Videoconferencing: IP or ISDN?

Hi everyone,
We are looking to implement video conferencing. We would like to communicate with a remote office in Europe, and also with external partners and customers.
I am interested in opinions or experiences with video conferencing over the Internet, vs. over ISDN. Is v-con over the web between Canada and Europe feasible? What are the majority of companies doing out there? We have a fibre link to the Internet, but the latency to Europe makes me question how the video will look. Help keep us from making a mistake before buying! :)
regards,
Dan

I have been using it for some time from California to Germany and Italy with Polycom equipment and have had good results.
The latency is there, but I think there is a an inherrant acceptance of turn-around time with Video Conferences. There is a sense or courtesy that even on local "calls" that develops - "now it is your turn to talk".
I can notice the latency, but the users have not complained. If there is packet loss or excessive jitter, they make more comments about the screen pixilating, which happens about the same frequency on ISDN. When you think about a 384K call, that is 6 - 64K ISDN "calls." With that many International calls going on at the same time, you can imagine that one of those could have static or other problems.
As for the purchase decision, I would try to get a unit that can do both. We use IP for most of our internal conferences, and ISDN for video conferences with outside partners.
--Michael

Similar Messages

Voice over IP: ISDN PRI Voice E1 - Calling Out problmes (PSTN)

Hi All,
Has anyone seen this before in a 'debug isdn q931' on a H323 gateway.
*Jun 15 07:39:03.893: ISDN Se0/2/1:15 **ERROR**: CCPMSG_OutCall: fails with cause 0x22.
When I am trying to call out side getting busy signal.The incoming calls working fine. But some times it working without any problems.
Please go through the attached configs...
Thanks
Ilyas

10.31.144.2#sh controllers e1
E1 0/2/0 is administratively down.
E1 0/2/1 is up.
Applique type is Channelized E1 - balanced
Cablelength is Unknown
No alarms detected.
alarm-trigger is not set
Version info Firmware: 20080918, FPGA: 13, spm_count = 0
Framing is CRC4, Line Code is HDB3, Clock Source is Line.
Data in current interval (351 seconds elapsed):
0 Line Code Violations, 0 Path Code Violations
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
Total Data (last 24 hours)
0 Line Code Violations, 0 Path Code Violations,
0 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins,
0 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs
10.31.144.2#
10.31.144.2#sh network-clocks
Network Clock Configuration
Priority Clock Source Clock State Clock Type
1 E1 0/2/1 GOOD E1
10 Backplane GOOD PLL
Current Primary Clock Source
Priority Clock Source Clock State Clock Type
1 E1 0/2/1 GOOD E1
10.31.144.2#
*Jun 25 04:49:30.483: ISDN Se0/2/1:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x1, Calling num 73845
*Jun 25 04:49:30.483: ISDN Se0/2/1:15 Q931: Sending SETUP callref = 0x02D4 callID = 0x8260 switch = primary-net5 interface = User
*Jun 25 04:49:30.483: ISDN Se0/2/1:15 Q931: TX -> SETUP pd = 8 callref = 0x02D4
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Calling Party Number i = 0x0181, '73845'
Plan:ISDN, Type:Unknown
Called Party Number i = 0x80, '141'
Plan:Unknown, Type:Unknown
10.31.144.2#
In this output the E1 was up - able to call outside..., will try to get the down time output if necessary

How to fake ISDN b-channels as busy out of the pri-group timeslots definition ?

