Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN-PSTN calls get this error...
-Greg
---
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN-PSTN calls get this error...
-Greg
---
On Mon, 2008-03-31 at 23:07 +0100, Razza wrote:
On 31/03/2008, J. Oquendo [EMAIL PROTECTED] wrote:
YMMV Change to reflect your firmware (e.g. P003-07-4-xx)
8 SNIP 8
I removed the following lines:
loadInformation8 model=IP Phone
7940P003-07-4-00/loadInformation8
On Mon, 2008-03-24 at 19:52 +0100, Philipp Kempgen wrote:
mark morreny schrieb:
I am working on deploying voip for my company and would like to seek some
advice on the number of E1 lines we need to rent.
E1 is not VoIP. :-)
It is if provisioned for 30 channels of concattenated data :)
On Feb 7, 2008, at 2:07 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, Feb 07, 2008 at 07:53:12PM +, Ben Willcox wrote:
Olivier wrote:
At the opposite, I think it could be useful for an Asterisk server
to
act as XMPP User Activity provider (ie update XEP-0108 field with
On Feb 2, 2008, at 2:11 PM, John Von Essen [EMAIL PROTECTED] wrote:
I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.
It turns out to be a phone/stun/linksys issue - not an asterisk issue.
Which brings up a couple of questions
I
get back home, I will login to the asterisk
servers and tell you what IPs the registration requests have in them.
From : Greg Oliver [EMAIL PROTECTED]
To : Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
You need $dnis.
On Jan 30, 2008, at 11:08 PM, Prashant Sharma
[EMAIL PROTECTED] wrote:
Hi,
I am new to asterisk configuration.
I want to get called number in features.conf.
I am defining a feature in features.conf and that feature got
executed on pressing a particular DTMF key sequence.
Cisco routers with DSPs as ip2ip gw will do it if you want to spend a
few bucks
On Jan 29, 2008, at 2:36 PM, Khaled Chehab [EMAIL PROTECTED]
wrote:
Dears
Any one knows a standalone voip transcoder software name,not an ip
pbx.
What I want is to transcode the incoming sip calls
On Mon, 2007-12-10 at 17:58 -0800, Robert McNaught wrote:
Hi
Does anyone have any recommendations of an SMS gateway which you can
just sign up for on a pay-as-you-go basis for testing, for use with
Asterisk?
Thanks
Robert McNaught
In and Out Bound SMS from *, or just * - SMS? If
On Thu, 2007-12-06 at 10:32 -0500, John Bittner wrote:
The fix for this is not to use the normal Cisco IOS. Must use 12.4T
version. It is a Cisco bug.
I would suggest jumping to greater than 12.4.11T as they introduced all
kinds of DTMF fixes there as well..
-Original Message-
On
On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote:
2007/11/14, Greg Oliver [EMAIL PROTECTED]:
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files
for the
Cisco
On Thu, 2007-11-15 at 05:34 +0100, Patrick wrote:
On Wed, 2007-11-14 at 09:06 -0500, Anciso, Roy wrote:
The Cisco Documentation states that you can modify standard and
nonstandard softkey templates. They may not be xml files. I just
assumed they were xml since that is what is used to
On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote:
Hello List,
Does anyone have access to the soft key configuration files for the
Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and
didn’t find much up there.
Thanks
Softkeys running both SCCP and SIP firmware are
On Thu, 2007-06-28 at 14:52 +0200, Olivier wrote:
2007/6/27, Greg Oliver [EMAIL PROTECTED]:
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
Hi,
Has anyone met any success, installing localized (ie
non-english)
menus within SIP firmware
On Tue, 2007-06-26 at 21:45 +0200, Olivier wrote:
Hi,
Has anyone met any success, installing localized (ie non-english)
menus within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco
Call Manager but as they are managed by an Asterisk
On Wed, 2007-06-27 at 14:32 -0600, Stephen Bosch wrote:
Hi, folks:
Snip
Thoughts? Who here has used BRI in North America? And when you did, what
interface hardware did you use?
