H323 Trunk - "dial-peer voice xy voip" - How many calls could be placed?

Hallo,
do anybody know a limit of simultaneous calls over a H.323 Trunk Connection?
Thanks a lot in advance
Best Regards Martin

This depends upon the bandwidth available and the codec used .
You can use voice bandwidth calculator in cisco tech support site for this if you are a register user.

Similar Messages

Cisco Dial Peer Voice VOIP

Dear Sir,
Due to mis-communications of two companies, My Cisco IP telephone Dial extension is conflicting with other newly installed Telephone extension, because its started on same 4.. number.
Example;
Riyadh Branch (4 Digits) 4119
dial-peer voice 551 voip
tone ringback alert-no-PI
destination-pattern 4...$
session protocol sipv2
session target ipv4:200.200.200.13:5068
session transport tcp
dtmf-relay rtp-nte
codec g711ulaw
fax protocol none
no vad
*Newly Added Configuration*
Jeddah Branch - (3 Digits) 411
dial-peer voice 302 voip
corlist incoming EMPInt
destination-pattern 4..
video codec h263+
session protocol sipv2
session target ipv4:172.16.22.2
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
What will be the possible soultion, they dont want to change the extension of both Branches/Offices.
All Calls from 4 digits will automatically forward & route to 3 digits extension.
Thanks in advance.
Michael
IT

Hi Michael,
Here is one solution from my side...
Apply access code to each site which are overlapping .
Suppose for Riyadh Branch access code is 7 and for Jeddah Branch code is 6. Change dial-peer according to access code and apply translation to convert those number to normal 4 and 3 digit numbers after it matches dial-peer.
voice translation-rule 1
rule 1 /^6$4..$$/ /\1/
voice translation-rule 2
rule 1 /^7$4...$$/ /\1/
voice translation-profile Jeddah
translate called 1
voice translation-profile Riyadh
translate called 2
Riyadh Branch (4 Digits) 4119
dial-peer voice 551 voip
translation-profile outgoing Riyadh
destination-pattern 74...$
Jeddah Branch - (3 Digits) 411
dial-peer voice 302 voip
translation-profile outgoing Jeddah
destination-pattern 64..$
Rate all the helpful post.
Thanks
Manish

Can you see how many calls are waiting in a queue using Supervisor.

Is there a way to see how many calls are sitting in a particular queue using supervisor or is there and additional software or hardware i need to use or purchase?
I am using Contact Center - 5.0(2)SR01_Build053

Do you see any CSQs listed in the upper left box in the Supervisor?
If so, when you click on one or the top-level tree item you should see stats relating to CSQs to the right. You may have to scroll that window accross so see some stats.
Regards
Aaron

My FaceTime icon in dock is not showing when and how many calls I have missed (similar to the email system where it shows how many emails (on the stamp icon) have been received

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Hey michelefromuk,
Thanks for using Apple Support Communities.
Mac Basics: Notifications keep you informed
http://support.apple.com/kb/ht5362
Check or uncheck "Badge app icon" option to show badges on the icon of the app in the Dock.
Have a nice day,
Mario

How to calculate how many calls are there in Queue

Hi All,
I have a requirement of calculating how many calls are there in queue at a particular moment, as the client wants to drop all the calls if there are 5 calls waiting in the queue.
I am using ICM 8.0
Any help would be great.
Thanks

Hi David,
I am trying to crete a IF condition with CallType.callsqnow property, however callsqnow is not coming up in the list for calltype properties.
But under Skillgroup's property list, there is a pooperty "callsqueuednow". I am trying with this property, let's see if it works.
However could you please advice why i am not seeing callsqnow property under call type?
What i am doing is
under if formula, select "call type" -> select name of the call type -> list of properties
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How many calls are held in the call log for the iPhone 3g?

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I have a 10.5.x 100 user deployment of UCCX (30 user licenses) and was wondering if there is a way to tell how many calls (roughly) it will be able to store before it starts purging old data. I've found the purge settings (90 months / 80%) but I don't know how to tell how many calls it takes to get to 80% of database capacity. Has anyone found any documentation on this?
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Hi Kyle
I hadn't seen it, but knew there must be...
from CLI:
show uccx dbserver disk
This appears to show the various informix databases and their files. I'm no Informix expert, but if I apply what I know about database servers generally then I'd interpret it like so:
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In how many ways could I connect different company for telepresence?