Hi, Cisco Support Community,
This is my first post here, so I hope I won't make mistakes... First, thanks for reading me.
I wanted to reuse an old 2811 to act as a SIP gateway on an ISDN primary rate interface. This router is featured with 2 PVDM2-8 modules, so that we're able to handle up to 16 simultaneous calls. So I ordered a pri with 10 channels, which is supposed to be more than enough for our usage.
Everything works fine, excepted that I discovered that the PRI which was delivered to us is a full one, meaning 30 B channels.
As a result, as incoming calls are dispatched round robin by the isdn line provider, when incoming calls start to be presented on b-channel 16, the call is rejected by the router with a "channel unavailable" message... The only way to have a successful incoming call is then to have 15 more unsuccessful tries so that the calls are presented on B-channel 1 again :-/
I have to wait for monday to deal with the line provider to sort this out, but meanwhile, I was trying to find a workaround.
My first try was to use B-channel negotiation on the serial interface (isdn negotiate-bchan resend-setup), but it seems that b channel negociation is disabled on the provider side.
So the other obvious solution would be to fake channels 17 to 31 as busy (isdn busy b_channel xx) ... But as the pri-group timeslots are 1-16, the related serial interface is serial0/0/0:15, and I'm unable to set busy B-channels over 15. And of course, I have no way to define pri-group timeslots 1-31 as I have not enough DSPs to handle more than 16.
I also tried to use the ds0 busyout directive within the E1 controller configuration, but while the command is accepted, it does not appear in the configuration
Below is my running configuration... If some of you can help, this would be much appreciated. Current IOS on the router is Version 12.3(11)T3
Kind regards,
Philippe Martorell
--- cut here ---
Current configuration : 4311 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
hostname RTR-xyz-abc
boot-start-marker
boot-end-marker
enable secret 5 xx
username user1 privilege 15 secret 5 xx
username user3 privilege 15 secret 5 xx
username user1 privilege 15 secret 5 xx
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
aaa new-model
aaa authentication login default local
aaa session-id common
ip subnet-zero
ip cef
ip name-server 8.8.8.8
ip name-server 8.8.4.4
no ftp-server write-enable
isdn switch-type primary-net5
voice-card 0
dspfarm
no local-bypass
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
no call service stop
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
controller E1 0/0/0
pri-group timeslots 1-16
translation-rule 1
Rule 1 ..% 900 international international
Rule 2 ..% 90 national national
translation-rule 2
Rule 1 ..% 02212345
interface FastEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-FE 0/0$
ip address 172.16.31.200 255.255.255.0
duplex auto
speed auto
no mop enabled
interface FastEthernet0/1
ip address 10.0.1.200 255.255.255.0
shutdown
duplex auto
speed auto
interface Serial0/0/0:15
no ip address
isdn switch-type primary-net5
isdn incoming-voice voice
isdn map address .* plan unknown type unknown
isdn negotiate-bchan resend-setup
isdn outgoing-voice info-transfer-capability 3.1kHz-audio
no cdp enable
ip default-gateway 172.16.31.254
ip classless
ip route 0.0.0.0 0.0.0.0 172.16.31.254
no ip http server
snmp-server community xx RO
control-plane
voice-port 0/0/0:15
no echo-cancel enable
playout-delay maximum 170
playout-delay nominal 80
playout-delay minimum low
cptone CH
bearer-cap 3100Hz
no mgcp piggyback message
dial-peer voice 1 pots
application session
incoming called-number .
direct-inward-dial
dial-peer voice 456389160 voip
destination-pattern 6[0-8]
translate-outgoing calling 1
translate-outgoing called 2
rtp payload-type cisco-codec-fax-ind 124
rtp payload-type nte 96
no modem passthrough
voice-class codec 1
session protocol sipv2
session target ipv4:10.200.3.21
dtmf-relay rtp-nte
fax rate disable
no vad
dial-peer voice 999999 voip
answer-address 0.
rtp payload-type cisco-codec-fax-ind 124
rtp payload-type nte 96
no modem passthrough
voice-class codec 1
session protocol sipv2
dtmf-relay rtp-nte
fax rate disable
fax protocol pass-through g711ulaw
no vad
dial-peer voice 9912 pots
destination-pattern 99123T
progress_ind alert enable 8
port 0/0/0:15
dial-peer voice 9920 pots
destination-pattern 99222T
clid restrict
clid restrict
port 0/0/0:15
dial-peer voice 456389162 voip
destination-pattern 69
translate-outgoing calling 1
translate-outgoing called 2
rtp payload-type cisco-codec-fax-ind 124
rtp payload-type nte 96
modem passthrough nse codec g711ulaw
voice-class codec 1
session protocol sipv2
session target ipv4:10.200.3.21
dtmf-relay h245-alphanumeric
fax-relay ecm disable
fax rate 9600
fax nsf 000000
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
no vad
gateway
timer receive-rtp 1200
sip-ua
max-forwards 15
sip-server ipv4:10.200.3.21
banner login ^CCC
WARNING: This is a private network.
Unauthorized access is prohibited.
Use of this system constitutes your
consent to interception, monitoring,
and recording for official purposes
of information related to such use,
including criminal investigations.
^C
line con 0
line aux 0
line vty 0 4
password 7 xx
scheduler allocate 20000 1000
end