-Stephen-
I grew up on BRI when the internet first started taking off here. All
terminated into Ascend
On Fri, 2007-06-01 at 10:18 +0530, Vamsi Pottangi wrote:
Hi Greg,
Narrowed the problem ot be that of codec mismatch ;-) Damn
CCM, doesn't provide proper debugs.
I have another query with CCM and Asterisk integration. In CCM cluster
Phones register to 1st CCM and they fallback to 2nd
On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
we are using 7941 with sip v8.2(2)SR3, it working quite well ;-)
Eric Lubow wrote:
All,
I am having a lot of trouble with the Cisco 7961G phones. I have
managed to get them up and running with Asterisk to the point where I
On Wed, 2007-05-30 at 18:03 -0500, Eric ManxPower Wieling wrote:
David Boyd wrote:
Does that mean that even when dynamic dns entries exist and the time
to
live is set to 15 minutes asterisk will continue to try using the
old
expired results?
I can also say that my experience in putting
On Wed, 2007-05-23 at 19:53 +0530, Vamsi Pottangi wrote:
Hi,
I was able to work out SIP trunk between Asterisk and CCM 4.x without
any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk.
Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not
replying. For the
On Wed, 2007-05-16 at 13:57 -0500, Bruce Reeves wrote:
How sad, cnet misspelled Polycom and Cisco didn't make the cut.
Yeah, Cisco and MSoft are on BAD terms since the inking of the deal with
Nortel.. MSoft got mad when they moved from Windows Server to Linux for
their CallManager platform, and
On Wed, 2007-05-16 at 23:19 +0100, Robert Lister wrote:
I was wondering if it is possible (in 1.2.x) to get the SIP response code
back after doing Dial().
Dial() seems to treat most call-setup problems as dialstatus CONGESTION, and
some are NOANSWER, but I want to know the SIP response
On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote:
I am actually getting DTMF over SIP when people call in to a clients system
that is running a2billing. They are using RFC2833.
If you are using a Cisco router anywhere in the loop, there is a known
bug that causes rfc2833 and inband
On Fri, 2007-05-11 at 18:44 -0400, Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't
On Fri, 2006-10-13 at 11:53 -0500, Tim Connolly wrote:
Does anyone know what triggers the 7970 to update its config? I
was able to get it to update to SIP, but the config I used initially
won't go away. I am making small changes to the SEPxxx.cnf.xml file and
rebooting the phone, the
On Fri, 2006-10-13 at 13:08 -0500, Jessee J Holmes wrote:
Actually, come to think of it, I don't know who will support it. Does
Asterisk support G.722? From what I know it doesn't, is it included in
the 1.4 beta? Will they support it? If Asterisk doesn't support it,
then the phone won't do HD
On Tue, 2006-10-10 at 15:16 -0700, Alyed Tzompa wrote:
What I want is to transfer some calls to a Cisco extension, so think I don't
need to do the upgrade to CM5.
I'm I right?
Alyed
Yes - you are right. On your CCM, go to a phone and check the CSS of
the device and the partition of the
When you do a factory reset on a 41/61/70/71, it actually deletes ALL of
the firmware except the bootloader from the phone. You would have to
have all of the 70s firmware files that come with them in order to boot
them. The term70.default.loads tells the phone what version of software
to tftp.
On Thu, 2006-09-28 at 07:54 -0500, Tom wrote:
At 05:39 AM 9/28/2006, you wrote:
Any pros / cons on getting one over the other ? I was wondering what
the main differences were.
New phones (7941) support 802.3af POE. Old phones only Cisco special
POE. New phones don't work with old SIP
4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that.
Anything over 4.0 supports SIP trunking.