Hello everyone,
I'm new in the video conference and telepresence field.
I'd like to know how many ways there are to connect different company to make a telepresence session.
Currently I'm managing a telepresence infrastructure in which all the different location are 30 Km distant and are all part of the same intranet.
Obviously we have a VCSc and VCSe and a MCU.
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Is it possible to have a 720p call considering the distances?

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And these links are not used exclusively for video, but these remote sites use them for all their internet traffic.
Just on a side note, I run HD videconferences from home, but then I have a 100/40Mbps fibre connection, but it still goes across the public internet - and I haven't had any issues yet. :)
As I said at the top, it's trial and error; stress test the link at various times without VPN first, then move to VPN if needed. It's not as simple as a "yes" or a "no".
/jens
Please rater replies and mark question(s) as "answered" if applicable.

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How many calls/texts do you miss with this phone?

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My phone intermittently does not come out of sleep when getting a call or text. I have rebooted, hard reset, erased, and restored the phone. My email is turned to manual. The switch is set to ring and volume turned up all the way.
When I say it doesn't come out, it doesn't light up, vibrate, or make a noise until I click on it to bring it out of sleep. Then it tells me I have a missed call, voicemail, and text.
I can't keep this phone if I miss every single call and text, it defeats the purpose of having it. I only have one phone, so getting calls can be pretty important.
I have perfectly fine network coverage. It's not always full bar, but it is ALWAYS enough to text, email, or call.
If the phone is already unlocked, like if I am looking at my calendar, then the call/text/etc will come through audible with vibrate in addition.
In addition, the phone is sooooo quiet. I can't feel it vibrate in my holster because the text message vibration lasts like a tenth of a second, and it's really not loud at all.
Potentially, I have three defects with this phone: WiFi broken, bad speaker/low sound, won't come out of unlock. Is it really THAT likely to have all three defects in ONE brand new phone? I hope these incidents don't repeat when I get my new phone.
Any thoughts are appreciated.

This is happening to me too! It is not the that the ringer's not loud enough. Calls don't ring through while it's asleep. I had someone stand right next to me and call me and... nothing. But once I wake it up & unlock it, it's fine and calls will ring through. What is the problem? I can't keep it unlocked all the time, the battery will burn down in no time. Can anyone help? Should I be returning it? Is it a hardware or software problem?

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Site to Site calling issue - Cisco 2911 Dial Peer Configuration

My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party.
At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue. We cut that site over to our service and the issue started right up. I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX. Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
Cisco equipment/Dialpeer config below ........
co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
dial-peer voice 100 voip
description --- VoIP Dial-Peer ---
translation-profile outgoing 7digit
huntstop
preference 1
service session
destination-pattern .T
progress_ind setup enable 3
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 99
dtmf-relay rtp-nte
fax-relay ecm disable
fax rate 14400
fax nsf 000000
ip qos dscp af41 signaling
no vad
dial-peer voice 150 voip
permission none
description 900 block
huntstop
destination-pattern 1900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 151 voip
permission none
description 900 block
huntstop
destination-pattern 900T
session protocol sipv2
session target sip-server
voice-class codec 99
dtmf-relay rtp-nte
ip qos dscp af41 signaling
no vad
dial-peer voice 101 pots
description --- INCOMING Calls from PBX ---
incoming called-number .T
direct-inward-dial
dial-peer voice 1001 pots
description --- Calls to the PBX ---
preference 3
destination-pattern .T
port 0/0/1:23
forward-digits 4
Here is some ISDN debug information
BAD CALL
Protocol Profile = Networking Extensions
0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
Component = Invoke component
Invoke Id = 66
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, ''6551''
Plan:Unknown, Type:Unknown
Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
Channel ID i = 0xA98381
Exclusive, Channel 1
Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
GOOD CALL
Protocol Profile = Networking Extensions
0xA116020144020100800E475245454E204D4F554E5441494E
Component = Invoke component
Invoke Id = 68
Operation = CallingName
Name Presentation Allowed Extended
Name = XXXXXXXXXXXXXXXXXX
Display i = ''XXXXXXXXXXX''
Calling Party Number i = 0x2180, ''XXXXXXXXXX''
Plan:ISDN, Type:National
Called Party Number i = 0x80, 'XXXX''
Plan:Unknown, Type:Unknown
Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
Channel ID i = 0xA98381
Exclusive, Channel 1