Hi,
Thanks for your answer, but unfortunately, this did not help. commands ds0 busyout are accepted, but not taken into account :
RTR-505-012#conf t
*Apr 7 10:23:04.361: %SYS-5-CONFIG_I: Configured from console by sysadmin9 on vty0 (172.16.0.88)
Enter configuration commands, one per line. End with CNTL/Z.
RTR-505-012(config)#controller e1 0/0/0
RTR-505-012(config-controller)#ds0 busyout 17-31
RTR-505-012(config-controller)#do sh run | begin controller
controller E1 0/0/0
pri-group timeslots 1-16
And the "isdn service b_channel state" commands in the serial interface context are rejected
RTR-505-012(config-controller)#interface Serial0/0/0:15
RTR-505-012(config-if)#isdn service b_channel 16 state 2
%The service state of reserved channel 16 on DSL 0 can not be changed.
While the "show isdn service" status shows channels 16 to 30 are out of service, some incoming calls are still directed to these channels by the provider :
RTR-505-012#show isdn service
PRI Channel Statistics:
ISDN Se0/0/0:15, Channel [1-31]
Configured Isdn Interface (dsl) 0
Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3
Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
Channel : 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
State : 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2
*Apr 7 10:20:45.509: %ISDN-6-CHAN_UNAVAILABLE: Interface Se0/0/0:15 Requested Channel 19 is not available
*Apr 7 10:20:53.353: %ISDN-6-CHAN_UNAVAILABLE: Interface Se0/0/0:15 Requested Channel 20 is not available
I'll try to sort this out with the provider today.
Thanks,
Philippe.

Connecting to an ISDN BRI on a router from a PSTN line from remote site..!!

Friends,
I have a 1812 at HO with WIC1 BS/T Card, connected it to an ISDN BRI line.One of my critical remote location uses Leased circuit to reach my HO Router.But in case if LL goes down,some of the important PC's at the remote site should be able to dial to HO Router BRI Number using PSTN line, get connected and continue, ip address assigned from HO Router and access the application... Is any way available...

Hi Arnab.
You need to configure dial backup through AUX port. You need to connect a normal PSTN line to a modem which in turn is connected to your AUX port. You will have to configure a chat script which you can find in the attachmnet. The attachment is a working example in one of my routers.
You also have to create an ASYNC interface. Check that in the attached file also
--Pls rate if useful---

HP Laserjet 4100 PCL 5E keeps "phoning home" and tying up our ISDN line

We have a couple of HP Laserjet 4100 printers that connect to the hospital via an ISDN line (can't upgrade at this time as that is all that is available at that location) and which we pay for by the minute (not much but still by the minute). The problem we are having is that the printer keeps "phoning home" about every 30 seconds or so unless we unplug them. The queue for them is located here at the hospital so I'll assume the printer is "phoning home" so that it can update the status in the queue and whether or not it is online/offline. We have other HP printers out in the field that don't have this problem and yes they do show offline in our queue they still work fine and have worked fine for over a year now.
My question is how does not stop the printer from constantly "phoning home" so that it can report its status. The current driver we are using the is the Laserjet 4100 PCL 5E. Thanks!