-Greg
On Thu, 2006-09-28 at 19:32 +0200, Yusuf wrote:
Hi,
I recently had to hook up to Cisco Call Manager 4.1.3, and it only
supports H323. SO I used ooh323, and a strange thing
On Fri, 2006-09-29 at 20:26 -0700, Dan Austin wrote:
Greg wrote:
4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that.
Anything over 4.0 supports SIP trunking.
While it is true that CCM 4.0 and up supports SIP trunking, it is not
all rainbows an butterflies. The 4.X
, there is a 'Calling Party
Selection' box. Changing the values in that drop down does not have
any affect on the callerid.
Thanks.
On 5/23/06, Greg Oliver [EMAIL PROTECTED] wrote:
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:
Hey guys,
When
/index.php?page=Asterisk+cmd+CallingPres
On 5/23/06, Greg Oliver [EMAIL PROTECTED] wrote:
I am trying to set CIDNum to nothing, but my outgoing PRI controlled
by another PBX seems to fill in something when asterisk does not.. If
I set a number either in the sip channel for the phone, or from
is
currently acting as our IVR. Would that make any difference?
Thanks.
On 5/24/06, Greg Oliver [EMAIL PROTECTED] wrote:
On Wed, 2006-05-24 at 07:30 -0700, Gary Richardson wrote:
On the route pattern configuration page, there isn't a
'redirecting
number
On Tue, 2006-05-23 at 09:32 +0100, Mark Ackroyd wrote:
Here in the UK on pri, setting the callerid to 0, withholds it.
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not.. If I
set a number either in the
On Tue, 2006-05-23 at 06:27 -0400, Steve Totaro wrote:
Greg Oliver wrote:
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not.. If I
set a number either in the sip channel for the phone, or from
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:
Hey guys,
When a call comes in via the PSTN to our Call Manager 3.2 and is
forwarded (via unity and H323), the caller id is set to our Unity
Voicemail instead of the caller id from the PSTN. We're using the
oh323 channel in this
On Mon, 2006-05-22 at 12:16 -0400, Greg Boehnlein wrote:
Hello,
I was wondering if anyone out there is successfully running
Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two
weeks that has me scratching my head and muttering strange things in the
wee hours of the
I am trying to set CIDNum to nothing, but my outgoing PRI controlled by
another PBX seems to fill in something when asterisk does not.. If I
set a number either in the sip channel for the phone, or from
extensions.con, it is realized.. If I try to leave them blank, or even
Not Defined, the main
On Sun, 2006-05-21 at 14:28 +0200, Olivier Krief wrote:
Hi,
I can't upgrade an old 7960 from SCCP 3.0 to SIP 7.5. Could you
help ?
From
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080094584.shtml#sccptosip2,
I got the following:
1. Copy the desired
On Mon, 2006-05-15 at 17:40 -0300, Gustavo Souza Queiroz wrote:
Hello,
I´m have a CCM 3.3 and Asterisk in my LAN.
I need connect my Asterisk in my CCM 3.3.
You can a help me?
I hate to say it, but your best bet is to upgrade to CCm 4.0 and use
SIP.. It is a free cisco upgrade assuming
On Sun, 2006-04-30 at 11:53 +0800, Ronald Wiplinger wrote:
One of my user is praising Skype!!!
I cannot figure out anymore what I can improve!
This users sip show peers is jumping from 65 msec to 1800 all the time.
Of course his voice quality is like a morse code with dashes or dots of
On Mon, 2006-04-10 at 22:42 -0400, Andres wrote:
Steven wrote:
You heard wrong. We have multiple PRIs from XO and they DO NOT send
caller name. We have discussed the issue with them on several
ocassions. The sales people will say whatever they want, but the tech
people who actually work
OK - I know this is expected behavior, but I am stuck.
Transferring calls from the * IVR to another SIP PBX ringing multiple
extensions simultaneously with call-forwarding set on a phone obviously
goes directly to the forwarded # since that phone answers first.
