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IP Telephony and Convergence | Forum profile
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Forum profile page for IP Telephony and Convergence on http://avayausers.com.
This report page is the aggregated overview from a single forum: IP Telephony and Convergence, located on the Message Board at http://avayausers.com.
This forum profile page summarizes the general forum statistics such as: Users Activity, Forum Activity, and Top Authors, which are reported in either a table or graph below for a given reporting time period.
Additional forum profile information for "IP Telephony and Convergence" on the Message Board at http://avayausers.com is also shown in the following ways:
1) Latest Active Threads
2) Hot Threads for Last Week
Warning: These statistics are generated using 'best efforts' and can experience delays and reporting errors at times. Please note that such statistics do not constitute a forum's popularity and/or exact posting volumes at any given reporting period.
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Posting activity on IP Telephony and Convergence:
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Week
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Month
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3 Months
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Threads:
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96
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360
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1,133
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Post:
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183
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680
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2,250
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IP Telephony and Convergence Posting activity graph:
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Top authors during last week:
user's latest post:
G350 VoIp Module firmware upgrade
Published (2009-11-07 08:28:00)
I thought on the G350 in the voip firmware was integrated into the gateway image.
user's latest post:
96xx Web Softkey
Published (2009-11-09 07:02:00)
Hi swmenzies, We use the Directory Thin Client Administration ourself for our 9620 phones, but user struggle to use it.. very complexed with all the browsing. We aslo converting cisco phones to use SIP with our Sip Enablement Servers, and for the ciscophones I found this one to use : http://croesnet.eu/trixbox/ldap-dire...7960-ip-phones and modified this for our usage and it works very good. I'd really like the same directorysearch for...
user's latest post:
Fax Over IP Issue, need help
Published (2009-11-09 13:12:00)
Only thing I can think of is if the number you're testing with is one of those combo voice mail/email boxes? And by the time it sends its fax tone the call is already disconnected? Or does this fail for all numbers?
user's latest post:
SIP issues - please help!
Published (2009-11-06 14:50:00)
Glad to hear it was just the codec and you got everything working. I especially thank you for posting all your troubleshooting to get the end results. That will help us all going forward. Most of the time the OP will just say they got it working and not provide the details.
user's latest post:
Avaya softconsole and Bluetooth...
Published (2009-11-06 13:00:00)
It should work as stated above but it will not work exactly like it does for a cell phone. You will not be able to push the button on the headset to pickup calls unless you have one that has some kind of software configuration to interact with Softconsole.
user's latest post:
s8300 C for CM5
Published (2009-11-09 13:35:00)
I think licensing comes into question... What did you generate your license files against? This was my high level process (without going to 20+ sites): Step 1. Generate License Files (BP did this against the SID's/MID's) Step 2. Back up Media Gateways config Step 3. Upgrade Firmware on all Media Gateways Step 4. Back up Media Gateways config again Step 5. Load License files on Media Servers Step 6. Load CM on HTTP server local to...
user's latest post:
s8300 C for CM5
Published (2009-11-09 12:03:00)
8300Cs will support 5.2.. so there should not be a huge problem.. would give you a platform to test against.. Dave
user's latest post:
Fax Over IP Issue, need help
Published (2009-11-09 14:20:00)
Thanks Kennyp, tried what you suggested but still get the same results. The call won't come up to g.711 and the trace shows it not even trying. I have arranged for Avaya Channel Partner to be on site Thursday to work this issue out with me.
user's latest post:
SIP issues - please help!
Published (2009-11-09 22:30:00)
No probs - hopefully it will help other SIP 'noobs' with troubleshooting if nothing else - well thats the intention anyway Jim
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Latest active threads on IP Telephony and Convergence::
Started 3 weeks, 1 day ago (2009-10-20 01:47:00)
by AdG Member
Hello,
We have sometimes the problem that there is no audio when a call is answered. The phone rings, I answer the phone, but I hear nothing.
I hang up, the phone rings again (same caller) and the audio works.
It happens only with internal calls, especially when the call is made from a 4690 conference station...
What can this be?
System: S8700+G650 with CM3.1
Phones: 4610SW and ...
Started 15 hours, 55 minutes ago (2009-11-10 18:09:00)
by kennyp
You can plug it 2 ports but you need some type of loop avoidance to prevent
network loops. You can use spanning tree or you can use port redundancy but
only one port will ever be active at any given time to again to prevent
loops.