The product you have seems to be a commercial printer. For the best chance at finding a solution I would suggest posting in the forum for HP Business Support!
You may find the Commercial Laserjet board here.
http://h30499.www3.hp.com/t5/Printers-LaserJet/bd-p/bsc-413
Hope this helps!
Help the community by marking this post as a solution if it solved your issue!.
If my post helped you in any way, please click the blue KUDOS star under my name! It would mean a great deal.

3241 ISDN Gateway--DTMF Relay Type & local vs. national signaling

Question 1:
Is there a way to configure the 3241 so that it will flag local vs. national call types on the d-channel appropriately? The local TSP sees all calls flagged national.. They do accept both 7 and 10 digit formats for local calls so we set dialing rules to add the local area code (without a +1) to 7-digit numbers and that seems to work OK.
Querstion 2:
Does anyone know what type of DTMF relay is being used by the Cisco-acquired Tandberg/Codian 3241 ISDN Gateway? I'm pretty sure the DTMF relay type is not configurable in that box. Does this fall under the alphanumeric H.245 relay type?
I have a customer that, for outbound long distance calls (Primarily ISDN voice, but some ISDN video), requires their ISP to provide a DTMF-based billing code service. We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately (although I also don't see any way to tell the Cisco 3241 GW to flag the call as Local vs. National
The PRI itself is provided through one local LEC and handed to a secondary TSP to provide the DTMF/billing solution.
The PRI has been works fine for local calls, but long distance (national) calls route throuth the primary provider and then pass to the secondary provider. We've worked with the local TSP to verifiy what they are seeing on the switch and it looks like all digits are passing appropriately. Logs from the ISDN gateway do not show opening any audio or h.245 channel, no 'biiling tone-prompt' is heard, and we cannot access/enable the touch tone menu on the endpoint during the time the ISDN GW is attempting to set up the call. I suspect that either the billing code service is not turned up yet, but I'm hoping its is not a H323/H.245 and/or endpoint cpability that is preventing this from working.
In order to troubleshoot with the service provider I assume I'll need to know what DTMF type we are sending, I am hoping they could also tell me by debuggin the switch for any signaling they are receiving from us.
If anyone has any experience setting up similar service on CUCM i'd love to hear what was required on that system for DTMF relay, and if there is anything else I may need to ask the TSP
Thanks!

DTMF relay for H323 is handled with 3 commands.
cisco-rtp Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal DTMF Relay via H245 Signal IE
Your best bet is h245-alphanumeric. The dtmf-relay rtp-nte command is used for SIP dial-peers.
The DTMF relay feature transports DTMF tones generated after call establishment out of band using either a standard H.323 out-of-band method and a proprietary RTP-based mechanism, or for SIP calls, an NTE RTP packet.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800d62d2.shtml#Solution4
Please rate any helpful posts
Thanks
Fred

Support for AVM ISDN PCMCIA Card in Solaris 10 ?

Hi all,
I would like to install Solaris 10 on my Toshiba Tecra S1 notebook. Today I use an ISDN-Card (AVM FritzCard, PCMCIA) for dial-up connections under Windows XP. Are there any drivers for ISDN and/or PCMCIA-Cards in Solaris10? Does anyone have any experiences with this issue?
Every hint would be appreciated.
Thanks a lot.
Kindly Regard,
Holger

Hello.
There only very poor support for ISDN cards in Solaris. As far as I know only the on-board ISDN card of one of the SparcStation computers is supported (this card is available as S-Bus card for other SparcStation computers).
I think there is no support for any ISDN card that can be attached to an x86 computer.
I wrote a driver for USB based ISDN modems that are based on the Winbond 6694 chip myself due to this lack of ISDN support. I tested it under Solaris 9/Sparc.
Martin