I need a way to make it where if
On Thu, 2006-04-06 at 18:57 -0700, Jay Wilton wrote:
Hello people,
370 Watts maximum output / 9.6 Watts/phone = 38 phones
Does this logic hold water or change with line loss?
Thank you,
JJW
All I can say is that if you oversubscribe POE devices to a cisco
switch, they have the tendency
On Wed, 2006-04-05 at 17:54 -0400, Jimmy Smith wrote:
does one know how to program so i can have 2 lines on one sip account
on that phone ?
im runnign my own asterisk
do i need 2 local accounts ? one for each line ? that rebounds to same
SIP forp VOIP provider ?
Yes.
On Mon, 2006-04-03 at 13:59 -0500, Doug wrote:
At 22:16 3/30/2006, Bill Gibbs wrote:
Use the codec command in your dial-peer. Or a voice-class so you can
have multiple supported codecs.
Thanks, Bill.
Could you please give an example of a voice-class
entry in the dial-peer file?
The
On Tue, 2006-03-28 at 13:08 -0500, Technical Support wrote:
You can't reliably run a real-time application (like asterisk) on a
virtual machine. You will get better performance from an old PC than
a VM on a new top-end PC. Sorry
MD
H, I would have to say a properly configured GSX
On Fri, 2006-03-24 at 13:07 +1100, James Harper wrote:
I haven't done any sort of research, but I've been told that GSM+DECT
phones are available, and while having them seamlessly switch network
types during a call probably isn't possible, they can function as a
cordless handset.
Can
On Wed, 2006-03-22 at 09:22 -0500, Ron Joffe wrote:
On Wednesday 22 March 2006 00:33, Nathan Alberti wrote:
Here is a dump of the configuration options, you will see there is a
few new, these are also documented on the wiki.
Nathan,
How did you go about obtaining the dump ?
You can
On Wed, 2006-03-22 at 11:52 +0100, Paul Brown wrote:
Hi,
I couldn't find the 7970 SIP image on the cisco.com site. Is it hidden :-)
Any pointers would be appreciated
On Thu, 2006-03-23 at 02:17 +, john wrote:
Hi,
Does anyone know how to define speeddials in XML for the 7970 sip
firmware?. I've played with the SEPmac.cnf.xml file that was posted
previously but can't find a way to do it. I can define them on the
phone usually (seems a bit buggy) but
On Thu, 2006-03-16 at 18:39 -0500, Alexander Lopez wrote:
I have offered but I don't think he (owner) id open to that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Kristian Kielhofner
Sent: Thursday, March 16, 2006 6:39 PM
To: Asterisk
On Fri, 2006-03-10 at 11:52 -0600, Aaron Daniel wrote:
Anyone have the 7970 xml config for sip yet?
Aaron
[EMAIL PROTECTED] ~ $ cat SEP0014A89EF5E3.cnf.xml
device xsi:type=axl:XIPPhone ctiid=203849429
uuid={96f8508b-10ef-f98c-d20d-0471777ec725}
fullConfigtrue/fullConfig
nice, please post the xml file for us :)
On Fri, 2006-03-10 at 13:56 -0600, Aaron Daniel wrote:
Awesome, that works, 'cept now the dialplan doesn't work lol. I've
programmed the voicemail button in, but anything I try to dial doesn't
make it past the first digit.
Aaron
Greg Oliver wrote
On Thu, 2006-03-09 at 10:36 -0500, Nabeel Jafferali wrote:
Is there a way to display the time of the 7960 running firmware 7.4? Im
unable to find any information.
Add the following to SIPDefault.cnf or SIPMAC.cnf:
sntp_server: time.nrc.ca
sntp_mode: unicast
time_zone: EST
You should
On Thu, 2006-03-09 at 13:11 -0600, Shane Young wrote:
Quoting Mailing List [EMAIL PROTECTED]:
I believe they've done that the entire time. I've never known them to be
real supportive of
competing third party solutions.