I done the configration via CCA and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
dial-peer voice 2100 voip
corlist incoming call-internal
description **CCA*INTERSITE inbound call to SITE 1
translation-profile incoming multisiteInbound
incoming called-number 82...
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
dial-peer voice 2101 voip
corlist incoming call-internal
description **CCA*INTERSITE outbound calls to SITE2
translation-profile outgoing multisiteOutbound
destination-pattern 81...
session target ipv4:192.168.50.1
voice-class h323 1
dtmf-relay h245-alphanumeric
fax protocol cisco
no vad
no dial-peer outbound status-check pots

Incoming Dial-peer

I have a SIP trunk set up and can make outgoing calls fine. However, at the moment, i'm having a little trouble getting the incoming calls to be recieved on my ephone-dn.
When an incoming call is recieved I get the following from "debug ccsip all"
The Call Setup Information is:
Call Control Block (CCB) : 0x4838BFD4
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +4418338xxxxx
Called Number : 4418336xxxxx
Source IP Address (Sig ): 83.x.x.x
Destn SIP Req Addr:Port : 87.x.x.x:5060
Destn SIP Resp Addr:Port : 87.x.x.x:5060
Destination Name : 87.x.x.x
Aug 25 08:24:38.529: //2100/FB9CA81483B9/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0
Source IP Address (Media): 83.x.x.x
Source IP Port (Media): 18736
Destn IP Address (Media): 87.x.x.x
Destn IP Port (Media): 12272
Orig Destn IP Address:Port (Media): 0.0.0.0:0
I believe my problem lies with me incorrectly setting up my dial-peer. This is what i currently have.
dial-peer voice 9999 voip
description Incoming call via SIP
translate-outgoing called 3
destination-pattern 4418336xxxxx
translation-rule 3
Rule 0 ^4418336xxxxx 2000
2000 is my extension on CME4.0 and i've x'd up IPs and phone numbers for privacy.
I would have thought that translation-rule would replace the called number with my extension, which would in turn cause my extension to ring. No such luck though.
I can get the call to be put through to a normal phone via a pots dial-peer with the following dial-peer
dial-peer voice 9999 pots
destination-pattern 4418336xxxxx
port 0/1/1
Thanks

Still pulling my hair out with this. I have tried your suggestion with the following setup:
voice translation-rule 10
rule 1 /^441833612345/ /2000/
voice translation-profile incSIP
translate called 10
dial-peer voice 1 voip
translation-profile incoming incSIP
incoming called-number .
Then with "debug voice dialpeer" activated, I get the following.
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=441833612345, Called Number=441833600026, Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=441833612345
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Aug 25 14:29:25.215: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
Aug 25 14:29:25.219: //-1/F1237AF8804E/DPM/dpAssociateIncomingPeerCore:
Calling Number=+441833912345, Called Number=441833600026, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:25.219: //-1/F1237AF8804E/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1
Aug 25 14:29:27.663: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=441833612345, Called Number=441833600026, Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:27.663: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=441833612345
Aug 25 14:29:27.667: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
Aug 25 14:29:27.667: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result=NO_MATCH(-1)
Aug 25 14:29:27.667: //-1/F29902FC8050/DPM/dpAssociateIncomingPeerCore:
Calling Number=+441833912345, Called Number=441833600026, Voice-Interface=0x0,
Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
Peer Info Type=DIALPEER_INFO_SPEECH
Aug 25 14:29:27.667: //-1/F29902FC8050/DPM/dpAssociateIncomingPeerCore:
Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1

H323 Trunk - "dial-peer voice xy voip" - How many calls could be placed?

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