ISDN Authorization with RADIUS using ISE 1.1.2

Hi,
I am trying to move my ISDN dialup branches authentication/authorization from old ACS 4.1 to ISE appliance. Before it was through ACS 4.2 with TACACS protocol but now since we are moving to ISE we are moving them to ISE with radius.
Problem is that isdn client gets authenticated and authorized but calls get dropped and they dont able to communicate with HO. IP address is assigned by Head End router to all remote isdn dialing branches..
I have used default "PermitAccess" in authorization policy and authentication policy is also default. I dont understand where I am going wrong as authentication and authorization is sucessful.
aaa authentication ppp default group radius local
aaa authentication network default group radius
aaa accounting network default start-stop group radius
radius-server host 12.18.22.41
radius-server key *****
below is the router configuration for AAA
can any one help in this

CoA is not needed, nor supported for ISDN aaa, i used ACS 3.3 for this a long time ago. I think you should do some debugging if ise does not give you any errors.
try doing some debug aaa / debug radius & deb ppp nego if your calls are authenticated and ip is assigned to the calling router, you should see some disconnect reason in the debug.

Best practice for Video over IP using ISDN WAN

I am looking for the best practice to ensure that the WAN has suffient active ISDN channels to support the video conference connection.
Reliance on load threshold either -
Takes to long for the ISDN calls to establish causing the problems for video setup
- or is too fast to place additional ISDN calls when only data is using the line
What I need is for the ISDN calls to be pre-established just prior to the video call. Have done this in the past with the "ppp multilink links minimum commmand but this manual intervention isn't the preferred option in this case
thanks

This method is as secure as the password: an attacker can see
the hashed value, and you must assume that they know what has been
hashed, with what algorithm. Therefore, the challenge in attacking
this system is simply to hash lots of passwords until you get one
that gives the same value. Rainbow tables may make this easier than
you assume.
Why not use SSL to send the login request? That encrypts the
entire conversation, making snooping pointless.
You should still MD5 the password so you don't have to store
it unencrypted on the server, but that's a side issue.

X.75 ISDN test setup with 2811 and PVDM2-12DM modem

Hi all,
We have a customer that requires ISDN-X.75 connection as a back-up for IP. This is used by emergency services that require redundancy. We have to build a test set-up. We have two routers with the modem cards and with HWIC-1CE1T1-PRI= cards. We need to build a test set-up to simulate an incoming call. With ISDN-V110 we could use one of the routers to place an outgoing call to the other router. This set-up is not possible with X.75, since the card can not dail out. What would be the best/cheapest/simplest approach to build this test set-up?
My involvement in this project is to build a Perl script that collects data via IOS using expect. If someone has the procedure to collect the calling and called line information for a connection with an example I would be very grateful. Our routers run Cisco IOS Software, 2800 Software (C2800NM-ADVENTERPRISEK9-M), Version 12.4(24)T1, RELEASE SOFTWARE (fc3).
For V110/V.34 and X.25 the tty/vty is collected by issueing a show tcp to relate an IP connection that is created to the tty/vty. The tty/vty port is then correlated to the calling and called line id. This is done with show x25 vc for X.25 or show modem csm for V.110/V.34. We need something similar for X.75 to distill the calling and called line IDs for a given TCP connection.
Thanks in advance!

It colud be Hardware(Digital Modem Module ) problem.
Try this link:
http://www.cisco.com/en/US/products/ps6441/products_feature_guide09186a00806b941f.html#wp1068376

ISDN E1 PRI incoming calls busy out after 2nd b channel picks up

I have an ISDN E1 PRI line coming in from the Telco. D Channel comes up nicely and B channels accept calls, but only the first two calls. Once two callers are online, any further incoming calls get a busy tone. ISDN Debug data reports:
**ERROR**: call_incoming: Remote Protocol Error: No b channel assigned by switch, call id 0xCE
Hardware I'm using? Cisco 2650, 12.3(7T) and NM-HDV, VWIC-2MFT-E1-DI
Anybody please help?