They support third-party partners such as Broadsoft.
Broadsoft is
On Mon, 2006-03-06 at 22:11 -0500, Darren Wright wrote:
InterestingI've upgraded the 7970 to SIP, but it is still saying
unprovisioned. I've got a SIPMAC file, but it is still looking for the
SEPMAC file...
That's correct - the CCM5 loads only look for SEP files. Even when you
give it
On Mon, 2006-03-06 at 12:38, Nabeel Jafferali wrote:
I have a service contract for my 7960 but I don't see 8.x SIP firmware for
it at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960.
I do see a .cop file for the 7941/7961 8.x SIP load, but nothing for the
7960.
You have to
On Mon, 2006-03-06 at 15:00, Jerry Geis wrote:
I am getting this error from call manager (4.0) and asterisk 1.2.4
I have canreinvite=yes on the call manager setup.
I can call into the asterisk box from call manager. THat seems to work.
When I am calling out of the box using a call file I
On Mon, 2006-03-06 at 15:42, Jerry Geis wrote:
here is some of the output. I am no longer the to spcifically do sip
debug but this is what I have.
along with my sip.conf snip.
The call to extension 3726 never rings. so it never gets answered.
Are you sure your sip trunk and route
On Mon, 2006-03-06 at 15:59, Mailing List wrote:
tar zxfv *.cop
- Original Message -
From: Aaron Daniel [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 06, 2006 4:00 PM
Subject: Re:
On Sat, 2006-03-04 at 10:34 +, Ron Wellsted wrote:
Unfortunately you have to make a choice:
SIP firmware - Easy to implement on *, but poor XML support
SCCP firmware - poor/non-trivial asterisk support, great XML support.
The newest SIP firmware (beta versions) allows the exact XML
It actually depends on the switch model. Some put the port into
trunking mode automatically with the sw voi command, and some do not.
Hopefully one day Cisco will finally make their own products and become
uniform instead of buying several companies and glue'ing them all
together to get an
Greg Oliver wrote:
It actually depends on the switch model. Some put the port into
trunking mode automatically with the sw voi command, and some do not.
Hopefully one day Cisco will finally make their own products and become
uniform instead of buying several companies and glue'ing them
On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote:
Does anyone have a way to do wake calls?
Jordan Novak
Communications Technician
Logistics Health Inc.
You could use cron and /var/spool/asterisk/outgoing scripts to dial
numbers, etc...
On Sat, 2006-02-25 at 13:42 -0500, Dean Collins wrote:
I know this is a OT but great article
http://www.theregister.co.uk/2006/02/23/rwanda_terracom/
Will be interesting to see how this project goes.
Hmmm - it is nice to see things like this happening, but I would have
thought that
The above concern have been a major issue with telephone equipment (eg,
central
offices) and the telco's spend a significant amount of money burying very
long
rods in the ground and interconnectng them with the CO hardware using cables
that are larger then 1/4 in diameter (don't
Depends on the type of satellite, but generally 1500 - 3000ms.
On Wed, 2006-02-01 at 18:28 +0100, Master_PE wrote:
What is a normal dealy on a satelite installation?
Regards,
Master_PE
Op 1-feb-2006, om 13:26 heeft Garth van Sittert het volgende geschreven:
Hi Cosmin
You should
I have found * with the ooh323 channel to be best for this.
On Mon, 2006-01-30 at 15:23 +0200, [EMAIL PROTECTED] wrote:
Hello,
I would like to find an appropriate solution for SIP to H323
translation (vice versa would be great too!), in an environment where
there's going to be 100+
You can have asterisk dial your Unity vmail pilot on busy or
unavailable, and have CCM use the last redirected number on the trunk to
determine the called extension, or pass the $RDNIS value and digit
add/strip from * to CCM.
We use * in the exact opposite fashion, but should suffice either
?