Started 3 months, 3 weeks ago (2009-07-19 23:15:00)
by dpupkov
Are you sure, that you correctly write ip addr ESS, CLID, Server ID of 7th ESS pair in cha sys ess ?
Started 1 week, 5 days ago (2009-10-30 01:06:00)
by alhafoudh Member
Hi all,
I have 3 SIP accounts which are registered from 3 asterisk instances. These asterisk translate this to H323 singaling for one avaya system using singaling groups/trunks. My problem is to detect an outage of one of SIP trunks of asterisks on avaya to put the singaling group to " out of service". Please can someone advice?
Thank you!
Started 18 hours, 9 minutes ago (2009-11-10 15:55:00)
by PBXJR
Outgoing trunk seizure failure, busy verify hit glare.
There is a possible problem with the selected trunk, or an incoming call was received on a trunk selected for an outgoing call.
Started 1 day, 21 hours ago (2009-11-09 12:13:00)
by cbirckhead
Are you sure the signaling group is using region 2? Is that the far end
network region? The processor by default is in region 1?
Started 1 month, 3 weeks ago (2009-09-17 09:45:00)
by smitty50
I've been having issues getting this setup as well. Any information would
be greatly appreciated.
Started 22 hours, 17 minutes ago (2009-11-10 11:47:00)
by bshamilton
If voicemail is Audix, you can change subscriber 5678 and add 1234 to it as the secondary extension.
If this is not possible, you could forward 1234 to the DID for 5678
(i. e. call forward to 9 xxx yyy 5678) The call will go out on a trunk, and come in on a trunk to the voicemail for 5678
Started 23 hours, 27 minutes ago (2009-11-10 10:37:00)
by David Tessari
all of the Avaya IP wireless products will work.. the Spectralink gear and
the 3631.. should not be a problem..
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Hot threads for last week on IP Telephony and Convergence::
Started 1 week, 2 days ago (2009-11-02 06:46:00)
by cbirckhead
Yes you can setup phones in groups... you add them to the group you want them in CM, and then you build out the settings.txt file to have the rules you want for different groups..
Also I don't mean to state the obvious on you, but COR and COS can also restrict different options which may or may not fit your needs, and that can be done directly in CM, without setting up groups.
Started 1 day, 21 hours ago (2009-11-09 12:13:00)
by cbirckhead
Are you sure the signaling group is using region 2? Is that the far end
network region? The processor by default is in region 1?
Started 1 week, 1 day ago (2009-11-02 12:14:00)
by cyberfalco
Started 1 week, 1 day ago (2009-11-03 05:44:00)
by LauraH
Not clear on whether you are asking about EAS or AES - two completely
different things.
Started 2 weeks, 5 days ago (2009-10-23 06:47:00)
by Trey
I made this a few months ago....it may help you.
http://kosmo.homedns.org/avaya/Setti...%20-%20SES. doc
My only other suggestion is to check your digit map on the phone's webpage.
Started 5 months, 2 weeks ago (2009-05-26 12:26:00)
by David Tessari
I think you can still use any URL based directory.. Avaya has a freeware for that.. but the integration note for the SES/Cisco config can be found here..
http://www.avaya.com/master-usa/en-u...scoUIP_SIP. pdf
in the TFTP file you will need to identify the location..
<directoryURL>http://10.0.0.20/cisco_voip/PhoneD irectory.xml</directoryURL>
Dave
Started 1 month, 3 weeks ago (2009-09-15 06:38:00)
by cbirckhead
Without knowing specifics, if we assume for a second it does require your
AES server to run, did you verify if the link to the AES server is up?
Might be an easy place to start...
Started 1 week, 1 day ago (2009-11-02 13:46:00)
by PBXJR
Once your License issue was resolved and the RSP was started; you'll be able to view updated reports. (You wont have to wait till tomorrow).
Do you see calls being processed in the RSP?
Make sure the call Processing Collection status has been is enabled; if not re-enable it.
Do you have records in rating queue waiting to be processed? If so then then calls are not being processed....
Started 2 weeks, 1 day ago (2009-10-27 08:02:00)
by hpenn1
Same problem on same server on version 5.01.2.416
Seen a software request 1 ( warm) restart at same start
Anybody any ideas?
Thanks,
Started 1 week, 2 days ago (2009-11-02 06:23:00)
by LauraH
Can you be more specific about exactly what you are trying to do? The number of licenses will control how many agents you can have logged in.
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