Problem sorted - believe it or not, it was the telco. They changed their switch setting for the line to preferrable B channel - and voila!
Thanks,
Brian

Cause 1 = 0x82A6 Network out of order - ISDN error

When we are try to make outgoing call, we are getting the below error :
cause 1 = 0x82A6 Network out of order
Incoming call there is no problem in the Same E1 PRI line, ISP saying there is no problem from thier side.

We are using h323 gateway. When I am using csim start the outgoing calls fails
ISA-VG1#un all
All possible debugging has been turned off
ISA-VG1#csim start 93XXX6306
csim: called number = 93XXX6306, loop count = 1 ping count = 0
csim err csimDisconnected recvd DISC cid(570049)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
Router#sh gateway
H.323 ITU-T Version: 4.0   H323 Stack Version: 0.1
H.323 service is up
This gateway is not registered to any gatekeeper
Alias list (CLI configured) is empty
Alias list (last RCF) is empty
ISA-VG1#csim start 93XXX6306
csim: called number = 934466306, loop count = 1 ping count = 0
*Feb 5 13:06:47.949 UTC: ISDN Se0/3/0:15 Q931: Sending SETUP callref = 0x3957
callID = 0xCE73 switch = primary-net5 interface = User
*Feb 5 13:06:47.953 UTC: ISDN Se0/3/0:15 Q931: TX -> SETUP pd = 8 callref = 0x
3957
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA9839F
                Exclusive, Channel 31
        Called Party Number i = 0x80, '34XXX6306'
                Plan:Unknown, Type:Unknown
*Feb 5 13:06:48.049 UTC: ISDN Se0/3/0:15 Q931: RX <- SETUP_ACK pd = 8 callref
= 0xB957
        Channel ID i = 0xA9839F
                Exclusive, Channel 31
csim err csimDisconnected recvd DISC cid(570048)
csim: loop = 1, failed = 1
csim: call attempted = 1, setup failed = 1, tone failed = 0
ISA-VG1#
*Feb 5 13:06:48.049 UTC: ISDN Se0/3/0:15 Q931: RX <- CALL_PROC pd = 8 callref
= 0xB957
        Progress Ind i = 0x8288 - In-band info or appropriate now available
*Feb 5 13:06:48.097 UTC: ISDN Se0/3/0:15 Q931: RX <- RELEASE pd = 8 callref =
0xB957
       Cause i = 0x82A6 - Network out of order
*Feb 5 13:06:48.097 UTC: ISDN Se0/3/0:15 Q931: TX -> RELEASE_COMP pd = 8 callr
ef = 0x3957
Incoming calls are perefect, Please let me know the problem is from ISP Side ?

Looking for high-quality videoconferencing to PC in Italy

I've been using Skype to do piano lessons with my teacher in Italy & between me and my son we are videoconferencing nearly 4hrs/wk with her. I have a wireless broadband connection (with ethernet hard-wire to router from my machine). She has a PC laptop & ADSL connection. Problems are increasing with very bad connections. It was noticeably worse after upgrading my OS in November, but I don't know why. We get unbearable static after a few seconds in audio, and frame rate degrades from about 15fps at the start to 4 or 5fps after half a minute or so with latency of more than half a second. The image breaks up frequently, and we often lose the connection entirely. We end up shutting down the audio and calling her on the landline phone for audio. I've tried various solutions, but am really looking for a much higher quality application, and am willing to pay up to $50/mo. We do not need multi-user capability, just the one connection.
Has anyone had personal experience with other applications? iVisit doesn't look much better, but I haven't tried it & their website doesn't compare it to Skype.