On 1/24/06, Greg Oliver [EMAIL PROTECTED] wrote:
You can have asterisk dial your Unity vmail pilot on busy or
unavailable, and have CCM use the last re
directed number on the trunk to
determine the called extension, or pass the $RDNIS value and
digit
Post your relevant config section and your CCM trunk settings as well as
route patter settings.
On Tue, 2006-01-24 at 12:16 -0800, sys read wrote:
Greg,
appending the number just gives me a fast busy.
Mike,
a) is out because the cheaper cisco sccp phones don't have two way
speaker
I am unsure of * capabilities on NFAS (we do not use PCs to terminate
any PRIs), but it allows bonding of desparate PRIs to use a single
d-channel. ie, you can have 1 d-channel (optional backups) for the
entire DS3. Not sure if * can communicate across cards like that in the
same bus though.
On
Is it set for DHCP - or static? If dhcp, just put option 150 in the
scope for a tftpserver on your network. The password can be changed in
the config file it asks for.
On Tue, 2006-01-17 at 13:38 -0800, Hoss Bazargani wrote:
Hi Cory
thanks, I bought many 7940 from e-bay for our internal use.
I know from everything in the past I have read, that Asterisk natively
bridges calls between endpoints.
We use * for only ACD and VMail purposes at this point, and I was
wondering if there was any way to get a call from:
PSTN-MGCP(cisco)-CCM-*(ACD)-Dial(SIP/)-CCM-(CCM phone)
to not be
I use:
# Enable_VAD (1-enabled, 0-disabled)
enable_vad: 0
dtmf_inband: 1
dtmf_outofband: never
dtmf_avt_payload: 101
and it works well for me. Sometimes going through a callmanager I have
to set outofband to avt to get dialtone sent though.
On Tue, 2005-12-27 at 16:05 +0100, Tomislav Parcina
If using CCM = 4.0, using SIP trunks will alleviate a lot of headaches.
On Tue, 2005-11-15 at 16:33 -0800, Dan Austin wrote:
I posted a couple weeks back about our experiences with H323 trunks on
CCM.
As of version 4.0, the Cisco documents state that a 3rd party H323
gateway
requires a
Do you have a XmlDrfault.cnf.xml file on your tftp server?
On Fri, 2005-11-11 at 16:02 -0700, Kris Edwards wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Still looking for any advice with this. I had given up with the upgrade
process (to SIP.. tftp won't send the files for some
Do a debug voip ccapi on the CME and look through it. It will have
detailed codec negotiations, etc in it.
-Greg
On Wed, 2005-11-09 at 16:10 +0100, Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
my topology is:
CME (Cisco) -- [sip trunk] -- Asterisk --
9, 2005, at 4:33 PM, Greg Oliver wrote:
Do a debug voip ccapi on the CME and look through it. It will have
detailed codec negotiations, etc in it.
thanks for your answer, Greg.
Could you help me?
http://www.nesys.it/snap/debug_voice_ccapi.txt
thanks for your support
Regards
Just put codec g729(whatever version you need) in your dialpeer.
I do not see what the voice-class codec 1 is without that section.
-Greg
On Wed, 2005-11-09 at 17:17 +0100, Andrea Riela wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've forgotten my dial-peer config:
dial-peer
Circuit timing is only to let the hardware know how to keep in sync with
framing and what it is supposed to be. T1 timing will always be the
same, so syncing your card to any of them will be fine. Syncing to 2 -
1 as backup would be best, etc..
Timing has nothing to do with the remote end - it
It is set by your SIPMAC.cnf file.
phone_password: password ; Telnet/Console Password
On Tue, 2005-11-08 at 08:51 -0500, Polycom User wrote:
i appear to misplaced my password for my cisco 7960 SIP Phone. Does
anyone know the procedure to recover this? I have read in the past
that you can
No - only 323 until CCM 5.0
On Tue, 2005-11-08 at 21:42 -0500, Jonathan k. Creasy wrote:
I thought there was a sip image for that phone?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John
Reynolds
Sent: Tuesday, November 08, 2005 4:28 PM
The 7970 when reset to factory will delete the firmware load leaving
just the bootloader.
