, depending upon the reseller. The registration process typically takes around 24-48 hours to process. Once registered, you will receive an email from Cisco with instruction on obtaining a Cisco TAC login. Once you have your login, you will be able to access and download the SIP firmware. If you look around on Google or on the Cisco website, there is a lot of documentation out there that describes the process for migrating the firmware. I agree, it is a lot of work. I do not see Cisco shipping phones with SIP firmware on them anytime soon, as obviously their vested interest is in their CCM and CCME platforms, and their native Skinny protocol. They are being dragged reluctantly into SIP and platforms such as Asterisk present a threatthey are not going to tailor their tools and channel practices toward folks using a non Cisco platform. Cory J Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of yonoko molomo Sent: Thursday, September 27, 2007 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk Hi, i bought this device and the cost of the 7040G itself was similar to the license. if im not wrong, the telephone cost around 80€. the sip license was around 80€ as well however, i am quite annoyed because the phone did not come with sip, but callmanager so i cant use it as i planned. i have read somewhere that I need to change the firmware, but i require a cisco account to download the firmware (but nobody provided me this account). we paid for the SIP license, but we did not get a SIP-capable device, and we do not have the way to download the firmware (yet). Regarding the power adapter, I had to buy them sepparately. since i do not have POE devices i cant answer your last question. 2007/9/27, Erick Perez [EMAIL PROTECTED]: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, -- Erick Perez ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] username/auth name mismatch
Hi, I have a asterisk/voip newbie and I am sorry if my quetion is banal. I used in my private LAN, Express Talk on Windows XP and Asterisk latest version on Fedora Core 4 , with this configuration in Express Talk Lines menu: Setting for Line: Default Line Settings Full 'friendly' Display Name: port SIP Numeber: 200 Server: 10.0.0.112 Password: mypassword In menu Network: Local SIP Port to Listen on: 5070 Local RTP ports: 8000 My sip.conf: [200] type=friend callerid=port username=200 secret=mypassword host=dinamic context=internal My extensions.conf: [internal] exten = 200,1,Dial(SIP/200,20) ..but in Asterixk log file I have: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.0.0.230 - Username/auth name mismatch and on Express Talk I have: Register attempt for sip:[EMAIL PROTECTED] failed 404 Not found ..where: 10.0.0.112 - asterisk ip address 10.0.0.230 -- express talk ip address ..where is my error ? thanks. Salvatore. ___ --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