Hello cynjon
Before you change software, you might want to try to determine whether the problem is with your teacher's system or ISP. Although I do not frequent the Skype Fora and have not checked there, as noted in this iChat AV 3 thread, iChat users have reported difficulty with Italian contacts who use the Italian "FastNet" ISP: http://discussions.apple.com/thread.jspa?messageID=6237921&#6237921
It might also be helpful to know whether your Italian teacher encounters problems similar to the one with you when he video chats with others. If possible, both you and teacher should try Skype video with other international contacts to try to determine whether one system has more trouble than the other. If one system is not working properly, switching to another video chat app may not help your video connection.
If you would like to try some other apps, several cross-platform video conference / chat apps of widely varying cost and capability are listed in Some Applications You Can Use With iSight. Use the links to each app that is of interest to you for info about its features, terms, and instructions.
You might also want to try using your browser with a web-based video chat-room like MeBeam to see if it might work for you. (Click here for Info / Help.)
If someone here has already found a reliable video connection to Italy, I am sure they will suggest it.
EZ Jim
PowerBook 1.67 GHz w/Mac OS X (10.4.11) G5 DP 1.8 w/Mac OS X (10.5.1) External iSight

NOT RECEIVING THE COMPLETE DIALED STRING FROM AN ISDN TRUNK PBX - AS5850

Hi
The problem i'm facing is the next:
I have a Cisco AS5850 with an Euro ISDN trunk connected to a PBX. The pbx send out the calls in order the A5850 terminate these calls in a remote gateway. I receive only a maximum of 12 digits of the complete string dialed in a phone hooked up in the PBX side.
Is it normal to receive only a max of 12 digits or there is something to do to receive as much digits as those come from the PBX??
My dial-peer looks like this:
dial-peer voice 13 pots
incoming called-number 0T
destination-pattern 0T
direct-inward-dial
port 2/7:D
dial-peer voice 14 voip
destination-pattern 0T
session target ipv4:209.210.174.17
tech-prefix 1051
fax rate 9600
fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback none
and the output from the debug dialpeer is this:
1w4d: destination pattn: 1234567891 expanded string: 1234567891
1w4d: MatchNextPeer: Peer 7 matched
1w4d: MatchNextPeer: Peer 8 matched
1w4d: dpMatchPeersMoreArg: Result=0 after MATCH_ORIGINATE
Thanks in advance for your help!
Oscar

Do a debug isdn q931 and see what comes in from the PBX. Post it here so we can see. They could be sending the digits using overlap sending , so the intial setup may not have all the digits that you want or need.
Also, your VOIP and POTS dial peers have the same destination pattern. There is nothing stopping an inwards ISDN call from being hairpinned back out the PRI if it has a leading 0. You might want to make your dial plan a bit more specific.

Isdn bchan-number-order ascending round-robin is applied but not working

isdn bchan-number-order ascending round-robin is applied but not working
ios c3825-spservicesk9-mz.124-21.bin
relevant config
controller E1 0/3/0
framing NO-CRC4
pri-group timeslots 1-31
interface Serial0/3/0:15
no ip address
encapsulation hdlc
no logging event link-status
no snmp trap link-status
isdn switch-type primary-net5
isdn overlap-receiving
isdn incoming-voice voice
isdn bchan-number-order ascending round-robin
trunk-group 1
no cdp enable
my problem is this:-
At the moment calls are leaving on the first available channel of my h323 voice gateway PRI, I can apply this command to the interface fine but it makes no difference the b cannels are always being picked on a first come first served basis . I have also tried combination "isdn bchan-number-order decending" and "isdn bchan-number-order descending round-robin" but it makes no difference , the first b channel is always selected .
Any ideas , if you think its a bug can u identify the bug id as I cannot find it .

Hello
I can see that you use ascending , that why you hit 1st channel.You have to choose one of these commands ascending or descending
I have a question here , for your incoming calls which channel hits ?.FYI isdn bchan-number-order descending is enable by default , so if you need to change you have to type ascending. Can you just share incoming calls which channel hits?.This is because before you define your outgoing calls , you have to check your service provider channel used, and this to avoid error channel is not available.
Command
Router(config-if)# isdn bchan-number-order {ascending | descending}
Purpose
Enables B channel selection for outgoing calls on a PRI interface (optional).
Thanks
please rate all useful information

Videoconferencing: IP or ISDN?

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