1. Hold down the # key
2. Power it on
3. Keep holding the power key until the line keys blink orange down the
tree
4. Have the firmware files on your tftpserver when it boots
5. Put the load into the
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
device xsi:type=axl:XIPPhone ctiid=581916804
Forgot to mention - it is 7.0.2-0S firmware
On Fri, 2005-11-04 at 11:35 -0600, Greg Oliver wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL
have the 7.1 images on the phone.
and the message waiting icon is nothing there too but i have a new message on
the server
On Fri, 04 Nov 2005 11:35:32 -0600
Greg Oliver [EMAIL PROTECTED] wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo
You probably do not need firmware. I have tried several versions on
70s, 60s, 12s, 05s and 20s (not 02s) with success.
If they are not even looking for TFTP, then from the phone, hit
Settings-2**#, and erase. Make sure your DHCP server is kicking out
option 150 right (the correct TFTP server) -
You will probably also need to change the media exchange timers in CCM
if you are going to use it as a PRI gateway - otherwise asterisk - 323
- CCM - PSTN calls will get dropped after 4 secs of ringing.
On Mon, 2005-10-31 at 14:41 +0100, Patrick Zwahlen wrote:
Hey Dan, and thanks a lot for your
With asterisk and call manager hooked up via the sip trunk, the calls
from ccm and asterisk can call each other. I have 2 problems.
1. Is it possible to route all calls via the call manager and not
via asterisk when I dial any number?
Yes
1. This is divided into 2
I would have to agree - your easiest route is to upgrade to CCM 4.0+
with SIP trunk support..
On Fri, 2005-10-14 at 16:55 -0500, Paul Davidson wrote:
Message: 13
Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT)
From: [EMAIL PROTECTED]
Subject:
What type of switch/hub is it connected to?
On Thu, 2005-10-06 at 15:40 -0700, Tom Tune wrote:
I saw a thread from 2003 that addressed this problem but they didn't
post a fix:
When I plug my PC into the 2nd ethernet jack on my Cisco 7960g it
loses connection on and off for ~30 seconds at a
Add direct-inward-dial to your dial peer and it should work fine.
-Greg
On Mon, 2005-10-03 at 15:48 -0700, Tim Pozar wrote:
I would think I could do this but for some reason I am stymied.
I have a PRI from RCN connected to a cisco 3640 (in my day cisco is
all lower case :-)). My config
Hm, I would have to disagree.
We use MGCP dial-peers and use it on PRIs with 3725s and 2851s
currently.
On Tue, 2005-10-04 at 12:12 -0700, Tim Pozar wrote:
Greg Oliver wrote:
Add direct-inward-dial to your dial peer and it should work fine.
That command is only supported for POTS
Glad to hear it!
On Tue, 2005-10-04 at 17:53 -0700, Tim Pozar wrote:
Greg Oliver wrote:
Hm, I would have to disagree.
We use MGCP dial-peers and use it on PRIs with 3725s and 2851s
currently.
Our config was fubar'ed. We were using dial-peer isdn instead of
pots. direct-inward
We have all Cisco - and they are pricey, but work great otherwise. Both
with chsn_sccp and SIP. 05 - 70s and a few 20s
-Greg
On Sat, 2005-10-01 at 14:47 +0100, Angus Comber wrote:
No it happens on our asterisk and at a customers. Not that noticeable but
not crystal clear. Didn't happen on
Whatever you have the voice vlan set it is what they operate on. You
cannot provision that on the phone manually. If they are small switches
(35xx, etc), then you need to configure without .1q trunking as those
switch imply it automatically. For the larger switches 1.q trunking in
the config is
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